- sr-users - kamailio.org

Rebooting kamailio.org server
by Daniel-Constantin Mierla 02 May '23

02 May '23

Hello, in about 1 hour the kamailio.org server will be rebooted for some system upgrades, it should be quickly back running, but if anyone accesses the web/wiki or mailing lists archives exactly during that time, they might not respond -- just wait for a minute and try again. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

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Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

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Loading the LCR Module Tables at startup/reload
by harneet singh 30 Apr '23

30 Apr '23

Hi All, Can someone point to a sample as to how we can *load the LCR Module tables* namely the following at startup from a csv file or any other mechanism. * lcr_gw_table * lcr_rule_table * lcr_rule_target_table The idea is to not need to manually feed in the entries into the table but instead add these in the beginning through either a *wrapper script *OR *through the kamailio cfg file* if possible. Any pointers are much appreciated. Thanks & Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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Route Header based routing for initial INVITE
by harneet singh 28 Apr '23

28 Apr '23

Hi Experts, I believe the loose_route() in kamailio can be used for initial Out of the dialog INVITE processing as well. Is that the correct understanding? If so, the request will be forwarded ahead to the destination corresponding to the Route Header remaining (if any) after Kamailio has stripped off it's own Address(es) from the Route Header(s). Is that correct? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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Strange RTCP issue
by Kiss Zoltán 27 Apr '23

27 Apr '23

Hello, We have a transparent proxy in front of Asterisk boxes. The media is handled by rtpengine, and the "manage" work is done by Kamailio in the route[NATMANAGE] section like this: if(nat_uac_test("8")) { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } else { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } Everything is working fine, but with some clients (like Grandstream phone) the RTCP session wants to go tot he private address of the phone. Here is the log of one of these strange calls: Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11088 <> 81.183.216.3:5068 , SSRC 77c19488, 625 p, 107500 b, 0 e, 29 ts Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11089 <> 10.0.5.192:5069 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 43 ts As you can see the RTP itself is okay, but the RTCP will somehow go to the private address. If we tracing the sdp messages in sngrep, then we can see that Kamailio transforming IP addresses for the backend servers to the public (and the private address of the rtpengine) addresses. We are missing something? Thank you! With kind regards, Zoltan

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Sensible values for pike mudule?
by Benoit Panizzon 27 Apr '23

27 Apr '23

Hi Before I go too deep into try and error, I guess others have been there too. What are sensible value you use for the pike module to detect / mitigate abusive behavior, especially dictionary attacks? Are there better solutions that the pike module? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Fwd: Sensible values for pike mudule?
by Stefan 27 Apr '23

27 Apr '23

Hi Benoît We use pike in combination with fail2ban. Log a message -> create a fail2ban rule to block the ip for xxx time. example: if ($ua =~ "(friendly-scanner|sipvicious)") { sl_send_reply("488", "Go away."); xlog("L_INFO","$ci Blocking traffic from $si\n"); exit; } Cool thing is, you can write a log entry from every part in your kamailio-config, and then fail2ban does the rest. Regards Stefan Am 27.04.23 um 16:07 schrieb Benoit Panizzon: > Hi > > Before I go too deep into try and error, I guess others have been there > too. > > What are sensible value you use for the pike module to detect / > mitigate abusive behavior, especially dictionary attacks? > > Are there better solutions that the pike module? > > Mit freundlichen Grüssen > > -Benoît Panizzon-

1 0

hooking mechanism for routing blocks
by Valentin Christoph 27 Apr '23

27 Apr '23

Dear all, I am looking for some hooking mechanism for user defined routing blocks. Is something like this already available? 1) a common part of the routing configuration shall not (or only slightly) be changed, after some hooks have been implemented Something like a function "engage_hooks()" of some "hook_module"? : Routing config that should not be changed by application developers engage_hooks(HOOK_TYPE); Routing config that should not be changed by application developers : 2) one application developer may "hook" new parts of the routing configuration into this, by modparam("hook_module", "hook", "name=MY_RB;type=HOOK_TYPE;prio=50); : : Route [MY_RB] { Do something at hook HOOK_TYPE; } : : 3) another application developer might want to "hook" into the same HOOK_TYPE with a higher priority modparam("hook_module", "hook", "name=ANOTHER_RB;type=HOOK_TYPE;prio=20); : : Route [ANOTHER_RB] { Do another thing with higher priority at hook HOOK_TYPE } : : Kr, Christoph

2 2

Kamailio use case
by jdprsaga＠gmail.com 26 Apr '23

26 Apr '23

Hello All, new subscriber here. I've been reading kamailio documentation to check if kamailio has a use case on the architecture for the solution for my company's software, here an small architecture diagram of our solution. sipclient <----> [ Middleware ] <-----> SBC Our middle ware is both a sip proxy and media proxy. we have a large number of sip clients but a small calling number, our current bottleneck is that the middleware can't handle a high number of registration, i would like to deploy Kamailio as a load balancer for registrations, making sure it is statefull so registrations and calls are hashed into the same middleware server. sipclients < ----> [ kamailio ] <------> [ middleware1, middleware2,...,middleware N] <-----> SBC Any thoughts or pointers?

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rtpengine module: Actions caused by rtpengine_manage() on reply-codes?
by Benoît Panizzon 26 Apr '23

26 Apr '23

Hi all I wonder, if there is a documentation on what action is performed by rtpengine_manage() on which kind reply. Am I right, that rtpengine_manage(), on any 4XX reply causes a delete? So if there is a re-invite (for example to switch to T.38) and the other party replies with 488 which in turn is passed through rtpengine_manage() the existing RTP stream is deleted from rtpengine instead of being left running with the previously agreed codec? Has anyone already figured out a list of what reply codes should be exempt from calling rtpengine_manage()? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Configuring dynamic Destination identification for Kamailio Routing
by harneet singh 25 Apr '23

25 Apr '23

Hi Experts, We have been using Kamailio as a sip proxy for a fair bit of time now(version 5.3.2 though) and do understand the native routing mechanism that can be instrumented through the kamailio cfg scripts. We initially had only two sides, where the traffic could come from into Kamailio and it just passed through to the other side(ie Traffic from A was routed to B and that from B was routed to A). Off late, a few more sip elements have been introduced in the deployment and now there is a need for a more formal/robust mechanism to select the intended destination to forward an initial SIP INVITE to. I believe that we could still keep all the logic in the native kamailio script and do the routing(for example, an incoming call where the *From URI* contains *sip.elementA.com <http://sip.elementA.com>* should be routed to destination *sip.elementB.com <http://sip.elementB.com>*, where as if the* To URI *contains *sip.elementC.com <http://sip.elementC.com>*, the request should be routed to *sip.elementC.com <http://sip.elementC.com>)**. *Since we have many sip elements like A,B.C,D,E etc where the requests might have to be routed to, based on certain different criteria, this logic would become cumbersome. *What would be the best way to accommodate this?* *Would it be possible to rewire this routing logic at runtime(*like in some csv file that can be reloaded*) without a Kamailio restart? * Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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sips to sip with TLS proxy
by Kiss Zoltán 25 Apr '23

25 Apr '23

Hi all, We have a working Kamailio setup, lets call it a transparent proxy for Asterisk boxes. Its based on domain and dispatcher modules and everything is working as expected with the test clients (more or less microsip, softphone for ios, etc). We are tried to register with a Grandstream deskphone today, and we see that the phone sending sips:xxx in the Reg Contact field for example. Because the sips schema, the register is working, but we cannot initiate calls from this phone. If we are turning SIP scheme to sip from sips in the phone, then everything is working as expected. I think we can transform those requests from sips to sip with Kamailio, but currently we dont know where can we start. Has anybody a suggestion about this issue? I know that we can transform ruri, contact, etc with textops, nathelper and a lot of other modules, but what is the best for this sips->sip translation? Thanks for your help. With kind regards, Zoltan

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TCP/TLS connection (id: 0) for WebSocket could not be found
by Joey Golan 22 Apr '23

22 Apr '23

Hello, I have 2 clients. 1 using websocket and 1 mobile using TLS. Both can call each other perfectly. After 24-48 hours calls from mobile to websocket failed with an error message: TCP/TLS connection (id: 0) for WebSocket could not be found. At the same time I can make calls from the websocket client to the mobile. I know that the search of the connection is done by target address so I printed the addresses + ports and they are matched to the ones that appear on the ul.dump and ws.dump. *kamcmd ul.dump output:* Info: { AoR: usrmt63ly_1040 HashID: 1830036546 Contacts: { Contact: { Address: sip:usrmt63ly_1040@example.com;transport=ws Expires: 283 Q: -1 Call-ID: tvne1pskgkm2a01pnqm677 CSeq: 1060 User-Agent: JsSIP 3.9.0 Received: sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws Path: <sip:10.218.154.228:50443;transport=ws;received=sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws;lr> State: CS_NEW Flags: 0 CFlags: 64 Socket: tls:10.218.154.228:50443 Methods: 7071 Ruid: uloc-61ebba84-64242101-89ba-1 Instance: <urn:uuid:7573726d-7436-116c-915f-313034304077> Reg-Id: 1 Server-Id: 1642838660 Tcpconn-Id: 100420 Keepalive: 1 Last-Keepalive: 1680417282 KA-Roundtrip: 0 Last-Modified: 1680417282 } } } *kamcmd ws.dump* { connections: { 100420: wss:*10.218.51.86:24061 <http://10.218.51.86:24061>* -> wss: 10.218.154.228:50443 (state: OPEN, last used 14s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } *Call from mobile (TLS) to websocket: Failed with error message: TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: request_route: method [INVITE] from [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [sip:usrmt63ly_1040@example.com] 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: [RELAY]: method [INVITE] from [sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [ sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] 30(35263) WARNING: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2863]: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2040]: build_req_buf_from_sip_req(): could not create Via header 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:479]: prepare_new_uac(): could not build request 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: No error (2/SL) 30(35263) NOTICE: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} acc [acc.c:287]: acc_log_request(): ACC: call missed: timestamp=1680417920;method=INVITE;from_tag=e168ffa6-e577-48ec-b9c9-a73f5be318e8;to_tag=0c3e625e9f223a7ee2b093e8e31b2526-b7165bcb;call_id=4248c155-cd92-49b8-b68a-eef5f2cb3e88;code=500;reason=Server Internal Error;src_user=usrrcv4o4_1040;src_domain=example.com ;src_ip=10.218.51.86;dst_ouser=usrmt63ly_1040;dst_user=usrmt63ly_1040;dst_domain= example.com *Call from websocket to mobile (TLS):* 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: request_route: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: [RELAY]: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [100] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [200] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) NOTICE: {2 7488 INVITE urcnete8r71dpkhmngd8} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1680417926;method=INVITE;from_tag=18q0p37nfo;to_tag=a88272a0-25ca-43c2-8ce9-8615831b7e9a;call_id=urcnete8r71dpkhmngd8;code=200;reason=OK;src_user=usrmt63ly_1040;src_domain= example.com ;src_ip=10.218.51.86;dst_ouser=usrrcv4o4_1040;dst_user=usrrcv4o4_1040;dst_domain=10.218.51.86 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: request_route: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: [RELAY]: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] *Version: kamailio 5.6.2 (x86_64/linux)* What could be the reason for that? Your help would be much appreciated. Thanks, Joey.

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How to log expiring dialogue?
by Benoît Panizzon 21 Apr '23

21 Apr '23

Hi alle I think I have an issue with the dialogue and session timer module with Kamailio 5.4 I sync the Session Expire Timer with the dialogue timeout via a common AVP. When I check a dialogue via kamcmd dlg.list start_ts: 1682066352 init_ts: 1682066346 end_ts: 0 timeout: 1682066952 lifetime: 600 Timeout is increased on the first session refreshing re-invite, as expected (in this case after 300 seconds). But it looks like the dialogue just disappears (from kamcmd dlg.list, not from the backing database) when the lifetime ist reached, leaving an orphan database entry. I would like to digg deeper into this. Is there a way to log more details from the dialogue module like when it triggers dialogue expiry? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Roadmap to next major release series v5.7.x
by Daniel-Constantin Mierla 20 Apr '23

20 Apr '23

Hello, it is time to make the roadmap to the next major release series v5.7.x. In my opinion, I would try to freeze the development during the first part of April 2023, tentatively proposing April 14 as freezing date, being like 4 weeks from now. Then testing phase and first release (v5.7.0) sometime during May 2023. Alternative timelines can be proposed, of course. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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Rtpengine: no audio after kernelization.
by Michel Pelletier 17 Apr '23

17 Apr '23

Hello, I am proxying all RTP through RTPEngine. Everything works fine until about 5 seconds into the call, when rtpengine enters kernelization, after which all RTP forwarding ceases. I've checked the required iptables entries, and all looks good. Here is a description of my environment: # cat os-release PRETTY_NAME="Debian GNU/Linux 11 (bullseye)" NAME="Debian GNU/Linux" VERSION_ID="11" VERSION="11 (bullseye)" VERSION_CODENAME=bullseye ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" # kamailio -v version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 16:02:49 Dec 1 2022 with gcc 10.2.1 # rtpengine -v Version: 11.1.1.3-1~bpo11+1 Michel Pelletier

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Docker repository removed
by Manel Villar 17 Apr '23

17 Apr '23

Hi, I was using the kamailio/kamailio docker repository in one of my projects, and now the image is not available as the repo has been removed. Is the "kamailio-ci" repo a replacement for it? Thanks! -- Manel Villar Steegerstr. 59 13359 Berlin Germany Tel: (+49) (0)162 6574604 skype: manel.villar manelvf(a)gmail.com *"Ars longa, vita brevis"*

4 5

Development for v5.7.x is frozen
by Daniel-Constantin Mierla 17 Apr '23

17 Apr '23

Hello, the formal notification that the development for the next major version 5.7.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.7 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.7. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.7. Once the branch 5.7 is created, new features can be pushed again to master branch as usual. From that moment, the v5.7.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.7 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-5.7.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.6.x-to-5.7.0/ Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.6.x to 5.7.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Assigning a new RTPEngine for a call for the sake of failover
by mehdithreem+kamailio＠gmail.com 15 Apr '23

15 Apr '23

Hello, I am trying to implement RTPEngine failover, where if a selected RTPEngine node fails in the middle of the call (and not responding NG requests), I want to select an alternative node. However, when I call rtpengine_manage(), it does not seem to select an alternative node in case of a failure. To address this, I have attempted to check if $avp(RTP_INSTANCE) is null after calling rtpengine_manage(), and then use rtpengine_delete() to forget the selected RTPEngine node for the specific call, with the expectation that Kamailio would assign a new RTPEngine node for subsequent SIP message retries. However, it appears that rtpengine_delete() does not have the desired effect in this regard. Here is an excerpt of my code for reference: ``` route[RTP] { #!ifdef WITH_RTPENGINE if (has_body("application/sdp")) { if ($avp(x_source) == "INTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=internal direction=external"); } else if ($avp(x_source) == "EXTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=external direction=internal"); } if ($avp(RTP_INSTANCE) == $null) { rtpengine_delete(); drop(); } } #!endif } ``` Can you please advise on how I can achieve the desired failover behavior? Is there another function similar to `rtpengine_reset_call` that can make Kamailio forget the RTPEngine node for a specific call? Thank you!

2 1

CLI & destination based LCR
by John Cahill 14 Apr '23

14 Apr '23

Hi, I'd like to build an LCR solution that routes on both CLI & destination e.g. CLI matches ^\+49*, destinations matches ^\+44* => route to gateway 5 for a large rule set (~ tens of thousands). Any recommendations for which modules to use? Can kamailio perform this particular task well or should I look at another solution? I'd be very grateful to hear about any experience of doing this. Many Thanks. Regards, John

5 5

Kamailio + PostgreSQL
by georgy.grigoryev＠snapcom.fr 14 Apr '23

14 Apr '23

Hello everyone, I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. They are on two different servers. I have created the Kamailio DB on the Postgre with the help of the command "kamdbctl create" I'm able to connect to the database with no problem. The problem is in the kamailio.cfg file, i have added this so far: #!define WITH_PGSQL #!ifdef WITH_PGSQL #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif modparam("db_postgres", "retries", 3) modparam("db_postgres", "timeout", 10) modparam("db_postgres", "tcp_keepalive", 600) modparam("db_postgres", "lockset", 6) modparam("db_postgres", "bytea_output_escape", 0) But my kamailio doenst like this configuration, can someone help me find the problem? Thank you in advance! Best Regards!

3 2

manipulate Authentication header
by David Villasmil 13 Apr '23

13 Apr '23

Hello guys, I can't find a simple way of changing the username and recalculate the has in a REGISTER... do i have to use textops? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

How to correctly use the SST module?
by Benoît Panizzon 11 Apr '23

11 Apr '23

Hi I still have odd behaviour with the SST module: modparam("sst", "min_se", 300) if (sstCheckMin("1")) { xlog("L_ERR", "$cfg(route) Session Timer too small\n"); exit; } setflag(FLT_DLG); # set the dialog flag setflag(FLT_SST); # Set the sst flag if(is_present_hf("Session-Expires")) { $avp(dlgtimeout) = $(hdr(Session-Expires){s.int}); } On an INVITE containing those header: Session-Expires: 3600;refresher=uac Min-SE: 600 Supported: 100rel,timer Expires: 330 Now Kamailio replies: SIP/2.0 422 Session Timer Too Small Min-SE: 300 What am I doing wrong? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

dlg_onroute(): unable to find dialog for INVITE with route param
by Benoît Panizzon 10 Apr '23

10 Apr '23

Happy Eastern Kamailio 5.4 used. On dialogues calls, I noticed this happening on the second re-invite to refresh the session from the B side. dlg_onroute(): unable to find dialog for INVITE with route param 'df9.da75' [2557:22445] and call-id '14148a68-f1b2-485d-b5a2-f9dd7773d2c0' The did df9.da75 is identical to the one in the route-header on the first re-invite from the B side, before this one. No CDR was (yet) generated, and the dialogues is still present and on-going according to the database: CallID [14148a68-f1b2-485d-b5a2-f9dd7773d2c0] H-Entry: 2557 H-ID: 22445 But most probably not in memory as kamcmd dlg.stat_active shows zero (this is on a development node, so no other calls running). There are no sip messages that end the call in between those two re-invites. I see no session timeouts or similar in the logs. Any hint what could cause Kamailio not to find the corresponding dialogue on this re-invite? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 2

Fwd: Kamailio + PostgreSQL
by Stefan 07 Apr '23

07 Apr '23

Hi Give it a try without modparams to see if it works, I have the same config but without modparams and it runs without problems. Else have a look into the kamailio logfile for errors. Or check config with "kamailio -c"? Regards Stefan Am 06.04.23 um 15:18 schrieb georgy.grigoryev(a)snapcom.fr: > Hello everyone, > > I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. > They are on two different servers. > I have created the Kamailio DB on the Postgre with the help of the > command "kamdbctl create" > I'm able to connect to the database with no problem. > The problem is in the kamailio.cfg file, i have added this so far: > > #!define WITH_PGSQL > > #!ifdef WITH_PGSQL > #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" > #!endif > > #!ifdef WITH_PGSQL > loadmodule "db_postgres.so" > #!endif > > modparam("db_postgres", "retries", 3) > modparam("db_postgres", "timeout", 10) > modparam("db_postgres", "tcp_keepalive", 600) > modparam("db_postgres", "lockset", 6) > modparam("db_postgres", "bytea_output_escape", 0) > > But my kamailio doenst like this configuration, can someone help me > find the problem? > > Thank you in advance! > > Best Regards! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only > to the sender! > Edit mailing list options or unsubscribe:

2 2

Malformed To Header
by Serdar GUCLUER 06 Apr '23

06 Apr '23

Hello all, In my situation, it is possible to call forwarding to registered users multiple times on same leg. For each forward i have to update To uri and i am using uac_replace_to to do that. But on second relaying, To uri was malformed. On below, Im attaching configuration to illustrate my problem. (version-5.4.9) First relay - /To: //sip:xyz@192.168.0.51/ Second relay - /To: sip:xyz@192.168.0.51sip:abc@192.168.0.51/ Thanks in advance. Serdar /loadmodule "uac.so"// //modparam("uac","restore_mode","auto")// // // //request_route {// // ...// // $avp(x) = "abc";// // $avp(x) = "xyz";// // $avp(myip) = "192.168.0.51";// // $avp(i) = 0;// // route(SERDAR);// // ...// //}// // //route[SERDAR] {// // if ($(avp(x)[$avp(i)])!=$null) {// // $avp(destination) = $(avp(x)[$avp(i)]);// // $avp(i) = $avp(i) + 1;// // $rU=$_s($avp(destination));// //uac_replace_to("$_s(sip:$avp(destination)@$avp(myip))");// // route(LOCATION);// // } else {// // send_reply("403", "Opps!");// // exit;// // }// //}// // //route[LOCATION] {// // $avp(oexten) = $rU;// // if (!lookup("location") ) {// // t_newtran();// // switch ($var(rc)) {// // case -1:// // case -3:// // send_reply("480", "Temporarily Unavailable");// // exit;// // case -2:// // send_reply("405", "Method Not Allowed");// // exit;// // }// // }// // t_on_failure("SERDAR_FAILURE");// // if(!t_relay()) sl_reply_error();// // exit;// //}// // //failure_route[SERDAR_FAILURE] {// // route(NATMANAGE);// // if (t_is_canceled()) //exit;// // route(SERDAR);// //}/

6 11

Presence module and privacy
by Ihor Olkhovskyi 06 Apr '23

06 Apr '23

Hello! Is there any possibility to control more-or-less privacy related parameters of pua_dialoginfo module (like include_localremote) per subscription? Main idea is I want to show some users full info on the call other user is making, for a different user - just show fact of the call without any details. For now I'm only seeing some manual manipulation over XML based on user who is receiving NOTIFY, but maybe I'm trying to invent a weel here. -- Best regards, Ihor (Igor)

2 2

Re: Repository for modules
by Henning Westerholt 06 Apr '23

06 Apr '23

(please keep the list in the communication) Hello, you can find the modules for example in the Debian package service: https://packages.debian.org/bullseye/amd64/kamailio-tls-modules/filelist Crypto is in TLS module package. Cheers, Henning From: Duarte Rocha <duarterocha91(a)gmail.com> Sent: Donnerstag, 6. April 2023 15:42 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Repository for modules Thanks for the quick answer. How can I see info regarding which modules are in the combined packages? One of the modules i'm looking into is the crypto module for example. Otherwise, where can I get a list of modules not included in Debian packages? Cheers, Duarte Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> escreveu no dia quinta, 6/04/2023 à(s) 13:17: Hello, the majority of modules should be packaged for Debian. What modules are missing for you? Keep in mind that some modules are combined in some packages. Some modules can not properly packaged due to missing dependencies, this needs to be installed from source, right. Cheers, Henning From: Duarte Rocha <duarterocha91(a)gmail.com<mailto:duarterocha91@gmail.com>> Sent: Donnerstag, 6. April 2023 13:55 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] Repository for modules Hello, I'm currently testing Kamailio on a Debian 11 machine. In order to install it I used apt with repositories from Kamailio (https://deb.kamailio.org/). Now I'm trying to add modules and I'm wondering the best way to do it. Using apt install i don't have access to all the modules. Is there another repository for this? Is my only solution downloading and compiling the modules? Cheers

1 0

Repository for modules
by Duarte Rocha 06 Apr '23

06 Apr '23

Hello, I'm currently testing Kamailio on a Debian 11 machine. In order to install it I used apt with repositories from Kamailio (https://deb.kamailio.org/). Now I'm trying to add modules and I'm wondering the best way to do it. Using apt install i don't have access to all the modules. Is there another repository for this? Is my only solution downloading and compiling the modules? Cheers

2 2

issue in kamailio of udp
by satyaprakash ch 06 Apr '23

06 Apr '23

Hi, I had upgraded Kamailio from 5.4.7 to 5.5.5, Then I restarted the Kamailio service. After upgradation im getting the below error, /usr/local/sbin/kamailio[21754]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock, buf: 0x7f7502289200, len: 61012, 0, dst: ( 10.122.45.189:5080), tolen: 16) - err: No buffer space available (105) /usr/local/sbin/kamailio[21754]: ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed /usr/local/sbin/kamailio[21754]: ERROR: tm [uac.c:678]: send_prepared_request_impl(): Attempt to send to precreated request failed ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed ERROR: tm [uac.c:678]: send_prepared_request_impl(): Attempt to send to precreated request failed Anyone have any idea on this errors.

2 3

SCSCF hangs randomly
by solomon.thoobe＠smilecoms.com 05 Apr '23

05 Apr '23

Hi All, We currently experience issues where scscf hangs randomly throughout the day. It started hanging once in a month, the issue has gotten worse over time and now it can hang more than 5 times in a day. I have no prior kamailio experience or experience debugging c applications. Kamailio version: 4.4.0 Running ps shows scscf processes are running. Service status also shows scscf is running but it does not respond to sip requests. Testing with sipsack hangs and print timeout errors. root@NGVI-PCSCF2:/var/log # service scscf status * Status of Kamailio SIP Server: * scscf is running sipsak -vv -s sip:NGVI-PCSCF2:6060 received ICMP message received ICMP message** timeout after 500 ms** Below is the extract from log files Apr 5 12:04:46 NGVI-PCSCF2 /usr/sbin/kamailio[5102]: {1 1 ACK isbc505ramjjjr3g23cs02r53cms0uxbg2es(a)10.18.5.64} WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Apr 5 12:04:46 NGVI-PCSCF2 /usr/sbin/kamailio[5104]: {1 3 REGISTER _A9LGREyW2miE7LhPJzv7g..} ERROR: ims_registrar_scscf [save.c:1034]: update_contacts(): Unable to remove contact <sip:+2347020246874@154.66.2.182:5060;rinstance=28bf37d2cce6e4a1;transport=udp> Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5131]: ERROR: rr [rr_mod.c:300]: w_record_route(): Double attempt to record-route Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5144]: ERROR: rr [rr_mod.c:300]: w_record_route(): Double attempt to record-route Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5108]: {1 1 ACK r8ug398d118i8iaaunrouggisntkigkn(a)10.18.5.64} WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5190]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 23 Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5062]: ALERT: <core> [main.c:741]: handle_sigs(): child process 5103 exited by a signal 11 Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5062]: ALERT: <core> [main.c:744]: handle_sigs(): core was generated Apr 5 12:06:12 NGVI-PCSCF2 kamailio: WARNING: <core> [daemonize.c:348]: daemonize(): pid file contains old pid, replacing pid Apr 5 12:06:13 NGVI-PCSCF2 /usr/sbin/kamailio[15060]: WARNING: interconnectroute [ported_cache.c:113]: initPortedNumberCount(): Requested records[726049]: Total allocated[871258]: Fat 20% :Total MemUsed [26 MB divided between primary[0x7f22b8667d60] and secondary[0x7f22b93b3390]] pr 5 04:38:24 NGVI-PCSCF2 /usr/sbin/kamailio[14018]: message repeated 107 times: [ ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table] Apr 5 12:50:38 NGVI-PCSCF2 /usr/sbin/kamailio[2604]: WARNING: interconnectroute [icncache_eventprocessor.c:76]: icnRefreshProcess(): total size 2147483648 qm_real_used 700707576 qm_used 387064680 1st_frag 0x7fd5f6d6a690 last_frag 0x7fd676d30fd0 Apr 5 13:00:25 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table Apr 5 13:00:26 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: message repeated 125 times: [ ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table] Apr 5 13:01:25 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: CRITICAL: <core> [main.c:650]: sig_alarm_abort(): shutdown timeout triggered, dying... Apr 5 13:02:30 NGVI-PCSCF2 kamailio: WARNING: <core> [daemonize.c:348]: daemonize(): pid file contains old pid, replacing pid Apr 5 13:02:30 NGVI-PCSCF2 /usr/sbin/kamailio[27629]: WARNING: interconnectroute [ported_cache.c:113]: initPortedNumberCount(): Requested records[726049]: Total allocated[871258]: Fat 20% :Total MemUsed [26 MB divided between primary[0x7f3093a3fd60] and secondary[0x7f309478b390]] Thank you in advance. Regards, Solly Thoobe

3 2

urn:emergency:service:sos as RURI
by a.izquierdo＠zaleos.net 05 Apr '23

05 Apr '23

Hello, I'm running a Kamailio instance with some routing logic in it. When a UA sends an INVITE with urn:emergency:service:sos as RURI, Kamailio responds with a 400 Bad Request URI. I have set debug log level and it looks like the problem is in the parse_uri.c when sanity_check("1511", "7") is called: parse_uri(): bad char ':' in state 3 parsed: <urn:emergency:service> (21) / <urn:emergency:service:sos> (25) According to RFC8141 (https://www.rfc-editor.org/rfc/rfc8141), this is a correct URN, but maybe I'm missing something. Is there a way to make Kamailio parse this RURI? Kamailio version used: 5.6.4 Thank you!

3 2

segfault during Identity verification (stirshaken)
by Yuriy Nasida 04 Apr '23

04 Apr '23

Hello, It's kamailio 5.6.4. I can say that there is no segfault in case Identity is OK but bad attempts sometimes are the reason for segfault. Not sure I can duplicate but it happens several times per day. Please look at logs. in messages: Apr 3 09:40:02 kam1 kernel: [6381565.762987] kamailio[9141]: segfault at 746c75616665 ip 00007f20c4cab6df sp 00007ffe7ec51470 error 4 in libcrypto.so.1.1[7f20c4c3b000+19e000] kamailio.log: Apr 3 09:40:02 kam1 kamailio[9141]: NOTICE: <script>: it is INVITE with Identity. Lets check Identity Apr 3 09:40:02 kam1 kamailio[9141]: CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x645f6d6574737973 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring Apr 3 09:40:02 kam1 kamailio[9182]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 53 Apr 3 09:40:02 kam1 kamailio[9100]: ALERT: <core> [main.c:774]: handle_sigs(): child process 9141 exited by a signal 11 Apr 3 09:40:02 kam1 kamailio[9100]: ALERT: <core> [main.c:777]: handle_sigs(): core was not generated Apr 3 09:40:02 kam1 kamailio[9100]: CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x645f6d6574737973 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring loadmodule "stirshaken.so" ####### STIRSHAKEN ###### modparam("stirshaken", "vs_verify_x509_cert_path", 1) modparam("stirshaken", "vs_ca_dir", "/etc/kamailio/stir_CA") #modparam("stirshaken", "vs_crl_dir", "/etc/kamailio/CRL") modparam("stirshaken", "as_default_key", "/etc/kamailio/key/private_key.pem") modparam("stirshaken", "vs_cache_certificates", 1) modparam("stirshaken", "vs_cache_dir", "/tmp/cert_cache") modparam("stirshaken", "vs_cache_expire_s", 100) if (is_method("INVITE") && is_present_hf("Identity")) { xlog("L_NOTICE", "it is INVITE with Identity. Lets check Identity\n"); if (1 == stirshaken_check_identity()) { xlog("L_NOTICE", "Shaken Identity is OK\n"); } else { xlog("L_NOTICE", "Shaken Identity is invalid\n"); } } Please give me some advice.

2 2

Handling stateless replies with kemi
by Ovidiu Sas 03 Apr '23

03 Apr '23

Hello all, How can we handle stateless replies with kemi? tm has the following param: modparam("tm", "on_sl_reply", "stateless_replies") What's the proper way to map the "stateless_reply" onreply_route to a kemi function. I'm playing with the app_jsdt module. Thanks, Ovidiu

5 7

How to check, if call is already handled by rtpengine module?
by Benoît Panizzon 02 Apr '23

02 Apr '23

Hi Running Kamailio Registrar Node + rtpengine without dialog module, only TRX module. Situation: Some customer are in ranges to which RTP can directly be routed, some are not. So rtpengine shall be engaged only when backhauling of the rtp stream is required. On a message from a CPE, this is done by checking $si and on a message to the CPE by checking $(ulc(aor=>addr)[$T_branch_idx]{uri.host}); On a BYE, rtpengine_delete() is called no matter if RTP was backhauled or not. If rtpengine does not find the call to delete, no harm happens. But now I struggle with Session Refresh Re-Invites TO the CPE as they are routed very early on and do not pass the whole location lookup stuff, therefore I have no access to the aor=>addr to determine if the destination needs backhauling or not. So the obvious solution which came to my mind is to call rtpengine_query() and see if it is finding the call and if so, call rtpengine_manage() to keep backhauling traffic on the Re-Invites refrreshing the session. Unfortunately, I found no way to check if rtpengine_query finds the call. Did I miss something? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Germany unsure about whether Kamailio will survive next winter
by Alex Balashov 01 Apr '23

01 Apr '23

For immediate release: ATLANTA, GA (1 April 2023)--The Federal Ministry for Economic Affairs and Energy of Germany was forced to disclose today that it has been tasked with assessing whether Germany will be able to operate its Kamailio through the next winter. This initiative comes amidst considerable uncertainty, shared in some other EU member states, about whether the enormous fossil fuel energy footprint of Kamailio is sustainable after the Russian invasion of Ukraine in early 2022. Pioneering Atlanta energy market analysts Evariste Systems were tapped to assist with forecasting whether German strategic natural gas reserves and liquefied natural gas (LNG) imports could keep Kamailio's enormous turbines spinning under a variety of stress-test scenarios outlined by regulators, including SIPP and SIPVicious. Alex Balashov, Evariste's principal, said in a news conference earlier this week: "Simply put, the political leadership wanted a dispassionate, technocratic kind of analysis, data-driven and the rest, free of the inflection of ideological bias and tendentious policy angles in charged German domestic politics. Whether it's the centre-right or the Greens, there are strong ideas about what to do with energy-greedy megaprojects like Kamailio." Florida-based energy turnaround vet Fred Posner, parachuted in to assist with the fast-tracked study, agreed: "It's well-understood that the German GDP is almost a perfectly linear function of gross annual SIP packets routed. The Nord Stream 1 shut-off situation in September only exposed Germany to further Kamailio-related energy vulnerabilities." At issue are the basic physics of SIP message routing in Kamailio. Due to a fixed-size worker process pool, Kamailio routing consumes about 1500 BTUs per packet-kilometre travelled. While 1500 BTUs/packet-kilometre is efficient in distance terms as compared to a typical automobile, which consumes roughly 3800 BTUs of fossil fuel energy per passenger-kilometre travelled (at typical load factors), the dizzying number of SIP packets routed through German territory in a typical business day greatly outstrips passenger-kilometres travelled. Balashov noted that sometimes, the activation energy requirement can be higher than 1500 BTUs when INVITEs with large SDPs are launched toward the next hop, though this varies with the altitude of the destination above mean sea level, wind direction, weather and other factors involved in ballistics. "It's a bit of a fool's errand to play these guessing games with averages. The energy budget can vary enormously depending on whether there is DTLS, video codecs, PASSporTs and other stuff. I find it helpful to think in kilowatt-hour terms: sometimes it's less than half a kWh, so like US $0.20 if you're counting the beans, and sometimes it's more like $0.38, we just don't know. The Bundestag always wants these big, round numbers, but if you've ever looked at the TM module, you know that's not how this works." A milder-than-expected winter, a 9% year-over-year increase in US LNG exports (to about 300 million cubic metres per day) in 2022, dependable imports from Norway and the Netherlands, and other favourable factors gave German SIP regulators a reprieve. Despite energy market volatility, premium LNG spot prices, and occasional 408 Request Timed Out scenarios, the country dodged a widely-feared Kamailio energy crisis. However, it is difficult to say whether energy market conditions will be as propitious next year. "Then there's the whole climate change goals thing. Nobody even wants to talk about that, especially for the WebRTC gateway side," says Posner. "There's a persistent fantasy that we can just power Kamailio with solar or wind if we just had enough generating capacity. I think that's really missing the forest for the trees. The worker processes stay running whether you need them or not, it all depends on how many listeners you have set and what the children config value is. Either way, I've got two words for you: base load. Even if we completely ignore that the energy density is just not there per hectare of solar or wind installation, how do you provide the constant power to the child processes?" Past feasibility studies published jointly by Balashov and Posner support this assessment. One such study, initiated in 2018 and concluded during the COVID-19 global pandemic, found that an area equivalent to the size of the entire state of North Rhine-Westphalia would need to be devoted to batteries, even using the latest lithium ferrous phosphate (LFP) technology. "But wait, there's more!" says Posner: "Show me the amperage." "The packet forwarding mechanism is similar to a railgun, at least if you're doing stateful. Where are you going to get the millions of amps? It's either setting up huge banks of capacitors god-knows-where, or the pulsed power system that's undergirded by the original OpenSER turbines. I think we already know the answer to that." Balashov noted that, even beyond the political and economic challenges that Kamailio downtime would introduce, there are other tightropes to walk. As the most economically dynamic, export-oriented EU member state, Germany is prone to flirt with SIP proxy privatisation schemes in its national discourse, all while messaging a stronger Euro to the ECB. "There is a vocal minority," says Balashov, "who push for research into more modular, regionally sited SIP gateways. They're always talking about sclerotic, unresponsive federal regulations and listless, bumbling Brussels bureaucrats and making it all smaller and leaner. Listen, I'm all for the Invisible Hand, but setting the Request URI and adding custom headers requires a truly integrated, national-scale infrastructure. It's the stuff of public-private partnerships and megaprojects. Even if you're a market zealot who is not sold on basic INVITEs as a public good, think about Presence or IMS. You really need the full capacity of a nation-state on deck for that." If devolution to smaller, more numerous SIP routing sites is not feasible--at least, at the unit cost and reliability level demanded by advanced economies--then excessive complexity offers a cautionary tale from the opposite extreme. In a particularly notorious example of technical and policy failure, the Electric Reliability Council of Texas (ERCOT), following the guidance of northern Californian advisors Unicorn & Moonshot, attempted to scale down traditional Kamailio turbine blades into newer, so-called Kubernetes "Pod" chasses to increase exhaust recapture. This made for a much more intricate installation with an exponential growth of moving parts, telemetry and site support skill requirements. This proved unmanageable, and the now-famous explosion that followed the deterioration of the site led to unprecedented casualties in US history. Furthermore, it also resulted in the long-term humidification of large expanses of East Texas, Louisiana, Mississippi and southern Alabama, rendering most of these areas uninhabitable. These events discredited the Unicorn & Moonshot approach in the eyes of most industry analysts. While it will take some months for Evariste to reach a conclusion regarding the resilience of the German Kamailio through the winter and beyond, and it will take still longer to issue policy recommendations, one factor is universally agreed upon already: this prolonged test of Germany's resilience is part of a Russian strategic calculation. A conservative MP from the CDU/CSU faction of the Bundestag, speaking on condition of anonomity, offered this summary: "The Russians think they can wait this out. They are waiting for us to fold and switch to an OpenSIPS reactor, knowing full well this will keep us busy with troubleshooting and diminish our ability to support Ukraine militarily." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 1

How to get top-most outbound via branch attribute from branched call?
by Benoît Panizzon 31 Mar '23

31 Mar '23

Hi all For rtpengine to correctly distinguish branched calls, the top-most via branch= parameter should be passed to it. I have attempted to use $via0 but that does not yet exist in Kamailio 5.4 So I did try to read $hdr(Via) which I guess returns an array and then access: but this returns the inbound via, not the one generated on the branched call. How can this be accomplished with kamailio 5.4? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 1

CANCEL timeout in branch route does not trigger any failure routes
by Richard Chan 31 Mar '23

31 Mar '23

Hi Kamailio users How can I get a failure route to trigger in timeout-to-CANCEL in a parallel forking scenario? Parallel forking test scenario 1. one(the main - 0) branch picks up the call 2. 2nd branch sends CANCEL (expecting 487 etc) - but the callee does not send any replies Result: No failure route handlers are called on timeout of this CANCEL - that is kamailio attempts the CANCEL 4 times but neither tm.t_on_failure_route, tm.t_on_branch_failure handlers are called. Any suggestions on how to resolve this? Regards Shih-Ping

3 2

Rancher, AWS, and Elastic IP
by Calvin E. 30 Mar '23

30 Mar '23

I'm looking for documentation//howto/advice/best practices for running Kamailio under Kubernetes on AWS via Rancher. Specifically, how to manage the public/Elastic IP. This is a load balancer (dispatcher) scenario with rtpengine for receiving inbound calls from a DID provider, but a similar configuration might be used for clients. I'm open to using an AWS load balancer, and would like to scale with multiple Kamailio pods behind a single public IP. I'd rather use Rancher-managed EC2 than pay extra for EKS, unless EKS can cancel out its own cost. This sounds like it should be a solved problem, just complex. I see blogs and videos out there, but it's not always obvious if their content is current. Anyone want to help me peel this onion? -Calvin

4 6

MongoDB query from Kamailio help needed
by Edo 28 Mar '23

28 Mar '23

Help is needed. I know I have seen a similar problem either via email or google search but for the life of me I cannot find it now. I am using: Ubuntu: 5.4.0-144-generic #161-Ubuntu SMP Fri Feb 3 14:49:04 UTC 2023 x86_64 x86_64 x86_64 GNU/Linux Kamailio: version: kamailio 5.3.9 (x86_64/linux) -- This is a dev platform and I can update kamailio if needed. I am trying to query a mongodb collection for the phone number stored under “did” as an integer and make a decision based on the returned result (boolean is sufficient for this decision). I don’t know what I am doing wrong here. Any help will be well appreciated ##### This is the insert statement into the telephone db from a linux script db.c$coll.insertOne({"did":parseInt($(cut -d_ -f 1 <<< $1)), "callDesc":$(cut -d_ -f 1 <<< $1)}))" >> $(cut -d_ -f 1 <<< $1).js did is stored as an integer and the callDesc is stored as presented by $rU { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 1XXXXXXXXXX, "callDesc" : 1XXXXXXXXXXXX } > db.c234.find() { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 2XXXXXXXXXXXX, "callDesc" : 2XXXXXXXXXXXX } > db.c234.find() { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 3XXXXXXXXXXXX, "callDesc" : 3XXXXXXXXXXXX } > Kamailio config: ..... ..... # --- mongodb module loaded 202201082322 loadmodule "ndb_mongodb.so" if (uri==myself) { sl_send_reply("100", "Trying to check if uri is myself $rU $ru"); #return; #Adding this to test connection to mongodb…. This worked and the reply was sent to the endpoint if(*mongodb_cmd_simple*("mongodbsrv1", "telephone", "cXXX”, "{ \"collStats\": \"cXXX\” }", "mgr1")) { sl_send_reply("183", "I got inside this message loop $rU found in mongodb"); xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } # End of the test #This is to run the actual db query to test if the phone number is in the mongodb…. I am not successful at this query even when the number is known to be in the database if(*mongodb_find_one*("mongodbsrv1", "telephone", "cXXX”, "{\"did\" : \"$rU\" }", "mongoResponse1")) { # Route to confirmed route here -- 202302212032 sl_send_reply("183", "I got inside this mongodb message loop"); route(CONFIRMED_ROUTE); xlog("response from mongodb is [[$mongodb(mongoResponse1=>value)]]\n"); } else { route(UNCONFIRMED_ROUTE); } } #append_hf("P-hint: outbound\r\n"); route(RELAY); #exit; } Please help. -- ----- Ekunwe Tel: 601.497.3932

2 2

Three year old issue with a new CVE vulnerability report being reported
by Olle E. Johansson 28 Mar '23

28 Mar '23

Hi Kamailians! A new CVE with a critical severity level was published recently for an almost three year old bug, which was also fixed and released three years ago (CVE-2020-27507). The issue was fixed in Kamailio 5.4.2 and is not present in newer releases. The Kamailio project has unfortunately not been involved in the CVE process or been informed about this old issue being published at this time. We take vulnerability handling seriously and our process is documented at: https://www.kamailio.org/wikidocs/security/policy/ The latest stable branch is 5.6, with v5.6.4 released out of it. Reference: https://cve.mitre.org/cgi-bin/cvename.cgi?name=2020-27507 Best regards and thanks for flying Kamailio! The Kamailio dev team through /Olle

4 3

json RPC syntax
by Volker Denneberg 28 Mar '23

28 Mar '23

Hello, I am trying to get json RPCs running. simpleequests seem to work, but I can't figure out how to pass parameters. For example curl --header 'Content-Type: application/json' --data-binary '{"id": 1,"jsonrpc": "2.0", "method": "ul.db_users" }' http://localhost:8081/RPC/ { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters (table to lookup)" }, "id": 1 } According to http://www.kamailio.org/docs/docbooks/3.2.x/rpc_api/rpc_api.html#rpc.gettin… parameters dont have names and according to http://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.r.db_… 1 parameter is required. I guessed adding "params":"location" => no error anymore, but still no result: curl --header 'Content-Type: application/json' --data-binary '{"id": 1,"jsonrpc": "2.0", "method": "ul.db_users", "params":"location"}' http://localhost:8081/RPC/ { "jsonrpc": "2.0", "result": { }, "id": 1 } Same result (no error) if I pass "params":"nonsense", so "params" seems to be kind of ignored ... What's the correct json syntax here or in general? Best regards, Volker

5 6

KEMI: app_jsdt_run_ex(): js loading state not initialized (call: ksr_htable_event)
by Ovidiu Sas 27 Mar '23

27 Mar '23

Hello all, While using KEMI with app_jsdt I noticed the following possible regression between 5.5 and 5.6. The following config is working ok on 5.5 but throws an error on 5.6: ERROR: app_jsdt [app_jsdt_api.c:555]: app_jsdt_run_ex(): js loading state not initialized (call: ksr_htable_event) kamailio.cfg: debug=2 children=1 log_facility=LOG_LOCAL0 log_name="k" mpath="/home/osas/src/kamailio/src/modules" loadmodule "htable/htable.so" modparam("htable", "event_callback", "ksr_htable_event") modparam("htable", "event_callback_mode", 1) loadmodule "kemix/kemix.so" loadmodule "app_jsdt/app_jsdt.so" modparam("app_jsdt", "load", "/usr/local/etc/kamailio/kamailio.js") cfgengine "jsdt" kamailio.js: function ksr_htable_event(evname) { KSR.info("ksr_htable_event(" + evname + ")\n"); } function ksr_request_route() { KSR.x.exit(); } If we load the htable module after app_jsdt, then the error is gone. Is there something new during init in 5.6 that requires specific module loading order? -ovidiu

2 2

Handling branch-level append_hf()
by Alex Balashov 27 Mar '23

27 Mar '23

Hi, I’ve got a situation where I catch a 302 redirect in a failure_route (branch #1), and use the Contact to set a custom header (append_hf()). This custom header is utilised on the subsequent branch (branch #2), but if that branch fails (e.g. due to 4xx-5xx response), the custom header is not preserved on subsequent branches (branch #3+). What’s the best way to preserve it across subsequent branches? The documentation for msg_apply_changes() says it cannot be used once the transaction is already created, and in any case, I have other branch-level changes in branch #2 that I _would_ like to revert on subsequent branch attempts (i.e. branch #3). I can certainly devise a transaction-persistent vehicle for this value which can be checked on every branch, but I wondered if there’s an easier and more best-practical way. Thanks! — Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 1

No voice for webrtc call
by Arun K R 25 Mar '23

25 Mar '23

Hello, I am using kamailio to register the webrtc client and routes the calls to asterisk using pjsip. I am having no voice on both ends. Logs are attached witht this email. Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'offer' from 10.13.1.127:59841 Mar 24 13:06:58 debian rtpengine[7345]: NOTICE: [8asmubtpv46ac7nrfk57]: [core] Creating new call Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'offer' from 10.13.1.127:59841 (elapsed time 0.016067 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'answer' from 10.13.1.127:52045 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'answer' from 10.13.1.127:52045 (elapsed time 0.000997 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: peer for component 1 is 10.13.1.170:15634 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 24 13:06:58 debian rtpengine[7345]: ERR: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'answer' from 10.13.1.127:52045 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'answer' from 10.13.1.127:52045 (elapsed time 0.000786 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS: Peer certificate accepted Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: peer for component 1 is 10.13.1.170:15634 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 24 13:07:02 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Confirmed peer address as 10.13.1.170:15634 Mar 24 13:07:02 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Kernelizing media stream: 10.13.1.170:15634 -> 10.13.1.127:10940 | 10.13.1.127:10960 -> 192.168.30.200:53489 Mar 24 13:07:02 debian rtpengine[7345]: WARNING: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] No support for kernel packet forwarding available (encryption cipher or HMAC not supported by kernel module) Mar 24 13:07:02 debian rtpengine[7345]: NOTICE: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Setting 'non-forwarding' flag for kernel stream due to lack of sinks Mar 24 13:07:03 debian rtpengine[7345]: ERR: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [rtcp] SRTCP output wanted, but no crypto suite was negotiated Mar 24 13:07:03 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Removing media stream from kernel: local 10.13.1.127:10940 Mar 24 13:07:03 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Kernelizing media stream: 10.13.1.170:15634 -> 10.13.1.127:10940 | 10.13.1.127:10960 -> 192.168.30.200:53489 Mar 24 13:07:03 debian rtpengine[7345]: WARNING: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] No support for kernel packet forwarding available (encryption cipher or HMAC not supported by kernel module) Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'delete' from 10.13.1.127:52045 Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of call branch '9693uq66f0' (via-branch '') in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of call branch '223df8fa-8106-48af-844f-33f97530a0d8' (via-branch '') in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of entire call in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'delete' from 10.13.1.127:52045 (elapsed time 0.001463 sec) Mar 24 13:07:17 debian /usr/local/sbin/kamailio[13362]: INFO: <script>: 10.13.1.170 authentication bypassed for sip:developer@10.13.1.127 Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Final packet stats: Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- Tag '9693uq66f0', created 0:36 ago for branch '' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscribed to '223df8fa-8106-48af-844f-33f97530a0d8' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscription for '223df8fa-8106-48af-844f-33f97530a0d8' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --------- Port 10.13.1.127:10960 <> 192.168.30.200:53489, SSRC 0, in 0 p, 0 b, 292 e, 36 ts, out 0 p, 0 b, 0 e Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- Tag '223df8fa-8106-48af-844f-33f97530a0d8', created 0:36 ago for branch '' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscribed to '9693uq66f0' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscription for '9693uq66f0' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using opus/48000/2 Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --------- Port 10.13.1.127:10940 <> 10.13.1.170:15634, SSRC 1ada0b2b, in 292 p, 40137 b, 1 e, 30 ts, out 2 p, 1449 b, 0 e

2 3

different sip response on database (mysql) timeout
by Leon de Rooij 24 Mar '23

24 Mar '23

Hi, I need to use a database that may sometimes timeout and I'd like to give a conditional sip response based on whether there was a timeout or not. I'm using sqlops with mysql modules. Does anyone know if that's possible ? Thanks, Leon

3 3

Route: comma separated HOP list
by Benoît Panizzon 24 Mar '23

24 Mar '23

Hi List Cisco-SPA112 does reply to an invite containing Record-Route header with one Route: Header containing a comma separated list of those hops. It looks like kamailio fails to parse them and correctly route for example an ACK to a 200 OK. Is this a known issue? Is there a solution? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

WebRTC "client did not present a certificate" error
by David Cunningham 23 Mar '23

23 Mar '23

Hello, We have a Kamailio 5.2.7 server with WebRTC enabled. However, a WebRTC client at https://tryit.jssip.net/ is unable to connect on either Chrome or Firefox. In the Kamailio log we see the lines below. In tls.cfg we have "verify_certificate = no" and "require_certificate = no" for both [server:default] and [client:default]. Would anyone be able to help us with this? Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:1159]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x14baf1cbb090: (nil) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:741]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:741]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:424]: tls_accept(): TLS accept successful Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from xx.xx.xx.xx:39816 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: yy.yy.yy.yy:8443 Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:1189]: tls_read_f(): Reading on a renegotiation of connection (n:532) (0) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1527]: tcp_read_req(): EOF Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x562ffde66d00, 17, -1, 0x10) fd_no=4 called Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1680]: release_tcpconn(): releasing con 0x14baf4cc1ec8, state -1, fd=17, id=665 ([xx.xx.xx.xx]:39816 -> [xx.xx.xx.xx]:8443) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x14baf289ea30 Mar 22 01:25:28 br /sbin/kamailio[25291]: DEBUG: <core> [core/tcp_main.c:3320]: handle_tcp_child(): reader response= 14baf4cc1ec8, -1 from 1 Mar 22 01:25:28 br /sbin/kamailio[25291]: DEBUG: tls [tls_server.c:683]: tls_h_close(): Closing SSL connection 0x14baf289ea30 Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 7

How to require session timer and set a max session timer value?
by Benoit Panizzon 23 Mar '23

23 Mar '23

Hi All I would like to avoid having run away calls as good as possible. For this, I would like to require timers and set the maximum expires timer to say 1800 I had a look at the sst module, but I see no way to require timers or to set a max timers value. Do my intentions make sense? How did others accomplish this task? Thanks for any hints! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

With TOPOH module enabled, transferred call hangup from callee side does not hang up the call
by Arsenijs Kabihno 22 Mar '23

22 Mar '23

Hello, we have faced an issue which appears when TOPOH module is enabled, call is made from WebRTC client, then transferred by callee. We have WebRTC client behind load balancer. We make call to softphone from WebRTC, call from softphone is transferred to external number using Asterisk. When call is hanged up on external number side, it doesn't hangs up on WebRTC client side. If TOPOH is not enabled, this issue doesn't occur. We have the following TOPOH-related sections of kamailio.cfg: <...> #!ifdef WITH_TOPOH loadmodule "topoh.so" #!endif <...> <...> #!ifdef WITH_TOPOH modparam("topoh", "mask_key", "xxxxxxxxxxxxxxxx") modparam("topoh", "mask_ip", "10.0.0.1") modparam("topoh", "mask_callid", 0) modparam("topoh", "sanity_checks", 1) modparam("topoh", "uparam_prefix", "") modparam("topoh", "vparam_prefix", "") modparam("topoh", "callid_prefix", "1337") #!endif <...> <...> #!ifdef WITH_DIALOG modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 800) modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_timer", 30) modparam("dialog", "send_bye", 0) modparam("dialog", "dlg_filter_mode", 1) #!ifdef WITH_TOPOH modparam("dialog", "lreq_callee_headers", "TH: dlh\r\n") #!endif #!endif <...> <...> #!ifdef WITH_DISPATCHER event_route[dispatcher:dst-up] { sql_query("ds_dburl", "select description from dispatcher where destination='$ru'", "ds_dbresult"); xlog("L_WARN", "Destination up: $ru, found name: $dbr(ds_dbresult=>[0,0])\n"); prom_gauge_set("trunk_reachability", "1", "$ru", "$dbr(ds_dbresult=>[0,0])"); } event_route[dispatcher:dst-down] { sql_query("ds_dburl", "select description from dispatcher where destination='$ru'", "ds_dbresult"); xlog("L_WARN", "Destination down: $ru, found name: $dbr(ds_dbresult=>[0,0])\n"); prom_gauge_set("trunk_reachability", "0", "$ru", "$dbr(ds_dbresult=>[0,0])"); } #!endif #!ifdef WITH_TOPOH event_route[topoh:msg-sending] { if(is_reply() and $rs=="302") { drop; } } #!endif (can provide the full config, if needed) We are receiving the following errors: On transfer: https://pastebin.com/H3FQ4Kpj On hangup from external number side: https://pastebin.com/wgfRzGbd On hangup from WebRTC client side: https://pastebin.com/sE0yhBae There are also differences in SIP traffic when TOPOH is disabled and when it is enabled. With TOPOH enabled, apparently, ACK from Asterisk is not reaching kamailio: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> (SIP flows attached - topoh_on.csv, topoh_off.csv) kamailio -v version: kamailio 5.6.1 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:11:15 Nov 15 2022 with gcc 9.3.0 We have dockerized kamailio running on Ubuntu 16.04.3 LTS Docker version 18.03.1-ce, build 9ee9f40 Please advise where to start looking.

1 0

OT: path support fix for asterisk
by Henning Westerholt 21 Mar '23

21 Mar '23

Hello, It's off-topic, but it was several times discussed here on the list. It seems that the upcoming 18.17 and 20.2 asterisk version seems to properly support path headers. From the announcement: * [ASTERISK-30100<https://issues.asterisk.org/jira/browse/ASTERISK-30100>] - res_pjsip: Path is ignored on INVITE to endpoint (Reported by Yury Kirsanov) This bug has been many years open. have not tried it yet, but it looks promising. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

7 8

Follow-up: Printing Global Parameter Values & Logging JSONRPC Requests over UDP
by sadik.oualla.mohamed＠gmail.com 18 Mar '23

18 Mar '23

Hello everyone, I am following up on my previous email regarding printing Global Parameter values and logging JSONRPC requests over UDP in Kamailio. I understand that everyone might be busy, but I would greatly appreciate any guidance or suggestions you can provide on these topics. To recap my questions: 1)Is there a way to print the Global Parameter values, such as debug, children, flags, etc., within the Kamailio configuration file, to ensure they have been changed? 2°How can I log in Kamailio that it has received a JSONRPC request over UDP? I have been successful in logging this information when using HTTP and JSONRPC but am unsure how to do it with UDP. Any input or advice would be truly valuable. Thank you in advance for your time and assistance. Best.

1 0

Issues/Vulnerabilities in OpenSipS that may affect Kamailio
by Dovid Bender 17 Mar '23

17 Mar '23

Hi All, OpenSipS just released an update to the audit that was done to OpenSips [1]. From my basic coding skills it seems like the changes that were done by the OpenSipS project were not implemented in Kamailio which means that Kamailio is potentially vulnerable? For example you can compare the changes made by OpenSips project here [2] and the Kamailio code here [3] I am not active much on the list so please don't roast me if I am completely wrong here. Regards, Dovid [1] http://lists.opensips.org/pipermail/users/2023-March/046849.html [2] https://github.com/OpenSIPS/opensips/commit/dd9141b6f67d7df4072f3430f628d4b… [3] https://github.com/kamailio/kamailio/blob/master/src/core/parser/digest/par…

4 4

RPC error: 500 - Failed to use table
by Volker Denneberg 17 Mar '23

17 Mar '23

Hi, what can be the reason why usrloc module sometimes can read from database, sometimes not? Normal operation looks ok, the location table is kept up to date. The problem seems to be related with RPC (kamcmd or jsonrpc). For example, kamcmd ul.dump # ok kamcmd ul.lookup location 6351*1 # ok kamcmd ul.db_users location # error: 500 - Failed to use table Error 500 correlates with syslog entry: ERROR: <core> [db.c:481]: db_use_table(): invalid connection parameter Who connects where? In my understanding usrloc via core to mysql. I suppose RPC calls do not fail if usrloc can operate on kamailio memory, does that make sense? If so, what usrloc parameter can I tweak to fix error 500 ? I don't find any usrloc parameter regarding db connection. (Opening another mysql session as user kamailio is no problem, so we re not running out of connections.) Thanks in advance! Volker [cid:fb71bb9b-ad4f-4110-824b-d6805406bf6a] Volker Denneberg Development Telefon: +49 241 980 986 41 Web: www.m3connect.de<https://mx.scientific.de/owa/redir.aspx?C=RW1TSCipXQC6K0Yw1m69CGqyTwJHXjiiN…> E-Mail: v.denneberg(a)m3connect.de<https://mx.scientific.de/owa/redir.aspx?C=BjBXIUrD2WCryLRZ2IrAAwEu0b0W0ciUY…> m3connect GmbH | Pascalstr.18 | 52076 Aachen | DE/Germany | Amtsgericht Aachen | HRB 8773 | USt.- ID: DE219 664 658 | Geschäftsführer/CEO: Emilio Dragas

2 1

Sync NONCE between registrars?
by Benoit Panizzon 16 Mar '23

16 Mar '23

Hi We have two registrars. Today I observed a CPE doing this: CPE REGISTER (no auth) => Registrar 1 => 407 Challenge NONCE A CPE REGISTER (auth to NONCE A => Registrar 2 => 407 Challenge NONCE B CPE REGISTER (auth to NONCE B => Registrar 1 => 407 Challenge NONCE A and so on. Time on both registrars is in sync (I tought NONCE was based on time) Is there a way to sync the NONCE so that both would accept authentication based on the same NONCE? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

How to Print Global Parameter Values and Log JSONRPC Requests Received over UDP in Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Dear everyone, I am wondering if there is a way to print the Global Parameter values such as debug, children, flags, etc. just to make sure that they have been changed. I am aware that I can use the `kamcmd cfg.get core debug` command to check the value of the debug parameter for example, but I am wondering if there is a way to do this in the configuration file of Kamailio. Additionally, I would like to know if there is a way to print in the log of Kamailio that it received a JSONRPC request over UDP. I have been able to print some logs when using HTTP and JSONRPC to know that the server has received a JSONRPC request, but I am unsure how to do this when using UDP. I would greatly appreciate it if someone could provide some guidance. Thank you in advance for your time and assistance. Best regards.

1 0

Increment in number of "200" responses while calling procedures over jsonrpc with http
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Hello, Thank you very much! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the procedures over jsonrpc with http. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which iscausing Kamailio to treat it as a "200" http responses code as "200" SIP codes. Could you advise on how to avoid this issue? Is there any further configuration that needs to be done? Best regards.

3 8

Kamailio store call statistics
by georgy.grigoryev＠snapcom.fr 16 Mar '23

16 Mar '23

Hello everyone, I'm trying to store the call statistics in my Kamailio server. I'm using SQLite just to understand what type of information i can store. I'm using the dialog module with the classic parameters: enable_stats => 1 dlg_flag => 4 hash_size => 4096 db_url => sqlite:///usr/local/etc/kamailio/kamailio.sqlite db_mode => 1 I'm able to open the DB but everything is empty, am i using the wrong modules? or i didn't configure the server well? The goal is to store CALL CDR in the DB, so that i can see who called & how many times etc

2 1

webrtc audio issue
by Arun K R 15 Mar '23

15 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but there is no sound for webrtc from public internet . Also i am getting warning (SRTCP /RTP output wanted, but no crypto suite was negotiated) webrtc from Local network works fine with VPN For normal udp call without webrtc works fine. Kamailio having two interfaces, interface1 is private 10.13.1.140 and interface 2 is publi ip 100.x.x.x webrtc client sends calls to the public ip interface 100.x.x.x and kamaiio routes the call to the asterisk via private interface 10.13.1.140. asterisk server sends the call to remote server and gets the rtp back from that Asterisk server ---converts RTP to SRTP and forward to ----> kamailio 10.13.1.140 but there is nothing happens after that, please help me on this i am new to kamailio and rtpengine. The issue is only for webrtc from public internet. But when using webrtc from LAN works fine below are the logs Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: from domain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: 09496381412 has fromDomain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: Request going to srtp customied arun Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [control] Adding a proxy for control ng stats:127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'offer' from 127.0.0.1:53374 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'offer' from 127.0.0.1:53374: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=group:BUNDLE 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:9 IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:NM2h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:BF2WmB5sOZZWVIFppZ3YTiSH Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 9F:FA:18:B0:8D:17:A5:14:FE:22:BA:1E:73:1E:3E:02:6F:17:EC:B5:FC:77:BD:E1:0B:51:DF:7F:61:B0:2F:55 Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "223.190.206.123" ], "from-tag": "7n38ig1acl", "command": "offer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: NOTICE: [ui7jc53b9lfbt9lf4nu7]: [core] Creating new call Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to '' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for offerer 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30020 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30021 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30000] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [1] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'offer' from 127.0.0.1:53374 (elapsed time 0.022450 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'offer' to 127.0.0.1:53374: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30000 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30001 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:1 AEAD_AES_256_GCM inline:czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDJWi/0EDtIq16vYCOM Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:2 AEAD_AES_128_GCM inline:5abyoU5PKLg1lNfmLg1aBLIqyNAgmKNEEo8J1w Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:i4O28fJjCyvBVQmwjCb7dySy5K20yeg5n1ky+uhPNSpbqkv8vxkcIaerkO9Gbg Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:OOydiBrJ9nMu23yaBP7JuYNddpbXX8sN/V9Ab/psnTZIBRZKXenHWVPvJSM9Fw Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:5 AES_192_CM_HMAC_SHA1_80 inline:UtkHFdpsaZSO0nwvmEotIoGN+paLt3LHt7yMG4MfDv5rueBsb80 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:6 AES_192_CM_HMAC_SHA1_32 inline:hgFJ9vS9Roz1vAMhnrCyviKC/TwCPgrk5jeKbxHlIEKtqSvLdK4 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:07Q9Z/8DguM0HJgDHv5HF3YDWHdp23hbz+TepHa8 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:djpscuSL/05sVtFNocQpHmr1x+NTcLXKu8ePb0q+ Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:9 F8_128_HMAC_SHA1_80 inline:5FqC2VY+q3vqaju3JwroIJO/OlCM1i6TNRE/3Atj Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:10 F8_128_HMAC_SHA1_32 inline:kUfTs7OFD6ltRNpDi1dSG5ZjuQAFDeEQD5IvOwme Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:11 NULL_HMAC_SHA1_80 inline:JZ2ruS0914NL4mmS7jxg+ngz0IQaE6FBRlet8abd Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:12 NULL_HMAC_SHA1_32 inline:h21D+oGRkr1dDEoMm1+vEKmKsn+4GxNBwgIew2qA Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:8100c85eb1c58be3580e8acd7f097000 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:hcqW22Sc Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:QjHJKT4qNsfZm80zrcyh8SfI76 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30000 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 2 UDP 2130706430 10.13.1.127 30001 typ host Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 between 10.13.1.127 and 10.13.1.152:14450, type prflx Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Triggering check for GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30000 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30020 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30001 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30021 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing '7n38ig1acl' from 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' from '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30000 remote 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30001 remote 10.13.1.152:14451 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Replacing previously learned prflx ICE candidate with Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Learning new ICE candidate H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:H6719a439:1 between 10.13.1.127 and 103.25.164.57:14450, type host Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Recalculating all ICE pair priorities Mar 9 20:21:02 debian rtpengine[11823]: WARNING: [ui7jc53b9lfbt9lf4nu7]: [ice] Priority collision between candidate pairs GtpEjl9r4IzNpiXr:H6719a439:1 and GtpEjl9r4IzNpiXr:Ha0d0198:1 - ICE will likely fail Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Sending ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.003226 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as succeeded Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Best succeeded ICE pair with all components is GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE not completed yet, but can use pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS: Peer certificate accepted Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS handshake successful Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Start nominating ICE pairs Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Nominating ICE pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Sending nominating ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting nominated ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as valid Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE completed, using pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate Ha0d0198 in favour of Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate H6719a439 in favour of H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.002244 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New ingress SSRC for: 10.13.1.152:14450 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31783 TS 528 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 00, length 16, result 6806f5cc... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 01, length 20, result 044b0cce... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 02, length 14, result 4426c95d... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New egress (direct) SSRC for: 0.0.0.0:9 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: ERR: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Error when sending message. Error: Inappropriate ioctl for device Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31784 TS 1488 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31785 TS 2448 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31786 TS 3408 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected:

2 1

Dispatcher behavior
by Jawaid Bazyar 15 Mar '23

15 Mar '23

Hi, is this expected behavior? I have two target hosts in a dispatcher set, 10.20.20.118 (prio 5) and 10.20.20.252 (prio 10). When I have 10.20.20.252 turned off, dispatcher still tries to send packets to that host, even when the status flags are IP. root@vs-rs01:/etc/kamailio# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:10.20.20.252:5060 FLAGS: IP PRIORITY: 10 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 62 } } DEST: { URI: sip:10.20.20.118:5060 FLAGS: AP PRIORITY: 5 LATENCY: { AVG: 3.876000 STD: 2.965000 EST: 5.058000 MAX: 14 TIMEOUT: 0 } } } } } }

5 14

Error on transfer when calling from WebRTC with TOPOH
by Arsenijs Kabihno 15 Mar '23

15 Mar '23

Hello, We have an issue which appears when TOPOH module is enabled, call is made from WebRTC, then transferred by calee: Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) INFO: {2 2 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK66807520, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) NOTICE: {2 2 INVITE g2rgvnioadg8m7l6i7k6} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881327;method=INVITE;from_tag=p9qqi7hrua;to_tag=ea7d2850-3e76-4c11-821c-a0d1e3cbb011;call_id=g2rgvnioadg8m7l6i7k6;code=200;reason=OK;src_user=92010;src_domain=10.181.0.1;src_ip=10.5.0.145;dst_ouser=27221;dst_user=27221;dst_domain=10.181.0.1 Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 31(37) INFO: {1 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} <script>: RTPENGINE branch_id:0 ruri: sip:97093@10.128.150.67:46058;rinstance=d403efb922420649;transport=UDP, method:BYE, status:<null>, extra_id: <null>, rtpengine_delete Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 28(34) NOTICE: {2 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881338;method=BYE;from_tag=0c29be45-6aff-4d4e-8503-650893a5a5ac;to_tag=b1a8a401;call_id=e034a187-0e4d-4c4e-ab4b-f2e5f1281330;code=200;reason=OK;src_user=22628;src_domain=10.5.2.192;src_ip=10.5.2.192;dst_ouser=97093;dst_user=97093;dst_domain=10.128.150.67 Mar 15 11:55:39 ap07stvp.xxxxxxx.net docker[2062]: 32(38) INFO: {1 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: sip:jgqtuev6@2idi5cucan2v.invalid;transport=ws;ob, method:INVITE, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address direction=priv1 direction=pub1 rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 27(33) INFO: {1 4728 INVITE 133ccd4c-031b-4d10-b073-745f35433132} <script>: RTPENGINE branch_id:0 ruri: sip:0037126048570@193.138.189.220, method:INVITE, status:<null>, extra_id: z9hG4bKPjedcf2d96-01cd-43fd-bed0-4e601e3e542f0, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 via-branch=extra Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 50(56) INFO: {2 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bKPj1cd87bda-7862-4ca9-8475-ac6b342c45ef0, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address via-branch=extra rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/resolve.c:1730]: sip_hostport2su(): could not resolve hostname: "2idi5cucan2v.invalid" Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/forward.c:516]: forward_request(): bad host name 2idi5cucan2v.invalid, dropping packet Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) We have the following setup: WebRTC client behind load-balanser -> kamailio -> asterisk. When call is transferred, asterisk makes re-invite. If TOPOH is disabled, it happens this way and issue is not present: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> WebRTC ACK example: 03/09/2023 09:35:48.126 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:98dn3met@0j62ln34srjn.invalid;transport=ws;ob;alias=10.5.0.145~27562~6 SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPj1d11dae2-10bf-466e-9221-5710f801d725 From: <sip:27221@10.181.0.1>;tag=b6b73135-efe1-4d86-bbc8-bf4313130eb5 To: <sip:92010@10.181.0.1>;tag=namgu127j9 Call-ID: t0hmmk50h5i4ok88asjj CSeq: 28632 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Route: <sip:10.181.0.1:4444;transport=ws;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 If TOPOH is enabled, kamailio doesn't pass this ACK to WebRTC: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> ACK example: 03/09/2023 09:33:03.544 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:10.0.0.1;line=Bn7Ipzh8eb8DxMSFyk.rWzitxzVFBoirxRG5xYeOxb7hpoXTlMGDBbuTCk3oBDAqlH** SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPjf00fc541-f011-4e52-b63a-2c569478ca39 From: <sip:27221@10.181.0.1>;tag=7d402872-7355-4269-b912-e52ee28ed415 To: <sip:92010@10.181.0.1>;tag=t0q4g9am9g Call-ID: t0hmmetickvlvn5k7nco CSeq: 25565 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=t0q4g9am9g;did=4c8.a182;rtp=bridge;rtp=ws> Route: <sip:10.0.0.1;line=Bn7IpzdIwzd8Vf8Iwzd9WkyFWkAFBELJBo.qBYyuCoVgBz1uxn8gx61uxn8geYXOeD3FV6dFeD7Oxj7YpnX5ekFFlDHJljd8VzATC60ulYi5ebC7poiFBk3oBI**> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 We have an assumption that issue might be related to the way how TOPOH handles alias in contact. Did someone have encountered something like this?

1 0

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Adil Mafzool Zaka 15 Mar '23

15 Mar '23

Hi I am using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address >> Kamailio AWS NATed address >> FreeSwitch 1 1X.2X.2X.2X >> 3X.1XX.1XX.2XX 10.0.1.4 >> 10.0.4.23 10.0.1.4 >> 10.0.4.23 Many thanks Regards Muhammad Zaka

2 1

Use Kamailio and RTPEngine to play wav file as early media to announce a server maintenance
by jhayhoe＠liquidweb.com 15 Mar '23

15 Mar '23

I am looking to configure Kamailio and RTPEngine to play a wav file for any new calls during a scheduled asterisk server maintenance. I have set up the following within route{} to send the call to a early_media route based on a variable that I can set on the server using kamcmd. route { ... if ($sht(config=>maint)=="1") { if (is_method("INVITE") && !has_totag()) { xlog ("L_INFO","Ongoing maintenance. Call-ID:$ci | Method: $rm"); route(early_media); } } ... } route[early_media] { xlog("Maintenance: handling early_media\n"); rtpengine_manage("direction=external direction=external"); $var(body) = $(rb{re.subst,/(IP4.).*/\1##server IP redacted##/g}); set_reply_body($var(body), "application/sdp"); t_send_reply("183", "Session Progress"); sleep(2); unblock_dtmf("all"); play_media("file=/etc/kamailio/media/maintenance.wav"); sleep(21); rtpengine_delete(); t_reply("480","Temporarily Unavailable"); } This mostly works as I hear the recorded message in place of the ringing tone when a call is processed by kamailio and then a busy signal is heard as part of the 480 reply at the end. The sleep's are in place to match the duration of the recorded message as otherwise the 480 is sent immediately following the start of the play_media. Feel free to let me know if there is a better solution to wait the duration of the media file and then proceed with processing instead of using sleep(). Is there a way to instead send a CANCEL or a BYE instead of a 480 after the wav file is played? I tried using t_cancel_callid("$ci", "$cs", "0") as well as a dlg_bye("all") at the end of the early_media route and neither seemed to send any headers out.

3 3

stuck with webrtc on kamailio
by Arun K R 14 Mar '23

14 Mar '23

hello, I have issue of no voice on kamailio configured for webrtc. I have configured rtpengine with public and private interface. My kamilio has two interface of pubic and private ip. Somebody has suggested me that there is no ice trickling happens. I am using jssip webrtc client. Kindly suggest me or help me with this. I have send another email for same issue to this group and didnt get any reply. Kindly help me

1 0

Re: Your message to sr-users@lists.kamailio.org awaits moderator approval
by Arun K R 14 Mar '23

14 Mar '23

I need a help on issue with rtpengine configuration and I didn't get any repy for it. Kindly check about it and help me. If i won't get any help kindly let me know, so I can stop waiting for your reply On Mon, Mar 13, 2023 at 12:50 PM <sr-users-bounces(a)lists.kamailio.org> wrote: > Your mail to 'sr-users(a)lists.kamailio.org' with the subject > > webrtc audio issue > > Is being held until the list moderator can review it for approval. > > The message is being held because: > > The message is larger than the 150 KB maximum size > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. >

1 0

send messages from source ip in Linux anyIP list
by mohsen khashei 14 Mar '23

14 Mar '23

1 0

Dispatcher Incoming
by nutxase 14 Mar '23

14 Mar '23

Hi All I have setup kamailio using dispatcher to proxy registrations from the UAC to asterisk but when asterisk sends an incoming call it does not seem to keep the path header and therefore kamailio sends 404 is there anyway around this? Thanks Sent with [Proton Mail](https://proton.me/) secure email.

5 18

Use MySQL connector instead of mariadb connector
by suchendra adiga 14 Mar '23

14 Mar '23

Hi, I have been working on setting up the database with the Kamailio server. But facing a problem when my db user has configured with SSL required. But I learnt that there is no support to enable the SSL in mariadb connector. So I am thinking of using the MySQL connector instead. Is there anyway I could configure the kamailio to use MySQL instead of Mariadb? Thanks and regards, Suchendra

2 7

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Muhammad Zaka 13 Mar '23

13 Mar '23

Hi We are using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address --> Kamailio AWS NATed address --> FreeSwitch 1 1X.2X.2X.2X --> 3X.1XX.1XX.2XX 10.0.1.4 --> 10.0.4.23 10.0.1.4 --> 10.0.4.23 [cid:image001.png@01D955CE.9F0C2EE0] Many Thanks Regards Muhammad Zaka Muhammad Zaka UCaaS Solutions Engineer Tel: 03330147908 Email: Muhammad.Zaka(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

1 0

(no subject)
by gui 10 Mar '23

10 Mar '23

Hello I have an issue with Rx in my VolTE call Flow. Please refer to picture and pcap in annex. I am using same rtp engine at origination and termination. I am sending rtpmanage for invite and 183 session progress at both origination and termination. My issue is that Rx is not sending for the proper media flow component to PCRF. At origination it is considering a media flow between A 50034 and RTPEngine 49186 (instead of considering RTP Engine Port 49152) At Termination it is considering a media flow between B 50054 and RTPEngine 49132 (instead of considering RTP EnginePort 49170) Any idea why?

3 4

use_dns_failover and dispatcher
by Lars Jansson 10 Mar '23

10 Mar '23

Hi, I had the intention to route calls to some SRV destinations via dispatcher and to other SRV destinations via DNS lookup and 'use_dns_failover=on'. Currently in a route block using dispatcher two out of three targets within the SRV destination are unavailable (by intention). During my tests to the dispatcher destination I have found that: A) If I have 'use_dns_failover=on' this setting "overrules" the dispatcher routing and the unavailable targets are also tried before DNS failover finds the available target. B) If I have 'use_dns_failover=off' the dispatcher routing logic works as expected and the call is always routed to the only available target directly, making use of the result of the dispatcher OPTIONS pinging. -- So my conclusion is now that you have to choose: Either use dispatcher routing, or use DNS failover based routing. - I'm fine with this, but I just would like to know if my conclusion is correct, or if there is actually a way to use dispacther for some SRV destinations and DNS failover for other SRV destinations. Regards, Lars

1 0

New 'reverse' transaction not using correct transport protocol
by Benoit Panizzon 10 Mar '23

10 Mar '23

Hi CPE registering with TCP (or TLS). 2nd leg is UDP. Within one transaction (INVITE to OK including all ACK and PRACK) transport is kept as desired for both legs. But when the B side disconnects (sending BYE, new reverse transaction), this is sent via UDP to the A CPE which initially talked TCP and thus ignored. record_route() is called on each leg. What could I be missing? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 6

Silo extra_hdrs duplication
by Dr. Barabás Péter 10 Mar '23

10 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

3 2

Re: [sr-dev] KEMI python 'KSR' has no attribute 'sanity'
by Henning Westerholt 10 Mar '23

10 Mar '23

(adding sr-users) Hello, maybe because you're spelling it wrong? Compare e.g., to here: https://kamailio.org/docs/tutorials/5.6.x/kamailio-kemi-framework/modules/#… Cheers, Henning -----Original Message----- From: agural5(a)gmail.com <agural5(a)gmail.com> Sent: Freitag, 10. März 2023 08:32 To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] KEMI python 'KSR' has no attribute 'sanity' Hello, I am trying to use KEMI KSR.sanity in python but got error: AttributeError: module 'KSR' has no attribute 'sanity' appreciate your help for solve this. Thanks! _______________________________________________ Kamailio (SER) - Development Mailing List To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org

2 1

webrtc from public internet have no voice
by mail4arunkr＠gmail.com 10 Mar '23

10 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but there is no sound for webrtc from public internet . Also i am getting warning (SRTCP /RTP output wanted, but no crypto suite was negotiated) webrtc from Local network works fine with VPN For normal udp call without webrtc works fine. Kamailio having two interfaces, interface1 is private 10.13.1.140 and interface 2 is publi ip 100.x.x.x webrtc client sends calls to the public ip interface 100.x.x.x and kamaiio routes the call to the asterisk via private interface 10.13.1.140. asterisk server sends the call to remote server and gets the rtp back from that Asterisk server ---converts RTP to SRTP and forward to ----> kamailio 10.13.1.140 but there is nothing happens after that, please help me on this i am new to kamailio and rtpengine. The issue is only for webrtc from public internet. But when using webrtc from LAN works fine below are the logs Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: from domain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: 09496381412 has fromDomain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: Request going to srtp customied arun Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [control] Adding a proxy for control ng stats:127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'offer' from 127.0.0.1:53374 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'offer' from 127.0.0.1:53374: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=group:BUNDLE 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:9 IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:NM2h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:BF2WmB5sOZZWVIFppZ3YTiSH Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 9F:FA:18:B0:8D:17:A5:14:FE:22:BA:1E:73:1E:3E:02:6F:17:EC:B5:FC:77:BD:E1:0B:51:DF:7F:61:B0:2F:55 Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "223.190.206.123" ], "from-tag": "7n38ig1acl", "command": "offer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: NOTICE: [ui7jc53b9lfbt9lf4nu7]: [core] Creating new call Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to '' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for offerer 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30020 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30021 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30000] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [1] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'offer' from 127.0.0.1:53374 (elapsed time 0.022450 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'offer' to 127.0.0.1:53374: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30000 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30001 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:1 AEAD_AES_256_GCM inline:czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDJWi/0EDtIq16vYCOM Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:2 AEAD_AES_128_GCM inline:5abyoU5PKLg1lNfmLg1aBLIqyNAgmKNEEo8J1w Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:i4O28fJjCyvBVQmwjCb7dySy5K20yeg5n1ky+uhPNSpbqkv8vxkcIaerkO9Gbg Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:OOydiBrJ9nMu23yaBP7JuYNddpbXX8sN/V9Ab/psnTZIBRZKXenHWVPvJSM9Fw Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:5 AES_192_CM_HMAC_SHA1_80 inline:UtkHFdpsaZSO0nwvmEotIoGN+paLt3LHt7yMG4MfDv5rueBsb80 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:6 AES_192_CM_HMAC_SHA1_32 inline:hgFJ9vS9Roz1vAMhnrCyviKC/TwCPgrk5jeKbxHlIEKtqSvLdK4 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:07Q9Z/8DguM0HJgDHv5HF3YDWHdp23hbz+TepHa8 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:djpscuSL/05sVtFNocQpHmr1x+NTcLXKu8ePb0q+ Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:9 F8_128_HMAC_SHA1_80 inline:5FqC2VY+q3vqaju3JwroIJO/OlCM1i6TNRE/3Atj Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:10 F8_128_HMAC_SHA1_32 inline:kUfTs7OFD6ltRNpDi1dSG5ZjuQAFDeEQD5IvOwme Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:11 NULL_HMAC_SHA1_80 inline:JZ2ruS0914NL4mmS7jxg+ngz0IQaE6FBRlet8abd Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:12 NULL_HMAC_SHA1_32 inline:h21D+oGRkr1dDEoMm1+vEKmKsn+4GxNBwgIew2qA Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:8100c85eb1c58be3580e8acd7f097000 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:hcqW22Sc Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:QjHJKT4qNsfZm80zrcyh8SfI76 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30000 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 2 UDP 2130706430 10.13.1.127 30001 typ host Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 between 10.13.1.127 and 10.13.1.152:14450, type prflx Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Triggering check for GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30000 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30020 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30001 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30021 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing '7n38ig1acl' from 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' from '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30000 remote 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30001 remote 10.13.1.152:14451 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Replacing previously learned prflx ICE candidate with Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Learning new ICE candidate H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:H6719a439:1 between 10.13.1.127 and 103.25.164.57:14450, type host Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Recalculating all ICE pair priorities Mar 9 20:21:02 debian rtpengine[11823]: WARNING: [ui7jc53b9lfbt9lf4nu7]: [ice] Priority collision between candidate pairs GtpEjl9r4IzNpiXr:H6719a439:1 and GtpEjl9r4IzNpiXr:Ha0d0198:1 - ICE will likely fail Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Sending ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.003226 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as succeeded Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Best succeeded ICE pair with all components is GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE not completed yet, but can use pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS: Peer certificate accepted Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS handshake successful Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Start nominating ICE pairs Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Nominating ICE pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Sending nominating ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting nominated ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as valid Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE completed, using pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate Ha0d0198 in favour of Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate H6719a439 in favour of H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.002244 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New ingress SSRC for: 10.13.1.152:14450 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31783 TS 528 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 00, length 16, result 6806f5cc... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 01, length 20, result 044b0cce... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 02, length 14, result 4426c95d... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New egress (direct) SSRC for: 0.0.0.0:9 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: ERR: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Error when sending message. Error: Inappropriate ioctl for device Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31784 TS 1488 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31785 TS 2448 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31786 TS 3408 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected:

2 1

Re: Kamailio stops processing the calls - restart fixes it.
by Henning Westerholt 10 Mar '23

10 Mar '23

Hello, please keep the list in the mail flow. Have you tried to set the acc_function modparam to an empty string, as commented in the linked issue? modparam("uac_redirect","acc_function","") If you don't need the acc function from the uac_redirect. Regarding the other warnings, they should be fixed - but different compilers can report different warnings, so it's sometimes happening. If they are just warnings, they should not create an issue. If you are using Debian/Ubuntu you can just use the prebuild packages from deb.kamailio.org - you don't need to build by yourself. Cheers, Henning -----Original Message----- From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> Sent: Donnerstag, 2. März 2023 14:25 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. Hi, I tried to install 5.6.4 (https://www.kamailio.org/pub/kamailio/5.6.4/src/) and get this error. Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [../../modules/acc/acc_api.h:191]: acc_load_api(): cannot find bind_acc Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [uac_redirect.c:259]: redirect_init(): cannot bind to ACC API Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: <core> [core/sr_module.c:975]: init_mod(): Error while initializing module uac_redirect (/usr/local/lib64/kamailio/modules/uac_redirect.so) It seems this issue was reported earlier and supposed to be fixed but apparently it's not. https://github.com/kamailio/kamailio/issues/3188 Besides that , I had seen warnings during compilation. What should be the most tested and supported version ? Example Warnings: dmq_funcs.c: In function ‘ki_dmq_send_message’: dmq_funcs.c:303:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:303:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] dmq_funcs.c: In function ‘ki_dmq_bcast_message’: dmq_funcs.c:373:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:373:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] CC (gcc) [M dmq.so] notification_peer.o CC (gcc) [M dmq.so] dmq.o dmq.c:61:1: warning: missing braces around initializer [-Wmissing-braces] sip_uri_t dmq_server_uri = {0}; On Wed, Mar 1, 2023 at 7:26 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > Hello, > > better take the latest one, e.g. 5.6.4 released yesterday. Minor releases only contains bugfixes, documentation enhancements and similar. Only rarely regressions happen. But you should of course test it. > > Cheers, > > Henning > > -----Original Message----- > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > Sent: Mittwoch, 1. März 2023 09:22 > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > Thank you for your email. > > Is v5.6.1 (July 6, 2022) stable? > > obtained from > > https://www.kamailio.org/pub/kamailio/latest-stable-version-number > > > On Wed, Mar 1, 2023 at 5:50 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > > > Hello, > > > > hard to say without more information, a backtrace etc... As I first step, I would suggest you to update the system to one of the supported releases, e.g. the latest 5.6.x or 5.5.x. > > > > Cheers, > > > > Henning > > > > -----Original Message----- > > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > > Sent: Mittwoch, 1. März 2023 01:39 > > To: sr-users(a)lists.kamailio.org > > Subject: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > > > Hi, > > > > I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. > > > > Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. > > > > Version is > > > > kamailio 5.5.3 (x86_64/linux) 473cef > > > > configuration is very simple , routing is based on tm.t_relay ( based on KEMI). > > > > Any help will be welcome. > > > > Danish > > > > > > > > -- > > Muhammad Danish Moosa > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to the sender! > > Edit mailing list options or unsubscribe: > > > > -- > Muhammad Danish Moosa > > " The core of mans' spirit comes from new experiences. "___ > Christopher McCandless -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

4 21

WebRTC Error
by Social Boh 09 Mar '23

09 Mar '23

Hello, I have this error when trying to register this softphone: https://github.com/InnovateAsterisk/Browser-Phone 61096#015#012Host: sip1.kamailio.xyz:8443#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Upgrade: websocket#015#012Origin: https://campus.voztovoice.org#015#012Sec-WebSocket-Version: 13#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/110.0.0.0 Safari/537.36#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: es-419,es;q=0.6#015#012sec-gpc: 1#015#012Sec-WebSocket-Key: uj5cXle8zzBFUzv4kWB2pg==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012> Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: WARNING: <script>: HTTP request received on 8443 Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: DEBUG: xhttp [xhttp_mod.c:426]: xhttp_send_reply(): sending out response: 403 Forbidden Any help is really apreciated -- --- I'm SoCIaL, MayBe

1 1

Location / Registrar, TCP: Use existing TCP connection
by Benoit Panizzon 09 Mar '23

09 Mar '23

Hi List CPE behind Firewall, registering to Kamailio via TCP (or TLS). When a call is sent to the CPE, kamailio attemts to open a new TCP connection and is blocked by Firewall. Is there an option to tell kamailio to use the existing registered TCP connection? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Call recovery for HA with sems
by Patrick Karton 09 Mar '23

09 Mar '23

1 0

TLS -> UDP ReInvite
by Social Boh 08 Mar '23

08 Mar '23

hello, I'm configuring Kamailio to permite calls like: UDP -> UDP TLS -> TLS UDP -> TLS TLS -> UDP Until now all is ok; the only problem is with Microsip SoftPhone. When make a call from Microsip TLS to another phone UDP, when the callee answer, Microsip send a new INVITE and Kamailio don't delete all references to SRTP so the call happens but without audio. I know I have to process the reINVITE where I do has_totag stuff but I don't know how. Regards -- --- I'm SoCIaL, MayBe

1 0

Re: sr-users Digest, Vol 213, Issue 52
by Mohamed SADIK 08 Mar '23

08 Mar '23

Hello, Thank you for your reply! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the *"sl.stats"* or any other procedures. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which is causing Kamailio to treat it as a "200" response code. Could you advise on how to avoid this issue? Is there any further configuration that needs to be done? Best regards. On Sat, Feb 25, 2023 at 12:00 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via email, send a message with subject or > body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..."Today's Topics: > > 1. Kamailio RPC configuration (sadik.oualla.mohamed(a)gmail.com) > 2. Re: Kamailio RPC configuration (Alex Balashov) > > > > ---------- Forwarded message ---------- > From: sadik.oualla.mohamed(a)gmail.com > To: sr-users(a)lists.kamailio.org > Cc: > Bcc: > Date: Fri, 24 Feb 2023 17:07:33 -0000 > Subject: [SR-Users] Kamailio RPC configuration > Dear Kamailio community, > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > I have already set up Kamailio (builed it from the source code 5.5.5) and > Python on my machines and I have basic knowledge of Kamailio configuration. > I just need some guidance on how to set up the RPC module in Kamailio and > how to create a Python script to communicate with it. I found that I can > use RPyC library of python. > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > Best regards, > Wild Coder. > > > > ---------- Forwarded message ---------- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 24 Feb 2023 13:22:16 -0500 > Subject: [SR-Users] Re: Kamailio RPC configuration > Hi, > > The easiest way to communicate with the JSONRPCS module is probably to use > HTTP, in the form of the `xhttp` module: > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html > > You will need... > > 1) To enable the HTTP transport in the `jsonrpcs` module; > > 2) A TCP or TLS listener (listen= config directive) to accept the traffic; > > 3) An xhttp `event_route[xhttp:request]` route, as detailed in that > module's docs: > > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html#xhttp.f.xhttp_re… > > 4) Probably some measure of security / authentication for #3. > > The example from jsonrpc_dispatch() is perfectly sufficient to get you > started: > > > https://kamailio.org/docs/modules/5.6.x/modules/jsonrpcs.html#jsonrpcs.f.js… > > You should then be able to do things like: > > # curl \ > -X POST \ > -d '{"jsonrpc": "2.0", "id": "abc123", "method": "ul.dump"}' \ > http://${KAMAILIO_IP}:5060/RPC <http://${kamailio_ip}:5060/RPC> > > -- Alex > > > On Feb 24, 2023, at 12:07 PM, sadik.oualla.mohamed(a)gmail.com wrote: > > > > Dear Kamailio community, > > > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > > > I have already set up Kamailio (builed it from the source code 5.5.5) > and Python on my machines and I have basic knowledge of Kamailio > configuration. I just need some guidance on how to set up the RPC module in > Kamailio and how to create a Python script to communicate with it. I found > that I can use RPyC library of python. > > > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > > > Best regards, > > Wild Coder. > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 e > > _______________________________________________ > sr-users mailing list -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >

1 0

Using jsonrpc_dispatch in xhttp:request - is it possible to restrict the RPC that can be invoked ?
by Chaigneau, Nicolas 07 Mar '23

07 Mar '23

Hello, I'm using jsonrpc_dispatch in a "xhttp:request" section (with xhttp and jsonrpcs modules), e.g.: event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("404", "Not Found", "", ""); exit; } This works, but I would like to restrict the RPC functions that can be invoked this way. Any advice on how I could do that ? The xhttp module provides "url_skip" and "url_match" but this is not useful because in the URL I only have something like "RPC". And the method is in the JSON body. I could parse the JSON body before calling jsonrpc_dispatch, but it's a bit complicated (I don't know if it's feasible in Kamailio script ?), and seems redundant because jsonrpc_dispatch already does that... Maybe somehow this could be checked from the "void *" second argument passed to the RPC function ? Thanks. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Info: Kamailio World 2023 - Announcing First Group Of Speakers
by Daniel-Constantin Mierla 06 Mar '23

06 Mar '23

Hello, it's now like 3 months till the start of the next Kamailio World Conference (taking place in Berlin, Germany, during June 5-8, 2023) and the first group of speakers has just been published, the details are available at: - https://www.kamailioworld.com/k2023/speakers/ Not all proposals were reviewed and Call for Speakers is still open, therefore if you want to share your interesting experiences or knowledge in the RTC space, submit it via: - https://www.kamailioworld.com/k2023/call-for-speakers/ All together, expect another edition of Kamailio World conference with great content covering topics such as RTC scalability, security, cloud-based deployments and elasticity, NG112/911, IoT, WebRTC, VoLTE and 4G/5G. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Remove silo extra_hdrs before store
by Dr. Barabás Péter 06 Mar '23

06 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

1 0

How to access rtpengine rtcp-metrics?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi all I'm trying to incorporate rtcp data from rtpengine into our CDRS. So trying with the first interesting value, the average mos. modparam("rtpengine", "mos_average_pv", "$avp(mos_average)") If I understood right, to get this variable set, I need to call rtpengine_manage() on the message that terminates the call. So on an established call, that is either BYE or the 200 OK to the BYE. Right? if ($rm == "BYE") { rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm: MOSS: $avp(mos_average)\n"); } onreply_route[MANAGE_REPLY] { [...] rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm reply MOS: $avp(mos_average)\n"); } Messages pass those blocks, $avp(mos_average) is 'null' no mater what. What am I missing? In the syslog output of rtpengine I see there is rtcp data. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 2

tcp_set_connection_lifetime => in respect to 'idle' connection?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi List Digging further into my tcp/tls issues... Upon successful authentication I call: tcp_keepalive_enable("60", "5", "5"); tcp_set_connection_lifetime("120"); I tought, this would do to keep the connection alive. I noticed, despite keepalive packets being exchanged, kamailio is closing the connection after a bit more than 2 minutes. I am right at assuming, that tcp_set_connection_lifetime is not related to how long the connection is idle, but is counted from the SYN, no matter if keepalives were successfully exchanged? Does calling tcp_set_connection_lifetime() again restart the lifetime like for 2 more minutes like in this example? Or would I have to keep increase the value of the lifetime on each successful re-register to keep it open? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio v5.6.3 Released
by Daniel-Constantin Mierla 05 Mar '23

05 Mar '23

Hello, Kamailio SIP Server v5.6.3 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.3. For more details about version 5.6.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/01/kamailio-v5-6-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

5 10

Kamailio v5.5.6 Released
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, Kamailio SIP Server v5.5.6 stable release is out. This is a maintenance release of the stable branch 5.5 that includes fixes since the release of v5.5.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.6. Note that 5.5 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.6, with v5.6.4 being release out of it. For more details about version 5.5.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/03/kamailio-v5-5-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Releasing Kamailio v5.5.6
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, I am considering to release Kamailio v5.5.6 soon, branch on branch 5.5, likely on Thursday or Wednesday next week (Mar 2/3, 2023). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Websocket closing
by David Villasmil 01 Mar '23

01 Mar '23

Hello guys, We're seeing corner cases where the following happens: On proxy-ws - IP1:PORT1 connects via websocket from Client1 - Registration happens on an upstream kamailio - for any reason, the TCP socket closes or times out. - IP1:PORT1 (same IP:PORT combination) connects via websocket from Client2 - Registration happens on an upstream kamailio Now a call comes in to Client1. Because the first connection was never cleaned up, it is sent to the proxy-ws and the proxy will send it to the IP1:PORT1 where Client2 is connected. Short story, proxy-ws doesn't check the IP1:PORT1 where it is sending the INVITE is the actual client it is supposed to be sending... It seems that when a socket is closed, the mapping IP:PORT to Address (i.e.: sip:d4f27e34@994f31243be9.invalid;transport=ws) doesn't seem to be cleared... is this by design? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 6

Kamailio stops processing the calls - restart fixes it.
by Muhammad Danish Moosa 01 Mar '23

01 Mar '23

Hi, I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. Version is kamailio 5.5.3 (x86_64/linux) 473cef configuration is very simple , routing is based on tm.t_relay ( based on KEMI). Any help will be welcome. Danish -- Muhammad Danish Moosa

3 5

Kamailio v5.6.4 Released
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, Kamailio SIP Server v5.6.4 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.4. For more details about version 5.6.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/02/kamailio-v5-6-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. An additional note to say that call for speakers and registration for Kamailio World Conference 2023 are now open, more details at: * https://www.kamailioworld.com Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Kamailio MS Teams TLS Issue
by iliusha.md＠gmail.com 27 Feb '23

27 Feb '23

Hello, We have one Kamailio Instance connected with MS Teams (based on this instruction: https://skalatan.de/en/blog/kamailio-sbc-teams), which worked fine for a while until recently we noticed that calls from teams are not working anymore. When I looked through the logs I found that Microsoft cannot establish a TLS connection to our server because of the cipher: TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: sbc.example.com - domain is obfuscated). Certificate is valid, the configuration is below: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt server_name = sbc1-teams.example.net ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt We use a certificate from Sectigo, but I've tried with Let's Encrypt - and it's the same. Any idea what could be the reason?

5 7

Releasing v5.6.4
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, I am considering to release Kamailio v5.6.4 (out of branch 5.6) next week (likely on Monday or Tuesday, Feb 27/28, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

2 1

OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

4 6

Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

3 2

What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

2 2

change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

2 1

Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

7 7

Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

2 2

topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 4

Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

3 11

enable topology hiding on one side or dynamic ip in topoh module Contact and Via Header
by Patrick Karton 17 Feb '23

17 Feb '23

Hello , im using kamailio with two interfaces external and internal. i need a way either to : * enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external * or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled. i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately. Thanks

4 8

Calling uac_replace multiple times?
by Benoît Panizzon 17 Feb '23

17 Feb '23

Hey all I have started using the uac module to be able to mirror username changes on both legs of a call. But now I found an issue: Assume this situation: Call IC => Customer IC is e164, Customer wants 'local' numbers. IC From: +41991234567 => Customer From: 0991234567 uac_replace_from("sip:" + $avp(from_local) + "@" + $(fu{uri.host})); Customer is absent, after timeout in failure_route call is dispatched to Voicemail Server, which is configured for e164 numbers: uac_replace_from("sip:" + $avp(from_e164) + "@" + $(fu{uri.host})); But then uac complains: replace_uri(): Already called uac_replace for this dialog Why isn't just the most from version kept and used? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

TLS version and cipher from WSS
by Michal Hajek 17 Feb '23

17 Feb '23

Hi, I can log incoming SIPS TLS version and cipher with PV $tls_version and $tls_cipher_info. How can I log these variables on incoming HTTP/WebSocket. Thanks Michal H.

2 2

Identify error in log
by Евгений Мастратов 16 Feb '23

16 Feb '23

Hello colleagues. I'm starting to use Kamailio for educational purposes, as an IMS test zone. Registration of subscribers, calls go successfully, but the following errors are constantly pouring into the log: Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Apparently they don’t carry anything critical, but they clog the log, natively I can’t understand the meaning of the errors. Please tell me what they mean and how to eliminate their cause? If necessary, I will provide the pcscf config.

2 1

uac reg reload frequency
by Dmytro M 16 Feb '23

16 Feb '23

When you do "kamcmd uac.reg_reload" frequently enough, you'll get an error: # kamcmd uac.reg_reload error: 500 - Failed to shift records - check log messages Logs: ERROR: uac [uac_reg.c:416]: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs What is the reason behind 150 sec limitation for how frequent reloads can be done? Is it configurable in any way? Use case: Remote app needs to disable uac registration. It deletes record in DB and calls kamcmd uac.reg_reload to activate the change. With 150 sec limit it can't apply DB changes immediately. Please advise. Thanks, Dmytro

2 2

ACK on RE-Invite is send to UAC NATed Internal port instead of public port
by Angelo Sipper 15 Feb '23

15 Feb '23

Hi, My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) -> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP) Both Kamailio are 4.3 with minor modifications on default cfg, and Topoh and TLS modules are used. I am having an issue on some outbound calls from nated UACs. On these calls our SBC sometimes receives reverse invites from the carrier side within the call. When this happens, my SBC(17.17.17.211) sends the invite to UAC and the UAC sends back 200 OK (SDP). Unfortunately my SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but instead to the UAC internal port making the call to crash with several retries from my SBC to send 200 OK as ACK was never received from UAC! If you look the below traces of reverse invite past within the call, the reinvite has port private 20125 and alias the public port 2314 of UAC but the ACK bellow is send to 20125 and not to 2314 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Any help will be much appreciated. Thank you! Regards, Angelo

2 5

SCA Presence : Record-Route Filed Not Update in sca_subscriptions Table
by Vinayak Makwana 14 Feb '23

14 Feb '23

Hello all, I am using the sca module in kamailio script.so when i subscribe with my ip phone for SCA Presence i get all the information in a sca_subscriptions table for the first time. When I reboot the device I get a new record-route header in trace but It does not update *record-route *Filed in sca_subscriptions table. Can anyone help me out with this issue ? *Here's modparam configuration * modparam("sca", "db_update_interval", 10 ) modparam("sca", "db_url", DBURL_LOCAL) modparam( "sca", "line_seize_max_expires", 30) modparam( "sca", "purge_expired_interval", 1800) -- -- * <https://www.ecosmob.com/itexpo/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

3 8

SCA Feature:: Record-route query.
by satyaprakash ch 13 Feb '23

13 Feb '23

Hello I have a query regarding the record-route field update in DB for SCA SUBSCRIBE. Here are my test cases *Initial SCA subscription,* We got the below SUBSCRIBE packet in kamailio SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081381rdb66 *Record-Route* : <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=CFFBAF19-19A68CBC Call-ID: bd4b50d9384523c14ea05c532d895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 also in the sca_subscriptions table record-route header is updated as per Initial SUBSCRIBE record-route. *record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> * But After rebooting the device, we got a new record-route in SUBSCRIBE Packet but in DB Record-route Field not updated after getting an updated record-route in Trace. *Kamailio Trace *SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081415rdb48 *Record-Route*: <sip:sansay1081415rdb48spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=35852031-CF749E20 Call-ID: 41fa031d79e35fae4a61c61984895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 as per logic it should update the record-route field in sca_subscriptions table . but it didn't update it : record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> So can you suggest to us how we can update the record-route field as per SUBSCRIBE's packet new record-route ?

2 1

Issues matching regex
by Nathan Stratton 11 Feb '23

11 Feb '23

I am having an issue matching toll-free. I tested my regex on regex101.com, and it looks good, but it does not match my if statement. Am I doing something wrong? if (method=="INVITE") { xlog("L_INFO", "Checking Routes $rU"); if ($rU=~"^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$") { xlog("L_INFO", "Tollfree call"); }; }; Logs show "<script>: Checking Routes +18002211212" so I don't get why it does not match. ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

2 1

kemi equivalent config
by mohsen khashei 11 Feb '23

11 Feb '23

2 1

KEMI Equivalent of script
by mohsen khashei 10 Feb '23

10 Feb '23

Hi I have an existing kamailio config and I want to migrate to kemi. I want to know what is the equivalent of this code in kemi javascript ? $avp(s:contact) = $hdr(Contact); $avp(s:contact_fix) = $hdr(Contact); avp_subst("$avp(s:contact_fix)", "/@([0-9]+).([0-9]+).([0-9]+).([0-9]+)/@$sel(cfg_get.kamailio.ss_ip)/gi"); remove_hf("Contact"); append_hf("Contact: $avp(s:contact_fix)\r\n"); t_on_reply("REG_FIX"); thanks for your attention.

1 0

Return value from lua/python/js/bash/etc... function
by Dmytro M 10 Feb '23

10 Feb '23

What is the simplest way to return values from external application into Kamailio config and assign it to a variable? I'm looking for the most light weight mechanism to do it and to offload some logic to external app. Something like this would be ideal: $var(myVar), $var(err) = lua_run("lua_func1", "$rU"); OR $var(encString) = exec_cmd("echo 'abc $rU' | base64"); There is this method: http_client_query("http://...", "", "", "$var(result)"); but there is an overhead of networking and HTTP protocol. Please advise. Thanks, Dmytro

3 4

ACK on Reverse Invite is send to NATed Internal port instead of public port
by Angelo Sipper 09 Feb '23

09 Feb '23

Hi, I am having an issue on some outbound calls from nated UAC. On these calls our SBC receive reverse invites from the carrier side within the call. When this happens, my SBC(Kamailio1) sends the invite to UAC and the UAC sends 200 OK (SDP). Unfortunately my SBC(Kamailio1) sends back the ACK not to the correct UAC public port but instead to UAC internal port making the call to crash! My setup has: (TLS)UAC -> (UDP)Kamailio1(UDP) -> (UDP)Kamailio2(TLS) -> Carrier(UDP) Bellow you can find the trace and you can see the ACK send to wrong port on 2023/01/26 09:48:31.254591 Your help will be much appreciated! 2023/01/26 09:48:23.225326 77.1.1.1:2314 -> 17.17.17.211:5060 INVITE sip:300987654321@17.17.17.211:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport Max-Forwards: 70 Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Remote-Party-ID: <sip:301234567890>;party=called Content-Length: 283 v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:23.240310 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314;received=77.1.1.1 To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:23.301454 17.17.17.211:5060 -> 17.17.17.181:5060 INVITE sip:300987654321@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 300 P-Asserted-Identity: <sip:301234567890@sip.mydomain.com> v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:23.302899 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0;rport=5060 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:24.077941 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.138348 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.139050 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:24.200575 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.115324 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.336507 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.432457 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.433017 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.457268 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.481470 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.555470 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.557824 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.530836 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.531333 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.609667 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.612312 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.63edf375f6b899dfa9d2d9a2dcee4dfd.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.536512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.536839 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.623207 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.641928 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.745a55762e847bc3cf56d27926f7e62d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.540735 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.541310 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.590013 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.595763 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.6c6654fa1c2f4b1991d5b6d792f2f081.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.561756 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.562039 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.599703 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.603466 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e022b4e31f8266525e62f213dc4dd8ae.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.584025 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.584257 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.622809 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.626545 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.0e211693731e67705c85dd4844e42ce8.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.629501 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.629961 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.676968 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.682458 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.cc80042ad9e1838346f4cf6d66a4c13f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.636998 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.637287 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.675272 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.681861 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.be45a1f76f0fcd5ce85292397ff68acc.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.677049 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.677633 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.721399 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.729950 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.365607f0beb6e7e5dcb047c1ad6fcf5f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.681010 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.681307 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.723730 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.731143 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.12bbae879e42215b50512c04c4fc0214.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.684483 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.684985 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.727999 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.732652 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.fb4ff819be250670479341711417a554.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.028200 77.1.1.1:2314 -> 17.17.17.211:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---7c113301b578bb42;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.090194 17.17.17.211:5060 -> 17.17.17.181:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.261453 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.261941 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Thank you! Regards, Angelo

2 4

DB_CLUSTER failure with carrierroute (and others?)
by Kaufman 08 Feb '23

08 Feb '23

All, I've been trying to chase down an issue with db_cluster and permissions in combination with something else, where with my full configuration using the same cluster in multiple modules has failures. I've been trying to create a simplified configuration to consistently reproduce the problem without much success, however I did stumble upon THIS particular problem. The configuration here works fine as is - avpops uses the cluster URL, and carrierroute uses the direct SQLite URL. If I define WITH_DBCLUSTER, then carrierroute fails fetching rows using the cluster connection. Note that AVPOPS is only used as a general test to see if the cluster config works. If I remove references to avpops , carrierroute still fails loading from the cluster URL. Thoughts? #!KAMAILIO loadmodule "db_sqlite" loadmodule "pv" loadmodule "db_cluster" loadmodule "carrierroute" loadmodule "avpops" modparam("db_cluster", "connection", "db1=>sqlite:////etc/db.sqlite") modparam("db_cluster", "cluster", "cl1=>db1=9s9s") modparam("db_cluster", "inactive_interval", 30) #!ifdef WITH_DBCLUSTER modparam("carrierroute", "db_url", "cluster://cl1") #!else modparam("carrierroute", "db_url", "sqlite:////etc/db.sqlite") #!endif modparam("carrierroute", "config_source", "db") modparam("avpops", "db_url", "cluster://cl1") modparam("avpops", "avp_table", "usr_preferences") request_route { exit; } Failure log: 0(1) ERROR: carrierroute [cr_db.c:345]: load_route_data_db(): Fetching rows failed 0(1) ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data 0(1) ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data 0(1) ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module carrierroute (/usr/lib/kamailio/modules/carrierroute.so) Kaufman

3 5

Drops on async_route with delay
by Alex Balashov 07 Feb '23

07 Feb '23

Hi, Some percentage of requests processed with async_route("REQ_PROCESS", "5") seem to end up with a resumed transaction, though most do. Not sure what the exact percentage is. The ones that don't Is this due to excessive requests? Is there a limit on internal IPC queue depth? Is it conceptually similar to a generic shared blocking queue internally, along the lines of 'mqueue'? Is there any reasonable way to troubleshoot this? Thanks! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

3 3

Dispatcher + ENUM
by Jawaid Bazyar 07 Feb '23

07 Feb '23

Hi, Can DISPATCHER be used in conjunction with ENUM? E.g., to avoid sending requests to SIP destinations that are returned by ENUM, but, which DISPATCHER knows are down? Thanks, Jawaid

2 1

nating private uas to public ip addr
by Jack Murphy 06 Feb '23

06 Feb '23

I have a multi-homed kam with a uas behind one private interface and a public interface for the uac+sip provider. When the uac initiates a call, kam sends the invite to the private uas which sets up a b-leg call back to kam with a Contact header that has the uas' private ip address. I want to change that private ip address to be Kamailio's advertised/public ip address before it gets forwarded to the sip provider. I am rewriting Contact in a route from request_route like this: > if ($rP != $null) { > $var(ct) = ";transport=" + $rP; > }else{ > $var(ct) = ""; > } > $var(cr) = "<sip:" + $fU + "@" + $RAi + ":" + $RAp + $var(ct) + ">"; > > remove_hf("Contact"); > append_hf("Contact: $var(cr)\r\n"); This seems to work fine for the INVITE going to the provider. There's also the 200 OK that the uas sends for the uac (a-leg) with a private ip address in Contact -- I want to rewrite that before sending it to the UAC, too. Using the same method as above (but inside onreply_route), what happens is the UAC receives the modified OK and sends back an ACK with the modified Contact (kam's public ip addr) in the ACK r-uri, and kam gets in a loop of sending an ACK from its private ip address to its public ip address and replying to itself. call flows: https://imgur.com/a/Yj2DJpn Is there a right way to do this?

2 2

htable: autoexpire vs updatedelete
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/htable#htable.p.htable says for modparam('htable', 'htable', '…autoexpire … updateexpire…') autoexpire -time in seconds to delete an item from a hash table if no update was done to it. If is missing or set to 0, the items won't expire. updateexpire - if set to 1 (default), the time until expiration of an item is reset when that item is updated. Certain uses of htable may dictate that updates should not reset the expiration timeout, however, in which case this attribute can be set to 0. My reading is that, when a value is updated and autoexpire is positive, autoexpire will reset the timer, but updateexpire may or may not reset the same timer. That is: modparam('htable', 'htable', 'a=>autoexpire=5;updatedelete=1') modparam('htable', 'htable', 'b=>autoexpire=5;updatedelete=0') If b was not updated for 5s, will it be removed, as the documentation for autoxpire suggests, irrespective of updatedelete? Greetings Dilyan

2 2

Depending on topoh.mask_key Kamailio either cannot parse RURIs or reads invalid data
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, when I use dns_try_naptr=on corelog=-1 debug=-1 enable_tls=yes use_dns_cache=off dns_cache_init=off modparam("topoh", "mask_key", "TEAI32l)- eauiDEUIA!?()") and run Kamailio under valgrind, Kamailio logs: 20(21) ERROR: kemix [kemix_mod.c:229]: ki_kx_get_ruri_attr(): failed to parse the R-URI 20(21) ERROR: rr [loose.c:1011]: loose_route_mode(): failed to parse Request URI The workflow is: I start Kamailio. A UDP-client registers. Then a websocket client calls the UDP-client. Finally the websocket client hangs up. Valgrind does not report anything suspicious. When I use instead modparam("topoh", "mask_key", "TEAI32l") for the same workflow, with the same configuration, valgrind logs: ==14== Invalid read of size 4 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) ==14== Address 0x0 is not stack'd, malloc'd or (recently) free'd ==14== ==14== ==14== Process terminating with default action of signal 11 (SIGSEGV): dumping core ==14== Access not within mapped region at address 0x0 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) The stacktrace with the values of the variables is available at https://github.com/kamailio/kamailio/issues/3350 . Or at least these things seem very similar to me. Any idea? I can share the OCI-image and the full configuration. Greetings Дилян

3 4

topoh: th_mask_decode (th_mask_encode (x)) ≠ x
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, after receiving INVITE sip:test-gnome-calls@bapha.be SIP/2.0 Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK974100 Max-Forwards: 70 To: <sip:test-gnome-calls@bapha.be> From: "Online https://sip.bapha.be" <sip:online@bapha.be>;tag=12q2efc423 Call-ID: fkhe8faq0fh80glmq59i CSeq: 5622 INVITE Contact: <sip:4q5ne5no@bapha.be;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.8 Content-Type: application/sdp Content-Length: 1964 v=0 o=mozilla...THIS_IS_SDPARTA-99.0 8263261229339246488 0 IN IP4 0.0.0.0 s=- … kamailio calls th_mask_encode( char *in = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60722~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\n"\ "Content-Type: application/sdp and so on", int ilen = 107, str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }, int *olen). It returns olen=167, and the string "sip:127.3.4.84;line=sr- if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p-1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*". Then Kamailio forwards (sends): Contact: <sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime" which at some point leads to ACK sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 and BYE sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 Below you can find th_test.c . The functions th_mask_encode() and th_mask_decode() there are identical to the same functions in src/modules/topoh/th_mask.c . main() provides some tests: Passing «sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime» to th_mask_encode() returns sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWXi-eo9* (this is what Kamailio does send out). Passing the last string to th_mask_decode() returns sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50-740fb9bdc562;alias=87.118.1 the ;alias= is incomplete, compared to the origin. The above ACK leads to passing "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ to th_mask_decode() and the result is callub.p.b;rn:s.d2lQg#cfa=bA07beQ7R409bcQ62;alac=87718*1P.15#~60R22~2 with the dots being unprintable characters. (result is garbage). I have seen that callub.p.b in the unparsable R-URI reports from Kamailio. That is an indication that th_mask_decode() does not decode the Contact encoded by th_mask_encode() correctly. Valgrind does not report improper memory usage. Greetings Dilyan gcc -g -o th-test ./th-test.c && ./th-test th-test.c #include <stdio.h> #include <stdlib.h> #include <string.h> #define TH_EB64I \ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.-" char _th_EB64[65] = "EFvXIzGq94xKcW126gV5wCdYpNSbiM8enRUou7LmhajBlPtAsk-OQTf3H0DyrZ.J"; int _th_DB64[256]; char *_th_PD64 = "*"; struct str_ { char* s; /**< Pointer to the first character of the string */ int len; /**< Length of the string */ }; typedef struct str_ str; str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }; char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen); char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen); char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen) { char *out; int left; int idx; int i; int r; char *p; int block; *olen = (((ilen+2)/3)<<2) + ((prefix!=NULL&&prefix->len>0)?prefix->len:0); out = (char*)malloc((*olen+1)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1)*sizeof(char)); if(prefix!=NULL&&prefix->len>0) memcpy(out, prefix->s, prefix->len); p = out + (int)((prefix!=NULL&&prefix->len>0)?prefix->len:0); for(idx=0; idx<ilen; idx+=3) { left = ilen - idx - 1 ; left = (left>1)?2:left; block = 0; for(i=0, r=16; i<=left; i++, r-=8) block += ((unsigned char)in[idx+i]) << r; *(p++) = _th_EB64[(block >> 18) & 0x3f]; *(p++) = _th_EB64[(block >> 12) & 0x3f]; *(p++) = (left>0)?_th_EB64[(block >> 6) & 0x3f]:_th_PD64[0]; *(p++) = (left>1)?_th_EB64[block & 0x3f]:_th_PD64[0]; } return out; } char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen) { char *out; int n; int block; int idx; int i; int j; int end; char c; for(n=0,i=ilen-1; in[i]==_th_PD64[0]; i--) n++; *olen = (((ilen-((prefix!=NULL&&prefix->len>0)?prefix->len:0)) * 6) >> 3) - n; if (*olen<=0) { return NULL; } out = (char*)malloc((*olen+1+extra)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1+extra)*sizeof(char)); end = ilen - n; i = (prefix!=NULL&&prefix->len>0)?prefix->len:0; for(idx=0; i<end; idx+=3) { block = 0; for(j=0; j<4 && i<end ; j++) { c = _th_DB64[(int)in[i++]]; if(c<0) { free(out); *olen = 0; return NULL; } block += c << (18 - 6*j); } for(j=0, n=16; j<3 && idx+j< *olen; j++, n-=8) out[idx+j] = (char)((block >> n) & 0xff); } return out; } int main() { //this is from th_mask_init for(int i=0; i<256; i++) _th_DB64[i] = -1; for(int i=0; i<64; i++) _th_DB64[(int)_th_EB64[i]] = i; int olen, o2; char *end; char *input = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>\";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\""; printf("input=[%s]\n", input); char* middle = th_mask_encode (input, 107, &prefix, &olen); printf("olen = %i, middle=[%s]\n", olen, middle); end = th_mask_decode(middle, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char * b1 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9*"; o2 = 0; end = th_mask_decode(b1, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char* b2 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ "Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK9069155\n"\ "Max-Forwards: 70\n"\ "To: <sip:test-gnome-calls@bapha.be>;tag=Pn4npAj\n"\ "From: \"Online https://sip.bapha.be\" <sip:online@bapha.be>;tag=12q2efc423\n"\ "Call-ID: fkhe8faq0fh80glmq59i\n"\ "CSeq: 5623 ACK\n"\ "Supported: outbound\n"\ "User-Agent: SIP.js/0.7.8\n"\ "Content-Length: 0\n"\ "\n"; o2 = 0; end = th_mask_decode(b2, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); return 0; }

2 1

no autocomplete for kamcmd for latest debian packages
by Ovidiu Sas 04 Feb '23

04 Feb '23

Hello, The latest debian packages are built without autocomplete for kamcmd. Do we have the libreadline-dev package installed on the build machine? I know that we have kamctl rpc and kamcli, but if we are still shipping kamcmd, it should have the autocomplete feature enabled. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 2

Sip messages responses counters per interface
by Patrick Karton 02 Feb '23

02 Feb '23

2 1

rtpengine - SRTP <> RTP
by kdrewicz+kamailio＠cludo.pl 01 Feb '23

01 Feb '23

I try to workout if - currently it would work, or - where and how to debug more: I face - 2 interfacec - public internet (so, TLS + sRTP) is desired and private - old infrastructure - i mus only use plain RTP 172.23.9.70 - private ip - from this endpoint of kamailio and rtpengine should send only basic RTP 172.23.210.75:5060 private - target for kamailio 1.2.3.24 obfuscated public IP (TLS + sRTP required) kamailio 5.4.4 (x86_64/linux) rtpengine -v Version: 11.1.1.4-1~bpo11+1 all i do is: if (proto==TLS) { rtpengine_manage("RTP/AVP ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=external direction=internal record-call=on"); } else if ($ru =~ "transport=tls") { rtpengine_manage("DTLS=on SRTP AVPF ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=internal direction=external record-call=on media-address=1.2.3.24"); } # 1.2.3.24 obfuscated public IP 172.23.210.75:5060 is in dispatch.cfg, as '11' route[SBC_CORE] { append_hf("X-My-SRTP: removed31337\r\n"); ### i see this text in invtes from kamailio 172.23.9.70 towards 172.23.210.75:5060 ### i see only RTP, so as expected if (!ds_select_dst("11", "0")) { xwarn("I:$var(i) DROP(DOWN!) FWD:$rm [$fU->$tU] [SBCVIP] to $du\n"); sl_send_reply("503", "Destination down"); exit; } what i did: certificate is a paid one (the public party needs it) TLS works i deleted - entries in (not kamailo) cryptosuite that caused this: 13:08:05 localhost rtpengine[15140]: ERR: [51ad8758-b64d-4d2f-9fd0-41d03a38f74d]: [core] Failed to parse a=crypto attribute, ignoring: unknown crypto suite Tried to search for any "ready" examples for this - only found old threads and - that this should be possible, but - no example for woking config. what i see: Jan 19 19:00:57 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000038 sec Jan 19 19:00:58 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000036 sec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Closing call due to timeout Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Redis delete_async=0 Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Final packet stats: Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag 'JVR5LTs', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/SAVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30136 <> 52.129.106.28:17030, SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30137 <> 52.129.106.28:17031 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag '', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30014 <> :0 , SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30015 <> :0 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Moved metadata file "/var/spool/rtpengine/tmp/rtpengine-meta-c17bab16-5eea-492e-b1c4-ac9387f3e265-7003946f152c6c8d.tmp" to "/var/spool/rtpengine/metadata" Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000828 sec Jan 19 19:01:00 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000053 sec route(SBC_CORE); maybe any hint or - someone has working exmaple of kamailio config + rtpengine settings ? i use only userspace daemon rtp forwarding (this is a test, dont need any performance here) Thanks,

5 9

transport=ws causing kamailio to use wrong listen socket
by nathan.bruning＠talksome.com 31 Jan '23

31 Jan '23

Hi all, We're hitting an issue while integrating secure websockets in our existing SIP infrastructure using Kamailio. When the registration comes in, it causes an entry in our AOR table, with ";transport=ws" appended. When we want to send a message to this client (using t_relay), kamailio seems to take 'ws' as being *unsecure* websockets. In turn, this makes Kamailio try to send out the message using a TCP listener - while it should have picked the TLS listener. There are some remarks in the sources about ws vs. wss, so i'm struggling to figure out where things go wrong. I've also created github issue #3340 with more details. Any help would be appreciated. If this turns out to be a Kamailio bug, i'm happy to provide a patch.

6 7

How can I setup Kamailio for failover on LCR ?
by scramatte 31 Jan '23

31 Jan '23

Hello, How can I setup Kamailio for failover on LCR ? Currently my Kamailio 4.x works with lcr but doesn’t care about gateway status. I mean that if a gateway match rule , it sends the call through without check if the gateway is available an never try the next one if the first fail. I’ve tried to implement inactivate_gw() But I received error that lcr_id_avp is not defined. I’ve added the parameters to lcr module for Ping and enable defunc but I’m unable to get it working. By the way I ‘ve trad that defunct is not a good practice as hacker sending malformed sip header can throw down the gateway. So what is the good and working way to setup outbound call failover ? Regards Sebastien

1 0

Evapi Dispatcher 100% cpu.
by Sebastian Sastre 30 Jan '23

30 Jan '23

Hello We are currently using EVAPI to push messages into kamailio from a go app. For the most part ift works without problems, messages get delivered into the event route and we use it to update presence status. Recently we started noticing delays on the evapi processing. After 500 requests x min the pid corresponding to the EVAPI dispatcher gets pegged at 100% CPU and processing of requests starts slowing down until in some cases the connection just drops and we have to reconnect. After that pid drops, the dispatcher goes back to working. Unfortunately the error we get back is very generic and is not telling us much logger.go:39: ERR Handler returned error (write tcp 127.0.0.1:48448->127.0.0.1:8228: write: broken pipe) This only happens on prod servers and with high load which makes it hard to debug. There is no other slowness or increase of traffic or drop SIP traffic, only evapi dispatcher at 100% and the EVAPI workers on idle. Curious if anyone has run into this issue. We've tried different versions (5.4 , 5.5 and 5.6) and it happens on all of them. Any feedback appreciated ! Thanks !

2 1

Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary?
by Дилян Палаузов 30 Jan '23

30 Jan '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/tls says: For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/utils/openssl_mutex_shared/README.md'. 'src/modules/tls/utils/openssl_mutex_shared/README.md says: **IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default.** This is a shared library required as a short term workaround for using Kamailio with OpenSSL (libssl) v1.1. It has to be pre-loaded before starting Kamailio. In v1.1, libssl does not allow setting custom locking functions, using internally pthread mutexes and rwlocks, but they are not initialized with process shared option (PTHREAD_PROCESS_SHARED), which can result in blocking Kamailio worker processes. Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary? If it is not necessary, why isn’t it removed from the source code? Greetings Дилян

1 0

Websocket-answers sent by K. log unparseable errors
by Дилян Палаузов 29 Jan '23

29 Jan '23

Hello, when I connect to Kamailio over websocket, according to the websocket protocol K. sends “HTTP/1.1 101 Switching Protocols” and K. logs: 17(18) DEBUG: websocket [ws_handshake.c:179]: ws_handle_handshake(): found Upgrade: websocket 17(18) DEBUG: websocket [ws_conn.c:195]: wsconn_add(): connection id [14] 17(18) DEBUG: websocket [ws_conn.c:213]: wsconn_add(): new wsc => [0x7fcad6e10160], ref => [0] 17(18) DEBUG: websocket [ws_conn.c:233]: wsconn_add(): added to conn_table wsc => [0x7fcad6e10160], ref => [1] 17(18) DEBUG: sl [sl.c:310]: send_reply(): reply in stateless mode (sl) 17(18) DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (22.222.222.222, 22.222.222.222, 0) 17(18) DEBUG: <core> [core/tcp_main.c:1644]: _tcpconn_find(): found connection by id: 14 17(18) DEBUG: <core> [core/tcp_main.c:2528]: tcpconn_send_put(): send from reader (18 (17)), reusing fd 17(18) DEBUG: <core> [core/tcp_main.c:2763]: tcpconn_do_send(): sending... 17(18) DEBUG: <core> [core/tcp_main.c:2796]: tcpconn_do_send(): after real write: c= 0x7fcad6da7b38 n=238 fd=9 17(18) DEBUG: <core> [core/tcp_main.c:2797]: tcpconn_do_send(): buf= 17(18) DEBUG: <core> [core/parser/parse_fline.c:255]: parse_first_line(): bad request first line 17(18) DEBUG: <core> [core/parser/parse_fline.c:258]: parse_first_line(): at line 0 char 22: 17(18) DEBUG: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parsed so far: HTTP/1.1 101 Switching 17(18) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 22) 17(18) DEBUG: <core> [core/parser/msg_parser.c:675]: parse_msg(): invalid message 17(18) ERROR: <core> [core/parser/msg_parser.c:749]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TLS 22.222.222.222:33776 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: NeswoJX5ZQp7ER8hTKM5B3a3HDQ= Content-Length: 0 > 17(18) ERROR: <core> [core/msg_translator.c:3256]: build_sip_msg_from_buf(): parsing failed 17(18) DEBUG: app_lua [app_lua_api.c:1924]: sr_kemi_lua_exit(): script exit call 17(18) DEBUG: app_lua [app_lua_api.c:789]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ 17(18) DEBUG: app_lua [app_lua_mod.c:164]: sr_kemi_config_engine_lua(): execution of route type 513 with name [ksr_xhttp_event] returned 1 22.222.222.222 is my public address (the address of my router). It is not the IP-address of Kamailio. While the returned address is not parseable in terms of SIP, it is a valid Websocket-answer message. This valid condition triggers logging an error, but does not represent an error condition. How can I avoid that the above non-error is logged as an error? Thanks for your feedback Дилян

1 0

Locally generated BYE sent to private IP
by James Lipski 27 Jan '23

27 Jan '23

Hi all, I'm doing some experimenting with kamailio -- currently looking for a replacement of our existing (aging) SIP infrastructure. One of the experiments that I'm doing is testing the dialog module's dlg_set_timeout function. The issue that I'm having is while I do see that kamailio is generating a BYE, and sends it out to the B-leg of the call properly, the BYE goes to a private address on the A-leg. To Note: - kamailio is behind a NAT (on AWS) - both UA endpoints are remote and are both behind a NAT - Not sure if this matters, but I do have the topos module enabled, though I see the same behavior regardless if it's enabled or not. - Have set nathelpers (fix_nated_register, fix_nated_contact,etc...) which helps calls between endpoints and the endpoints themselves are able to send/receive BYEs when they generate them; however doesn't seem to help if kamailio locally generates the BYE. The only thing I do see within my logs is when the BYE is being generated by kamailio, it uses the private IP of my UA and not public for the caller. Callee, the public address is referenced. If more information is needed, please let me know (just didn't want to spam my script/ logs). Thanks. Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40484 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): SIP Request: Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:680]: parse_msg(): method: <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:682]: parse_msg(): uri: <sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:684]: parse_msg(): version: <SIP/2.0> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:1300]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK42e9.4a962087000000000000000000000000.0>; state=16 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [To] type 3 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=ueisi99vbf Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): <To> [61]; uri=[sip:1000@fakedomain.com] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (44)[<sip:1000@fakedomain.com>], to tag (10)[ueisi99vbf] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [From] type 4 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [CSeq] type 5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Call-ID] type 6 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Content-Length] type 12 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:187]: get_hdr_field(): content_length=0 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Max-Forwards] type 8 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1132060721 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1013]: tps_request_sent(): handling outgoing request (1, 1) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:417]: tps_pack_message(): compacted headers - x_via1: [SIP/2.0/UDP <KAMAILIO PUBLIC IP>:5060;branch=z9hG4bK42e9.4a962087000000000000000000000000.0](85) - x_via2: [](0) - x_vbranch1: [z9hG4bK42e9.4a962087000000000000000000000000.0](46) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:539]: tps_pack_message(): compacted headers - a_rr: [](0) - b_rr: [](0) - s_rr: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:544]: tps_pack_message(): compacted headers - as_contact: [](0) - bs_contact: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1027]: tps_request_sent(): no x-uuid header - nothing to do Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/msg_translator.c:1799]: check_boundaries(): no multi-part body Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [topos_mod.c:567]: tps_msg_sent(): new outbound buffer generated Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:686]: send_prepared_request_impl(): uac: 0x7f22f94448e8 branch: 0 to 10.0.0.47:38606 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:432]: send_bye(): BYE sent to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to callee Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1001@<UA PUBLIC IP>:5066>Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40482

3 3

Kamailio not responding after stress test
by Igor Olhovskiy 27 Jan '23

27 Jan '23

Hello, I'm continuing investigations on Kamailio and stress test. Got it again in the state where it's not accepting any new TCP/TLS connections (UDP still works though), but all looks good from lsof/netnstat part, like system is not reporting any zombie connections. Restart of Kamailio process helps This time I got output of kamctl trap Put it here: https://pastebin.com/iYrNZ8U9 kamailio --version version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 14:01:01 Oct 18 2022 with gcc 4.8.5 It's statically linked with tlsa pointing on openssl-1.1.1q Settings related to TLS are: fork=yes children=4 tcp_children=12 enable_tls=yes tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Can you please help to read gdb output and understand where I missed in config? Thanks in advance! -- Best regards, Ihor (Igor)

3 7

Kamailio 5.6 on AlmaLinux 8 TLS crash
by Ihor Olkhovskyi 27 Jan '23

27 Jan '23

Hello! Currently testing Kamailio on Almalinux 8 (stated to be RHEL 8 binary compatible) and got this while trying to get TLS working: /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: ca_path='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:361]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:365]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:372]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:379]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:383]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:386]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:389]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:736]: set_verification(): TLSc<default>: Server MUST present valid certificate /usr/sbin/kamailio[32862]: ERROR: rtpengine [rtpengine.c:3070]: send_rtpp_command(): can't send command "ping" to RTPEngine <udp: 127.0.0.1:19999> /usr/sbin/kamailio[32862]: ERROR: rtpengine [rtpengine.c:2919]: rtpp_test(): proxy did not respond to ping /usr/sbin/kamailio[32883]: INFO: snmpstats [snmpstats.c:519]: spawn_sysUpTime_child(): A snmpVersion parameter was not provided. Defaulting to 3 /usr/sbin/kamailio[32883]: ERROR: snmpstats [snmpstats.c:561]: spawn_sysUpTime_child(): snmpget failed to run. Did you supply the snmpstats module with a proper snmpgetPath parameter? The kamailioSIPServiceStartTime is defaulting to zero /usr/sbin/kamailio[32881]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) /usr/sbin/kamailio[32870]: CRITICAL: <core> [core/mem/q_malloc.c:145]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(0, abcdefed)[0x7fe8375d3820:0x7fe8375d3858]! Memory allocator was called from tls: tls_init.c:293. Fragment marked by tls: tls_init.c:293. Exec from core/mem/q_malloc.c:391. /usr/sbin/kamailio[32870]: CRITICAL: <core> [core/mem/q_malloc.c:153]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten [0x7fe8375d3790:0x7fe8375d37c8] - fragment marked by tls: tls_init.c:293 systemd[1]: Started Process Core Dump (PID 32923/UID 0). systemd-coredump[32924]: Process 32870 (kamailio) of user 0 dumped core.#012#012Stack trace of thread 32870:#012#0 0x00007fe841786aff raise (libc.so.6)#012#1 0x00007fe841759ea5 abort (libc.so.6)#012#2 0x00000000006eec2f qm_debug_check_frag (kamailio)#012#3 0x00000000006f09a6 qm_malloc (kamailio)#012#4 0x00000000006fc930 qm_shm_malloc (kamailio)#012#5 0x00007fe83b5b9c31 ser_malloc (tls.so)#012#6 0x00007fe83fa4090d CRYPTO_zalloc (libcrypto.so.1.1)#012#7 0x00007fe8364b8163 sc_hash_type (libnetsnmp.so.35)#012#8 0x00007fe8364b49df hash_engineID (libnetsnmp.so.35)#012#9 0x00007fe8364b4b11 search_enginetime_list (libnetsnmp.so.35)#012#10 0x00007fe8364b4f8e set_enginetime (libnetsnmp.so.35)#012#11 0x00007fe8364b4885 init_snmpv3_post_config (libnetsnmp.so.35)#012#12 0x00007fe8364b9c92 snmp_call_callbacks (libnetsnmp.so.35)#012#13 0x00007fe836f3601b register_with_master_agent (snmpstats.so)#012#14 0x00007fe836f37f14 run_alarm_check (snmpstats.so)#012#15 0x0000000000531cde compat_old_handler (kamailio)#012#16 0x000000000053284e slow_timer_main (kamailio)#012#17 0x000000000042e3e1 main_loop (kamailio)#012#18 0x0000000000438d14 main (kamailio)#012#19 0x00007fe841772d85 __libc_start_main (libc.so.6)#012#20 0x000000000041cfce _start (kamailio) /usr/sbin/kamailio[32915]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 24 /usr/sbin/kamailio[32858]: ALERT: <core> [main.c:774]: handle_sigs(): child process 32870 exited by a signal 6 ... version: kamailio 5.6.3 (x86_64/linux) fe0c4d flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: fe0c4d compiled on 11:02:26 Jan 19 2023 with gcc 8.5.0 Same config is starting without issues on CentOS 7. OpenSSL version is 1.1.1k Kamailio is installed from official RPM repos. Is core dump would be useful here or best to compile from source and redo? -- Best regards, Ihor (Igor)

3 6

topos vs topoh, any reason for putting a private IP in the Contact header?
by Benoit Panizzon 27 Jan '23

27 Jan '23

Hi I'm trying to work around Route-Header and Via Issues with the two topology hiding modules topoh and topos and trying to figure out, which one works better for our environment. My conclusion so far: topos creates very clean header, but needs a database or redis. I'm always reluctant in adding more components which could fail or cause load. topoh also works, but it worries me a bit that according to the manual: https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask… a private IP is used to mask the contact header. I have come across a CPE which, as far as I see, is misbehaving by ALWAYS sending a PRACK to the Hostname or IP found in the Contact header, and ignoring Route and Via Header. So why using a private IP as 'mask' and not the IP or Hostname of the actual kamailio instance? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Re: [sr-dev] Is there any Function or parameter to check Initial Invite Response on Kamailio
by Henning Westerholt 25 Jan '23

25 Jan '23

(removed sr-dev from CC) Hello, you can do that with the reply_route and/or onreply_route, for an example check out example cfg or this: https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_reply Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: HimaBindu G <himabindu.garadareddy(a)gmail.com> Sent: Tuesday, January 24, 2023 3:16 PM To: sr-users(a)lists.kamailio.org; sr-dev(a)lists.kamailio.org Subject: [sr-dev] Is there any Function or parameter to check Initial Invite Response on Kamailio Hi All, Can you please suggest any function or parameter to check Initial Invite Response from the backend. Thanks & Regards, Hima Bindu.

1 0

RTPengine multiple setID sockets
by John Hardiman 24 Jan '23

24 Jan '23

Hi All, Hope you are all well? I am hoping you can help with a bit of syntax I am struggling to get working in the intended way. (We are using Kamailio 5.5.3 and using Kemi/python.) The scenario is in AWS where we have separate Kamailio and RTPengine instances, with a cluster of FreeSWITCH EC2's on private addresses only. In the current setup, all RTP routes via the NAT gateway and back to the private IP of the selected FreeSWITCH (and vice versa). This works in the current config, but I would want this traffic not going via the NAT gateway and use the private address and route internally on the VPC. (These EC'2 have only 1 network interface and that cannot be changed for this work.) So anything between FreeSWITCH and Kamailio/RTPengine uses the private IP's for media, the rest use public IP's. It is good to note that the RTPengine's are the same in each set (3 total RTPengine's), just using the public/private IP's based on set ID. My first aim is to use KSR.permissions.allow_source_address() to check if the source address is from the FreeSWITCH cluster and set the rtpengine setID based on this. I have used this method for different parts of the config and it works. I am failing at this first step it seems, as no matter what I use in the function (below) it defaults and uses the public IP's in SDP. (No matter if I use the same setID in each set) kamailio.cfg snippet: # ----- rtpengine params ----- modparam("rtpengine", "setid_avp", "$avp()") # Public IP's ID 1: modparam("rtpengine", "rtpengine_sock", "1 == udp:5.5.5.5:2223 udp:6.6.6.6:2223 udp:7.7.7.7:2223") # Internal IP's ID 2: modparam("rtpengine", "rtpengine_sock", "2 == udp:10.0.0.1:2223 udp:10.0.0.2:2223 udp:10.0.0.3:2223") Kemi file snippet: # RTPengine Options for different SetID def ksr_route_rtpengine(self, msg): if KSR.textops.has_body_type("application/sdp") > 0 : # Source address is private subnet - use RTPengine set 2 (private IP's) if KSR.permissions.allow_source_address(400) == 1: KSR.rtpengine.set_rtpengine_set(2) KSR.rtpengine.rtpengine_manage( "RTP/AVP replace-session-connection replace-origin port-latching ICE=remove") else: # Source address is NOT private - use RTPengine set 1 (public) KSR.rtpengine.set_rtpengine_set(1) KSR.rtpengine.rtpengine_manage( "RTP/AVP replace-session-connection replace-origin port-latching ICE=remove") KSR.xlog.xlog("L_INFO", "RTPengine handling Block") return 1 Group 400 is the private subnet of the FreeSWITCH clusters we have behind on private addresses and the list contains the subnet of private IP's. Using the above, the Kamailio starts fine and understands/loads each set of RTPengine's and calls work with audio. but regardless of the set number used (even only using set 2 in both) it defaults to public IP's in the SDP. (This is the only place in the config where this function is defined) My SDP from FreeSWITCH egress to Kamailio/RTPengine has private IP's in the SDP. The only thing I do not fully understand is the setid_avp - specifically $avp() block and if this needs to stay empty, or be duplicated with the set ID in each one. Any help would be greatly appreciated in understanding where I am going wrong here. Thanks, John.

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[SR-Users]: Port number in To: field
by Olivier 24 Jan '23

24 Jan '23

Hello, While trunking with an Oracle SBC, I was lately required to include a port number in INVITE To: field. Would you qualify this as a "legal requirement" (according to SIP Standards) ? Best regards

4 3

dialog: db_mode => 1, enable_dmq => 1 but entries not replicated to 2nd node DB
by Benoit Panizzon 24 Jan '23

24 Jan '23

Hi I am struggling with DMQ and dialog DB storage leading to orphan or duplicate entries in database and not loading all dialog information upon restart. usrloc on the other side, also using local db storage and dmq, works flawlessly even when restarting one node. All location information is replicated between the databases via DMQ. So trying to find what is different. usrloc module: modparam("usrloc", "db_url", DBLOCAL) modparam("usrloc", "db_mode", 2) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) dialog module: modparam("dialog", "db_url", DBLOCAL ) modparam("dialog", "db_mode", 1 ) modparam("dialog", "db_skip_load", 0) modparam("dialog", "enable_dmq", 1) via kamcmd dlg.list I see that dialog information is being replicated between the two nodes. But I can only find the dialog in the local database of the node which is handling the call. Is this the expected behavior? I also just found how to create duplicate entries in the database... * Run a call on Node A, verify the dialog information is store in the local database and replicated to node B via DMQ. * Restart Kamailio on Node A. There I suspect, node A is restoring Dialog Information via DMQ from Node B and loading the same information from database, leaving duplicates in dialog_vars. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio World 2023 - Call For Presentations
by Daniel-Constantin Mierla 23 Jan '23

23 Jan '23

Hello, I would like to announce that Call for Presentations at Kamailio World 2023 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k2023/call-for-speakers/ The 11th edition of the conference returns to Berlin, Germany, during June 5-7, 2023, as an in-person event. Expect a large range of participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Performances issue when freeing shared memory in custom module (Kamailio 5.5.2)
by Chaigneau, Nicolas 23 Jan '23

23 Jan '23

Hello, I'm encountering performance issues with Kamailio (5.5.2). I'm using a custom Kamailio module that loads routing data in memory, using Kamailio shared memory. This routing data is very large. It can be fully reloaded through a Kamailio RPC command (which is done once each day). When reloading, two sets of data are maintained, one "loading" and another "current" (the latter being used to handle SIP requests). When loading of the new data is finished, it is swapped to "current". Then, memory of the old (now unused) data is freed. I've noticed that when Kamailio is freeing the old data, there is a very significant performance impact on SIP requests. This is surprising to me, because the SIP requests do not use this old data. This is not a CPU issue, idle CPU% is at about 99% at that moment. I'm using the following functions : - shm_mallocxz - shm_free From what I understand, shm_free is actually "qm_shm_free" defined in "src\core\mem\q_malloc.c" (the default shared memory manager being "qm"). I've noticed that there is also a variant shm_free_unsafe ("qm_free"), which does not perform locking. I'm wondering if the lock could be the cause of my performances issues ? (But I'm not sure how this could be possible, because although the SIP requests need to access the shared memory allocated, they do not use directly the functions from the share memory manager.) If the performances issues are causes by the lock, could I use the unsafe version "safely" ? (considering that it is guaranteed that the old data cannot be used by anyone else) Thanks for your help. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 6

dialog.dlg_filter_mode has ambiguous definition with double negation
by Дилян Палаузов 23 Jan '23

23 Jan '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/dialog#dialog.p.dlg_fil… says for dialog.dlg_filter_mode (int) Set dialog fitering mode, which can specify what dialogs are processed. Its value can be a combination (the sum) of following flags: 1 - do not send keepalives and do not execute timeout function if dialog is not local (if the associated bind address is not a local socket). Default value is “0”. As you can see, the definition of 1 is a double negation and there is no definition for 0. What means 0: - do not send keepalives and do not execute timeout function if dialog is local - send keepalives and execute timeout function if dialog is not local - send keepalives and execute timeout function if dialog is local

1 0

Does tm.t_relay() create a transaction?
by Дилян Палаузов 20 Jan '23

20 Jan '23

Hello, the documentation for tm.t_newtran() says „This is the only way a script can add a new transaction in an atomic way.“ Moreover tm.t_send_reply() “creates the transaction if it does not exist (executing internally t_newtran()) and sends a stateful reply (executing internally t_reply())”. This means that only t_newtrans() and t_send_reply() can create transactions in an atomic way. The documentation of tm.t_relay() does not say whether the function creates a transaction. It also does not say what happens if t_reply() is not called after t_newtrans(). But for tm.t_set_retr() and tm.t_set_retr() is written “If the transaction is already created (e.g called after t_relay() or in an onreply_route) …” This suggests that tm.t_relay() does create atomically or not-atomically a transaction. - What does it mean to create a transaction in a non-atomic way? - Does tm.t_relay() create a transaction? Greetings Dilyan

2 2

sslmode in db_postgres
by Centelles Martin, Joel 20 Jan '23

20 Jan '23

Hello everybody, I'm working on a kamailio instance as a basic registrar service and I thought to use db_postgres to connect a postgres service only allowing SSl connections. I see that that module doesn't provide any parameters to select sslmode and I was wondering if this module actually supports TLS connections. Does anybody know if this is supported and, if so, how to enable it? or should that support be added into db_postgres module code? Thank you very much in advance. BR, Joel Centelles RTC Lead Engineer | Care Communications R&D joel_centellesmartin(a)baxter.com<mailto:joel_centellesmartin@baxter.com> | baxter.com<https://baxter.com/> [cid:9ee6bcbd-9c30-45ec-bb43-8f439f19f724]<https://baxter.com>

1 0

[core/tcp_main.c:617]: _wbufq_add(): (1244714 bytes): write queue full or timeout
by Benoît Panizzon 20 Jan '23

20 Jan '23

Hi I'm trying to get dlg.list on a busy system via jsonrpc and xhttp. I already increased: tcp_conn_wq_max=2048 tcp_wq_max=20480 modparam("ctl", "binrpc_max_body_size", 2048) modparam("ctl", "binrpc_struct_max_body_size", 2048) But it looks like the tcp write queue overflows up at shy over 1MB reply no matter what I change. Any ideas what else to toggle? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 9

What are extended flags?
by Дилян Палаузов 20 Jan '23

20 Jan '23

Hello, The corex module manages extended flags with functions the setxflag(), isxflagset(), and resetxflag(). tmx.t_flush_xflags() does “flush the extended flags from the current SIP message into the already created transaction. It makes sense only in routing block if the transaction was created via t_newtran() and the extended flags have been altered since. It is not needed to execute this function when using t_relay() (or similar tm relay functions, xflags are synchronized automatically in that case).” The documentation of the tm module does not say anything about extended flags - in particular it does not say that t_relay() synchronizes them. Moreover, while the corex module utilizes the extended flags, the documentation of tm and tmx do not state, that they depend on the corex module. So, what are extended flags? Greetings Дилян

1 0

SIP Trunk with multiple phone numbers for account
by Carlos Escalona 19 Jan '23

19 Jan '23

Hello all! I have a question. In my system we use a series of VoIP providers to receive external calls to our telephone numbers, for the same provider we have several registered numbers and we need our providers to send that information in some way, so far it does not seem that there is a common way to do it (report the number that was dialed by the person who made the call). From what I understand, SIP does not have a place for that information because there are no things like phone numbers in the specification, my question is, is there a correct place to send such information, or is it entirely up to each provider? (Now, the most common is to have in the RURI and To header, the id of the trunk registered in the uacreg table) I have seen a couple of options that seem to me the most “correct”: send the information in the To header or send in a custom header. I really appreciate any help you can provide. Atenciosamente, [image: photo] <https://www.techer.com.br/> *Carlos Escalona* IT - Development 41 3073-0091 | R 1011 www.techer.com.br -- O conteúdo desta mensagem é confidencial. Se você não se encontra na lista de destinatários ou tenha recebido por engano, não a copie, imprima, envie, ou utilize, de qualquer forma, seu conteúdo. Neste caso, destrua a mensagem e, por favor, notifique o remetente.

3 3

Setting socket_workers does not work if using interface name
by Kaufman 19 Jan '23

19 Jan '23

At a quick glance it looks like using socket_workers when the listen parameter uses the interface name does not work. For example: children = 4 socket_workers=2 listen=udp:10.10.10.10:5060 Gives two listeners on udp:10.10.10.10:5060 children = 4 socket_workers=2 listen=udp:eth0:5060 Gives four listeners on the socket. Note this is just a quick cursory observation. Is it by design or a design limitation? Kaufman

3 3

RTC/Kamailio dinner at Fosdem 2023
by Daniel-Constantin Mierla 17 Jan '23

17 Jan '23

Hello, Fosdem 2023 returns to Brussels, Belgium, for a physical event and I'm writing to see if there are many people from the community going to the event and want to organize again the usual RTC/Kamailio dinner on Saturday evening (Feb 4, 2023). Each person should pay for himself, likely to be some price increase comparing with the last one. If there is enough interest, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like in the past, if available). Reply if you want to join the dinner and say how many other people are joining you. Note that RTC Devroom got at this edition a half a day room allocation, on the afternoon of Feb 5, 2023 (Sunday). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

7 8

Re: rtp lost package in 300 concurrent calls and above
by Henning Westerholt 16 Jan '23

16 Jan '23

Hello, please keep the list in CC. Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine: https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com> Sent: Monday, January 16, 2023 3:35 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above thanks for your reply my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server. I've selected rtp media server because of playback of a voice file over incoming calls in the future usage On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, do you observe (too) high load on the system with the standard system management tools? If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com<mailto:keshavarzianpoorm@gmail.com>> Sent: Monday, January 16, 2023 2:50 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] rtp lost package in 300 concurrent calls and above Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

4 10

rtp lost package in 300 concurrent calls and above
by Mohammad Reza Keshavarzianpoor 16 Jan '23

16 Jan '23

Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

4 5

Authenticated ACK: $au not set?
by Benoit Panizzon 16 Jan '23

16 Jan '23

Hi Gang I have this code snipplet: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); if ($aU == $null) { xlog("L_INFO", "$cfg(route): no auth user, send challenge\n"); } auth_challenge("$fd", "0"); exit; } } Please don't ask if this makes sense, it's a constructed snipplet to show the situation. If $rm is INVITE, this works fine, no challenge is being sent, $au is populated. If $rm is ACK (some CPE seem to send the Credentials with each message) then $au is $null, auth_challenge is called and fails. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Parallel call forking with call authorization
by Mathias Schneuwly 13 Jan '23

13 Jan '23

Hi As a first notice; I already wrote to the mailing list, but without registration. I read afterwards that I should register first because of spam reasons. If this is a duplicate, then I'm sorry. I'm new to mailing lists... I've an issue with parallel call forking together with call authentication. I hope somebody can give me some hints about what the problem could be and how I could solve this problem. I'm using Kamailio 5.5.4 with the python KEMI binding. Kamailio is in front of an Asterisk, which requires call authorization. RTPEngine is also inplace which does SRTP to RTP bridging. Simple calls from and to Asterisk through Kamailio and RTPEngine work without any issue. So the basic setup seems to work fine. The problem starts when I do parallel call forking where both calls are sent to Asterisk. If I use "KSR.tm.t_on_failure("trunk_authentication")" for authentication, the first branch is authenticating and the device rings, but the second branch does not authenticate. Well, that's not 100% true, if the SIP timeout appears on the first branch, I can see in the log, that the second branch enters the trunk_authentication method, but then, the call is already terminated. Call forking to two devices (not behind Asterisk and without call authentication) works also as expected. I made some trials with "KSR.tm.t_on_branch_failure("trunk_authentication")" but in this case, the behavior is strange. Then the authentication for both branches work and both devices are ringing. If a device hooks up, the second device does not stop ringing. And after a few seconds, the first device cancels the call, because it did not get the ACKs. I can also see that the CANCEL is sent from Kamailio to Asterisk, but Asterisk does not react to it. For trunk authentication, I'm using the implementation from the well known examples plus a few own stuff. Btw, NAT is not enabled...: def trunk_authentication(self, msg): scode = KSR.pv.get("$T_reply_code") self.log('notice', 'trunk_authentication {}'.format(scode)) if WITH_NAT and self.ksr_route_natmanage(msg) == -255: return 1 # Call manage_rtp_engine to destroy RTPEngine if it has been used. call_context = self.get_call_context(msg=msg) callee_object = call_context.callee.get_callee_object() if callee_object.connection_type: call_context.manage_rtp_engine() if KSR.tm.t_is_canceled() > 0: return 1 if scode in [401, 407]: KSR.pv.sets('$avp(auser)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_username)')) KSR.pv.sets('$avp(apass)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_password)')) if KSR.uac.uac_auth(): # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if KSR.is_method_in('IBSU'): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") self.log('notice', 'Authentication Asterisk trunk') if KSR.tm.t_relay() < 0: KSR.sl.sl_reply_error() return 1 Any ideas? Best regards Mathias

3 6

src/modules.lst - difference between skipping a module from compile list and excluding a module from compile list
by Дилян Палаузов 13 Jan '23

13 Jan '23

Hello, the generated src/modules.lst file contains: # the list of modules to skip from compile list skip_modules= … # the list of modules to exclude from compile list exclude_modules= … For me excluding a module from the compile list and skipping the module when compiling is the same. Please describe what is the difference between skipping and excluding, either in modules.lst, at https://www.kamailio.org/wikidocs/devel/makefile-system/ or at https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . Thanks Дилян

2 1

Newbie question
by Joao Paulo Vaz 12 Jan '23

12 Jan '23

Hello masters, I am a newbie who recently got to know Kamailio. I have a scenario and I would like to know if this tool would suit: a central asterisk server with all user accounts and three remote units with Kamailio acting as a proxy. Is there any module or configuration that if the central asterisk server fails to respond, would the remote proxies act in survival mode, allowing local communication by users until the central asterisk server returns a response? Att, Joao Paulo

3 4

unsubscribe.pls
by Stanly Levyy 11 Jan '23

11 Jan '23

*** THIS E-MAIL MESSAGE AND ANY DOCUMENTS ATTACHED IS/ARE INTENDED ONLY FOR THE USE OF THE INDIVIDUAL(S) OR ENTITY TO WHICH IT IS ADDRESSED; AND MAY CONTAIN INFORMATION THAT IS LEGALLY PRIVILEGED, CONFIDENTIAL AND/OR EXEMPT FROM DISCLOSURE UNDER APPLICABLE LAW, including the Electronic Communications Privacy Act, 18 U.S.C. Secs 2510-2521 *** On Wednesday, January 11, 2023, 01:36:44 AM PST, <sr-users-request(a)lists.kamailio.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Performing async_http calls on stateless Loadbalancers (Julius Flohr) 2. Re: Releasing v5.6.3 (Daniel-Constantin Mierla) 3. Performing async_http calls on stateless Load balancers (Lewis Hutchinson) ---------------------------------------------------------------------- Message: 1 Date: Tue, 10 Jan 2023 18:50:44 +0100 From: Julius Flohr <flohr(a)sipgate.de> Subject: [SR-Users] Re: Performing async_http calls on stateless Loadbalancers To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Alles wird besser <alleswirdbesser(a)sipgate.de> Message-ID: <34C7487C-0900-4C49-BBB4-49A0291D83E6(a)sipgate.de> Content-Type: multipart/alternative; boundary=Apple-Mail-0AF7C672-1C85-4A98-9DDC-EED95FFB4B7E A message part incompatible with plain text digests has been removed ... Name: not available Type: text/html Size: 7117 bytes Desc: not available ------------------------------ Message: 2 Date: Wed, 11 Jan 2023 08:30:17 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [SR-Users] Re: Releasing v5.6.3 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org> Message-ID: <454ae3b5-a810-f393-4196-e71c4482e9d3(a)gmail.com> Content-Type: text/plain; charset=UTF-8 Hello, the usual note that preparing the release of Kamailio v5.6.3 starts soon, therefore any commits to branch 5.6 should be announced first on sr-dev mailing list and wait for the ack before pushing to repo. Once the announcement is out, commits can go in the branch directly. Cheers, Daniel On 06.01.23 16:44, Daniel-Constantin Mierla wrote: > Hello, > > I am considering to release Kamailio v5.6.3 (out of branch 5.6) next > week (likely on Wednesday, Jan 11, 2023). If anyone is aware of > issues not yet on the bug tracker, report them there asap in order to > have a better chance to be fixed. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com ------------------------------ Message: 3 Date: Wed, 11 Jan 2023 09:20:28 +0000 From: Lewis Hutchinson <lewis.hutchinson(a)missionlabs.co.uk> Subject: [SR-Users] Performing async_http calls on stateless Load balancers To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Cc: "flohr(a)sipgate.de" <flohr(a)sipgate.de>, "alleswirdbesser(a)sipgate.de" <alleswirdbesser(a)sipgate.de> Message-ID: <CWXP265MB2517CB6E3E8188F3E3BB5B2AC5FC9(a)CWXP265MB2517.GBRP 265.PROD.OUTLOOK.COM> Content-Type: text/plain; charset="us-ascii" Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list -- sr-users(a)lists.kamailio.org To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org ------------------------------ End of sr-users Digest, Vol 212, Issue 33 *****************************************

1 0

Performing async_http calls on stateless Load balancers
by Lewis Hutchinson 11 Jan '23

11 Jan '23

Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis

1 0

Releasing v5.6.3
by Daniel-Constantin Mierla 11 Jan '23

11 Jan '23

Hello, I am considering to release Kamailio v5.6.3 (out of branch 5.6) next week (likely on Wednesday, Jan 11, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 1

Help with rtpengine_delete()
by Jose Figueroa 10 Jan '23

10 Jan '23

Hello everyone! I've been struggling with a kamailio scenario with the rtpengine_delete. My scenario is the following: I've setup a Kamailio multihomed, since some of my vendors has provided us a direct link using a different local subnet in eth2 and I have my public IP in eth1. The kamailio i've been working on is using app_lua module. SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth2) 10.203.5.8 <------> Vendor 10.203.5.20 100.XXX.XXX.22 Basically, I receive a call from public IP eth1 and according to the logic from the vendor I forward statefully the call to 10.203.5.20 in second branch. I've been reading a lot of issues and documentation such as:[#840 rptengine], [#875 rtpengine] and [#2222 kamailio], but I don't really get working the rtpengine_delete() correctly, since I go to netstat -plan|grep rtpengine and I see some sessions still that are not freed. My second case is the following when I receive a 503 from another GW: SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth1) 200.XXX.XXX.14 <-----------> Vendor GW 233.XXX.XXX.19 100.XXX.XXX.22 In that case I still see the rtpengine ports, mostly with my public IP 200.XXX.XXX.14 but some of them with the LAN 10.203.5.8 IP. Also, something curious I see on my 2nd use case is the following: https://pastebin.com/raw/B4vCgExX So, it would be great if someone can point me to the right direction, how and when I should use rtpengine_delete in order to finish the rtpengine sessions correctly and the ports can be freed up. Any help will be great! [840] https://github.com/sipwise/rtpengine/issues/840 [875] https://github.com/sipwise/rtpengine/issues/875 [2222] https://github.com/kamailio/kamailio/issues/2222 -- Regards, Jose Figueroa

3 2

Why do htable Keys have two colons?
by Noah Mehl 10 Jan '23

10 Jan '23

Hey everyone, Whenever I read documentation for htable usage, and other examples of other scripts in Kamailio, htable keys are typically named: something::something Examples are: $au::auth_count, $ci::srcip, join::$rU Why are there two colons (as opposed to a | )? Is this just a standard semantic thing in Kamailio? Is there something about that double character that’s unusual and easy to parse with SIP values? Or is it just what people do, and that’s a good choice to continue for all other Kamailio script writers to have understanding? Any insight would be appreciated. Thanks! ~Noah

3 6

Kamailio Crypto module and 3rd party openssl
by Arsenijs Kabihno 10 Jan '23

10 Jan '23

Hello, I am trying to make use of crypto module. The scenario is the following: 1. Encrypt plaintext passwords on kamailio side using crypto_aes_encrypt function of crypto module 2. Decrypt them outside of kamailio using 3rd party tools, such as PHP openssl_decrypt Crypto module parametrs are the following: modparam("crypto", "salt", "") modparam("crypto", "key_derivation", 0) If encryption and decription are handled by kamailio, the initial password matches the decrypted string (initial password -> encrypted string -> decrypted string) For example: sql_query("ds_dburl", "select secret from kamailio_sip_buddies where username='$au'", "secret"); $avp(secret_plain) = $dbr(secret=>[0,0]); //value: 'kamailio' crypto_aes_encrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(encrypted)"); //value: 'qfOqTMoJMgGAherGCqLRRQc4zTlqsEj3MEtcORurFf0=' crypto_aes_decrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(decrypted)"); //value: 'kamailio' But in case I try to decrypt the value of $var(encrypted) using PHP openssl_decrypt or online tool like https://www.devglan.com/online-tools/aes-encryption-decryption, I am getting the result 'kamailio' prepended by 16 random bytes, like: 6“¤›¿Ð+IÊ\ñžïZkamailio or 6��+I\��Zkamailio I believe this 16 bytes are actually random initialization vector in some encoding. Possibly kamailio is trimming them from result here: https://github.com/kamailio/kamailio/blob/master/src/modules/crypto/crypto_… Moreover, if secret was encrypted using crypto_aes_encrypt and random IV, you can use any IV while decrypting and get 16 different bytes+result Please advice if someone have used crypto module in similar way and what are the possible ways to acheive results consistency

2 5

pkg memory leak when acc module cdr_enabled
by Juha Heinanen 10 Jan '23

10 Jan '23

In latest stable K release, we noticed pkg memory leak (pgk memory usage increases by each processed call). It turned out that the leak goes away if acc module cdr_enable is not enabled. Could be a bug in dialog or acc module. Any debug instructions if the bug is not obvious? -- Juha

4 14

wrong char / bad PVE format
by Nathan Stratton 09 Jan '23

09 Jan '23

Trying to rename rtpengine files with: exec_msg("mv /var/spool/rtpengine/rtpengine-recording/$ci* /var/spool/rtpengine/rtpengine-recording/$ci.wav"); But get this error: Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching pvar "ci.wav" Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [v/118] in [$ci.wav] at [6 (0)] Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/sr_module.c:1157]: fix_param(): bad PVE format: "mv /var/spool/rtpengine/rtpengine-recording/$ci* /var/spool/rtpengine/rtpengine-recording/$ci.wav" Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/mod_fix.c:564]: fixup_spve_null(): Cannot convert function parameter 1 to space Any ideas? ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

3 2

RABBITMQ and AWS
by Fred Posner 06 Jan '23

06 Jan '23

Has anyone had any success connecting to RabbitMQ via a TLS 1.2 connection (such as the AWS MQ product)? Fred Posner https://pgpx.io

3 3

TypeError exception in Kemi with python3
by Michel Pelletier 06 Jan '23

06 Jan '23

Hello, I am using Kemi with Python3. I am getting a TypeError exception when using the following: KSR.presence.handle_subscribe() 40(208) INFO: PY3 {SUBSCRIBE}: app_python3 [apy_kemi.c:1839]: sr_apy_kemi_exec_func(): alert - action KSR.presence.handle_subscribe(...) took too long [2004761 ms] (file:(null) func:(null) line:733) 40(208) ERROR: PY3 {SUBSCRIBE}: app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 152, in ksr_request_route if self.ksr_route_presence(msg) == -255: File "/etc/kamailio/kamailio.py", line 733, in ksr_route_presence KSR.presence.handle_subscribe() SystemError: <built-in function handle_subscribe> returned a result with an error set Using the following appears to be a good workaround, as it does not exhibit the issue at all: KSR.presence.handle_subscribe_uri(KSR.pv.get("$fu")) Cheers, Michel Pelletier

1 0

Call drop when reinvite is proceed
by nathan.castillo＠snapcom.fr 05 Jan '23

05 Jan '23

Hello all, I'm trying to figure out one problem on my Kamailio server. When kamailio receive a reinvite from the infra, I get the error message "477 - Unforunately error on sending to next hop" and the call drop. I don't understand why this message appears as the first invite is well answered and the call is going well. Any ideas how to resolve this issue ? Thanks in advance for your help. Regards, Nathan

6 6

Too many 200:OK sip responses
by Waqar 40 05 Jan '23

05 Jan '23

Dear All, For a normal call, Kamailio receives 200:OK sip response for the method invite only once but for some calls, it receives 200:OK too many times around 6 to 7. And I also do not receive BYE or any other packets for this kind of call. How can I detect such calls in Kamailio and cancel them? I am inserting data of the calls in the database when the call occurs and deleting it when BYE is received but for these calls as I don't receive BYE so I am unable to delete the record of such calls. Is there a way to detect such calls? Like How can I count if the number of 200:OK responses is more than one or anything like that? Regards Vicky

4 4

How to have Kamailio retry backends
by Unai Rodriguez 04 Jan '23

04 Jan '23

Dear List, We’re using Kamailio to load balance MRCP requests to multiple backend groups with a configuration as follows: # kamailio.cfg ... ... route[DISPATCH] { if($ua=="mrcp_backend_1") { if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } } if($ua=="mrcp_backend_2") { if(!ds_select_dst("2", "4")) { send_reply("404", "No destination"); exit; } } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } ... ... # dispatcher.list 1 sip:mrcp01.server.int:8060;transport=tcp 1 sip:mrcp02.server.int:8060;transport=tcp 2 sip:mrcp03.server.int:8060;transport=tcp 2 sip:mrcp04.server.int:8060;transport=tcp With this configuration, Kamailio load balances the initial SIP INVITE among the MRCP servers. After the INVITE, the service communicates directly to the MRCP servers via SIP (for hanging up the call), MRCPv2 (for sending speech control messages), and RTP (for sending audio). We would like to implement a configurable number of retries, so that if a particular backend times out, Kamailio would retry X times to other backend(s). In short, something equivalent to HAProxy’s retries, but for Kamailio. This probably implies having Kamailio always as part of our communication (not just load balancing the initial SIP INVITE). I haven’t been able to find much information about this, could someone provide some pointers? Thank you so much With best wishes, Unai Rodriguez

3 6

Recover call in ringing state
by Devang Dhandhalya 04 Jan '23

04 Jan '23

Hello All I have below setup : SBC -> kamailio -> Media server(remote) -> web-client Call comes from sbc to kamilio and it relays to Remote server to end user, In Ringing state if remote server goes down then in kamailio there is any way to Recover that ringing call like we can generate new invite for other Remote server which is active state, for relaying call i am using dispatcher module.. I tried with tm.cancel RPC command but it works only if the remote server is running , when I crash the remote server at that time tm.cancel is not generating cancel to remote server. Please suggest if there is any other way to generate INVITE. Regards Devang Dhandhalya -- * <https://www.ecosmob.com/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 3

Job offer Kamailio/VoIP developer
by Henning Westerholt 03 Jan '23

03 Jan '23

Hello and Happy New Year, we are looking for a new Kamailio & VoIP developer colleague to join and grow our international team. More detail about the position can be found at: https://gilawa.com/en/about/careers If you are interested, please send me a (direct) email. Please include the information described in the linked job description. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Localy generate BYE message from ims_charging
by masoud96＠mail.ru 03 Jan '23

03 Jan '23

Hello! I have a problem with writing the BYE message from the ACC module to the database, because the messages are generated locally by the ims_charging module at the end of the credit and the flag will not work. You can help catch such local messages and make a record in the database.

2 2

Question about UAC and PUA modules routing behavior
by Jean-Francois Lemieux 03 Jan '23

03 Jan '23

Hello, I have some trouble to figure out how routing works when the request coming from the kamailio itself (used as a user agent per example) .I try to use presence PUA... modules. The issue I have is the NOTIFY requests aren't sent to the good destination; it doesn't seems to obey to the contact or other informations when the device did a SUBSCRIBE. I don't want to ask to troubleshoot the presence module directly I just want to know how Kamailio deals with request coming from Kamailio itself (in the case of a NOTIFY generated by PUA_... modules My mains questions are: Does the pua modules when generating request like NOTIFY will pass through request_route ? Does the pua modules when generating request like NOTIFY will be sent only when t_relay() is used or UA kamailio based request bypass all the logic of the Kamailio proxy behavior ? I'm trying to understand if I can modify the NOTIFY request to change destination before the "routing decision" is made by Kamailio Kamailio 5.6.0 used as : -proxy to remote pbx for call dialog -registrar/location server -media relaying and nat management -...and I try to use it as a presence server instead to use the pbx behind kamailio

3 2

Steering egress on replies
by Alex Balashov 03 Jan '23

03 Jan '23

Hi, Happy New Year! From what I understand, replies are always sent back out of the listener on which they were received, though `reply_to_via` (off by default) can override this and cause the exit listener to be recalculated based on a real-time assessment of the best way to reach the last Via hop. Two questions arise: 1) If turning `mhomed` off, can `$fs` be used to override the exit interface of locally generated replies, too? If so, where should it be done? In a request_route prior to a send_reply() / sl_send_reply() / t_reply() call? Or should it be done inside a local reply route or onsend_route? 2) Does this differ for locally generated replies vs. relayed replies, either statefully or statelessly? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Registration Proxy
by nutxase 02 Jan '23

02 Jan '23

Hi Guys, can someone point me in the right direction as to how i can use kamailio as a registration proxy similar to how opensips does mid_registrar? i want to proxy all sip registrations through kamailio to asterisk or freeswitch Sent with [Proton Mail](https://proton.me/) secure email.

3 2

Happy New Year 2023!
by Daniel-Constantin Mierla 01 Jan '23

01 Jan '23

A Happy, Peaceful And Prosperous Year 2023 to everyone using Kamailio and helping to improve the project! Looking forward to meeting many of you around this world during 2023 and collaborating in a friendly environment! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
by Lewis Hutchinson 30 Dec '22

30 Dec '22

Hi Brandon Apologies if you have provided this but do you have a PCAP of the INVITE / 400 Bad Request rather than the Pastebin of the INVITE The last time I had this was because of a missing /r/n on the SDP granted it was not with teams but still difficult to track down. Also, what is the MTU Size, have you checked to see if the is causing packet fragmentation and manifesting in a Bad Request. Are there any clues in the Teams - Call Analytics. Regards Lewis -----Original Message----- From: sr-users-request(a)lists.kamailio.org <sr-users-request(a)lists.kamailio.org> Sent: 29 December 2022 20:17 To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 211, Issue 67 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) (Brandon Armstead) 2. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) (Brandon Armstead) ---------------------------------------------------------------------- Message: 1 Date: Thu, 29 Dec 2022 12:15:16 -0800 From: Brandon Armstead <brandon(a)cryy.com> Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Message-ID: <CABAX3EoF+hfruLiV2Nh_g117_q8diaKxVEyp5hMFO=wyKbQidQ(a)mail.gmail.com> Content-Type: multipart/alternative; boundary="0000000000005692d905f0fd2538" Henning, There was no reply (reason) available in my case. - Brandon On Thu, Dec 29, 2022 at 11:48 AM Henning Westerholt <hw(a)gilawa.com> wrote: > Hello, > > > > as mentioned before, have a look to the 400 reply reason phrase. This > was done quite good from their side, better than many other vendors. > > Usually, Teams is not too bad if you are using a defined > infrastructure that do not introduce too many variables. We are > usually using a B2BUA in our customer projects for that purpose. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt - > https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskal > atan.de%2Fblog%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=QU%2BzOgPdbmRYh51a9PvptIvVjnTZrGxT8z1G7b2RRGQ%3D&reserved=0 > > Kamailio services - > https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgila > wa.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7C0% > 7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQI > joiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=PuDqkZ > QCj2IlxvRsLQGoKFv19K3uVE5RBxvcPKPODwQ%3D&reserved=0 > > > > *From:* Brandon Armstead <brandon(a)cryy.com> > *Sent:* Thursday, December 29, 2022 7:52 PM > *To:* Kamailio (SER) - Users Mailing List > <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 > BAD > REQUEST) > > > > Alex & Kaufman, > > > > Appreciate the feedback. The pastebin is one example of about > 1000+ iterations literally of changes. So I'm finally reaching out > :). I haven't tested the socket attribute in record-route change, I > will try this now. As for the angle brackets I've tried with and > without, etc (already) to no effect. The SDP unknown media type seems > to be a Poly thing, so I > *want* to rule this out as its not in my other SDP's and still receive > 400 bad request. > > One more iteration here we go... please feel free to let me know if > you have any other thoughts on the matter :). > > > > Thanks! > > > > - Brandon > > > > On Thu, Dec 29, 2022 at 8:08 AM Alex Balashov > <abalashov(a)evaristesys.com> > wrote: > > Yeah, the grammar says that rr-params are just generic-params, in > which case it's not. I didn't see that -- nicely spotted! > > I think that may be the basis of the 400 Bad Request. I'd be shocked > if it weren't. > > The broken clock of "Microsoft SIP" can still be right twice a day. > > > On Dec 29, 2022, at 10:31 AM, Kaufman <bkaufman(a)bcmone.com> wrote: > > > > In your top Record-Route you have: > > > > socket=; > > > > Not sure if that is legal. > > > > Kaufman > > > > -----Original Message----- > > From: Alex Balashov <abalashov(a)evaristesys.com> > > Sent: Thursday, December 29, 2022 9:02 AM > > To: Kamailio (SER) - Users Mailing List > > <sr-users(a)lists.kamailio.org> > > Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 > > BAD > REQUEST) > > > > Sorry to hear you're having to interoperate with Teams. It's a > > unique > form of sadism I wouldn't wish upon anyone. > > > > A few theories: > > > > 1) Microsoft doesn't like the "bare" Contact header-value here: > > > > Contact: > sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE:5061;transport=tls > > > > Unlike the careted one right above: > > > > P-Asserted-Identity: > > <sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE> > > > > A bare URI absent other header-params is of course completely legal, > > but > I'm really trying to get inside the imaginary world of antisocial > "Microsoft SIP" here. > > > > 2) Could it be that antisocial "Microsoft SIP" sends 400 as a way of > objecting to something in the SDP, e.g. where a non-demented SIP stack > would send "488 Not Acceptable Here" or "415 Unsupported Media Type"? > > > > I know you've said you tried multiple clients to rule that out, but > > I > wonder if the thing it's objecting to hasn't been ruled out that way. > > > > 3) I saw this media line in the SDP: > > > > m=application 41356 <unknown media type> > > > > What's that? > > > > -- Alex > > > >> On Dec 29, 2022, at 9:51 AM, Brandon Armstead <brandon(a)cryy.com> wrote: > >> > >> Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams) > >> always > result in a 400 BAD REQUEST. > >> Example invite below: > >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn > >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25 > >> 2Fpast&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7 > >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA > >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda > >> ta=5v20CvNnmdiHwfWMBX6tgjMlE1i2aKpFgTnuVFfpyl0%3D&reserved=0 > >> ebin.com%2FF1G1Ce59&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf > >> 739 > >> a4ff447f008dae9b0c9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C > >> 638 > >> 079242759152531%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjo > >> iV2 > >> luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=Cbff6B4g > >> Nr8 > >> A9zs89smJ129y8IyNM9%2B3zVkGhlzpa54%3D&reserved=0 > >> I've taken care to make sure numbers are all E.164 format in > From/To/Contact. I've also taken care to make sure that FQDN is used > in Contact and Record-Route header. > >> I've tried many different variations and have followed the SIP > information here: > >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn > >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25 > >> 2Flear&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7 > >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA > >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda > >> ta=ELk0UD1Pleq%2FG9j1%2BT18tiuFe0yjflimR9ztnW%2F7oPM%3D&reserved=0 > >> n.microsoft.com%2Fen-us%2Fmicrosoftteams%2Fdirect-routing-protocols > >> -si > >> p&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf739a4ff447f008dae9 > >> b0c > >> 9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C638079242759152531 > >> %7C > >> Unknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6I > >> k1h > >> aWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=GO26A9FOOoS42yMCWPPGo5PkKo > >> 75c > >> kjSJSkjAeYRsU0%3D&reserved=0 I've also tried several different > >> clients (Bria, Polycom CCX 600, Grandstream, etc) to see if maybe > >> it was > something in the SDP or otherwise causing an issue. > >> SIP Transport is TLS, RTP is SRTP > >> I might also add that OPTION pings are active and Direct Routing > Endpoint is active, so this is successful. I also am able to receive > calls FROM teams to my IP phone(s) without issue. It is only when I > try and call INTO teams (INVITE -> Microsoft Teams) that I always > receive a 400 BAD REQUEST to my INVITE. > >> Any help is appreciated, thank you! > >> > >> - Brandon > >> > >> __________________________________________________________ > >> Kamailio - Users Mailing List - Non Commercial Discussions To > >> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > >> Important: keep the mailing list in the recipients, do not reply > >> only > to the sender! > >> Edit mailing list options or unsubscribe: > > > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > > Web: > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e > varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0, > > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c > srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7 > Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7 > C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMD > AiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata > =4E7a5ZxSvttBiMxl1VTtolmTvS4T5VGZHA00CsbPR2c%3D&reserved=0 > > > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply > > only to > the sender! > > Edit mailing list options or unsubscribe: > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply > > only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e > varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0, > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c > srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7 > Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7 > C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMD > AiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata > =4E7a5ZxSvttBiMxl1VTtolmTvS4T5VGZHA00CsbPR2c%3D&reserved=0 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only > to the sender! > Edit mailing list options or unsubscribe: > >

1 0

Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
by Brandon Armstead 29 Dec '22

29 Dec '22

Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams) always result in a 400 BAD REQUEST. Example invite below: https://pastebin.com/F1G1Ce59 I've taken care to make sure numbers are all E.164 format in From/To/Contact. I've also taken care to make sure that FQDN is used in Contact and Record-Route header. I've tried many different variations and have followed the SIP information here: https://learn.microsoft.com/en-us/microsoftteams /direct-routing-protocols-sip I've also tried several different clients (Bria, Polycom CCX 600, Grandstream, etc) to see if maybe it was something in the SDP or otherwise causing an issue. SIP Transport is TLS, RTP is SRTP I might also add that OPTION pings are active and Direct Routing Endpoint is active, so this is successful. I also am able to receive calls FROM teams to my IP phone(s) without issue. It is only when I try and call INTO teams (INVITE -> Microsoft Teams) that I always receive a 400 BAD REQUEST to my INVITE. Any help is appreciated, thank you! - Brandon

5 9

Remove all Via Header or create custome Via Header
by Benoit Panizzon 27 Dec '22

27 Dec '22

Hi out there! While experimenting with the listen and adverize 'hostname' option I came across a voice switch, which I suppose fails if the Via header is not an IPv4 address. There is a record_route_advertised_address(address) in the RR module, to set a customer Record-Route Header. Is there something similar for the Via header so I could try if having an IP address instead of a hostname in the top-most Via would solve the issue? Or is there a way to remove all via header? Remove_hf("Via") in the branch route trigger garbles the From: HF. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 4

Proxy edge
by Giovanni Iamonte 26 Dec '22

26 Dec '22

Dear List We have configured kamailio to work as a proxy edge and register. As a proxy edge cfg file we have used the example *edge_websocket.cfg.** * We are developing two different client the first one based on *PJSIP* and the other one based *JsSIP*. Client PJSIP (no websockets) everything work perfectly, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, INVITE Client JsSIP (with websockets) The REGISTER, SUBSCRIBE works, the NOTIFY does not work. In more in details we have: Browser Proxy Edge Register Client JsSIP Websocket | | | | | | | ----------Register-----------> | -----------Register----------> | | <----------200 OK ----------- | <----------200 OK ----------- | | | | | ----------Subscribe----------> | -----------Subscribe---------> | | <----------200 OK ----------- | <----------200 OK ----------- | | | | | X<--- | <----------NOTIFY ----------- | | | | As you can see the NOTIFY start from Register and reach the proxy, beloe the Register log file and the sip messages. Could somebody put me in the right direction. Thanks Bye *Log File* Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/tcp_read.c:1533]: tcp_read_req(): content-length=0 Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 2 (reply(status)) flags 1 Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:689]: parse_msg(): SIP Reply (status): Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:690]: parse_msg(): version: <SIP/2.0> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:692]: parse_msg(): status: <478> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:694]: parse_msg(): reason: <Unresolvable destination (478/SL)> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 *Sip Messages* *REGISTER* sip:sip-register.quintetto.it SIP/2.0 Via: SIP/2.0/WSS 13569868356.invalid;branch=z9hG4bK7825787 Max-Forwards: 69 To: <sip:quintetto00@sip-register.quintetto.it> From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=2tnuk9btmp Call-ID: unrnseei8924hf67khoi3g CSeq: 2 REGISTER Authorization: Digest algorithm=MD5, username="quintetto00", realm="sip-register.quintetto.it", nonce="Y6GUq2Ohk3//HTza7ZYMfd85XMdavgnc", uri="sip:sip-register.quintetto.it", response="c9cc56d2d5e36b0a68873c1d5258d062" Contact: <sip:quintetto00@13569868356.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:64e71c57-fd55-4868-be2b-a4a3fa546ac8>";expires=3600 Expires: 3600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE,PUBLISH Supported: path,gruu,outbound User-Agent: JsSIP 3.10.0 Content-Length: 0 SIP/2.0 *200 OK* Via: SIP/2.0/WSS 13569868356.invalid;rport=37566;received=192.168.100.192;branch=z9hG4bK7825787 To: <sip:quintetto00@sip-register.quintetto.it>;tag=2b789d0fa67ad6c2da65f0cce35dd60b.71a84df9 From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=2tnuk9btmp Call-ID: unrnseei8924hf67khoi3g CSeq: 2 REGISTER Contact: <sip:quintetto00@13569868356.invalid;transport=ws>;expires=3600;+sip.instance="<urn:uuid:64e71c57-fd55-4868-be2b-a4a3fa546ac8>";reg-id=1 Server: kamailio (5.6.2 (x86_64/linux)) Content-Length: 0 *SUBSCRIBE* sip:quintetto03@sip-register.quintetto.it SIP/2.0 Via: SIP/2.0/WSS 13569868356.invalid;branch=z9hG4bK3648599 Max-Forwards: 69 To: <sip:quintetto03@sip-register.quintetto.it> From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 Call-ID: 4beh90hur8hdhptnbjt8 CSeq: 5813 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="quintetto00", realm="sip-register.quintetto.it", nonce="Y6GU+mOhk86y3fnrMzoKgiVAdkaTvBId", uri="sip:quintetto03@sip-register.quintetto.it", response="04ca6d83661ac338810baa95ffaa568b" Event: presence Expires: 3600 Accept: application/pidf+xml, application/xpidf+xml Contact: <sip:quintetto00@13569868356.invalid;transport=ws> Content-Type: application/pidf+xml Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE,PUBLISH Supported: outbound User-Agent: JsSIP 3.10.0 Content-Length: 35 SIP/2.0 *200 OK* Record-Route: <sip:192.168.100.24:15006;transport=tcp;r2=on;lr> Record-Route: <sip:192.168.100.24:16005;transport=ws;r2=on;lr> Via: SIP/2.0/WSS 13569868356.invalid;rport=37566;received=192.168.100.192;branch=z9hG4bK3648599 To: <sip:quintetto03@sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 Call-ID: 4beh90hur8hdhptnbjt8 CSeq: 5813 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.100.23:15006;transport=tcp> Server: kamailio (5.6.2 (x86_64/linux)) Content-Length: 0 *NOTIFY*.sip:quintetto00@13569868356.invalid;transport=ws.SIP/2.0 Via:.SIP/2.0/TCP.1 92.168.100.23:15 006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0 To: <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03@sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq : 2 NOTIFY Call-ID:.4beh90hur8h dhptnbjt8 Route : <sip:192.168.1 00.24:15006;transport=tcp;r2=on;lr>,.<sip:192.16 8.100.24:16005;transport=ws;r2=on;lr> Content-Length:.0 User-Agent: kamailio.(5.6.2.(x86_64/li nux)) Max-Forwards:.70 Event: presence Contact:.<sip:192.168.100.23:15006;transport=tcp> Subscription-State: active;expires= 3600 SIP/2.0.478.Unresolvable.destination.(478/SL) Via: SIP/2.0/TC P.192.168.100.23 :15006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0;rport=38019 To:.<sip:quintetto00@s ip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03 @sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq: 2 NOTIFY Call-ID: 4beh90hur8hdhptnbjt8 Server :.kamailio.(5.6.2.(x86_64/linux)) Content-Length: 0 SIP/2.0.478.Unresolvable.destination.(478/TM) Via: SIP/2.0/TC P.192.168.100.23:15006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0;rport=38019 To: <sip:quintetto00@s ip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03 @sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq: 2 NOTIFY Call-ID: 4beh90hur8hdhptnbjt8 Server : kamailio.(5.6.2.(x86_64/linux)) Content-Length: 0 -- ------------------------------------------------------------------------ Ing. Giovanni Iamonte Developments and technologies area Quintetto Srl Via Monte Navale, 1 10015 - Ivrea (TO)  mobile: +39 393 9196310  tel: +39 0165 1845290  e-mail: giovanni.iamonte(a)quintetto.it  web: www.quintetto.it

3 4

ACK received instead of Status Code
by Waqar 40 26 Dec '22

26 Dec '22

Dear All, Attached in this email, is the screenshot of Kamailio logs for a call that did not originate. When Kamailio sends the call to the media destination which is an asterisk server, it should start sending status codes 100, 183, and 180 but instead, it sends ACK and the call did not originate. This happens only for some calls and I want to know why this is happening. Also, I want to identify these cases and do some further logic like trying some other media destination. How can I do that? [image: image.png] Regards, Vicky

1 0

Re: Call drops after 1 minute
by James Browne 23 Dec '22

23 Dec '22

Hi Christian I read your trace. There's no ACK after the SIP-200 response. It's cumbersome to read the trace because the client and server are at the same IP address, but I did it anyway. There's no ACK after the SIP-200 response, so it looks like the client (213.52.37.107) either doesn't sent it, or sends it directly to the server (213.52.37.107, the same IP address) without properly using the RR header field in the SIP-200 response. If you can set up a similar test and have a different host set up for the client and server (instead of 213.52.37.107 for both), and if you then trace the traffic again at all hosts, then you may find the rogue ACK request or be able to prove that the client never sent it (which might prove a broken client). When you use the client and server at the same IP address, it's not always straightforward to trace any traffic that might go directly from client to server, because it won't leave any network interface. Also I find strange that your call ends after "about a minute or so". I expect that they should end after about 31.5 seconds. It's almost certainly a SIP problem, and not any RTP/rtpproxy problem, so I recommend that you focus on the SIP. (I wrote this message before noticing that the thread is two weeks old.) James On Wed, 7 Dec 2022 at 08:41, Henning Westerholt <hw(a)gilawa.com> wrote: > > Hello, > > > > it looks that the 200 OK is not properly transmitted, as they seem to be retransmits. > > Maybe you should read a bit about SIP if you have not done it yet. It will help you debugging this problem (and probably others in the future). 😉 > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com > > > > From: Christian B Wiik <cbw(a)itf-as.no> > Sent: Wednesday, December 7, 2022 9:13 AM > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Call drops after 1 minute > > > > Link to entire trace: > > > > https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf… > > > > -- > > Regards > > Christian > > > > > > ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt <hw(a)gilawa.com>: > > Hi Christian, > > > > this ACK is the reply to the 407 and not the relevant one for the dialog. > > > > Please have a look to the full SIP dialog. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com > > > > From: Christian B Wiik <cbw(a)itf-as.no> > Sent: Wednesday, December 7, 2022 8:14 AM > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Call drops after 1 minute > > > > Thanks Henning. > > > > These are the first 3 packets filtering on my user. I see the ACK but I'm not able to spot the error. > > > > U 213.52.37.107:50336 -> 10.1.2.10:5060 #1 > INVITE sip:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9 > 706413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d1 > 4fb4c6..To: <sip:kmm@sip2.itf-as.com>..Contact: <sip:cbwlap@213.52.37.107:35270;ob>..Call-ID: b3dd380f0c1d4e > 0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: <sip:sip2.itf-as.com;lr>..Allow: PRACK, INVITE, ACK, BYE, CAN > CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: replaces, 100rel, timer, norefersu > b..Session-Expires: 1800..Min-SE: 90..User-Agent: MicroSIP/3.21.3..Content-Type: application/sdp..Content-Le > ngth: 345....v=0..o=- 3879388988 3879388988 IN IP4 213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m= > audio 35276 RTP/AVP 8 0 101..c=IN IP4 213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send > recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ssrc > :1053777612 cname:28d400de4b7d5918.. > # > U 10.1.2.10:5060 -> 213.52.37.107:50336 #2 > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 213.52.37.107:35270;rport=50336;branch=z9hG4bKPj > 398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e > 4531a7a39f45d14fb4c6..To: <sip:kmm@sip2.itf-as.com>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: > b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: Digest realm="sip2.itf-as.com", no > nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 (x86_64/linux))..Content-Length: 0.... > # > U 213.52.37.107:50336 -> 10.1.2.10:5060 #3 > ACK sip:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706 > 413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d14fb > 4c6..To: <sip:kmm@sip2.itf-as.com>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: b3dd380f0c1d4e0eb > dd7fc223710d938..CSeq: 23860 ACK..Route: <sip:sip2.itf-as.com;lr>..Content-Length: 0.... > > > > -- > > Regards > > Christian > > > > > > ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt <hw(a)gilawa.com>: > > Hello, > > > > as you’ve guessed, this can be a common problem related to the routing of the ACK message. > > > > Have a look e.g. with ngrep or sngrep to the SIP signalisation on the server side and check if everything is correct in the SIP messages. > > > > > > From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Christian B Wiik > Sent: Wednesday, December 7, 2022 7:43 AM > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Call drops after 1 minute > > > > Greetings! > > > > I have a CentOS setup in AWS where all my calls are dropped after about a minute or so. I realize this typically is a NAT problem, but I can't see where my error is. > > Sound is fine both ways. > > > > Kamailio is set with WITH_NAT and I use rtpproxy like this: > > OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722 -d INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48" > > (10.1.2.10 is the local IP for CentOS) > > > > Tested with MicroSIP and Linphone and tried numerous configurations. It seems the receiving client is not able to verify the call has been set up, and disconnects. MicroSIP has the status "Connecting..." until it disconnects. > > > > All tips appreciated. Will post configuration and logs if needed. > > Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source. > > > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > sr-users(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender! > Edit mailing list options or unsubscribe: > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio Lagging or Unreachable in Asterisk Peer
by Waqar 40 23 Dec '22

23 Dec '22

Dear All, I have three Kamailio in my production environment and I have written some code in Kamailio.cfg using SqlOps to insert data into the database on another remote server for every call. I deployed the code in one of the Kamailio and it inserted data perfectly fine into the database but as soon as I deployed it in the other two Kamailios. All three Kamailio's become unreachable or lagging in asterisk. I don't know if it has anything to do with SqlOps or queries being stuck somewhere as I am receiving logs consistently for every call in Kamailios. How can I identify if the queries are being stuck somewhere and that is why Kamailio is unreachable or lagging? Calls are not dependent on that database, this I know because even if I stop the mysql service calls will still be dialed. If it is due to the SqlOps, is there any other way to save information on every call without Kamailio being stuck? Regards VIcky

2 1

Dockerized VoIP Stack
by Terrance Devor 22 Dec '22

22 Dec '22

Hello Everyone, Does anyone have experience in deploying Kamailio, Asterisk and RTPProxy to a dockerized environment with K8s? In short, we are looking to deploy the following applications to our Kubernetes environment to support a SIP based PBX for our company: - Kamailio (Sip Proxy with DID, Outbound, edge routing and LB capabilities) - Asterisk Realtime (PBX, IVR) - RTPProxy (Media Relay) Has anyone been successful, or failed, in doing this? I would love to hear from you. Terrance

8 21

Re: sr-users Digest, Vol 211, Issue 47
by Luke Escudé 22 Dec '22

22 Dec '22

It's not easy and takes a long time to perfect, but our entire VoIP infrastructure is Dockerized, and composed of Asterisk, MySQL, Redis, NodeJS and Kamailio+RTPEngine (not rtpproxy, since rtpproxy isn't maintained anymore). No docker-level port forwarding required anywhere, since the entire system is designed to be volatile (see below). We have over 500 Asterisk docker containers running, supporting many thousands of users across the world at this point, and it works quite well, so it can be done. The concept behind the Asterisk containers is designing everything to be completely volatile: Outages happen, even in datacenters, so assume a container will reboot at some point on some random node at some random datacenter location. You don't know Asterisk's private IP, you don't even know where it might pop back up in the cluster. If you can ensure service continuity in that scenario, then you've built a solid setup. Our uptime has been 100% for over 4 years now (this is 3rd-party monitored and validated, in addition to our internal monitoring systems) so I assure you it is very possible. I also HIGHLY recommend not using a cloud provider for any of the core infrastructure. We tried this, with AWS and GCP, and the voip quality was... meh. The cloud outages were frequent, although customers generally don't notice them when you're using lightweight containers to run everything. So we are fully operating on our own hardware across multiple colo locations, and only using cloud for file storage (eFaxes, voicemails, greetings/recordings, MMS content, phone LCD background images for provisioning, etc). I am unable to help with your project, but I can at least provide the above pointers/experiences in case they get you in the right direction. An alternative to building your own infrastructure might be reselling (white label) an existing service provider as well, in case the decision to build your own infrastructure is still up in the air. Luke Escudé 972.600.1150 support(a)primevox.net Schedule a meeting!<https://outlook.office365.com/owa/calendar/PrimeVOXCommunications@primevox.…> View the PrimeVOX R&D Roadmap here<https://trello.com/b/pzpoyn8m/primevox-engineering> View the PrimeVOX Status Page here<https://status.primevox.net/> ________________________________ From: sr-users-request(a)lists.kamailio.org <sr-users-request(a)lists.kamailio.org> Sent: Thursday, December 22, 2022 9:44 AM To: sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org> Subject: sr-users Digest, Vol 211, Issue 47 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Dockerized VoIP Stack (Terrance Devor) 2. Re: Dockerized VoIP Stack (Terrance Devor) 3. Re: Dockerized VoIP Stack (Terrance Devor) ---------------------------------------------------------------------- Message: 1 Date: Thu, 22 Dec 2022 09:31:33 -0500 From: Terrance Devor <ter.devor(a)gmail.com> Subject: [SR-Users] Re: Dockerized VoIP Stack To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Message-ID: <CA+w+2f8CtdwkiDtz+Md-pZz5xA4-59gv768b_vq4bkdjf2V3hw(a)mail.gmail.com> Content-Type: multipart/alternative; boundary="000000000000398ece05f06b87af" I did see the CyCore project and tried to connect with Laurel Lauson and Sean McCord on LinkedIn On Wed, Dec 21, 2022 at 9:18 PM Fred Posner <fred(a)palner.com> wrote: > On Dec 21, 2022, at 8:41 PM, Terrance Devor <ter.devor(a)gmail.com> wrote: > > > Does anyone have experience in deploying Kamailio, Asterisk and RTPProxy > to a dockerized environment with K8s? > > > Check out some past Kamailio World and Astricon presentations, including > some from Sean McCord. He even has a GitHub demo: > > * https://github.com/CyCoreSystems/asterisk-k8s-demo > > -- Fred > The Palner Group, Inc. > +1 (212) 937-7844 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: >

2 1

Which KEMI-language has the lowest runtime overhead?
by Дилян Палаузов 22 Dec '22

22 Dec '22

Hello, KEMI can be programmed in Java, JavaScript, Lua(sr), Squirrel, Mono, Perl, Python(2,3,3s), Ruby. Which language binding has the lowest runtime overhead in CPU-terms? How about lowest overhead in terms of RAM? Kind regards Дилян

4 3

IMS integration with AS
by Oleg Belousov 22 Dec '22

22 Dec '22

Hi. Can you please advise if existing ims implementation supports integration with ims AS (Application Server), in particular for the third party registration: - upon UE registration is completed, S-CSCF constructs Register message (which keeps all details of subscriber register message) and submits it to AS server as regular SIP-message (TS 24.229 5.4.1.7 Notification of Application Servers about registration status). S-CSCF obtains AS address from SAA (Server Assignment Answer) diameter message, received from HSS upon successful registration. -- obelousov.tel

2 2

How to make a branch trigger persistent (still being active after failure route is triggered)?
by Benoît Panizzon 20 Dec '22

20 Dec '22

Hi List Some time ago, during my learning curve with kamailio, I learned that a branch route is where you make modifications to callerId's. So I extensively re-engineered the config to move as much as possible of callerid modification into branch routes. But now I discovered, when a call is dispatched via dispatch group with multiple destinations and one destination fails, therefore the failure route is triggered, the branch route is not being called on the second attempt, thus callerid being in the wrong format. Example: dispatcher.lst As an example, consider having dispatcher group 2001 for national traffic over two dedicated links and Dispatcher group 2002 for international traffic. 2001 sip:192.168.1.1:5060 0 0 name="Carrier1" 2001 sip:192.168.2.1:5060 0 0 name="Carrier1" 2002 sip:192.168.1.3:5060 0 0 name="Carrier2" request_route { # We determine where we want to send the call # and choose dispatch group 2001, the 'national' one. # But there are are other dispatch groups and branch triggers # set this way and routed to DISPATCHCALL $avp(dispgroup) = 2001; t_on_branch("BR_TO_NATIONAL"); route(DISPATCHCALL); } route[DISPATCHCALL] { t_set_fr(120000,1500); # Fast Failover t_set_max_lifetime(300000,0); t_set_retr(200,800); if (ds_select_dst("$avp(dispgroup)", "6")) { if (ds_is_from_list(-1,0,"$du")) { t_on_failure("DISPATCH_FAILURE"); route(RELAY); } } } branch_route[BR_TO_NATIONAL] { # Here I make sure, all CallerId and Header are correct for the # national Carrier } failure_route[DISPATCH_FAILURE] { if (t_check_status("(5[0-9][0-9])") or (t_branch_timeout() and !t_branch_replied())) { # Try next DS. if (ds_next_dst()) { xlog("L_INFO", " --> retargeting request\n"); t_on_failure("DISPATCH_FAILURE"); route(RELAY); #### RIGHT HERE! exit; } } # We fail xlog("L_ERROR", "Call Failed\n"); } My issue is not, that if the call is routed after the first path failed (marked with #### RIGHT HERE!) then the appropriate branch route is not triggered again and the headers are wrong. But I can not set t_on_branch within route[DISPATCHCALL] or route[DISPATCH_FAILURE] because this is called with a different pre-set branch trigger, depending on the target dispatch group. Is there a way to make a branch trigger more persistent? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 7

Topoh module not decoding the BYE received from the callee
by Michel Pelletier 20 Dec '22

20 Dec '22

Hello, I am using the topoh module and it works fine except for the BYE requests coming from the callee (the person receiving the call). When a BYE is received from the destination of the call, then Kamailio relays it to the dummy IP 127.0.0.8. This is exemplified in the topoh:msg-outgoing and topoh:msg-sending events where the $sndto(ip) is set to 127.0.0.8. I am using all default modparams with topoh. The only modparam I have is for setting the event_callback. I am using Kamailio 5.6.2 with Kemi using python3. Michel Pelletier

2 3

dialog_vars accumulating in database?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi List I noticed that the database is accumulating entries on dialog_vars. I don't know when and why this is happening. Are there any special measures that have to be taken like making sure to unset variables after using them to prevent this? Or is this something which is supposed to happen when kamailio is restarted? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 5

How to unset_dlg_profile in request_route or react more quickly to CANCEL?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi List Testing failure situations, I discovered unset_dlg_profile can not be used in request_route: I count the channels per customer in a dlg_profile to know when they are busy. Residential POTS customer have 1 channel. Now this situation (Trying to mimik POTS behavior) Kamailio <=> CPE of 'John-Doe' => INVITE set_dlg_profile of John-Doe is 0: Not busy set_dlg_profile of John-Doe +1 <= 100 trying <= 180 ringing == LINK to CPE DISRUPTED == X<= 200 OK (not reaching kamailio) CPE is sending 200 a couple of times and fails. Caller is still hearing ringing tone as it never got 200 OK. The caller does not want to wait longer and hangs up =>X CANCEL (NOT reaching CPE) This is the moment, on which I would like to unset_dlg_profile of John-Doe to mark his channel available again. But the CPE is never going to acknowledge this CANCEL with 487. Unfortunately: unset_dlg_profile(): dialog delete profile cannot be used in request route Only when kamailio has sent CANCEL a couple of times and the failure_route for this CANCEL is triggered, I can remove the call from the dlg_profile of that customer within failure_route. If I am unlucky, in the meantime other calls get 'busy' instead of being re-routed to the configured backup number of that customer. Agreed, after a couple of seconds, the call goes to the failure route and the dialogue is ending thus calls work as expected. But wouldn't it be nice to be able to unset the dlg_profile upon receiving the CANCEL instead on when the CANCEL transaction is successful? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

DNS Host-Names in Via and Record-Route for more redundancy?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi Folks We use two 'core' instances and two 'registrar' instances. The pairs sync their information (Dialogues + Locations) via DMQ. But I noticed that they record the Via and Record-Routes by IP addresses. So I guess if one fails, signaling would not automatically switch to the remaining host as the IP of the failed one is in the header, right? So, is there a way to put a DNS resolvable host-name in the Via and Record-Route header, so that DNS round-robin would make sure requests reach the available server if one of the pairs fails? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 6

Multiple websocket connections with the same port
by Benjamín Visón 19 Dec '22

19 Dec '22

Hello! I'm examining the WebSocket module and I am wondering how it handles multiple connections from the same port. For context, imagine the following scenario: UA1 ==> Load Balancer:41914 ==> Kamailio (WS) UA2 ==> Load Balancer:41914 ==> Kamailio (WS) Now, imagine Kamailio perceives UA1 from port 41914. What happens if that UA1 suddenly drops and another UA2 comes in from the same port? (before the connection closed event is fired for UA1) I understand by looking at the source code that `wsconn_add` is called and that this new connection is added to the list. But what happens with the previous UA entry? does it gets overwritten or does it stays there? The reason for my question is: I am perceiving the above scenario like this: Let's say UA1 is *sip:d3315cf8@9a431b1f90e5.invalid;transport=ws SIP/2.0* and UA2 is *sip:asd08a08@555343439666.invalid;transport=ws. *If UA2 comes in and takes over the port that UA1 was using, and I send an INVITE to UA1 UA2 ends up receiving the invite (obviously because it now has that port allocated). So what I am thinking is that both *d3315cf8(a)9a431b1f90e5.invalid *and *asd08a08(a)555343439666.invalid *are mapped to the same port for a short period of time. Thoughts? Saludos, [image: Twitter] <https://twitter.com/benjaminvison>[image: Instagram] <https://instagram.com/bvisonl/> Benjamín Visón / IT Engineer / Software Developer bvisonl(a)gmail.com / (829)-664-5163 https://www.linkedin.com/in/bvisonl/

3 2

auth_challenge() and async
by Alex Balashov 19 Dec '22

19 Dec '22

Hi, Is there any good solution for the scenario of an auth_challenge() with two separate transaction suspensions? To clarify (simplified): request_route { ... t_newtran(); if(!is_present_hf("Authorization") && !is_present_hf("Proxy-Authorization")) { auth_challenge("$fd", "1"); exit; } # TM suspend/continue t_newtran(); async credentials query("RESUME_AUTH"); } route[RESUME_AUTH] { # Credentials received into PVs. if(!pv_auth_check("...")) { auth_challenge("$fd", "1"); exit; } # Create transaction shell if not exists already from auth query. if(!t_lookup_request()) t_newtran(); # TM suspend/continue async route query("RESUME_ROUTING"); } route[RESUME_ROUTING] { # Unmarshal etc. t_relay(); # etc. } What happens here is that the first auth_challenge() results in a retransmission of its 407 challenge without absorbing the negative ACK. This is despite the ostensibly stateful behaviour of auth_challenge() without "force_stateless_reply" enabled. In this case, it eclipses the second, unrelated 407 challenge from the subsequent routing query (407 challenge with +1 CSeq): -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 17

Dealing with failed SRV peers
by Sebastian Damm 17 Dec '22

17 Dec '22

Hi, we have some Kamailios working as outboundproxy. So they get requests from internal systems and send them to different providers. From time to time, one provider returns a server as primary resource which is currently unavailable. I guess if the internal systems connected directly to the target, they would remember the failed server and remember to always use the server with second priority at least until DNS refresh time. In our setup, since every request is "new" for Kamailio, it doesn't remember, which host is reachable or not. Example: target: example.com _sip._udp.example.com SRV resolves to: 10 192.0.2.42 5060 20 198.51.100.42 5060 30 203.0.113.42 5060 192.0.2.42 is unavailable. Still, Kamailio uses it for every new request and failover to 198.51.100.42 occurs only after timeouts hit. Is there a best practice for solving this? I have played around with the dst_blocklist settings, but that caused even more trouble because Kamailio started blocking requests to peers that have only one server in the record having a short hickup. Thanks in advance for every input, as this is causing trouble every time we run into such a situation. Regards, Sebastian

4 7

Redundancy
by Jawaid Bazyar 16 Dec '22

16 Dec '22

Hi, I am exploring different redundancy / load-balancing models for a Kamailio cluster. When I say cluster, I mean, a number (N) of Kamailio nodes acting as stateful proxies. Each node is configured the same as the others, and all have access to the same lookup data to make routing decisions. I would appreciate any advice or experience any of you can share on these different models. Overall model: Direct to proxies Redirect servers first, which redirect to proxies Selecting the first node to talk to. Each model could use either type of selection. DNS-based (SRV or NAPTR, client makes call to dns name) Anycast with ECMP (equal-cost multi-path routing) Cluster with a mobile IP and service-down detection (this would just provide 1:1 protection) Have clients make calls through the proxy using a DNS record containing an SRV record for each node (or, alternatively, done with NAPTR). Would rely on the client to switch nodes in the event of a node failure mid-call. (Is that even possible?) Anycast would only work with UDP signaling. Use Anycast to find the first proxy, then use stateful responses to direct client back to same node for subsequent messages in a dialog. So for anyone who has tried any of these methods, I would love to hear the pros and cons.. Thanks in advance! Jawaid

2 4

Must openssl_mutex_shared.so be preloaded?
by Дилян Палаузов 16 Dec '22

16 Dec '22

Hello, The documentation on using openssl_mutex_shared.so is contradicting. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/README#L293 says: > For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/openssl_mutex_shared/README.md'. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/utils/open… says: > IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default. It is unclear, if openssl_mutex_shared.so must be preloaded. In case the library is not needed anymore, it shall not be compiled and installed. Kind regards Дилян

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Logs suddenly inundated with qm_malloc errors
by George Diamantopoulos 16 Dec '22

16 Dec '22

Hello all, I'm testing out release 5.6.2 on one production kamailio instance, and today our monitoring system triggered an alarm for lack of disk space on /. Indeed, kamailio.log started growing at an alarming rate at some point. At about the same time disk space started being depleted, the following line was frequently logged in the logs (no instances of it before that time): qm_malloc(): qm_malloc(0x7f5ade9c0010, 536) called from rtpengine: bencode.c: __bencode_piece_new(79), module: rtpengine; Free fragment not found! They don't only concern rtpengine, core has produced similar "free fragment not found" errors as well. There's additional multiple "had header field", "bad header" and "invalid header name" errors appearing after that too, but I'm guessing that's only a symptom of the qm_malloc issue? kamailio is started with -m 2048 -M 16 and there's another instance with almost identical traffic and configuration, but running on 5.6.1, which has never exhibited such failures in several months uptime. So I'm wondering, is there anything in 5.6.2 that could explain this? Thanks! Best regards, George

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NAT issues with Kamailio + Asterisk with 2 NICs
by Sergio Charrua 15 Dec '22

15 Dec '22

Hi all! Got a random issue with NAT which I really have no clue how to solve. Setup is as follows: - Client A sends SIP INVITE through Public IP 1.2.3.4, where a Cisco Firewall forwards NATed data to a Kamailio instance (acting as a SIPProxy with dispatcher) listening with internal address 192.168.0.1 on port 5060. RTPEngine is in the same server and address too. - Kamailio will forward the invite to Asterisk NIC 1 (192.168.0.2) which will do some provisioning and accounting stuff and redirect the call (if authorized) to Client B's public IP address (all clients are outside the network, reachable via WWW) using NIC 2 (192.168.1.2) and public address 4.3.2.1. The issue is that from time to time, calls will not have audio. Here is what I have found: - when Client A sends INVITE, Asterisk NIC 1 will reply SIP 183 with SDP IP Address set to the internal IP of NIC 1 (192.168.0.2) - when Asterisk forwards call to Client B public IP address the SDP IP address is 192.168.1.2 (NIC 2) And I think the issue is the SDP address is wrong: when Asterisk forwards call to Client B, the SDP address should be the public IP 4.3.2.1 but it sends 192.168.1.2. And when Asterisk replies to Kamailio the SDP is 192.168.0.1. Correct me if I am wrong but i think the solution should be: - when Kamailio receives a reply from Asterisk to be forwarded to Client A, it should modify the SDP address to public IP 1.2.3.4 - when Asterisk forwards call to Client B, the sip.conf should have the parameter externip set to the public IP 4.3.2.1 Questions: - are my analysis & solution correct? - is there an example of the above setup where I could get some ideas? - how do I set Kamailio to modify the SDP address for me? Any examples? Thanks in advance. *Sérgio Charrua*

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kamailio.org server maintenance - Dec 14, 2022 (tomorrow)
by Daniel-Constantin Mierla 15 Dec '22

15 Dec '22

Hello, not that tomorrow, Dec 14, 2022 starting after 09:00UTC (10:00Berlin), is planned some maintenance work on kamailio.org server. It will be unavailable from time to time, hopefully not for long time. Affected services: - web site - mailing lists - wiki site (alternative use the github repo: https://github.com/kamailio/kamailio-wiki) - html version of modules' documentation (text variant README is every module folder of git clones or on github.org repository) Short notifications will be sent when it starts and ends. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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