- sr-users - kamailio.org

$Au being set to $fu instead of $fU
by Joel Serrano 05 Sep '19

05 Sep '19

Hi, Testing on latest master deb nightly build from last night I've noticed that when $au is not available, $Au is being set to $fu (uri) instead of $fU (username). Docs: https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables ... $Au - Acc username: username for accounting purposes. It's a selective pseudo variable (inherited from acc module). It returns auth username ($au) if exists or From username ($fU) otherwise. ... I have the following log in several places in a test box: xlog("L_NOTICE", "[DEBUG] Au=$Au au=$au aU=$aU fU=$fU fu=$fu - \n"); And when I run through them (on both request and reply routes) I get the following: Aug 8 14:37:43 csbc01 csbc[9478]: NOTICE: {1 21564 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9479]: NOTICE: {1 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=1002366 aU=1002366 fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9484]: NOTICE: {2 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Those 3 logs belong to the same flow from different places in the config script. I would expect $Au to never have a "@some.domain" in it, so I assume in some place the replacement is not being done with $fU but with $fu. @Devs: I'm happy to create a GH issue for tracking etc if required, please let me know! Cheers, Joel.

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How to detect miss call and accomplished call
by Anuran Barman 05 Sep '19

05 Sep '19

Hi, I want to show call logs in my SIP app. How can I do this with Kamailio? I want to do it from the *kamailio.cfg *file. Is it possible to do it from where? I guess Dialog module should be used for this, but can not figure out how to do it. I want minimalist feature like who call whom with timestamp, thats all. How can I do this?

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Kamailio-IMS system
by Hạnh Nguyễn 05 Sep '19

05 Sep '19

Dear Mrs/Mss, I am Hanh, I am from Viet Nam. I am writting this email because I have an issue when I download The log-file and the config-files on website https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/. When I click on https://www.dropbox.com/s/ehkzi9pbhlp7njc/Kamailio-IMS%20config-files.zip?d… to download The log-file which I really need on this time, the page displays 404 Error - Link not found. So I can not download. I hope I will receive a new link from you soon. I look forward to hearing from you. Sincerely, Hanh Nguyen

2 1

Planning the next major release - v5.3.0
by Daniel-Constantin Mierla 04 Sep '19

04 Sep '19

Hello, it's about the time to set the milestones to the next major release - v5.3.0. There are 6 new modules and many other enhancements to existing components in git master branch, also during the last irc devel meeting it was proposed to freeze after summer holidays. At this moment I would propose Wednesday, September 4, 2019 to freeze the development. Then the release should be about one month later, after testing interval. In case there are people willing a bit more time for development, we can freeze one week later, September 11, or another date -- your suggestions here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Using Kamailio IMS for making a call
by supreeth herle 04 Sep '19

04 Sep '19

Hello Everyone, I have successfully setup a VM in OpenStack with Kamailio IMS, rtpengine and FoHSS using the instructions mentioned in the guide which i wrote (attached to this mail), all running in the machine. I am using the existing alice and bob IMPUs created in FoHSS. But, when I try to call each other the call doesnt go through and I observe in the wireshark traces that S-CSCF is replying to I-CSCF that "403, Forbidden - Domain not served". Can somebody please tell me do I need any additional components to make a call or am i missing some configuration? It would be great if someone can help me troubleshooting this issue. I have attached the pcap file for your reference. Many thanks in advance. Regards, Supreeth

1 0

Kamailio stops processing requests over UDP
by Juha Heinanen 04 Sep '19

04 Sep '19

I'm returning to an old issue, where Kamailio stops processing incoming requests. Now it turned out that only requests over UDP are not processed. Requests over TCP work fine. Debug log doesn't show anything about the requests over UDP, but wireshark sees them coming. core.ps shows that main process - attendant and all udp receiver processes are alive. Any ideas what could cause Kamailio (5.2 version) to stop processing requests over UDP? What additional info should I try to get? -- Juha

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Hello Group - I have a "challenge"
by Kevin Greene 04 Sep '19

04 Sep '19

I recently did the upgrade from v4.4 to 5.2 (via Yum). When starting Kamailio, I get these three errors. Not sure if they are related or not but I do not think so. Have spent a lot of time searching on Google for the answer and have not found the solution. [root@ids kamailio]# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 44608) [root@ids kamailio]# tail -f /var/log/sipserver.log Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/pvapi.c:1119]: pv_parse_spec2(): wrong char [t/116] in [$mt] at [2 (0)] Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/dprint.c:451]: log_prefix_init(): wrong format[{$mt $hdr(CSeq) $ci} ] Sep 4 00:39:06 ids /usr/sbin/kamailio[44626]: ERROR: <core> [core/receive.c:319]: receive_msg(): no request_route {...} and no other config routing engine registered The last error is repeated many times (each time a remote user tries to connect I think). I can send any additional files if needed. Thanks. I am burning out on this upgrade. Kevin

2 1

Unexpected BYEs from dialog module
by Alex Balashov 03 Sep '19

03 Sep '19

Hi, I am running 4.3.5:1b0c0a, which I am aware is an EOL'd release train, and have a problem with the dialog module I am baffled by. On many calls - I can't find any correlation to particular kinds of endpoints - I see spoofed BYEs come out of Kamailio after a minute and a half, as if the call had hit a dialog timeout or the dialog module's dead peer detection (OPTIONS pinging) kicked it out. The thing is, neither of those explanations seem to be borne out by the parameters of the calls under investigation. The dialog timeout on these calls is set to 7200, and the dialog keepalives aren't enabled in either direction. If they were, it'd be logged. Calls that are terminated in this manner also see the following log message: WARNING: dialog [dlg_req_within.c:214]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7f9997317e48 [3390:9644] with clid '1996679936_133218050(a)x.x.x.x' and tags '2bb17663-co4006-INS001' 'as7925780d' But I don't think this is unusual for a dialog-spoofed BYE; presumably this is due to a 200 OK for the BYE from both ends. So anyway, I am trying to track down anything else that could inadvertently cause dialog to hang up calls relatively quickly in that release, or inadvertently set the dialog timeout parameter to something lower than is apparent from the logging and the config. If anyone has any pointers, that would not go unappreciated! Many 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/

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crash/mem leaks on low memory
by Gaurav Bmotra 03 Sep '19

03 Sep '19

hi i m using kamailio 5.1 on Ubuntu 18.04 i m geting this error in logs and it stop registration form soft-phone plz help me -------------------------------------------------------------------------------------------------------- WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory th -------------------------------------------------------------------------------------------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Tue 2019-09-03 12:09:19 UTC; 26min ago Docs: man:systemd-sysv-generator(8) Process: 27237 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 7077888 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 536870912 - pkg size: 8388608 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7f0b41b001f4, 16) on 172.31.35.49:5091 : Address already in use Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- thank you *Regards:* Gaurav Kumar

3 3

evapi mismatch version
by bill＠novatrope.us 02 Sep '19

02 Sep '19

Hi All Installed Kam 5.2.4, but now getting version mismatch on evapi. ERROR: <core> [core/sr_module.c:348]: version_control(): module version mismatch for /usr/lib64/kamailio/modules/evapi.so; core: kamailio 5.2.4 (x86_64/linux); module: kamailio 5.1.6 (x86_64/linux) Anybody know how to get the correct evapi version. I am running centos 7 Bill

3 2

Issue with uac_replace
by David Costa 01 Sep '19

01 Sep '19

Greetings, I have a Kamailio proxy with uac_replace for To and From Headers. So far everything has worked correctly. However, I now have a client who encodes my Calling Number too and I'm getting some issues in the restore mechanics. I have the restore setup as "auto" and it's info is stored in "vsf" and "vst" parameters of Record-Route. On replies sent by my client I have the following Record-Route and From Header (Note : My Kamailio has two IPs which makes it insert two RR headers ): Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> From: "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this case the restore is done correctly When the client sends me a BYE request I have the following Route Header and To Header Route : <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2>,<sip:xxx.xxx.xxx. xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_o=128_11_Y;vsf=AA AAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcH wIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> To : "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this scenario, the To after restore is : To :"+351411110097" <sip:Q4525417L_<>Gxxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4, which throws an internal error on Kamailio about bad URI and also makes our MediaGateway reply with a "Bad Request". Can this error be caused by the restore of the uac_replace mechanics? Both Record Route URIs are on different headers while the both Route URIs are on the same Route header. Can this be the reason? If you need more information please let me know. Best Regards David Costa

2 1

Re: [SR-Users] controlling presence with new module `pua_json`
by Henning Westerholt 30 Aug '19

30 Aug '19

Hello Bill, (please keep the list in CC) yes, this seems to be the case. Just looked briefly to the code, it does an assignment in some cases. E.g. if no Presentity* data is given it will just use the From* information. And then it should later output an error from pua_json_update_presentity(..) if there is not the necessary data. Do you see any error message like this? Another idea - does it works if you add more "missing" data in the curl requests? Cheers, Henning Am 28.08.19 um 21:16 schrieb bill(a)novatrope.us: > Henning: Good point. > > I do see a number of errors in the log. It looks like the json routine > is looking for data that is not in the http query as specified by the > author. > > > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Name: [update] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Package: [dialog] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-User] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-User: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-Realm] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-Realm: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): From: > [sip:1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-User: [1020101] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Realm: [<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To: > [sip:1020108@<IP of server>:50060] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > To-User: [1020108] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Call-ID: [1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Direction] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Direction: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): State: > [Confirmed] > 17(125579) DEBUG: presence [event_list.c:348]: search_event(): start > event= [dialog/5] > > On 8/28/19 12:00 PM, Henning Westerholt wrote: >> Hello Bill, >> >> just giving some generic debug pointers here: >> >> - do you see any errors in the kamailio log file? >> >> - do you see any SIP traffic (e.g. by observing with ngrep, wireshark >> etc..) to the phone after your curl command? >> >> Cheers, >> >> Henning >> >> Am 28.08.19 um 19:04 schrieb bill(a)novatrope.us: >>> Hi all: >>> >>> I am trying to use the procedure for setting BLF lights described in >>> https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… >>> >>> >>> WIthout success. >>> >>> My configuration is 1 Grandstream GXP2130 phone behind a NAT on a >>> public IP. (IPP in example below) >>> >>> Kamailio is running on public IP (IPK in example) listening on port >>> 50060 >>> >>> The command I am sending: >>> >>> curl -d >>> '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' >>> >>> http://localhost:8080/presence/ >>> {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} >>> >>> >>> >>> >>> Kamailio appears to be processing the command OK, so I think I am just >>> not setting it up properly> >>> >>> Anybody having any luck with this procedure, let me know how you are >>> getting it to work. >>> >>> >>> Bill >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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Info: Upgrading kamcli to work with Python3
by Daniel-Constantin Mierla 30 Aug '19

30 Aug '19

Hello, with the perspective that Python2 is going to be removed from the major Linux distributes starting with the beginning of next year, the work to upgrade kamcli to run with Python3 has been started. >From now on, master branch of kamcli is expected to work only with Python3: * https://github.com/kamailio/kamcli There is still work to be done to upgrade all the code, but any use of it should be done with Python3 and eventual issues to be filed on bug tracker. Of course, pull requests are welcome as always! Who wants to use the version for Python2 has to use the code from git branch v1.2-python2. That branch is probably not going to receive any new feature, it will be kept for use with older systems (as well as a running option during the upgrade period to Python3). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

IMC module not sending messages to members of a conference room
by david＠aslo.us 29 Aug '19

29 Aug '19

2 1

Kamailio 5.0.8 | authentification issue only with TCP/TLS
by Laurent Schweizer 29 Aug '19

29 Aug '19

Dear all, I have a kamailio running in version 5.0.8 and since fee weeks we have an issue with different users connected in TCP or TLS, sometimes authorization like for REGISTER are rejected and after a moment (can be few minute or hours) it work again and of course no change was done in the password .... We see this issue with different device, snom swyx, ... and on UDP we have no issue. I can see that when the Register is rejected it's with the error -2, so wrong password... # Authentication route route[AUTH] { if (is_method("REGISTER")) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { switch($retcode) { case -1: sl_send_reply("503","Service not available"); exit; case -2: xlog("L_WARN", "auth error -2 username $au - src ip: $si \n"); auth_challenge("$fd", "0"); exit; I have attached an example of a trace where we can see a first REGISTER accepted and less than 2 minutes after a new one is rejected. ( in between they is a REGISTER without any Authorization header) Any idea ? BR Laurent

2 3

Connect Kamailio to ITSPs?
by Aingaran Thirunadarajah 29 Aug '19

29 Aug '19

Hello, Newbie here! Have Kamailio set up on a 512MB VPS. Created a couple of users, registered with Zoiper and can call between them. I am looking for documentation to connect the server to other VoIP providers (Signalwire, Anveo, VOIP.ms etc.) to make & receive calls from the external network (DIDs) / PSTN Can someone point me in the right direction? Thanks in advance.

2 1

Define properly IP in record-route header
by przeqpiciel 29 Aug '19

29 Aug '19

Hi, I have some issue and dont have enought knowledge to fix it. I have simple infrastructure. Internet -> router (NAT) -> kamailio -> asterisk Kamailio and Asterisk are inside the network and Kamailio places his own private address in the "record-route" headers instead of who's public. There is configuration to said to him what IP is proper ?

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kamailio+asterisk VoiceMail implementation
by Zhan Bazarov 28 Aug '19

28 Aug '19

Hello, what's best practice for VoiceMail implementation in Kamailio+Asterisk's scheme? should I forward register into asterisk? Or I can keep register on kamailio? Could you, please, shed any lights on this

2 1

Disable GW's on runtime
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, I have two Kamailio's working on a HA scenario. Both Kamailio's have DB replication and each machine has one HA IP. In order to set this up, i have every GW on the DB duplicated, one with Socket IPA and another with socket IPB. I have both GWs in a active state configurated in the replicated DB. They also both have the same URI and the same group id Example : GW1 : Socket = IPA ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 GW2 : Socket = IPB ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 In order to avoid errors, i want to disable on runtime the GWs that have a socket that doesn't belong, at the moment, to that machine (GW with Socket IPB on machine with IPA). To do that i would use the dispatcher.set_state, however it works with group id and address, which in this case it is the same. Is there other method to disable the GWs on runtime using other arguments like duid or socket ? I thought about using probing for this, since it will disable automatically the GWs with non-existent socket, but not all my clients support probing, so it isn't feasible. Thanks in advance

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UAC ha1 password
by Alexandru Covalschi 28 Aug '19

28 Aug '19

Hello, I'm trying to make Kamailio to SIP register on a remote server. However storing plaintext password looks very unsecure. Is there a possibility to store password for uac module using ha1 hash instead of the plaintext password? I see there's a row for it in the database, but in the source code it's not used and it seems i can't neither use it in db neither set up in-memory as avp. Maybe there's a workaround to directly access the in-memory uac registration htable? ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

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controlling presence with new module `pua_json`
by bill＠novatrope.us 28 Aug '19

28 Aug '19

Hi all: I am trying to use the procedure for setting BLF lights described in https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… WIthout success. My configuration is 1 Grandstream GXP2130 phone behind a NAT on a public IP. (IPP in example below) Kamailio is running on public IP (IPK in example) listening on port 50060 The command I am sending: curl -d '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' http://localhost:8080/presence/ {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} Kamailio appears to be processing the command OK, so I think I am just not setting it up properly> Anybody having any luck with this procedure, let me know how you are getting it to work. Bill

2 1

[tm-local-request] usage
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, Does "event_route[tm:local-request]" works like onsend route ? I'm having some problems and i'm using local-request to debug and i found this on the documentation : "Executed after the tm module has sent a local generated, transaction stateful request." I'm logging all the requests that go through this route. Is this really the last place before kamailio send the request? If the request is logged here, is there anything more in kamailio that can stop the request (pipelimit, etc) ? If a packet gets logged on this route does that mean that it was already sent? Best Regards, Duarte Rocha

3 4

Kamailio 4.4.7 - LCR modules
by Laura 28 Aug '19

28 Aug '19

Hi guys, on our enviromet we have 130+ gateway configured over the LCR_gw tables. when we try to reload it.. we get error.. "Aug 28 09:40:36 vm-nextaitz-02 /usr/sbin/kamailio[3800]: ERROR: lcr [lcr_mod.c:1511]: reload_tables(): too many gateways" We reached some top values.. how can we modify it ? Regards

2 2

Dispatcher - FQDN with multiple IPs
by Sergiu Pojoga 27 Aug '19

27 Aug '19

Hi ppl, The problem I'm facing is that function *ds_is_from_list* seems to return false when a gateway's address is a FQDN that resolves into multiple IP addresses (I'm talking about A records), consequently in-dialog requests like BYE and re-Invites get blocked since they don't pass the *ds_is_from_list* validation. Am I missing some parameter or something? Thanks, --Sergiu

3 3

RFC: about sl and tm local-response event routes
by Daniel-Constantin Mierla 27 Aug '19

27 Aug '19

Hello, just discovered what I consider to be an inconsistency in naming and behaviour for event_route blocks for local-request and local-response and starting a discussion here to see how to move on. The event_route[tm:local-request] is executed before sending there local generated request out (allowing also to change its content, drop, etc...). This event route is quite popular event route used when the local generated requests need to be checked or updates. The event_route[tm:local-response] is executed after the response is sent out, obviously no possibility to change anymore the content. I haven't checked the code for event_route[sl:local-response], but based on commit message should be the same. I haven't used the local-response so far at all, today after a discussion on sr-users I wanted to enable kemi callback for tm:local-response and I noticed that behaviour in the code, even I expected to be like local-request (before sending out). I am not sure how much used are the event routes for tm:local-response and sl:local-response, I haven't seen any questions about them so far on mailing lists, that's why I am asking here if would make sense to rename them like tm:local-response-sent and sl:local-response-sent to properly reflect when they are executed. I am expecting that they are very few used so far, so no big head ache with upgrades and bringing some consistency around (this change to be part of next major release). It can stay like now with proper documentation, however in the future if one want and event route for local responses before being sent out, there will be more confusion, imo ... Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 5

Too many hops 483 - unable to setup call
by Jack R 27 Aug '19

27 Aug '19

Hi, I installed Kamailio SIP server: Following article: https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… By following step # 12 (as per above article) when we initiate the call we get - Too many hops 483 Created two user earlier: $ kamctl add test testpasswd $ kamctl add test2 testpasswd syslog show: Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56276]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (45.45.0.16:42965) to sip:test1@139.54.102.135, 04e06b6352f7a2b8f97c1a3733b3334a(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: ims_registrar_pcscf [save.c:339]: save_pending(): Will save pending contact without security parameters Aug 27 09:37:05 localhost /usr/sbin/kamailio[56374]: ERROR: <script>: Diameter: AAR failed on subscription to signalling Server IP 139.54.102.135 where Kamailio is installed, UE IP address 45.45.0.16, 45.45.0.1 Followed this https://www.kamailio.org/wiki/tutorials/faq/main#why_the_sip_requests_are_r… Thought to test using SIPp - installed and ran default command: ./sipp –sn uac 127.0.0.1 and ./sipp –sn uac 139.54.102.135 Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 2-60232(a)139.54.102.135) Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist The user sipp and service does'nt exist Added user: kamctl add sipp testpasswd kamctl add service testpasswd Still same error: Aug 27 10:52:17 localhost /usr/sbin/kamailio[56289]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist Aug 27 10:52:17 localhost /usr/sbin/kamailio[56284]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 977-61357(a)139.54.102.135) Please advise. Thanks and Regards.

1 1

kamailio stop registering agents
by Gaurav Bmotra 27 Aug '19

27 Aug '19

hi i m using kamailio 5.1,,, on ubuntu 18.4 installed form source .... after few days it stop registering client to kamailio.... it resolved after restart service ... same after few day loges ---- --------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Wed 2019-08-21 08:07:44 UTC; 5 days ago Docs: man:systemd-sysv-generator(8) Process: 1936 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 67108864 - pkg size: 838 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7fbabac2c1f4, 16) on 172.31.35.49:509 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 21 08:07:44 talk kamailio[1936]: * already running Aug 21 08:07:44 talk kamailio[1936]: ...done. Aug 21 08:07:44 talk systemd[1]: Started LSB: Start the Kamailio SIP proxy server. plz help thank you -- *Regards:* Gaurav Kumar

3 3

Re: [SR-Users] [sr-dev] Failed to catch some reply message with reply_route block
by Henning Westerholt 27 Aug '19

27 Aug '19

(moved the discussion to user list) Hello Salah, I see the this output in the log message: 6(26) INFO: <script>: on_reply_route [SIP/2.0 200 OK Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bKe68a.99dc492fe74dcfe1017828521a5fa362.0 2(22) INFO: <script>: on_reply_route [SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bK178a.79cddb135aabfc42736fd9e9fd826ba8.0 and so on.. So you are seeing the replies correctly, or I wrong? Cheers, Henning Am 22.08.19 um 21:29 schrieb Salah Ahmed: Hello Henning, We already added xlog in the onreply_route and found all messages in between kamailio and UAS. The attached log files in first email has those logs. If any other specific log needed I am here to provide them. Thanks, Salah On Thu, Aug 22, 2019 at 2:24 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, ok, see if you are able to output some easy log with "xlog" in this route, to make sure there is no cfg problem somewhere. Cheers, Henning Am 22.08.19 um 21:15 schrieb Salah Ahmed: Hello Henning, Thanks for quick reply, We armed the t_on_reply just before t_relay() in the route block. t_on_reply("LOGRPL"); if (!t_relay()) { sl_reply_error(); } Thanks, Salah On Thu, Aug 22, 2019 at 2:03 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, the replies are going from the UAS to the UAC over Kamailo as a proxy, with the exception of the hop-by-hop 100. So you should see the 200 OK in the Kamailio in reply_route and onreply_route. Maybe you can check if you armed the onreply_route with t_on_reply for the INVITE routing. Cheers, Henning Am 22.08.19 um 19:50 schrieb Salah Ahmed: Hello, We facing an issue on capturing sip message in Kamailio(Version: 5.2.3). The scenario is very simple. UAC Kamailio UAS |--------INVITE--------->| | | |-------INVITE------->| | |<------100 Trying----| |<-------100 Trying------| | | |<------200 Ok--------| |<-------200 Ok----------| | In this simple scenario, we can't catch any reply messages in between the Kamailio and UAC. We have tried reply_route, onreply_route, and onsend_route. But no one work to grep that reply on that side. onsend_route was bad try as its only for forwarded reply message. Is there any other magic to capture those replies. A debug=3 log message attached here. Thanks, Salah _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

4 11

exit stop procesing or not after a conditional? based on "exit" core docs seems does not explain behaviour
by PICCORO McKAY Lenz 26 Aug '19

26 Aug '19

in my kamailio-asterisk integration i have a exit after the authentication logic, BUT; there's another conditional after the "exit" sentence, so my ask, if there's a exit in that piece of code .. that next conditional will not be executed? Docs seems said "stop execution of script" so all the rest of the logic in file will not be routed? The documentation seems does not explain in good behavior what happened or the kamailio-asterisk wiki page does not have good explanations for that! it's that an error? https://www.kamailio.org/wiki/cookbooks/4.3.x/core#exit - - ``` if (from_uri==myself) {#!ifdef WITH_ASTERISK if (!proxy_authorize("$fd", "sipusers")) {#!else if (!proxy_authorize("$fd", "subscriber")) {#!endif proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } ``` 8:52 am Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

reply cannot be forwarded - no 2nd via
by Ryan McEvoy 25 Aug '19

25 Aug '19

Hello, I have been having an issue using kamailio to load balance to a SIP/MRCP server. I believe that the issue actually lies with the reply being sent from the final recipient, but I wanted to be sure that I understood the problem and to see if there were any possible solutions within Kamailio. I am using the dispatcher module to dispatch SIP INVITES to media resources, in this case I am just using a single host in a single dispatcher group. The configuration that I have has worked just fine for other SIP/MRCP servers, which is why I think the issue is with this target in particular. The INVITE is routed correctly and contains the headers that it should. The invite comes originally from a Freeswitch, arrives at Kamailio, who should then dispatch to a host in the group. x.x.x.x:4901 is Kamailio x.x.x.x:8090 is Freeswitch y.y.y.y:8000 is the final target This is the invite, as received by the final target after dispatched via Kamailio INVITE sip: x.x.x.x:4901 SIP/2.0 Record-Route: <sip: x.x.x.x:4901;transport=tcp;lr> Via: SIP/2.0/TCP x.x.x.x:4901;branch=z9hG4bK3a9a.7f9a4b588cd75108cfea9a2bd9ffa829.0;i=3 Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S Max-Forwards: 69 From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ To: <sip: x.x.x.x:4901> Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE User-Agent: tts Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 281 v=0 o=IPSoft 3367790083977298880 2789991278648035021 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:existing a=resource:speechsynth a=cmid:1 m=audio 32038 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly a=mid:1 The reply that the server sends back appears to be missing a second Via header, which is preventing Kamailio for routing it correctly (based on the error message from core/forward.c) This is the reply sent by the target destination SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S To: <sip: x.x.x.x:4901>;tag=ndszib3k From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE Contact: "Voiceware MRCP Server" <sip:vwmrcp@ y.y.y.y:8000;transport=tcp> Content-Type: application/sdp Content-Length: 324 v=0 o=- 1566492047 1566492047 IN IP4 y.y.y.y s=Voiceware MRCP Server 2.15.0.7. Dec 28 2018 c=IN IP4 y.y.y.y t=0 0 m=application 8002 TCP/MRCPv2 1 a=channel:12951125463@speechsynth a=connection:new a=setup:passive a=cmid:1 m=audio 30158 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendonly a=ptime:20 a=mid:1 It is correct that there should still be a 2nd Via header in the 200 OK reply from the target host, right? If so, is there anything else that can be done in Kamailio to handle this, or is this mrcp server simply not complying properly with SIP? There’s no funky natting between these hosts or anything like that. Thanks in advance!

2 1

Wrong ACL to Provider kamailio+asterisk+uac
by Zhan Bazarov 25 Aug '19

25 Aug '19

Hello! Please, is it possible to change from/to headers in ACK message to provider? this is exactly our case, please, look it when u get a chance. http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html…

2 13

set_rtpengine_set question
by Juha Heinanen 24 Aug '19

24 Aug '19

When set_rtpengine_set() is called with two parameters before rtpengine_offer(), e.g. set_rtpengine_set("1", "2"); rtpengine_offer(); it was not clear to me from rtpengine README, if the sets need to reversed when set_rtpengine_set() is called before rtpengine_answer(), i.e., set_rtpengine_set("2", "1"); rtpengine_answer(); or can the order of the sets be the same as it was in case of rtpengine_offer()? -- Juha

1 1

Dealing with optional SRTP
by Mark Boyce 24 Aug '19

24 Aug '19

Hi all (Using Kamailio + RTPEngine, both current stable versions) Just wondering what current best practice is for dealing with “optional SRTP” such as those generated by Yealink devices. (I know this breaks the RFC but lots of things seem to use this method.) Where the device SDP message looks like; m=audio 50004 RTP/AVP 8 97 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhJOWOXPuzMeBmrbx2wAKohuveAvZeMEJTzfE7GT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yF+EhR3RtxSDCGtLCnEAl3Zrdut4b8cfFQP+hFqf An unencrypted RTP/AVP media line but with crypto lines showing that the device will support SRTP encryption but it is not mandatory. The system of offer SRTP and try and catch on not supported/acceptable here fail feels a little clumsy and relies on how gracefully the device sends “not acceptable here”. I’m wondering if I just short cut it and say that if a device offers a acceptable crypto line then Kamailio sends back a SAVP response. Naturally this question is then closely followed by how to do optional SRTP invites originating from Kamailio without just directly manipulating lines? Feels like there should be an RTPEngine setting for Optional? RTP/AVP+SAVP ? Thoughts? Mark

1 0

Kamailio as an Outbound Proxy - Strange Sip Flow
by Diogo Hartmann 23 Aug '19

23 Aug '19

Hello Kamailio experts, We have a Kamailio as outbound proxy in front of Asterisks PjSIP. There is a strange behavior after 200 OK: - ACK comes from Asterisk Public IP instead of coming from the internal IP address. - BYE can not be processed (404 not here) SNGREP print attached. Has anyone ever experienced this? I can provide more information if needed, just let me know what info. Thanks! Diogo Hartmann

2 5

Cant finish call
by przeqpiciel 23 Aug '19

23 Aug '19

Hi all, I am totally new with Kamailio but i have needs to build small cluster of Asterisk. Right now I could succesfully register to asterisk via kamailio and even make a call to hear tt-weasels but Asterisk cant close call My infrastructure diagram is here http://www.asciidraw.com/#Draw6816041626716434245/903175935 This is what probably I need is to know how to tell to Asterisk that send BYE to private IP of Kamailio instead public IP of Kamailio, am I right ?

1 0

TCP: Strange Device Behaviour
by Carsten Bock 23 Aug '19

23 Aug '19

Hi, I have a device here, which does some sort of strange things: I have SIP Message, e.g. with a length of 1100 Bytes and the devices adds "0" to the end to meet the MTU of the network (for a single TCP frame), some sort of TCP padding. The following TCP Frame contains only the next message. Looking at the trace in Wireshark, sngrep or similar, all of them try to re-assemble the next message in the TCP stream, but however don't decode them as SIP due to to first couple of zero's in the message. Looking at Kamailio, it seems not to have any issues with this and decodes/parses the message properly (they show up on Homer properly); however it also forwards the message INCLUDING the first couple of zero's even on UDP. To make it complitely irritating, the phone itself would ignore such packets, if we would forward them to the phone with starting zeros. Has anyone seen anything like this and is it standards-compliant? Any proper and easy way to get rid of the starting "zero's" from a message before forwarding these messages? I could always "hack" something into src/core/tcp_read.c, but I wonder if there is already a proper way? Any ideas anyone? Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

About STIR/SHAKEN - Caller Identity
by Daniel-Constantin Mierla 23 Aug '19

23 Aug '19

Hello, at couple of events I participated during the past few months, I was asked about support of STIR/SHAKEN (caller identity authentication/verification), which is a hot topic these days at least in USA, aiming to combat "fraudulent" robo-calling. Therefore I thought of share some details with everyone in the community about the state in Kamailio, writing to both devs and users, the information being relevant for everyone. We already have the (related) module named auth_identity, available since 2008 (iirc): - https://www.kamailio.org/docs/modules/stable/modules/auth_identity.html But it implements the previous iteration of the specs for caller identity, respectively RFC 4474: - https://tools.ietf.org/html/rfc4474 However, that RFC is obsoleted by 8224 (the latest core specs for STIR/SHAKEN): - https://tools.ietf.org/html/rfc8224 Then, there are also RFCs 8225 and 8226 to add to the core specs. Should anyone be interested to implement STIR/SHAKEN specs in a modules, I would suggest to start from auth_identity -- might not be much work to update it to become conform with latest specs (a new module can be created, of course, even when starting from auth_identity). However, these specs are about signing the SIP request (the INVITE) with special PKI certificate. It can be done easily with embedded scripts such as Lua or Python (inline execution in native kamailio.cfg or using kemi scripts). At Kamailio World 2019, one of the participants I discussed with told me they already implemented using Lua. That's it for a starting point, if anyone wants to discuss more, just reply to sr-users and add your comments or ask the questions. If someone wants to go ahead and work on a C module, announce yourself to avoid duplicate work of others, and use sr-dev if you need assistance on module development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Unable to start kamailio_pcscf.service
by Jack R 23 Aug '19

23 Aug '19

1. Followed steps from Step # 3 to Step # 16 : https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… 2. When I run last command >systemctl start kamailio_icscf kamailio_pcscf kamailio_scscfthe root@ubuntu18-04:/etc/init.d# *systemctl start kamailio_icscf kamailio_pcscf kamailio_scscf* Job for kamailio_pcscf.service failed because the control process exited with error code. See "systemctl status kamailio_pcscf.service" and "journalctl -xe" for details. 3. All services start but not kamailio_pcscf: root@ubuntu18-04:/etc/init.d# *systemctl status kamailio_pcscf.service* ● kamailio_pcscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_pcscf; generated) Active: failed (Result: exit-code) since Mon 2019-08-19 09:05:30 UTC; 45min ago Docs: man:systemd-sysv-generator(8) Process: 52271 ExecStart=/etc/init.d/kamailio_pcscf start (code=exited, status=1/FAILURE) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: ERROR: bad config file (1 errors) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: * Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Control process exited, code=exited status=1 Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Failed with result 'exit-code'. Aug 19 09:05:30 ubuntu18-04 systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. lines 1-16/16 (END) Please advise. Thanks and Regards.

2 10

basic Kamailio frontend to Asterisk
by Travis Ryan 22 Aug '19

22 Aug '19

All, I would like to use Kamailio as a frontend/proxy for an asterisk machine/cluster. I'm very well versed in Asterisk, but VERY noob to Kamailio. I've watched so many Kamailio presentations and feel like I've scoured the web for any article that could help me. There is alot out there, but I'm not finding my exact situation. I believe I understand the dispatcher stuff that I'll need down the road, but I'm having issues understanding the setup of my SIP trunk and Kamailio. I used UAC to setup the trunk connection in Kamailio and it sort of worked. I'm not sure what I do for Asterisk. I'd suspect that in this situation Kamailio would handle the registration and trunk connection for me. What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need to still register the trunk from each Asterisk box "thru" the Kamailio proxy, etc? Also, I'm merely accepting outside calls and then validating the caller and bridging them back out to the PSTN, so I don't have any local SIP clients, etc., so no need to register the sip devices, etc. Thanks! Travis

4 5

Docker image for 5.2.4
by Colin, Andrew 22 Aug '19

22 Aug '19

Hey All Who updates the images on https://hub.docker.com/r/kamailio/kamailio/tags i dont see the latest image kamailio:5.2.4-stretch --

1 0

{param.count,delimiter} issue
by Juha Heinanen 20 Aug '19

20 Aug '19

Looks like there is bug in {param.count,delimiter} transformation: $var(test) = "a=1;b=2"; xinfo("param count = $(var(test){param.count,;})\n"); $var(test) = "a=1,b=2"; xinfo("param count = $(var(test){param.count,,})\n"); syslog: Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 2 Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 1 -- Juha

3 10

Slowly increasing PKG memory usage
by Andrew White 20 Aug '19

20 Aug '19

Hi all, I had a Kamailio crash the other day, and some debugging showed I ran out of PKG memory. Since then I’ve run a simple bash script to compile the amount of memory used by all child processes, effective /usr/local/sbin/kamcmd pkg.stats | grep real_used summed together. I’ve graphed out the data, and there’s a clear growth of PKG memory going on, mostly increasing during our busier daytime hours. https://i.imgur.com/UTzx2k1.png Based on this, I suspect either a module loaded or something within my app_ruby conf is leaking memory. I’ve been reading through https://www.kamailio.org/wiki/tutorials/troubleshooting/memory <https://www.kamailio.org/wiki/tutorials/troubleshooting/memory>, but I’m a bit nervous, as I’m not really a C/deep memory type of guy. I can see a GDB script I can attach to Kamailio, but is that going to use significant resources to run or impact the running process? Is there a newer/better/alternative way to do this, and to help me break this down? Thanks! Andrew

3 17

Evapi dispatcher is consuming all the private memory
by José Seabra 20 Aug '19

20 Aug '19

Hello there, I'm using kamailio 5.2.1 with evapi module to send and receive events from an external application. During my stress tests to kamailio i have been noticing that at some point kamailio is getting out of memory: edge-sip-proxy[20565]ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/parser/msg_parser.c:328]: parse_headers(): memory allocation error edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: pv [pv_core.c:704]: pv_get_callid(): cannot parse Call-Id header edge-sip-proxy[20565]: INFO: ESP_LOG: <null>: [evapi:connection-received] EVAPI msg received -> {"aor":"sip:FUZECI1-x1516@uc.fuze.domain ","contact":"<sip:FUZECI1-x1516@10.225.121.206:5060;transport=UDP>"} edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! The process id 20565 is running out of memory as you can see bellow: output of kamcmd pkg.stats entry: 14 pid: rank: -2 used: 6058696 free: 824 real_used: 8387784 total_size: 8388608 total_frags: 30 This process ID is related with Evapi Dispatcher kamcmd ps | grep 20565 20565 EvAPI Dispatcher After stop the sipp that was generating traffic to kamailio the evapi process still have all the memory consumed Can it be some memory leak in Evapi module? Best Regards José Seabra -- Cumprimentos José Seabra

2 12

HSS download - open source
by Jack R 20 Aug '19

20 Aug '19

Please could you advise where can I get open source HSS that works with latest Kamailio 5.2 release for OS: ubuntu18-04:~$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 18.04.3 LTS Release: 18.04 Codename: bionic If possible please share the steps to integrate with Kamailio. The link in this article for FHoSS repository is not working: https://www.kamailio.org/wiki/tutorials/ims/installation-howto If not Open source then maybe an evaluation software. Please advise. Thanks and Regards.

2 1

kamailio doesn't reply on internal interface
by Zhan Bazarov 19 Aug '19

19 Aug '19

Hello, we have interesting issue... Some time kamailio comes unreachable on private interface for asterisk, but port checked via telnet and everything is ok... ps. connection via public ip works good root@kamailio:/usr/local/etc/kamailio# lsof -i :5068 | grep LISTEN kamailio 6014 root 13u IPv4 8341589 0t0 TCP kamailio.local.ip:5068 (LISTEN) kamailio 6014 root 14u IPv4 8341590 0t0 TCP kamailio.public.ip:5068 (LISTEN) root@kamailio:/usr/local/etc/kamailio# telnet from asterisk machine: root@asterisk:/etc/asterisk# telnet kamailio.local.ip 5068 Trying kamailio.local.ip... Connected to kamailio.local.ip. Escape character is '^]'. But peer is unreachable in asterisk... More than, kamailio comes reachable when I change transport from TCP to UDP, or when I restart kamailio daemon(but it situation repeats every day).. Any ideas how I can fix it? thanks!

2 1

set_rtpengine_set(set1, set2) issue
by Juha Heinanen 19 Aug '19

19 Aug '19

My sip proxy has two rtpengine sets: # sip-proxy_ctl rtpengine.show all { url: udp:192.26.134.1:6050 set: 0 index: 0 weight: 1 disabled: 0 recheck_ticks: 0 } { url: udp:192.26.134.40:6050 set: 1 index: 1 weight: 1 disabled: 0 recheck_ticks: 0 } Config makes callBefore calling rtpengine_offer(), config calls set_rtpengine_set("1", "0"); rtpengine_offer("..."); As result, debug shows that sip proxy sends TWO offer commands to the rtpproxy in set 1 and NONE to the rtpproxy in set 0. Debug at set 1 engine: Aug 19 18:06:08 buster-10-0 rtpengine[1314]: NOTICE: [5e1407211cc50fd9]: Creating new call Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.000812 sec) Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Received command 'offer' from 192.26.134.1:33284 Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.005247 sec) I haven't been able to figure out, why that is the case. It is a bug, or am I missing something? I added INFO call to set_rtpengine_set_f(): INFO("set1/set2 = %u/%u\n", selected_rtpp_set_1->id_set, selected_rtpp_set_2->id_set); return 1; and it correctly printed: Aug 19 18:06:08 salmon /usr/bin/sip-proxy[26911]: INFO: rtpengine [rtpengine.c:3574]: set_rtpengine_set_f(): set1/set2 = 1/0 Any ideas? -- Juha

1 0

MOH setup
by Maryam Baghdadi 19 Aug '19

19 Aug '19

Hello everyone, I need to setup Music on hold feature and I did all modifications in configuration file and did all steps for setup: 1- load MOH and all requirement modules . 2- rtpproxy setups and works correctly 3- mohqueue and mohqcalls tables created. 4- audio file with type index (novanet.8) exists 5- parameters of MOH module setup and mohq_process() function called in request_route()‌. 6- table mohqueue initiated like this‌: +----+------+----------------------------+-------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------+----------------------------+-------------------+---------+-------+ | 1 | main | sip:main at 191.38.245.2 | /var/kamailio/MOH | novanet | 1 | +----+------+----------------------------+-------------------+---------+-------+ Kamailio starts successfully and no error appears , call setup but on hold doesn't work and no audio file plays. I review all previous mailing lists responses but no one answers completely and before me others had this problem.

4 5

Replacing Caller ID
by Pete Kay 18 Aug '19

18 Aug '19

Hi Could anyone tell me if there is a module for replacing the caller ID in the SIP messages? I need to replace the caller ID when the call comes in and leaves Kamailio. When the egress sends back any response, I need to revert the caller ID back to its normal form. I have a long list of numbers to be used as caller-ID, and I need to keep modifying them without having to restart Kamailio. Do I need to write my own modules for maintaining the list of numbers to be used as caller-id? I am running at 2000 CPS, so I think I need to implement some sort of hash table in order for the caller id to be retrieved in round roubin fashion very quickly. Is there any modules for this purpose or I need to write a cusom one? Pete

2 1

Re: [SR-Users] Wildcarded IMPU
by Henning Westerholt 17 Aug '19

17 Aug '19

Hello Maryam, (please keep the list in CC) understand. There is work ongoing to update the IMS modules (e.g. refer to https://www.kamailio.org/wiki/features/new-in-devel). I don't know about a current effort to add wildcarded AVPs to them, though. If you like to implement something in this regards, it is always good to announce it quickly on the developer list sr-dev to get some feedback and to to synchronize with other developers. Cheers, Henning Am 17.08.19 um 08:04 schrieb Maryam Baghdadi: Hello Henning, Thank you for your response . I read all ETSI‌ standards , latest version of standards are according to Release15 IMS but IMS modules in kamailio support Release 7 IMS. for example there is Wildcarded_IMPU AVP in AUR in R15 IMS which isn't implemented in IMS_icscf module in kamailio . I know all AVP lists are exist in kamailio and I can add my requirements to modules code . I just wanted to know ims modules in kamailio will be updated or not? On Wed, Aug 14, 2019 at 11:55 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Maryam, Kamailio supports AVPs and also an extended version called XAVP. About your question - I quickly looked into ETSI TS 129 329 and TS 129 229 but they did not define how this actually works.. Can you elaborate what you actually are looking for? Cheers, Henning Am 05.08.19 um 12:41 schrieb Maryam Baghdadi: hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

uac_replace_from() and multiple branches
by John Quick 16 Aug '19

16 Aug '19

v5.2.3 of Kamailio. Serial forking is used to route calls to a multiple destinations as described in documentation for the TM module. Also using the DIALOG module - my script calls dlg_manage() before loading the serialised branches. I need to be able to substitute the username part of the From header and have been trying this with calls to uac_replace_from(). modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) There is no problem if all branches use one substituted From username, in which case my script calls uac_replace_from() in the main route. uac_replace_from("sip:" + $avp(route:src) + "@" + $fd); However, I may need to substitute a different value for each branch. Can this be done? I tried moving the call to uac_replace_from() into the branch route, but then I get an error when the final BYE is received: ERROR: uac [replace.c:281]: replace_uri(): decline FROM replacing in sequential request in auto mode (has TO tag) Is there a simple solution to this please? Hoping I can still use UAC auto restore mode because manual restore looks quite tedious/complex. Thanks. John Quick Smartvox Limited www.smartvox.co.uk

3 3

pike parameters doup when have dinamyc ip clients and scanners
by PICCORO McKAY Lenz 15 Aug '19

15 Aug '19

I read documentation for pike usage.. and have a doub.. what it's the best for very dinamyc ip of my devices.. i mean, i'm just curious of that very ironic and problematic scenario: for the scanners i setup fail2ban but only when the scaning are detecte.. but if i have the pike option how this: # this it's my setup for pike due the dinamyc ip and devices over the internet: modparam("pike", "sampling_time_unit", 4) modparam("pike", "reqs_density_per_unit", 80) modparam("pike", "remove_latency", 60) ... route { if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike block $rm from $fu (IP:$si:$sp)\n"); exit; } ... } I put the remove latency in 60, so then due are dinamycally must remian in memory more (due any one will be a possible clilent), just ban if there are 180 (60*3) request each 4 seconds, it's a good configuration or maybe i'm wrong please help me! -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 6

RFC 3860 support - Kamailio is sending SUBSCRIBE back with extra Via header
by Praveen Vatti 15 Aug '19

15 Aug '19

Hi Kamailio support, Basically, I am looking for RFC3860 support - i.e., accept SUBSCRIBE for reg event and send 200OK, NOTIFY to client. With some browsing on Internet I came across Kamailio, which is supporting RFC3860. I have installed and attached is my CFG file. I am sending a SUBSCRIBE request with reg event to Kamailio and it is sending back the same request with extra Via header. Via now contains Kamailio system IP and the source of SUBSCRIBE IP. Please help me in resolving this issue. I have loaded db_mysql, dialplan, tm, sl, presence, presence_reginfo modules. Thanks, Praveen

2 6

Wildcarded IMPU
by Maryam Baghdadi 14 Aug '19

14 Aug '19

hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks

2 1

Uac_replace for header's parameters
by Duarte Rocha 14 Aug '19

14 Aug '19

Greetings, I'm using uac_replace_to() and uac_replace_from() to do some number manipulations. It allows me to modify and automatically restore Display Number and URI. This works nicely, however it only allows to manipulate URI's parameter. Is there any method to the same and include the Header's parameters too? Best regards, Duarte Rocha

2 1

Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation
by Joel Serrano 14 Aug '19

14 Aug '19

Hello, Simple doubt regarding dispatcher module when you have the ds_default_socket() modparam set: Say you have 2 kamailio boxes, active + passive, one shared IP via keepalived.... On the actrive node, you have dispatcher.send_ping to 1, and the passive has it set to 0. So far OK, only the active box is sending out probing OPTIONS requests, and it's using as outbound socket the virtual IP out of all the available ones (as defined with ds_default_socket() modparam) I have seen in a capture that the outbound OPTIONS look like: OPTIONS sip:190.14.203.219:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bK7abb.6fbaccc1000000000000000000000000.0 To: <sip:190.14.203.219:5060> From: <sip:dspinger@my.domain>;tag=e2bdd495733c59fd91487a137fccad4e-8a73 CSeq: 10 OPTIONS Call-ID: 2019f8491329c382-31419(a)A.B.C.181 Max-Forwards: 70 Content-Length: 0 A.B.C.180 -> VRRP A.B.C.181 -> Physical IP of the box My doubt is, shouldn't ds_default_socket also update the IP used to generate the Call-ID used in the OPTIONS request? Currently, I believe it's using the first defined listen= IP from config script as I switched the order the listen= are defined and I get the desired result: OPTIONS sip:186.188.220.174:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bKc97e.a8d9e1c2000000000000000000000000.0 To: <sip:186.188.220.174:5060> From: <sip:dspinger@my.domain>;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323 CSeq: 10 OPTIONS Call-ID: 7d9a92c218fc1ba0-32111(a)A.B.C.180 Max-Forwards: 70 Content-Length: 0 Is this expected? Cheers, Joel.

3 4

Kamailio/Freeswitch auth user and UserID
by Petarr Jones 14 Aug '19

14 Aug '19

Hi I am new to Kamailio and would like some guidance on linking up with Freeswitch. I have an exiting hosted Asterisk PBX that uses a combination of extension and pbx username (eg 1001-126252581276) and I want to move it over to Kamailio/Freeswitch without re-configuring the SIP phones. I have tried Freeswitch "number-alias" but it does not seem like a good fit due to things like BLF not working and the added overhead etc. Is there a way to configure Kamailio to authenticate a user based on the above username and then send an extension to Freeswitch as the UserID instead, thereby separating the Kamailio authentication from the Freeswitch UserID? As I don't yet have a full understanding of the Kamailio/Freeswitch interaction, I don't know whether this will cause other issues? Thanks in advance Petarr

2 8

Re: [SR-Users] Dispatcher round-robin problems in 5.2
by JR Richardson 14 Aug '19

14 Aug '19

Hey Alex, After I recently upgraded, I had to include more response codes, 5.2 by default only 200 OK is accepted. I had some upstream carriers sending 405 and were taken off-line as a result until I added that code in modparm: modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=405") Not sure if this applies to what you are doing, but my dispatcher round robin distribution acts as it did prior to upgrade to 5.2, pretty even, no anomalies with any nodes getting more calls than any other. JR JR Richardson Engineering for the Masses Chasing the Azeotrope Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- Alex

2 2

Wrong Record-Route and Via header fields when TCP is used
by Leonid Fainshtein 14 Aug '19

14 Aug '19

Hi, Kamailio server has two legs that are connected to different networks. I'm using Kamailio v.5.2.3 and the "enable_double_rr" is implicitly set to "1". The leg "A" IP address is 10.159.65.1 The leg "B" IP address is 192.168.0.31 The call is initiated from 10.159.65.18 According to the "rr" module documentation, function record_route() should insert two "Record_Route" header fields when a request is accepted on one leg is should go out via the second leg. This works as expected in case of UDP protocol: INVITE sip:2000@192.168.6.106:5460;transport=UDP SIP/2.0 Record-Route: <sip:192.168.0.31;r2=on;lr;did=e2c.a191> Record-Route: <sip:10.159.65.1;r2=on;lr;did=e2c.a191> Via: SIP/2.0/UDP 192.168.0.31;branch=z9hG4bKcfa5.d64ecbd87d5315b5993c4ccf16f86537.0 Via: SIP/2.0/UDP 10.159.65.18:5060;rport=5060;branch=z9hG4bK3a9e9a4d But when the TCP protocol is used then the outbound message looks like this: INVITE sip:2005@192.168.0.178:35058;transport=tcp SIP/2.0 Record-Route: <sip:10.159.65.1;transport=tcp;lr;did=bb6.7dc1> Via: SIP/2.0/TCP 10.159.65.1;branch=z9hG4bKc85a.14afc3867dd3220826f9b9940f78168f.0;i=3 Via: SIP/2.0/TCP 10.159.65.18:5060;rport=58616;branch=z9hG4bK1469331f There are two problems there: a) only one Record-Route with leg is inserted b) the added "Via" header field contains the leg "A" IP address instead of expected leg "B" IP address (192.168.0.31). In the LAN trace I see that in reality the message was sent from leg "B". Is it a bug? Best regards, Leonid Fainshtein

2 13

Kamailio v5.2.4 Released
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, Kamailio SIP Server v5.2.4 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.4. For more details about version 5.2.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/08/kamailio-v5-2-4-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

1 0

Planning the release of Kamailio v5.2.4
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, I am considering to do soon a new release out of branch 5.2, respectively v5.2.4. I am aiming for August 12 or August 14, a matter of how backporting is finished, given that next week some devs travel to ClueCon. As usual, if you are aware of issues not reported yet on bug tracker or fixes not yet backported to branch 5.2, act during the next days to have enough time to approach them. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 3

Strange behavior for unregister() function. Wrong docs? Bug?
by Joel Serrano 13 Aug '19

13 Aug '19

Hello, I'm playing around with the unregister() function from the registrar module: https://www.kamailio.org/docs/modules/devel/modules/registrar.html#registra… Docs say that that it can be used in REQUEST_ROUTE or FAILURE_ROUTE. I needed it in REPLY_ROUTE, and before reading the docs I tried it and got this error: Aug 12 15:52:44 csbc01 kamailio: CRITICAL: <core> [core/cfg.y:3526]: yyerror_at(): parse error in config file /etc/kamailio/csbc/reply_routes.cfg, line 27, column 71: Command cannot be used in the block#012 Aug 12 15:52:44 csbc01 kamailio[12737]: ERROR: bad config file (1 errors) But, I have this working on another setup, I compared, and my difference is that I'm running unregister() inside a route, and calling that route from reply_route, and that works (and by works, I mean it also does what it is supposed to do, which is unregister the AoR). So this doesn't work: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } route(DEL_SIPSERVER); } ... But this works: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ route(UNREG_CUSTOMER); route(DEL_SIPSERVER); } ... ... route[UNREG_CUSTOMER] { if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } } ... So I'm in no-mans-land right now, docs say it's not allowed in REPLY_ROUTE, and if I try it it will fail, but if I try it inside a route, it woks (hack?). What is the correct approach here? Should unregister() be valid in reply_route (like other similar functions, for example: save()) and therefor docs should be updated and the error in the logs should also be updated, or if it's really not supported, shouldn't Kamailio not let me use it anywhere that a ends up being called from reply_route? I hope I have explained myself correctly. As always, please let me know if this should be a GH issue instead and I'll happily create it. Thanks, Joel.

3 5

Dispatcher round-robin problems in 5.2
by Alex Balashov 12 Aug '19

12 Aug '19

Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- 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 4

Kamailio DRA - Unable to get diameter_request() function to send
by Nick Jones 10 Aug '19

10 Aug '19

Hi Kamailio users, I'm trying to setup Kamailio as a Diameter Routing Agent, but I just can't seem to get the diameter_request() function to actually send a diameter request, I just get a warning about no JSON Response: *WARNING: ims_diameter_server [avp_helper.c:341]: addAVPsfromJSON(): No JSON Response* Inside my event route I've got one line to send the data to the peer "ims-hss.localdomain": *event_route[diameter:request]{ diameter_request("ims-hss.localdomain", $diameter_application, $diameter_command, $diameter_request); xlog("Forwarded Diameter Request");}* Hard as I try I can't seem to get this request forwarded to this peer and see that next xlog() line. In the config file I've included the cdp_check_peer & cdp_has_app checks to confirm the peer is up, and able to accept the request, (full config in pastebin below) I've defined the peers in the diametercfg.xml config file, and they're all showing as online when I do a "kamcmd cdp.list_peers": * FQDN: ims-hss.localdomain Details: { State: I_Open Disabled: False Last used: 0 Applications: { appid:vendorid: 16777216:10415 appid:vendorid: 16777216:4491 appid:vendorid: 16777216:13019 appid:vendorid: 16777216:0 appid:vendorid: 16777217:10415 appid:vendorid: 16777221:10415 } }* The source of avp_helper.c shows the warning is called if the length of the JSON is <= 0, but as I'm feeding back out what I've received it's not 0, an no recent major changes to the source, so I'm stumped as to why it's catching this. I'm running Kamailio 5.1.2 on Ubuntu 18.04 installed from Repos. Full Kamailio Config: https://pastebin.com/afgqUfWr Diameter CDP Config XML: https://pastebin.com/bVrBG8mG Relevant Syslog: https://pastebin.com/ZY8z2kd4 Kamcmd cdp.list_peers: https://pastebin.com/cKi4JAHC Any ideas as to what am I missing? Thanks in advance! Nick

1 0

need to Configure S-CSCF, I-CSCF and P-SCF
by ganesh nanaiah 10 Aug '19

10 Aug '19

Hi how do we configure the S-CSCF, I-CSCF and P-CSCF in kamalio 5.2.3 to authenicate with the HSS/PCRF, please point us to document to configure the IMS modules regards Ganesh

3 2

kamailio as UAC and NAPTR / sendsocket question
by Karsten Horsmann 09 Aug '19

09 Aug '19

Hi List, I try to register to Deutsche Telekom and there product Deutschland Lan siptrunk. Thats works find but i see an intressting behaviour on selecting the right outgoing interface. Kamailio is sending out with tcp the REQUEST via first private ip configured on that server (172.20.120.53). There is no listen directive for that. I forced NAPTR to use tcp or udp and i assume that kamailio got the right dns answers. On the list i read also that i can use force_send_socket to force the outgoing request. Now my idea - hey i use the $rP for the outgoing to select the right outgoing listen directive. $rP - reference to transport protocol of R-URI But to my surprise the logfile told me thats "UDP" - it sends out via TCP (thats okay). Whats an good transport selector variable from kamailio that works? event_route[tm:local-request] { if(!is_method("OPTIONS")) { xlog("L_INFO", "[tm:local-request] request [$rm] from [$fu] to [$ru] [$rP]\n"); } } INFO: <script>: [tm:local-request] request [REGISTER] from [ sip:+49XXXXXXXX@sip-trunk.telekom.de] to [sip:sip-trunk.telekom.de] [UDP] listen=tcp:2xx.xx.xx.xx:5060 listen=udp:2xx.xx.xx.xx:5060 listen=tls:2xx.xx.xx.xx:5061 advertise CFG_EXT_NAME:5061 listen=udp:172.20.120.55:5060 listen=udp:172.20.120.56:5060 listen=udp:172.20.120.57:5060 listen=udp:172.20.120.58:5060 listen=tcp:172.20.120.58:5060 use_dns_cache=on # use internal DNS cache use_dns_failover=on # depends on internal DNS cache dns_srv_loadbalancing=on dns_try_naptr=on dns_retr_time=1 # seconds before retrying a DNS request dns_retr_no=3 # number of DNS retransmissions dns_naptr_ignore_rfc=yes # ignore target NAPTR priority dns_tcp_pref=30 # TCP has second-highest priority dns_udp_pref=10 # use UDP with least priority tcp_connection_lifetime=3605 # set higher than registration expires #dont' restore modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) ## UAC REGISTER #!ifdef WITH_UAC_REGISTER modparam("uac", "reg_contact_addr", "CFG_PROD_IP") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_db_url", DBURL) modparam("uac", "restore_mode", "none") modparam("uac", "auth_username_avp", "$avp(AVP_AUTH_USERNAME)") modparam("uac", "auth_password_avp", "$avp(AVP_AUTH_PASSWORD)") modparam("uac", "auth_realm_avp", "$avp(AVP_AUTH_REALM)") #!endif ip a l 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN group default qlen 1000 link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo valid_lft forever preferred_lft forever inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: ens192: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP group default qlen 1000 link/ether 00:50:56:b5:c1:48 brd ff:ff:ff:ff:ff:ff inet 172.20.120.53/24 brd 172.20.120.255 scope global ens192 valid_lft forever preferred_lft forever inet 2xx.xx.xx.xx/29 scope global ens192 valid_lft forever preferred_lft forever inet 172.20.120.56/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.57/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.58/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.55/24 brd 172.20.120.255 scope global secondary ens192 valid_lft forever preferred_lft forever inet6 fe80::250:56ff:feb5:c148/64 scope link valid_lft forever preferred_lft forever default via 172.20.120.253 dev ens192 172.20.120.0/24 dev ens192 proto kernel scope link src 172.20.120.53 2xx.xx.xx.xx/29 dev ens192 proto kernel scope link src 2xx.xx.xx.xx -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

4 9

failure_route - no UAC support
by Attila Czébán 09 Aug '19

09 Aug '19

Hello, I'm trying to restart phones with local-requests (from config or kamctl cisco_restart). I can enable t_on_failure("REMOTE_AUTH") in event_route[tm:local-request] but when 401 Reply comes in I get this warning: "tm [t_reply.c:968]: run_failure_handlers(): no UAC support (1, 0)". kamailio -V version: kamailio 5.1.8 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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_BLACKLIST, 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: unknown compiled with gcc 6.3.0 How can I make it work? Thank you! Regards, Attila

1 0

Kamailio auth via http post.
by Zhan Bazarov 09 Aug '19

09 Aug '19

Hello, Its possible to made authentication via http request to third party Web-API instead of storing users credentials into Dataqbase? Now I looking on auth_ephemeral, but not sure is that my case... Don't see when I have to put API url.. Could anyone, please, shed any light on this issue?

2 1

XMLRPC request URL
by David Villasmil 08 Aug '19

08 Aug '19

Hello guys, I can't find any reference to a pseudovariable containing the request url, i.e.: http://myserver/XMLRPC?server=1.2.3.4 I'd like to get that server variable and manipulate the dispatcher list... the http client can't do XMLRPC or add a body, just an https request.. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Breaking HTABLE changes from v5.2 to v5.3? Function renamed
by Joel Serrano 08 Aug '19

08 Aug '19

Hi all, Just noticed that when updating from 5.2 to 5.3 I saw some errors in Kam logfile regarding HTABLE module commands failing, the reason is the function 'sht_has_name' has been renamed to 'sht_match_name': https://www.kamailio.org/docs/modules/stable/modules/htable.html#htable.f.s… https://www.kamailio.org/docs/modules/devel/modules/htable.html#htable.f.sh… Not sure if updating this doc is automatic or manual but I think it's definitely worth mentioning: https://www.kamailio.org/wiki/install/upgrade/stable-to-devel Just FYI! Cheers, Joel.

2 1

Re: [SR-Users] Sqlops
by arish haque 08 Aug '19

08 Aug '19

Hi, Passing the value within single quotes works. Thanks for the help Fred. With Regards, Arish

1 0

Reload geoip2 database without restart
by Joel Serrano 07 Aug '19

07 Aug '19

Hi guys, Is there a way to instruct the geoip2 module to reload its database without a full restart of kamailio? Maybe some rpc command or something? I haven't found anything in the docs other than the comment about "Database is loaded at startup in cache." which makes me assume that just overwriting the db file wouldn't be enough. Any suggestions? Thanks!! Joel.

2 2

Sipt redirection number retrieves old values
by Duarte Rocha 07 Aug '19

07 Aug '19

Greetings, I'm doing VoiceMail detection using $sipt(redirection_number) from SIPT module and i've been having some false positives on the detection. I inspected further and noticed that in some cases $sipt(redirection_number) returns old values from previous messages, which makes me think that it's value is not being correctly refreshed. I'm using this variable on a on_reply_route. As far as i know, $sipt(redirection_number) is read only and can't be refreshed by me in the code. Here is an example : Aug 7 16:51:49 dev_sbc_a_vts01 /usr/local/kamailio-5.2/sbin/kamailio[4746]: ERROR: <script>: VMD2 : Reply Code : 180 ; Sipt Redir = 9610050000 and Call ID = 00158FED-98B7-E911-8B3B-4DA50AE2ED72(a)192.168.24.86 and Cseq = 83 Here is the JSON from the 180 message : https://pastebin.com/zFHKxyav As you can see, there isn't any 9610050000 in the message. What can be causing this issue ? Best Regards, Duarte Rocha

2 2

Sqlops
by arish haque 07 Aug '19

07 Aug '19

Hi all, I am querying mysql db inside routing block in kamailio.cfg ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column '8d77' in 'where clause' (1054) Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [select * from vca_asterisk where front_id=8d77] I'm getting the above error message when, variable front_id=8d77 is alphanumeric. It's working fine when front_id is pure numeric. eg: front_id=877 Database column front_id is VARCHAR(255). Any suggestions to solve this error? Thanks, Arish

2 1

ul.rm_contact usage
by Володимир Іванець 07 Aug '19

07 Aug '19

Hello, I'm trying to use *ul.rm_contact* RPC command to unregister spesific Contact but can not figure out "contact" parameter. I tried different combinations but received "error: 404 - Contact not found" message every time. Could please someone tell how this command should be called for one of registered Contacts? Thank you! *# kamcmd ul.dump* *{* * Domains: {* * Domain: {* * Domain: location* * Size: 1024* * AoRs: {* * Info: {* * AoR: 200-test* * HashID: 406934013* * Contacts: {* * Contact: {* * Address: sip:200-test@1.1.1.1:52181;transport=TCP;rinstance=5d1a0b6b677f269d* * Expires: 114* * Q: -1.000000* * Call-ID: YmNlMmJlNDMwMGY5NGFkYzkzOTU2NDFhNzc5NWE3Mzc.* * CSeq: 4* * User-Agent: 3CXPhone 6.0.26523.0* * Received: sip:1.1.1.1:52181;transport=tcp* * Path: [not set]* * State: CS_NEW* * Flags: 0* * CFlags: 65536* * Socket: tcp:172.16.30.205:5060 <http://172.16.30.205:5060>* * Methods: 5119* * Ruid: uloc-5d4983fb-40b5-4* * Instance: [not set]* * Reg-Id: 0* * Server-Id: 0* * Tcpconn-Id: 3* * Keepalive: 0* * Last-Keepalive: 1565099252* * Last-Modified: 1565099252* * }* * Contact: {* * Address: sip:200-test@1.1.1.1:58166;rinstance=5996b96740d52a3a* * Expires: 118* * Q: -1.000000* * Call-ID: 97566MWMxYmVhMmE1OGUyOTlkMzVmYTYzZmE2ZDE5YWJkYTc* * CSeq: 4* * User-Agent: X-Lite release 5.5.0 stamp 97566* * Received: sip:1.1.1.1:58166 <http://1.1.1.1:58166>* * Path: [not set]* * State: CS_NEW* * Flags: 0* * CFlags: 65536* * Socket: udp:172.16.30.205:5060 <http://172.16.30.205:5060>* * Methods: 5087* * Ruid: uloc-5d4983fb-4093-2* * Instance: [not set]* * Reg-Id: 0* * Server-Id: 0* * Tcpconn-Id: -1* * Keepalive: 0* * Last-Keepalive: 1565099256* * Last-Modified: 1565099256* * }* * }* * }* * }* * Stats: {* * Records: 1* * Max-Slots: 1* * }* * }* * }* *}*

2 2

100% CPU usage after NTP date update
by Benjamin Marty 07 Aug '19

07 Aug '19

Hello, I'm facing the same issue. Is there a solution for this in the meantime? Benjamin https://lists.kamailio.org/pipermail/sr-users/2015-November/090487.html

2 2

Problem with tcp connections
by logosvlad＠gmail.com 07 Aug '19

07 Aug '19

Hello! we have kamailio 5.2.3 and openssl 1.1.0f. wss connections incoming to kamailio and outgoing sip connections, after some time http request check not getting answer from kamailio. Netstat show increase recieve tcp buffer on connections of kamailio. We set up script which after not response of http check, run SIGSEGV of main process of kamailio. gdb) bt full #0 0x00007fb779279450 in futex_wait (private=<optimized out>, expected=1, futex_word=0x7fb6f78cb964) at ../sysdeps/unix/sysv/linux/futex-internal.h:61 __ret = -512 err = <optimized out> #1 futex_wait_simple (private=<optimized out>, expected=1, futex_word=0x7fb6f78cb964) at ../sysdeps/nptl/futex-internal.h:135 No locals. #2 __pthread_rwlock_wrlock_slow (rwlock=0x7fb6f78cb958) at pthread_rwlock_wrlock.c:67 waitval = 1 result = 0 futex_shared = <optimized out> #3 0x00007fb77995eee9 in CRYPTO_THREAD_write_lock () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #4 0x00007fb779907896 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #5 0x00007fb77990807e in CRYPTO_free_ex_data () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #6 0x00007fb779c65a09 in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 No symbol table info available. #7 0x00007fb777c022d6 in tls_h_tcpconn_clean (c=0x7fb6fbb73d58) at tls_server.c:655 extra = 0x7fb6fba1a6d0 __func__ = "tls_h_tcpconn_clean" #8 0x0000559ddff2966f in _tcpconn_free (c=0x7fb6fbb73d58) at core/tcp_main.c:1339 __func__ = "_tcpconn_free" #9 0x0000559ddff3c5f7 in tcpconn_put_destroy (tcpconn=0x7fb6fbb73d58) at core/tcp_main.c:3017 __func__ = "tcpconn_put_destroy" #10 0x0000559ddff3e0f2 in send_fd_queue_run (q=0x559de0518bc0 <send2child_q>) at core/tcp_main.c:3210 p = 0x7fb77e19e2c8 t = 0x7fb77e19e2c8 __func__ = "send_fd_queue_run" #11 0x0000559ddff4f61c in tcp_main_loop () at core/tcp_main.c:4548 si = 0x0 r = 10 __func__ = "tcp_main_loop" #12 0x0000559ddfe4e85d in main_loop () at main.c:1753 i = 3 pid = 0 si = 0x0 si_desc = "udp receiver child=2 sock=127.0.0.1:5060 (127.0.0.1:5060)\000\065\060\066\060)\000\000\000\000\200\000\000\000\000\000\000\000\002\000\000\000\000\006\000\000\000\000\000\000\000O \254\177\267\177\000\000p\324\067?\375\177\000\000\060\000\000\000\060\000\000\000\020\325\067?\375\177\000\000\360k\255}\267\177\000" nrprocs = 3 woneinit = 1 __func__ = "main_loop" #13 0x0000559ddfe554ce in main (argc=13, argv=0x7ffd3f37d7a8) at main.c:2696 cfg_stream = 0x559de1c10010 c = -1 r = 0 tmp = 0x7ffd3f37deba "" tmp_len = 32765 port = 1060623840 proto = 32695 options = 0x559de01c2a50 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 33232859 ---Type <return> to continue, or q <return> to quit--- rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0xf0b5ff <error: Cannot access memory at address 0xf0b5ff> st = {st_dev = 19, st_ino = 13222, st_nlink = 2, st_mode = 16832, st_uid = 108, st_gid = 112, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = { tv_sec = 1560780032, tv_nsec = 539855841}, st_mtim = {tv_sec = 1565076521, tv_nsec = 998570962}, st_ctim = {tv_sec = 1565076521, tv_nsec = 998570962}, __glibc_reserved = {0, 0, 0}} __func__ = "main" Maybe our problem correlate with https://github.com/kamailio/kamailio/issues/1860

1 0

app_ruby - invalid ruby environment attributes or parameters
by Andrew White 07 Aug '19

07 Aug '19

Hey all, I’m setting up a dev-7 (bd477f) server to test as per the other thread, however I’m getting the following loading my config: ERROR: app_ruby [app_ruby_api.c:465]: sr_kemi_ruby_exec_func_ex(): invalid ruby environment attributes or parameters I’m unsure what this error means - are there expected environment variables or other values that must be set? I didn’t see this issue in dev-4. I thought it might be not able to find the library, but I ran with debug=4 and saw: DEBUG: app_ruby [app_ruby_api.c:193]: ruby_sr_init_child(): test execution without error ERROR: app_ruby [app_ruby_api.c:107]: app_ruby_print_last_exception(): exception: undefined method `>' for false:FalseClass I changed kamailio.cfg to load my generic routing script, and during the onreply_manage it runs: scode = KSR::PV.get("$rs”) This now returns false (which appears to be the second error above, I check if scode is > 599). Has anything changed in the way ruby is loaded in dev-7? I’m using the same ruby/ruby-devel library versions on the same OS. I did this deploy the same with devel-4 a few weeks ago and this didn’t happen. Thanks! Andrew

3 8

using in almost one worker
by logosvlad＠gmail.com 06 Aug '19

06 Aug '19

Hello! We have in configuration : c <sr-users(a)lists.kamailio.org>hildren=3 tcp_children=30 But in 90-95% connections kamailio use one worker (one process) by statitistics netstat -anp. Why not using more other workers (processes) ?

2 2

what version openssl recomnded for kamailio 5.2.3
by logosvlad＠gmail.com 06 Aug '19

06 Aug '19

Hello! what version openssl 1.1.x did you recomnded for kamailio 5.2.3 for more stable operation ?

1 0

The domain module in 5.1.8 has a memory leak
by Mack Hendricks 06 Aug '19

06 Aug '19

Based on my tests, the memory leak is only exposed when you are using domain attributes, which are stored in the domain_attrs table. I wanted to see if anymore else is running into this issue. *Reproducible use case:* 1) Insert the following into the database insert into domain values (null,"aprilco.com","aprilco.com",now()); insert into domain_attrs values (null,"aprilco.com","pbx_type",0,"0",now()); insert into domain_attrs values (null,"aprilco.com","pbx_id",0,"65",now()); 2) Open a new terminal window and execute: watch "kamcmd core.shmmem" 3) Open up another terminal windows and execute: while true; do kamcmd domain.reload; done *Expected Results:* You will see the shared memory being allocated, but never freed. The more entries you have in the domain_attrs table, the faster it will run out of memory. Thanks, *Mack Hendricks / Head of Support / dOpenSource* web: http://dopensource.com support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

2 3

Error Registering the user and finding the AVP
by jespinosa 06 Aug '19

06 Aug '19

Hello, My name is Javier, I must first apologize for my little knowledge about the tool and my poor English. I am trying to connect a functional core network LTE with Kamailio and I have used the Polaris Networks Solution that brings an iso with both functionalities. The EPC is correctly configured and working because I can have data connection in my UE, I have also activated all the IMS options in the EPC: I have created the APN with ims identifier, the default carrier QCI assigned is 5 (for SIP signaling), the MME has activated the option to support IMS Voice over the PS session and the HSS and PCRF nodes are configured with the interfaces at the addresses and ports Cx-HSS 127.0.0.1:3868 and Cx-PCRF 10.11.0.35:3868 . There are several parameters that I do not know if it is necessary to activate, for example, in the PCRF the customization of the QCI for the media type, if I must include the OCS and OFCS as nodes, or if in the profile of my APN I must include authentication by PDN AAA. The configuration I use for the IMS is as follows: imsrealm = net1.test imsport = 5060 pcscfgmip = 10.77.0.35 pcscfgmipintf = eth6 pcscfrxip = 10.102.81.111 pcscfrxipintf = eth1 icscfmwip = 127.0.0.21 icscfmwipintf = lo:21 icscfcxip = 127.0.0.20 icscfcxipintf = lo:20 scscfmwip = 127.0.0.23 scscfmwipintf = lo:23 scscfcxip = 127.0.0.22 scscfcxipintf = lo:22 audiobandwidth = 64 videobandwidth = 128 epcrealm = epc.mnc001.mcc001.3gppnetwork.org pcrfhost = pcrf2.morsecore pcrfrxip = 10.11.0.35 pcrfrxport = 3868 hsshost = hss6.morsecore hsscxdxip = 127.0.0.1 hsscxdxport = 3868 Both cores are running on the same machine but I have disabled the option of PCRF Co-located with IMS to be able to edit the pcrf host name since if I enable it it does not reach the peer pcrf.localhost even if the DNS is well configured. My UE is not able to launch the REGISTER by itself when it is going to make a call, so I am using the Google IMSDroid application to be able to launch the SIP requests and check the operation of the IMS. In the application of the UE the configuration is as follows: Identity: Alice Public Identity: sip:<MSISDN>@net1.test (MSISDN=8800000451 configurado previamente en el EPC) Private Identity: sip:<IMSI>@net1.test (IMSI incluyendo MCC y MNC) Password: alice Realm: net1.test (Aunque no sé si debo poner aquí epc.mnc001.mcc001.3gppnetwork.org, pero entonces la petición da error) Network: Proxy-CSCF Host: 10.77.0.35:5060 Transport UDP and NO Proxy-CSCF discovery. So, I launch my request and the log of the P-CSCF node is as follows: Aug 2 12:50:04 morsecore kamailio[53656]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:04 morsecore kamailio[53656]: INFO: cdp [authstatemachine.c:200]: auth_client_statefull_sm_process(): after callback of event 1 Aug 2 12:50:04 morsecore kamailio[53802]: INFO: cdp [authstatemachine.c:200]: auth_client_statefull_sm_process(): after callback of event 7 Aug 2 12:50:04 morsecore kamailio[53655]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:05 morsecore kamailio[53664]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:07 morsecore kamailio[53659]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:11 morsecore kamailio[53666]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) I understand that everything is correct until now, but in the I-CSCF node it appears: Aug 2 12:50:04 morsecore kamailio[53397]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_capabilities: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: ERROR: ims_icscf [cxdx_uar.c:152]: async_cdp_uar_callback(): RC_IMS_DIAMETER_ERROR_USER_UNKNOWN Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Aug 2 12:50:04 morsecore kamailio[53465]: ERROR: <script>: UAR failure - error response sent from module Where I believe that the REGISTER message is not even forwarded, however it seems that it gives an error that the user is unknown or not registered but if I access the MySQL database, the registered users are: +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | alice | net1.test | alice | | 91cc8c652ef5f37e91c468c7319d0086 | 044776963e2661788e8db076b393333e | NULL | | 2 | bob | net1.test | bob | | 6ef7f9c924f1385d83cf22cbb94cfbbb | 0d893d2d8c10a044b918e4bd73b6bee5 | NULL | | 3 | 8800000451 | net1.test | alice | | 46811c4e58f298953a6f9efccf6c32f1 | 49c7a7ad84fe797fbefa2185be4cd226 | NULL | +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ However, I do not know if it is accessing the mounted MySQL database or accessing the HSS and on this node, in the databases it does not contain the user (although it must be because it is registered in the network correctly). In case it works, this is the trace of wireshark filtered by the protocols S1AP, SIP and Diameter just when a REGISTER request is launched: "596","2019-08-02 12:50:00.705985000","127.0.0.20","127.0.0.1","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "597","2019-08-02 12:50:00.706133000","127.0.0.1","127.0.0.20","DIAMETER","210","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "599","2019-08-02 12:50:00.786878000","127.0.0.22","127.0.0.1","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "600","2019-08-02 12:50:00.787019000","127.0.0.1","127.0.0.22","DIAMETER","210","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "602","2019-08-02 12:50:00.914405000","10.102.81.111","10.11.0.35","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "603","2019-08-02 12:50:00.914549000","10.11.0.35","10.102.81.111","DIAMETER","214","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "622","2019-08-02 12:50:04.027238000","10.77.0.196","10.77.0.35","GTP <SIP>","1034","Request: REGISTER sip:net1.test (1 binding) | " "623","2019-08-02 12:50:04.027347000","10.77.0.196","10.77.0.35","GTP <SIP>","1034","Request: REGISTER sip:net1.test (1 binding) | " "624","2019-08-02 12:50:04.027448000","10.77.0.196","10.77.0.35","SIP","998","Request: REGISTER sip:net1.test (1 binding) | " "625","2019-08-02 12:50:04.028850000","10.102.81.111","10.11.0.35","DIAMETER","426","cmd=AA Request(265) flags=RP-- appl=3GPP Rx(16777236) h2h=fedcb8a e2e=4df0bbf6 | " "626","2019-08-02 12:50:04.029862000","10.11.0.35","10.102.81.111","DIAMETER","350","cmd=AA Answer(265) flags=-P-- appl=3GPP Rx(16777236) h2h=fedcb8a e2e=4df0bbf6 | " "630","2019-08-02 12:50:04.031113000","10.77.0.35","127.0.0.21","SIP","1280","Request: REGISTER sip:net1.test (1 binding) | " "631","2019-08-02 12:50:04.031794000","127.0.0.21","10.77.0.35","SIP","440","Status: 100 Trying | " "632","2019-08-02 12:50:04.032382000","127.0.0.20","127.0.0.1","DIAMETER","354","cmd=User-Authorization Request(300) flags=RP-- appl=3GPP Cx(16777216) h2h=fedcb8a e2e=4df0bbf6 | " "633","2019-08-02 12:50:04.033101000","127.0.0.1","127.0.0.20","DIAMETER","298","cmd=User-Authorization Answer(300) flags=-P-- appl=3GPP Cx(16777216) h2h=fedcb8a e2e=4df0bbf6 | " "636","2019-08-02 12:50:04.033557000","127.0.0.21","10.77.0.35","SIP","504","Status: 403 Forbidden - HSS User Unknown | " I don't know if any information or configuration details are missing but I can attach it at any time without any problem. Regards and thank you very much in advance, Javier Espinosa.

2 3

Kamailio Process CPU 100%
by M. Salman 05 Aug '19

05 Aug '19

Hi Team, Facing 100% Kamilio CPU issue, here are the details, any pointers/hints would help. Thanks *Scenario:* Random Kamailio CPU shoot, restart fixes it. Happens once or twice in a day. It's a pre-production server with around 30 extensions, not much load. For now, it has 2 CPUs and 4 G memory. *[Info]* uname -a Linux ip-X-aws #46-Ubuntu SMP Thu Jul 4 13:38:28 UTC 2019 x86_64 x86_64 x86_64 GNU/Linux kamailio -V version: kamailio 5.2.3 (x86_64/linux) c36229 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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_BLACKLIST, 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: c36229 *(gdb) bt full* #0 0x00007f0069d97aab in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #1 0x00007f0069d7d6bd in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #2 0x00007f0069d657c7 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #3 0x00007f0069d3ebce in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #4 0x00007f0069d3fc4c in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #5 0x00007f0069d33edc in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #6 0x00007f0069d34fa8 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #7 0x00007f0069d36d71 in mysql_real_query () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #8 0x00007f006a31e4b4 in db_mysql_submit_query (h=0x7f006a6cdb20, s=0x7f0069af92d0 <sql_str>) at km_dbase.c:116 t = 1564528172 i = 0 code = 0 _func_ = "db_mysql_submit_query" #9 0x00007f00698e8fb3 in db_do_submit_query (h=0x7f006a6cdb20, query=0x7f0069af92d0 <sql_str>, submit_query=0x7f006a31dd3c <db_mysql_submit_query>) at db_query.c:58 ret = 10 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 2077076196242373120} tz = {tz_minuteswest = 0, tz_dsttime = 0} tdiff = 0 _func_ = "db_do_submit_query" #10 0x00007f00698ec1dc in db_do_delete (h=0x7f006a6cdb20, k=0x7ffcdb440fa0, o=0x0, v=0x7ffcdb440fb0, _n=2, val2str=0x7f006a340429 <db_mysql_val2str>, submit_query=0x7f006a31dd3c <db_mysql_submit_query>) at db_query.c:299 off = 68 ret = 36 ---Type <return> to continue, or q <return> to quit--- _func_ = "db_do_delete" #11 0x00007f006a322f6a in db_mysql_delete (h=0x7f006a6cdb20, k=0x7ffcdb440fa0, o=0x0, v=0x7ffcdb440fb0, _n=2) at km_dbase.c:515 No locals. #12 0x00007f0065b6da70 in remove_dialog_from_db (cell=0x7f005c5ed3d8) at dlg_db_handler.c:722 values = {{type = DB1_INT, nul = 0, free = 477963473, val = {int_val = 1460, ll_val = 140720308487604, double_val = 6.9525070095907518e-310, time_val = 140720308487604, string_val = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, str_val = { s = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, len = 0}, blob_val = { s = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, len = 0}, bitmap_val = 1460, uint_val = 1460, ull_val = 140720308487604}}, {type = DB1_INT, nul = 0, free = 0, val = {int_val = 8020, ll_val = 139637976735572, double_val = 6.899032715982534e-310, time_val = 139637976735572, string_val = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, str_val = { s = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, len = 0}, blob_val = { s = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, len = 0}, bitmap_val = 8020, uint_val = 8020, ull_val = 139637976735572}}} match_keys = {0x7f0065dfc6c0 <h_entry_column>, 0x7f0065dfc6b0 <h_id_column>} vars_match_keys = {0x7f0065dfc7f0 <vars_h_entry_column>, 0x7f0065dfc7e0 <vars_h_id_column>} _func_ = "remove_dialog_from_db" #13 0x00007f0065b49375 in destroy_dlg (dlg=0x7f005c5ed3d8) at dlg_hash.c:371 ret = 1 var = 0x7f0065b45442 <futex_get+40> _func_ = "destroy_dlg" #14 0x00007f0065b50ca0 in dlg_unref_helper (dlg=0x7f005c5ed3d8, cnt=2, fname=0x7f0065be1eff "dlg_handlers.c", fline=1182) at dlg_hash.c:961 d_entry = 0x7f005c45b4c0 _func_ = "dlg_unref_helper" #15 0x00007f0065b7f708 in unref_dlg_from_cb (t=0x7f005c5b5dc0, type=131072, param=0x7ffcdb4412a0) at dlg_handlers.c:1182 dlg = 0x7f005c5ed3d8 iuid = 0x7f005c568c38 #16 0x00007f0068295b05 in run_trans_callbacks_internal (cb_lst=0x7f005c5b5e38, type=131072, trans=0x7f005c5b5dc0, params=0x7ffcdb4412a0) at t_hooks.c:260 cbp = 0x7f005c606118 backup_from = 0x5569cc3da8b0 <def_list+16> backup_to = 0x5569cc3da8b8 <def_list+24> ---Type <return> to continue, or q <return> to quit--- backup_dom_from = 0x5569cc3da8c0 <def_list+32> backup_dom_to = 0x5569cc3da8c8 <def_list+40> backup_uri_from = 0x5569cc3da8a0 <def_list> backup_uri_to = 0x5569cc3da8a8 <def_list+8> backup_xavps = 0x5569cc3da2c0 <_xavp_list_head> _func_ = "run_trans_callbacks_internal" #17 0x00007f0068295c2c in run_trans_callbacks (type=131072, trans=0x7f005c5b5dc0, req=0x0, rpl=0x0, code=0) at t_hooks.c:287 params = {req = 0x0, rpl = 0x0, param = 0x7f005c606128, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = { s = 0x0, len = 0}} #18 0x00007f00681cbf45 in free_cell_helper (dead_cell=0x7f005c5b5dc0, silent=0, fname=0x7f00682de882 "timer.c", fline=689) at h_table.c:166 b = 0x7f005c5b5dc0 "" i = 1 rpl = 0x7ffcdb4413a0 tt = 0x0 foo = 0x7ffcdb441390 cbs = 0x7f005c378700 cbs_tmp = 0x1681ca80f _func_ = "free_cell_helper" #19 0x00007f0068283bf6 in wait_handler (ti=855479619, wait_tl=0x7f005c5b5e48, data=0x7f005c5b5dc0) at timer.c:689 p_cell = 0x7f005c5b5dc0 ret = 1 unlinked = 0 rcount = 1 _func_ = "wait_handler" #20 0x00005569cbdb0661 in timer_list_expire (t=855479619, h=0x7f005c238af0, slow_l=0x7f005c23ae08, slow_mark=5646) at core/timer.c:874 tl = 0x7f005c5b5e48 ret = 0 #21 0x00005569cbdb0b09 in timer_handler () at core/timer.c:939 saved_ticks = 855479619 run_slow_timer = 0 i = 526 _func_ = "timer_handler" #22 0x00005569cbdb0fd2 in timer_main () at core/timer.c:978 ---Type <return> to continue, or q <return> to quit--- No locals. #23 0x00005569cbcf1382 in main_loop () at main.c:1727 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=172.31.2.33:6000 ( 13.54.252.29:6000)\000\034\004\000\000\000\000\000\000\000\321$}\034\000\000\000\000P0\316\313iU\000\000 \031D\333\374\177", '\000' <repeats 18 times>, "`\026D\333\374\177\000\000\360S\361\313iU\000" nrprocs = 8 woneinit = 1 _func_ = "main_loop" #24 0x00005569cbcf8a7b in main (argc=13, argv=0x7ffcdb441928) at main.c:2696 cfg_stream = 0x5569cd7c7260 c = -1 r = 0 tmp = 0x7ffcdb441f39 "" tmp_len = -616294480 port = 32764 proto = 3 options = 0x5569cc074190 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3114863458 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7f006af01787 p = 0x9c0 <error: Cannot access memory at address 0x9c0> st = {st_dev = 23, st_ino = 456, st_nlink = 2, st_mode = 16877, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1564229247, tv_nsec = 809557407}, st_mtim = {tv_sec = 1564229229, tv_nsec = 217250917}, st_ctim = {tv_sec = 1564229279, tv_nsec = 666090748}, __glibc_reserved = {0, 0, 0}} _func_ = "main" (gdb) -- Regards M. Salman

3 4

alias="server" from db?
by David Villasmil 05 Aug '19

05 Aug '19

Hello guys, I can't find a module that does this, i want to load my hostname aliases from db, instead of alias="host1:5060" alias="host2:5060" alias="host3:5060" I assume it is possible, but how? first time i need to implement this Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 6

Re: [SR-Users] Kamailio v5.2 kamctl alias add problem
by Igor Potjevlesch 04 Aug '19

04 Aug '19

Hi kamailio Users, I meet a problem with the " kamctl alias add" command in the 5.2 version, I saw that the module used to add an alias had changed, and now the command result with an error : [kamailio]# kamctl alias add XXXX@XXXX sip:XXXXX@XXXX { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters or wrong format" }, "id": XXXXX } So in the version 5.2, the table aliasdb replace the table alias ? We are using the table alias for the incomming call, and we implemented scripts to add with the command kamctl alias add the information needed in the tavble alias to route the call. But since the version 5.2 we need to add an argument "path" in this command, and we don't know how to put it at "NULL" or how we can use this argument ? Can you help us with this? Bests Regards Igor Le ven. 5 juil. 2019 à 16:57, Igor Potjevlesch <igor.potjevlesch(a)gmail.com> a écrit : > Hi Daniel-Constantin, > > So in the version 5.2, the table aliasdb replace the table alias ? > We are using the table alias for the incomming call, and we implemented > scripts to add with the command kamctl alias add the information needed in > the tavble alias to route the call. > But since the version 5.2 we need to add an argument "path" in this > command, and we don't know how to put it at "NULL" or how we can use this > argument ? > > Can you help us with this? > > Bests Regards > >>

3 3

Reverse Proxy Asterisk - SIP Provider
by Mihai Cezar 04 Aug '19

04 Aug '19

Hi all, I've tried to create a reverse proxy to forward incoming request that came from SIP provider to Asterisk PBX and forward the requests from asterisk to kamailio then sip provider. What i get is that I see the invite, but is like no ACK. Thanks in advance. M kamailio.cfg: #!KAMAILIO # ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=3 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=1 server_id = 10 xavp_via_params = "via" disable_tcp=yes auto_aliases=no listen=udp:0.0.0.0:5060 ####### Modules Section ######## loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE|REFER")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); sl_send_reply("100","Trying"); if ($si == "172.16.16.1") { sl_send_reply("183","Incoming session from Avoxi"); rewritehost("10.1.1.10"); #exit; } else if ($si == "10.1.1.10"){ # receiving response from client sl_send_reply("183","Outgoing session to Avoxi"); #rewritehost("172.16.16.1"); drop; exit; } else { sl_send_reply("500","No configured IP!"); drop; exit; } } if ($rU==$null) { sl_send_reply("484","Address Incomplete"); exit; } # received from main server - send to client and add via tokens for anycast handling via_add_srvid("1"); $xavp(via=>node) = "10.1.1.4"; via_add_xavp_params("1"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if ($si == "10.1.1.4") { xlog("L_WARN", "$ci|end|dropping message"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("400","Loop detected"); exit; } # TM manage for outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # TM manage for incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } # TM manage for failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } asterisk - sip.conf [opensips-trunk](sip-provider) fromdomain=10.1.1.10 host=10.1.1.4 context=from-trunk type=friend insecure=invite,port trunk=yes

4 9

Kamailio OPTIONS Round-Trip
by Eugeniu Istrati 04 Aug '19

04 Aug '19

I'm interested in this feature too, would it be available in the 5.3 release? Thank you. -- Eugeniu Istrati

2 1

location table
by Maryam Baghdadi 03 Aug '19

03 Aug '19

Hello How can I save subscribers information in location table? why after calling save("location") in configuration file , still location table is empty?

2 1

(no subject)
by jay binks 02 Aug '19

02 Aug '19

Hey all, looking for a little help. I've been tracking an issue, where a number of SIP Messages ( typically INVITE ) are sent, but Kamailio only see one of them ( the last one). I have verified that the messages are definitely received on the host, as verified with TCPDUMP, but the INVITES never hit the request route block. ( the very first thing is to log the message ) What i've managed to find is, that if I increase worker children, the problem goes away. So I've got a problem, I think I've found the solution, but what I'm struggling with is how to monitor better, so I can be alerted to this in the future. I started by looking at "kamctl stats | grep drop" "core:drop_replies = 14", "core:drop_requests = 10577" however drop_requests seems to include explicit drops in my config. which I do for many reasons, but mainly bad UA or blocked IP's, so this dosnt seem to be what im after. I then went digging in the code ( often the best way to find things ). in receive.c receive_msg() we find some conditions, where we drop the packet ( before calling request route ). right now im looking around line 251 at LM_DBG("dropping the received message\n"); goto error00; Following error00 I see it calls STATS_RX_DROPS which increments stats->received_drops. This is where it gets fun, I cant find where received_drops is ever returned in the stats output. I can see where -SIGUSR1 will trigger this to be dumped, if Kamailio is compiled with that option ( dosnt seem the debian packages are ). however, what im seeking clarification on is, does this get reported in "kamctl stats" Am I just going crazy or is something not quite right here? TLDNR; how do I monitor for Kamailio not having enough worker threads to process incoming messages. FYI I'm seeing about 50Mbps of incoming SIP on this box, and I do perform a decent amount of work on some message types. -- Sincerely Jay

5 7

Deleting extra location attribute(location_attrs table) when contact is expired
by Pintu Lohar 31 Jul '19

31 Jul '19

Hi All, Regarding modparam("tm|usrloc", "xavp_contact", "ulattrs") #ref : https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.xa… . There is some requirement where we need to save some extra location attributes to location_attrs <https://kamailio.org/docs/db-tables/kamailio-db-4.3.x.html#gen-db-location-…> tables $xavp(ulattrs=>deviceIdentify)= "Some str values" and calling save(“location”) , It works well and saved location attribute table but But when contact expired callback is received, udomain_contact_expired_db() does not it delete the extra location attribute from database but deleting the record from location table only ? If not, is there any recommended approach to delete the extra location attributes from the database? or am I missing something to set? Alternate I was thinking could be running timer from Kamailio config and check last_modified time and delete those(based on our UAC expiry setting), is it the right approach. Kamailio Version: 5.2.0. Thanks for your valuable suggestion in advance. Regards Pintu Lohar

2 10

Register without user create
by José Antonio Gutiérrez Delgado 29 Jul '19

29 Jul '19

Hello, I would like to know if it is possible to enable registration for users that have not been created with kamctl. For example, if I can register a user with a random name without a password, or obtain from another database other than kamailio; I would also like it to be allowed at the same time as there are users with password registered with kamctl. Thanks you, José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com> *De conformidad a lo establecido en el REGLAMENTO (UE) 2016/679 DEL PARLAMENTO EUROPEO Y DEL CONSEJO de 27 de abril de 2016 relativo a la protección de las personas físicas en lo que respecta al tratamiento de datos personales y a la libre circulación de estos datos y por el que se deroga la Directiva 95/46/CE (Reglamento general de protección de datos), le informamos que todos los datos pasaran a formar parte de un fichero automatizado cuyo responsable es AUGMENTED REALITY SOFTWARE S.L siendo la única finalidad de dicho fichero la gestión de carácter comercial y el posible envío de comunicaciones comerciales sobre nuestros productos. Si lo desea puede ejercer sus derechos de acceso, rectificación, cancelación y oposición de sus datos en C/ Duero 12, Parque Científico, ARSOFT, 37185 o enviando un mensaje electrónico a info(a)arsoft-company.com <info(a)arsoft-company.com> indicando en el asunto el derecho que desea ejercitar.Este mensaje y sus archivos adjuntos van dirigidos exclusivamente a su destinatario, pudiendo contener información confidencial sometida a secreto profesional. No está permitida su reproducción o distribución sin nuestra autorización expresa. Si usted no es el destinatario final por favor elimínelo e infórmenos por esta vía.*

3 4

Dialog missing when BYE is received from trunk
by Andrew White 28 Jul '19

28 Jul '19

Hi all, I’ve got a call flow where PBX makes a call via Kamailio, which uses the UAC modules to make the call to the trunk. This works great, however I’ve noticed recently that when the BYE comes from the trunk, Kamailio doesn’t recognise the dialog and throws back a 404. Due to this, the PBX never knows the session has ended, and keeps its session open. Here’s the flow from Wireshark: https://i.imgur.com/BxPRrXP.png <https://i.imgur.com/BxPRrXP.png> As you can see, the BYE is received from the trunk, and Kamailio throws a 404. I checked with an rpc dlg.list and at this point the dialog had gone. 30 seconds later, I threw a BYE from the PBX, and because the dialog no longer existed, it got a 481 back. The INVITE to the trunk has the following Record-Route header: Record-Route: <sip:kamailio.public-ip;lr;ftag=mjKtcKaBteQ4Q;vst=AAAAAAYFBQUHAwYHBgRzMwIRGRMcAgJBWBEVCwAbBhtADxBjdGVkLmNvbS5hdQ--;did=624.b2e1> When the first BYE comes in, it still has same did in the Route header. Below I’ve put the output from the dialog list rpc call (before the first BYE) and the relevant log lines. { h_entry: 1062 h_id: 7723 ref: 2 call-id: a9d32620-2b68-1238-048a-0cc47a18705a from_uri: sip:61400123123@pbx.local:5060 to_uri: sip:61400123123@10.0.0.2 state: 4 start_ts: 1564269942 init_ts: 1564269929 end_ts: 0 timeout: 1564313142 lifetime: 43200 dflags: 643 sflags: 0 iflags: 16 caller: { tag: mjKtcKaBteQ4Q contact: sip:pbx.local cseq: 7581172 route_set: socket: udp:10.0.0.2:5060 } callee: { tag: SD5ptp899-1177933201-1564269931305 contact: sip:SDd3mt8-nn9jnirnmqcrvfmvkpm0hmhvgggjgv2io788of70s84u9@trunk.remote:5060;transport=udp cseq: 0 route_set: socket: udp:10.0.0.2:5060 } profiles: { } variables: { { cseq_diff: 1 } } } Jul 28 09:27:30 ip-10-0-0-2 /usr/local/sbin/kamailio[11061]: NOTICE: {2 450917085 BYE a9d32620-2b68-1238-048a-0cc47a18705a} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1564270050;method=BYE;from_tag=SD5ptp899-1177933201-1564269931305;to_tag=mjKtcKaBteQ4Q;call_id=a9d32620-2b68-1238-048a-0cc47a18705a;code=404;reason=Not here;src_user=0400123123;src_domain=trunk.remote;src_ip=trunk.remote;dst_ouser=0400123123;dst_user=0400123123;dst_domain=kamailio.public-ip I’m assuming the 404 is because the Contact header I sent from Kamailio -> Trunk is rewritten, that Contact is Kamailio’s public contact IP. However the dialog has the caller contact within it referencing back to the local PBX, so I would think when Kamailio gets this dialog ID back with a BYE it would relay it back to the original contact, wouldn’t it? Thanks! Andrew

2 3

IMS_ISC technical specification
by Ali Shirvani 26 Jul '19

26 Jul '19

Hi all, I read some portion of imc_isc_mod.c, specially isc_from_as function. In that function there is some function call that searching for some marks in the SIP message. Also I read section 5.7.3 from TS 24.229 but I didn't find the marking mechanism that used in isc_from_as function. Would you please point me to the technical specification that implemented in IMS_ISC module? Regards, Ali

3 4

Concatenating strings in $hdr
by David Cunningham 26 Jul '19

26 Jul '19

Hello, I'm having an issue where this doesn't work: #!trydef PRODUCT "Foo" if( $hdr("X-" + PRODUCT + "-Transport") != "" ) { add_uri_param( $hdr("X-" + PRODUCT + "-Transport") ); } It seems Kamailio doesn't like the concatenation within the $hdr(). Concatenation does work in other places though: append_hf( "X-" + PRODUCT + "-Transport: tcp\r\n" ); Can anyone advise how to make the concatenation within $hdr() work? Thanking you in advance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

IM Chat facility in Kamailio
by Gaurav Bmotra 25 Jul '19

25 Jul '19

hi i m using kamailio 5.1.2 , on Ubuntu 18. LTS existing setup is working fine with audio and video call ,,, ut now i wan to add chat feature in it ... so plz help me how can i implement chat IM in to it thanks -- *Regards:* Gaurav Kumar

2 1

Malformed Route Header
by Duarte Rocha 25 Jul '19

25 Jul '19

Greetings, When a in-dialog request has Route headers, Kamailio will remove the Route Header with it's address when routing the packet. This works perfectly for me in every situation. However, in one of the calls I found a weird behaviour in the removal. So, this is the scenario : Kamailio has two IP's 1.1.1.209 and 1.1.1.205. This is BYE request received in the on 1.1.1.209 The header Route looks like this when it arrives at 1.1.1.209 : Route : <sip:1.1.1.209;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2>,<sip:1.1.1.205;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2> It reaches 1.1.1.205 like this : Route : As you can see, the header is empty and it should contain the 1.1.1.205 URI. Is the header Router malformatted and creating this error ? Best Regards, Duarte Rocha

2 1

Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports -RESOLVED
by Woscek, Martin W. 24 Jul '19

24 Jul '19

Hi Ali, Looking through the source code, its not that the port for I-CSCF is hard coded to 5060, its just that it defaults to 5060 if the following is set in the Kamailio.cfg file: dns_try_naptr=no which we had left it disabled. Well I assumed I had all the DNS worked out, I had NAPTR records set for the P,I and S-CSCF. Looking though the pcscf.cfg there is a setting for NETWORKNAME which is set to ims522.ims.org, and is a separate FQDN from the P,I,S-cscf FQDN’s. I always kept the NAPTR rec for ims522 as the same port of the P-CSCF (pcscf522) in DNS. Two things I needed to fix: 1. Enable the DNS NAPTR lookups: dns_try_naptr=yes 2. Modify the NAPR for ims522 to be that of 4060 Thanks to you and Amar for the pointing to DNS. _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

kamailio 4.4.4 fills IPv6 contact address when sending packet over IPv4
by Pali Rohár 24 Jul '19

24 Jul '19

Hi! I detected strange problem that sip.linphone.org refuse to accept presence information re-transmitted by kamailio 4.4.4 server. I debug this problem with tcpdump and I found out that problem is in kamailio which fills IPv6 address into UDP datagram and that datagram is sent via IPv4 socket to IPv4 address, to sip.linphone.org server. And sip.linphone.org server does not have IPv6 connectivity, so correctly return over IPv4 to sender just "400 Bad Contact Header" error. On my server is running kamailio 4.4.4 from Debian Stretch and I can 100% reproduce this problem against public sip.linphone.org server. My server has both IPv4 and IPv6 connectivity and kamailio is listening for both IPv4 and IPv6 connections. So why is kamailio sending IPv6 address over IPv4 and therefore makes it impossible to communicate with non-IPv6 enabled servers? Looks like a problem with choosing default/correct socket for Contact header. And how to fix this problem? Can you help me? I would like to have working interconnection with linphone servers. Just to note I'm seeing this problem only for presence information packets. Other requests, like INVITE or MESSAGE seems to work. Below is relevant tcpdump output. Some parts were replaced by {VAR}. PS: I'm not subscribed to list, so please CC my address when sending reply. Thank you! 17:22:58.121719 IP (tos 0x10, ttl 64, id 21629, offset 0, flags [none], proto UDP (17), length 1266) {MY_IPV4_ADDRESS}.5060 > 91.121.209.194.5060: [bad udp cksum 0xa099 -> 0x9825!] SIP, length: 1238 NOTIFY sip:{REMOTE_NAME}@{REMOTE_USER_IPV4_ADDRESS}:5060;registering_acc=sip_linphone_org SIP/2.0 Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 CSeq: 2 NOTIFY Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 Route: <sip:91.121.209.194:5060;lr> Content-Length: 597 User-Agent: kamailio (4.4.4 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:{MY_IPV6_ADDRESS}:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="Pali <sip:{MY_SIP_URI}>"> <tuple id="sg89ae"> <status><basic>open</basic></status> <contact priority="0.8">Pali <sip:{MY_SIP_URI}></contact> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="TA0C538B2"> <status> <basic>closed</basic> </status> <contact priority="1">sip:{MY_SIP_URI}</contact> <timestamp>2019-04-19T17:20:36+02:00</timestamp> </tuple> </presence> 17:22:58.151188 IP (tos 0x0, ttl 52, id 22949, offset 0, flags [none], proto UDP (17), length 373) 91.121.209.194.5060 > {MY_IPV4_ADDRESS}.5060: [udp sum ok] SIP, length: 345 SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0;rport=5060 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 CSeq: 2 NOTIFY Content-Length: 0 -- Pali Rohár pali.rohar(a)gmail.com

4 10

How to hook to Register And Unregister event in Kamailio
by Anuran Barman 24 Jul '19

24 Jul '19

Hi, I am integrating Kamailio into my application. I want to hook to the successful REGISTER and unregister event into Kamailio into my application. For now, I am able to hook into INVITE event and can hit my server to send a email to the callee user that user X is calling you. The way I am doing is by this if (is_method("INVITE")) { xlog("LOG_LOCAL3","L_INFO","invite came ($fU) ($tU)"); $var(res) = http_connect("sipnodejsserver", "/","text/plain","src_user:$fU,dst_user:$tU" ,"$avp(route)"); xlog("LOG_LOCAL3","L_INFO","request sent $avp(route) $var(res)"); setflag(FLT_ACC); # do accounting } I need to show into my application that their SIP phone is online or not. How can I hook into this?

4 24

Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

DNS is fine. I flip the NAPTR records when I flip back and forth the 4060/5060 btwn the P and I. Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin

3 2

Debugging kamailio module
by Ali Shirvani 23 Jul '19

23 Jul '19

Hi all, I want to trace function calls in a kamailio module, e.g. ims_isc module. I installed kamailio-dbg package and started the kamailio with gdb: `gdb kamailio` I also set break point on target function, but it seems gdb doesn't stop on that break point. How should I start gdb and set break points? it seems gdb couldn't handle kamailio child processes that forked in start. Regards, Ali

2 2

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23 Jul '19

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2 1

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23 Jul '19

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3 2

Two GWs with the same IP
by Duarte Rocha 22 Jul '19

22 Jul '19

Greetings, I'm building some testing procedures for my Kamailio proxy. In order to do that i have a SipTester (Acting as Calling Number) from where i want to originate calls to Kamailio. After Kamailio processes the call, i want it to route it to SipTester again so it can act as Called Number. This should work with only one SipTester in one machine, so i created two SipTester GW's on Kamailio, belonging to different users and GW groups. Also, the two GWs use different Kamailio addresses to communicate with the proxy. However, in this combination, when routing i get this error " ERROR: dialog [dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/". If i change the destination or the origin of the INVITE the calls work without a problem. Does this happen because both GW's have the same address? Best Regards

2 1

Reading presence state
by Gertjan Wolzak 19 Jul '19

19 Jul '19

Hello Kamailions, I have been able to get the presence configured, subscribers see the status of their subscriptions, this is working fine, for internal, incoming external and external calls. This was solved by enabling use_pubruri_avps in the pua_dialoginfo module. Now I want to be able to route calls based on the presence state. Using the presentity table could be used by doing a query based on the userpart of the ruri, which works fine when the call is still active, but it can take a while for the record to be removed from the presentity table, as that is dependend on the expires value. So when a query is made after a call is finished, a record would still be present in the presentity table till clean_period timer has hit again and the expires value is expired. So I should get the status from memory, but I have not been able to find the information on how to do that. Can someone enlighten me on where to find that information? Rgds, Gertjan

2 4

I need to hold Leg A of a call for a random amount of time
by Gaurav Bmotra 19 Jul '19

19 Jul '19

hi i m using kamailio where i m tring to hold the call bitween switch A and Switch B PSTN -> Switch A -> SWITCHB -> SwitchC -> DESTINATION When PSTN hangs up, I need the call to stay up between SwitchA and B for an arbitrary number of seconds or minute plz help thank you -- *Regards:* Gaurav Kumar

2 1

Kamailio 5.2.3 failed to start due to incompatible openssl version
by Andrew Chen 18 Jul '19

18 Jul '19

Hey support, Today I ran into this error on my latest 5.2.3 Kamailio box: Jul 18 15:05:28 sjomaintpsg51 /usr/sbin/kamailio[5525]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed “OpenSSL 1.1.1 11 Sep 2018” (0x1010100f), compiled “OpenSSL 1.1.0g 2 Nov 2017" (0x1010007f).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) There's not support for openssl1.1.1 ? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 3

kamailio + asterisk integration RTP flow question
by Satish Patel 18 Jul '19

18 Jul '19

I am configuring kamailio as registar and asterisk for media services. in kamailio + asterisk integration how two SIP clients RTP traffic will flow, are they go via Asterisk or directly from kamailio ?

3 2

SIP-DTLS support
by Martin Meadows 17 Jul '19

17 Jul '19

Hello, Has there been any movement on the support of SIP-DTLS (datagram-TLS) within Kamailio since: * sr-users:2012-October:075185.html <https://lists.kamailio.org/pipermail/sr-users/2012-October/075185.html> I'm interested to see if Kamailio can proxy 5061/udp to a FreeSWITCH 5060/udp. In general, I have been unable to find any material on implementations of: * draft-jennings-sip-dtls-05.html <https://tools.ietf.org/id/draft-jennings-sip-dtls-05.html> Cheers - Martin

4 6

opt_reply(): called for non-OPTIONS request
by Alexandru Covalschi 17 Jul '19

17 Jul '19

Hello list, if KSR.is_OPTIONS() and KSR.is_myself_ruri(): KSR.x.modf("options_reply"); exit(); with a valid URI (no username) gives the subj output (opt_reply(): called for non-OPTIONS request) How could that be? Python KEMI, 5.2.3, docker. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

2 5

Kamailio IMS to IP PBX - how?
by David Dean 16 Jul '19

16 Jul '19

I'm trying to connect a VoLTE smartphone to an IP PBX using Kamailio IMS. I've managed to get the smartphone to connect to Kamailio IMS and make calls with our devices registered with Kamailio. The problem is that I need the smartphones to register with the IP PBX which is a VigorBX 2000n. This is so they can use an internal extension number and make calls to other local extensions with hard wired desk phones. Should I be using Kamailio IMS to "proxy" the smartphone SIP connection to the VigorBX PBX? Or would it be better to connect the smartphone directly to the VigorBX using SIP? If so, how? I've read absolutely every how-to document I could lay my hands on but none of them describe integrating IMS with a PBX. Thanks for any help / advice you can offer. Dave

1 0

KEMI beginner question on alpine Linux
by Karsten Horsmann 16 Jul '19

16 Jul '19

Hi List, I play around with alpine Linux 3.10.x and Kamailio 5.2.x and I search for app_python3 and KEMI support. But I don't found it. AFAIK if I want to use KEMI with Python3 I need of course Kamailio and app_python3 (maybe something else?). Is the package maintainer on this list? Cheers Karsten

4 4

How to get total number of dialogs in profile with value
by Nathan Angelacos 15 Jul '19

15 Jul '19

In kamailio < 5.1.x it used to be possible to get total number of items in a dialog_with_profile by skipping the "value" paramter, such as: modparam ( "dialog", "profiles_with_value", "outbound" ) . . . get_profile_size ( "outbound", "$var(myvar)" ); However, 5.1.x and 5.2.x report: ERROR: dialog [dialog.c:912]: w_get_profile_size_helper(): invalid value parameter The documentation ( https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.f.get_pr… ) Indicates the "value" paramter is still optional. I'm sure I must be missing something obvious... Any ideas? Thanks!

2 3

UDP fragmentation issues
by Gerry | x164 15 Jul '19

15 Jul '19

Hi, I have setup Kamailio on Centos 7 as a proxy in front of Asterisk in the same datacenter, using the dispatcher module. I have noted that large INVITEs sent by Kamailio do not arrive at the Asterisk box. It seems the network is dropping fragmented UDP traffic. This does not happen however when I contact the Asterisk box directly from outside the datacenter. Then large INVITES are arriving are Asterisk. So the problem is either in Kamailio or in the datacenter. I have analysed the problem with ngrep and tcpdump/wireshark. In normal mode (udp4_raw=no setting) Kamailio is producing UDP packets above 1500 bytes and leaves all fragmentation to the operating system. Enabling raw udp in Kamailio and lowering the MTU in the server didnt help. I even disabled offloading of UDP fragmentation to the NIC to get to the bottom of this. However nothing helps. Only forcing TCP in the dispatcher works when having large INVITES going to Asterisk. In any case, it appears that there is no guarantee that fragmented UDP arrives at all. As far as I am aware, the recommendation is to use TCP if messages are bigger than 1400 bytes. If so, how can I force Kamailio to send messages over 1400 bytes only in TCP mode? Or is there some other way how to deal with this? Thanks for any hints. Gerry

3 3

Kamailio Asterisk integration with docker DNS Problem
by Amir 15 Jul '19

15 Jul '19

Hi I have created a kamailio container with docker and two asterisk container And this is my dispatcher list: *1 sip:asterisk:50601 sip:asterisk2:5060* and this is my both asterisk SIP.conf *[Kamailio]host=kamailioport=5060insecure=invitetype=friendcontext=from-internal* The problem is that I use User-defined networks which has its own DNS when a container starts a dns record would be set If I start a kamailio first then start two asterisks the asterisk works fine because the name "kamailio" has been set in docker dns but I should restart the kamailio because at startup It could not resolve astersik dns records after the everything works fine but if the any of the asterisk instances crash/stop kamailio can not detect that because a dns record would not be available: *17(22) ERROR: <core> [core/resolve.c:1684]: sip_hostport2su(): could not resolve hostname: "asterisk2"17(22) ERROR: tm [ut.h:309]: uri2dst2(): failed to resolve "asterisk2"17(22) ERROR: tm [uac.c:452]: t_uac_prepare(): no socket found17(22) ERROR: dispatcher [dispatch.c:3110]: ds_ping_set(): unable to ping [sip:asterisk2:5060]* It is kind of loop , I think maybe I can fix asterisk problem with asterisk realtime and set the sip.conf in database. anyone has any suggestion for this problem?

8 14

Wrong $rm value
by Alexandru Covalschi 14 Jul '19

14 Jul '19

Hello list, I use kamailio 5.2.3 from repo's on stretch in docker. I use python KEMI + async and encountered weird behaviour: after t_continue $rm shows OPTIONS instead of REGISTER/INVITE. I wonder if that's correct behaviour, because for me it looks like a bug. Repeating is easy - just check $rm after t_continue. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

1 2

Check if an INVITE comes from uac_reg endpoint
by Diogo Hartmann 14 Jul '19

14 Jul '19

Hello Kamailio experts, We were already able to register correctly at a SIP Provider using uacreg table. Now, we would like to validate inbound calls coming from this registered provider, sending them to a specific dispatcher group. I tried reading the uac module docs, found *uac_reg_lookup* function, but i'm not sure if that's we need. Could someone please help me to understand if UAC has a function to do that, or if we need to make some? Thanks!!

3 4

kamailio as ICSCF connection with FHoss
by Maryam Baghdadi 14 Jul '19

14 Jul '19

hello, I use Kamailio as my Icscf in IMS network . after receiving Invite Request from SCSCF to ICSCF‌, a LIR request sent from ICSCF to FHoss. User_Unknown response received in all states. I checked every thing carefully , users are registered and defined in HSS and every configuration done . What's wrong? is it possible that icscf has problem? or it can be FHoss bug?

4 14

call stuck and disconnect in kamailio
by Gaurav Bmotra 13 Jul '19

13 Jul '19

hi i m using kamailio 5.1.2 with ubuntu 18.4 LTS , and rtpproxy (Basic version: 20040107 ) we are using it for video calling everything is working fine but call get stuck after about 2 min and disconnect plz help -- *Regards:* Gaurav Kumar

2 3

Long distance rtpengine communication
by Koray Vatansever 12 Jul '19

12 Jul '19

Hi all, We have kamailio servers which communicate with the geographically distributed rtpengine servers. Sometimes udp packets are lost or delayed between kamailio and rtpengine. And this causes call drops... Since rtpengine communication is synchronized and tcp worker processes are blocked, I couldn't set higher timeout values and more retry count. Are there any suggestions to overcome this problem? Thanks, Koray

2 1

Variables for P-Called-Party-ID Header
by Benjamin Tietz 12 Jul '19

12 Jul '19

Hi, I have some INVITES where the called number is in the P-Called-Party-ID header: P-Called-Party-ID: <sip:+123456789@sip-trunk.example.com;user=phone> Is there any function or variable where the user is stored or made available? In this case +123456789 Thanks, Benjamin

2 3

SIP URI Evaluation and Forwarding
by Ronald Hartmann 12 Jul '19

12 Jul '19

Good Day List, I am very new to Kamailio and trying to understand at 10,000 feet how it may be able to be used. I was wondering if the following Scenario would be possible and if so what modules would accomplish this aspect. (Any URLs for further reading would be appreciated) Scenario: Given 1) SoftPhones will register as follows sip uri, user(a)subdomain.myvoipserver.com 2) The following users are authorized to connect (this would be stored in a config file etc): 1001,1003,1007 are valid users for apples.myvoipserver.com 1001,2213,5817 are valid users for oranges.myvoipserver.com Is it possible for Kamailio to perform the following evaluation and then take the appropriate steps: If [domain is apples.myvoipserver.com AND user portion is 1001] then Forward/Proxy the SIP/RTP packets to Asterisk server responsible for registration and audio for the domain apples.myvoipserver.com Else if [domain is oranges.myvoipserver.com AND user portion is 2213] then Forward/Proxy the SIP/RTP packets to Asterisk server responsible for registration and audio for the domain oranges.myvoipserver.com Else Silently Drop packets etc. given the above evaluation rules, a user who tries to register as 1001(a)apples.myvoipserver.com would be granted access and their SIP Packets and subsequent RTP would be forwarded to the asterisk server at apples.myvoipserver.com, where said server would take care of the Registration, and any subsequent calls, subscriptions, Options etc. 1004@ apples.myvoipserver.com Would not match the authorized list for apples.myvoipserver.com (1001,1003,1007), therefore this registration attempt would be silently dropped 2213(a)apples.myvoipserver.com would be granted access and their SIP Packets and subsequent RTP would be forwarded to the asterisk server at oranges.myvoipserver.com, where said server would take care of the Registration, and any subsequent calls, subscriptions, Options etc. Thanks for any information to assist me in understanding how this would work. ~ron

3 2

Re: [SR-Users] kamailio warning
by Henning Westerholt 10 Jul '19

10 Jul '19

Hello Ali, (please keep the sr-users list in CC, that other people can also participate) Remove the “-D” from your Kamailio command line – then the warning will disappear. This is also necessary because you want to have Kamailio listen on several addresses. If you want to test it more in a production like setting, you should also remove the “-E” from the command line. You will find the logging in your system log then. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services From: ali mahfouz <arm037(a)usal.edu.lb> Sent: Wednesday, July 10, 2019 11:54 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] kamailio warning Dear Mr.Henning, I am working on high availability on two Kamailio servers connected to the database using keepalived and when I am testing Kamailio I got this issue and I have no idea about fixing it i am trying to see by this command kamailio -E -D -f /etc/kamailio/kamailio.cfg 0(16101) INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 10.10.10.20:5060<http://10.10.10.20:5060> udp: 10.10.10.3:5060<http://10.10.10.3:5060> tcp: 10.10.10.20:5060<http://10.10.10.20:5060> tcp: 10.10.10.3:5060<http://10.10.10.3:5060> Aliases: tcp: 10.10.10.3:5060<http://10.10.10.3:5060> tcp: 10.10.10.20:5060<http://10.10.10.20:5060> udp: 10.10.10.3:5060<http://10.10.10.3:5060> udp: 10.10.10.20:5060<http://10.10.10.20:5060> *: sip.domain.local:* WARNING: no fork mode and more than one listen address found (will use only the first one) 0(16101) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(16101) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(16101) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(16101) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(16101) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(16101) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(16101) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(16101) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 6(16131) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 5(16130) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/16130 ^C 6(16131) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(16130) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(16126) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16131 exited normally, status= 3(16128) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16126 exited normally, status= 4(16129) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16128 exited normally, status=0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16129 exited normally, status=0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16130 exited normally, status=0 0(16101) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(16101) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(16101) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(16127) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(16101) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Please help thank you for giving time from your time On Wed, Jul 10, 2019 at 12:40 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Ali, the first warning is related about the starting mode of Kamailio. You should not start it with “no forking mode” if you want to get rid of it. Have a look to your command line parameters: “-D” is the one to change. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of ali mahfouz Sent: Wednesday, July 10, 2019 8:46 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>; sr-users-request(a)lists.kamailio.org<mailto:sr-users-request@lists.kamailio.org>; sr-users-owner(a)lists.kamailio.org<mailto:sr-users-owner@lists.kamailio.org> Subject: [SR-Users] kamailio warning WARNING: no fork mode and more than one listen address found (will use only the first one) 0(8777) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(8777) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(8777) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(8777) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(8777) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(8777) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(8777) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(8777) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 5(8782) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/8782 6(8783) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) ^C 6(8783) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(8782) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 4(8781) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 3(8780) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(8778) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8782 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8778 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8781 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8783 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8780 exited normally, status=0 0(8777) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(8777) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(8777) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(8779) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(8777) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

3 3

(no subject)
by Ilie Soltanici 10 Jul '19

10 Jul '19

Hello, Could someone help me to set a variable as a string in Kamailio? At the moment i'm using the following statement: *$var(account)=$(au{s.substr,0,8}); * *xlogl("L_INFO", "[1234] Account: $var(* *account* *)\n");* which is working fine, but i need to change it into: *$var(account)=$(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-});* By doing this I'm getting the error: *[core/rvalue.c:1007]: rval_get_int(): automatic string to int conversion for "AT" failed[core/rvalue.c:1912]: rval_expr_eval_int(): rval expression conversion to int failed (729,30-729,59)* I tried also: *$var(account)="" + "$(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-})";* But, then I'm getting: *Account = $(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-})* instead of the value. What is the correct way to set it? Thank you.

2 3

kamailio warning
by ali mahfouz 10 Jul '19

10 Jul '19

WARNING: no fork mode and more than one listen address found (will use only the first one) 0(8777) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(8777) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(8777) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(8777) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(8777) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(8777) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(8777) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(8777) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 5(8782) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/8782 6(8783) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) ^C 6(8783) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(8782) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 4(8781) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 3(8780) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(8778) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8782 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8778 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8781 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8783 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8780 exited normally, status=0 0(8777) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(8777) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(8777) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(8779) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(8777) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

2 1

mi_datagram module in ipv6
by Albert Petit 09 Jul '19

09 Jul '19

hi i would need to replicate behaviour of refresh watchers command at mi_datagram module in an ipv6 system. but in such system i am not able to bind this module in an iov6 address desoite its doc claims is possible https://kamailio.org/docs/modules/4.4.x/modules/presence.html#idp21723212 hence not sure i will be able to use mi_datagram using ipv6 address unless some user tells me he/shr could do succesfully as module is quite old question is: h¥w could i force kamailio to notify watchers of a subscriber in some other way from an external application? any hints? thanks

2 2

kamctl ul command seems NOK when using an IPv6 Address
by Albert Petit 09 Jul '19

09 Jul '19

Hi, I am trying to create an identity with permanent location in my system using kamctl ul add command. Usually i have no issue doing that in IPv4 ,example kamctl ul add mcpttcontrolling sip:mcpttcontrolling@127.0.0.1:5070 it is OK However when i do that using an IPv6 address i get following error: kamctl ul add mcpttcontrolling sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 error: invalid SIP URI: sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 Let me know what you would need to investigate or what i am doing wrong for adding a location with IPv6 Address using kamctl tool Thanks Albert

2 2

Jansson_get issues
by Igor Olhovskiy 09 Jul '19

09 Jul '19

Hi! I'm trying to get jannson_get working, but seems missing something. Example is quite simple, actually taken from wiki $var(result) = $null; jansson_get("pi", '{"mystr":"my input string", "count":9000, "mybool":true, "pi":3.14159}', $var(result)); When running this part of code getting ERROR: <core> [core/action.c:1139]: do_action(): runtime fixup failed for jansson_get param 3 ERROR: <core> [core/action.c:1518]: do_action(): run action error at: /etc/kamailio/kamailio.cfg:1030 Also tried with jannson_get(".pi", ...); jannson_get('/"pi"', ...); and a bit others. What am I missing? kamailio 5.2.3 Thanks!

2 3

KW 2019 presentation files
by Denys Pozniak 09 Jul '19

09 Jul '19

Hello! Where can I find fresh presentation files from the KW 2019? PS. Nothing here: https://www.kamailio.org/events/ -- BR, Denys Pozniak

3 3

where can i find siremis user guide?
by quxiao＠unioncast.cn 09 Jul '19

09 Jul '19

hi�� i have installed siremis successfully for the first time, but there is no guide to use siremis, where can i find it, thanks Best Regards & Thanks�� ֻ��13401129519 ��䣺quxiao(a)unioncast.cn ְλ��߼�C++��ʦ ��ţ��з�� Vҕͨ�_��־W�j�Ƽ��޹�˾ Unioncast Digital Network Technology Co., Ltd. Beijing ��˾��ҳ��http://www.unioncast.cn/ һ��ƶ��Ӧ�÷��ṩ�� ַ��к��йش��ϴ��3�ź��Ƽ��9��910�� ʱࣺ100081

1 0

Kamailio new in devel for upcoming 5.3.0
by Henning Westerholt 08 Jul '19

08 Jul '19

Hello, I have updated the "new in devel" features document in our wiki. Pretty amazing stuff already in for the upcoming 5.3.0: * 5 new modules * Changes in 32 existing modules * Many changes in core, API refactoring and tools * About 1000 commits from 62 authors (development still ongoing!) Full list at: https://www.kamailio.org/wiki/features/new-in-devel A summary can be found in my today talk at COMMCON, slides 10,11: https://skalatan.de/de/archiv/presentations/commcon-2019-presentation.pdf Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

How to statelessly send a reply to custom destination?
by George Diamantopoulos 05 Jul '19

05 Jul '19

Hello all, I've been trying to handle OPTIONS statelessly, but proper handling fails for UACs behind NAT without ALG. I thought I'd overcome this by setting the $du pseudovariable, but apparently I'm missing something. The following does not work: if ($Rp == "6050") { force_send_socket(udp:10.10.10.10:6050); } $du = "sip:" + $si + ":" + $sp; sl_send_reply("200","OK"); exit; I'm now realising that I put this in request_route() so force_send_socket() and $du have no use whatsoever, since the request is not forwarded anywhere. Is this the reason? So the question is, how can I do the following things for a stateless reply: a) choose a sending socket for the reply and b) change the destination for the reply to something other than the URI in the first Via header. Thanks! BR, George

3 3

Append a new branch while the other one is in ringing state
by Koray Vatansever 04 Jul '19

04 Jul '19

Hi all, I want to fork a new branch, while the first one is ringing. Forking will take place in a rtimer process. So I will save the invite transaction in a htable, and using this transaction info I need to append a branch. I saw TSILO module provides some methods to append branch. However documentation says that it can be used when register message received. What is the best way to implement this? Thanks, Koray

3 5

people complain Kamailio not handling stale nonce correctly
by Juha Heinanen 03 Jul '19

03 Jul '19

Several people are complaining on baresip git that Kamailio is among those SIP servers that don't handle stale nonce properly, i.e., don't include stale=true flag in WWW-Authenticate header of 401 response. I have tried to tell that Kamailio does what it has been configured to do, but the response is that "by default" Kamailio works incorrectly. "By default" I guess refers to the example Kamailio kamailio.cfg in etc dir. How about adding stale nonce handling to the example config, for example, like in below? -- Juha *** kamailio.cfg 2019-06-13 17:17:17.182148989 +0300 --- /tmp/kamailio.cfg 2019-07-02 18:02:02.947183503 +0300 *************** *** 745,753 **** if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ! if (!auth_check("$fd", "subscriber", "1")) { ! auth_challenge("$fd", "0"); ! exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) --- 745,759 ---- if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ! switch (auth_check("$fd", "subscriber", "1")) { ! case 1: ! break; ! case -4: ! auth_challenge("$fd", 17); ! exit; ! default: ! auth_challenge("$fd", "0"); ! exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH"))

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Change
by Samir Kundu 03 Jul '19

03 Jul '19

Please note change of my new email address – samirkundu(a)samirkundultd.plus.com Thank you Samir ====================================================================== COMPUTING & COMMUNICATION HARDWARE & SOFTWARE SOLUTIONS PROVIDER ====================================================================== e-mail is confidential --- "SAVE PAPER - THINK BEFORE YOU PRINT!" Samir Kundu Ltd | Tel: 0191 538 8280 | Email: samir_kundu(a)samirkundugroup.co.uk Address: 3 Beechwood Terrace, Sunderland SR2 7LY Registered in England & Wales No. 09051111 ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

2 1

DMQ: dmq_t_replicate()
by Lars Olsson 02 Jul '19

02 Jul '19

Hi, I am trying to replicate a REGISTER request between several nodes. When I am using dmq_t_replicate(), it ends the execution of the script. It this correct or I have missed anything? I can not find anything in the documentation of the DMQ module the says that it will end the script execution. Is it recommended to use the DMQ module or should I simply forward the request instead as described in https://medium.com/@tumalevich/kamailio-registration-replication-without-dm… Best Regards, Lars

2 3

[DMQ] Sync DLG_STATE_CONFIRMED_NA dialog state ?
by Mathieu Bodjikian 02 Jul '19

02 Jul '19

Hello, We are using DMQ module to sync dialogs between 3 Kamailio boxes. We are using last Kamailio version (5.2.3). >From our understanding, the state DLG_STATE_CONFIRMED_NA is not synced over DMQ (https://github.com/kamailio/kamailio/blob/a84a3ea618f0e602a8892c37fce7f4e72…) With containerized environement, the ACK's can go to any Kamailio instance, which lead to : On first box : -> INVITE received -> Dialog created + KDMQ sent (state DLG_STATE_UNCONFIRMED) -> 180 | 183 received -> Dialog updated + KDMQ sent (state DLG_STATE_EARLY) -> 200 received -> Dialog updated + KDMQ NOT sent (DLG_STATE_CONFIRMED_NA) On second box : -> ACK received : - next_state_dlg(): bogus event 6 in state 2 for dlg - dialog not updated - KDMQ with state DLG_STATE_CONFIRMED not sent On first box, after timeout : -> After timeout : - dialg with incorrect timeouts (since box2 didn't send dialog update) - tm sends BYE in both ways - bye_reply_cb(): inconsitent dlg timer data on dlg We see two solutions : - We add DLG_STATE_CONFIRMED_NA to states we sync on dlg_dmq_replicate_action - We manage to route the ACK on the same instance that received first INVITE (could be tricky) Which one is preferable ? Kind regards, Mathieu Bodjikian

3 2

question about reply route
by Iván Aponte 02 Jul '19

02 Jul '19

Hello, I have a question about record route and via headers. My scenario is the following: [asterisk] ---- [kam] -------- [trunk] 10.142.0.27 10.142.0.6 200.x.x.x I have an asterisk an a kamailio in a private ip network (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also natted 1:1 to a public ip (35.0.0.6). My problem is that when the invite are replied the asterisk replies to the external ip (35.0.0.6) instead of the internal ip (even though the kamailio replies goes through the internal network) . I tried adding a reply route to change the advertised address to no avail. any advice would be appreciated. this is my conf: LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060" disable_tcp=yes loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "permissions.so" loadmodule "ipops.so" modparam("jsonrpcs", "pretty_format", 1) modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) reply_route { xinfo("source $si"); if(!is_in_subnet($si, "10.0.0.0/8")) { set_advertised_address("10.142.0.6"); } else { xinfo("not in subnet"); } } request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } # Wrapper for relaying requests route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { send_reply("405", "Method Not Allowed"); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } -- Iván Aponte Office: +58(212)9923193 Mobile: +58(412)2774713

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Re: [SR-Users] sr-users Digest, Vol 170, Issue 1
by ali mahfouz 02 Jul '19

02 Jul '19

chat over sip where and how save it On Mon, Jul 1, 2019 at 1:03 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 the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, 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: ATCF/ATGW functionality (Mojtaba) > 2. Re: ATCF/ATGW functionality (Mojtaba) > 3. Re: ATCF/ATGW functionality (Maryam Baghdadi) > 4. kamailio wiht PJSIP IOS client (Gaurav Bmotra) > 5. Re: ATCF/ATGW functionality (Mojtaba) > 6. Re: Late offer ACKs not passing through branch_route > (Enrico Bandiera) > 7. kamailio 4.4.4 fills IPv6 contact address when sending packet > over IPv4 (Pali Rohár) > 8. Re: install mongoDB module with kamailio-cli doker image > (Gaurav Bmotra) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 2 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 3 > Date: Sun, 30 Jun 2019 16:32:01 +0430 > From: Maryam Baghdadi <maryam.baghdadi(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > <CAPrdAbAnnTULTA1rZ-b= > PG1FE+4HfqRHtKg_3R49WEBUXbM+Fw(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Thank you for your response, > If we need to handover of call in our network, What's the solution? is it > possible to add external MSC Server to out network? or add a SBC with this > ability ? > > On Sun, Jun 30, 2019 at 3:40 PM Mojtaba <mespio(a)gmail.com> wrote: > > > Unfortunately you can't, because SRVCC is a capable between MME and > > MCS in your network. When the UE is moving on, the MME send "Take > > handover of the call" to MSC and move the surrect session control to > > MSC. You could read more about SRVCC in 3GPP TS 29.280. > > And FHOSS does not support interface between MME and HSS, > > With Best Regards.Mojtaba > > > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > > <maryam.baghdadi(a)gmail.com> wrote: > > > > > > Hello, > > > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > --Mojtaba Esfandiari.S > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > >

1 0

kamailio wiht PJSIP IOS client
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i na using kamailio 5.1.2 and i m developing ios client with PJSIP audio is working fine ..but on video it work one side but not showing other side it is working fine on other client like zoiper for audio/video both can anyone help me -- *Regards:* Gaurav Kumar

2 1

IMS strange behaviour
by Amar Tinawi 01 Jul '19

01 Jul '19

Hello list i'm facing strange behavior in my lap which consists of p-cscf, i-cscf, s-cscsf, FHoSS for HSS Registration is ok and ICSCF is performing well but when it comes to calls, INVITE Call between two ims users is not passing from scscf to icscf the flow should be : P-->S-->I-->hss i can see invite massage is going out scscf to icscf with all monitoring tools, but no incoming Invite massage to icscf not on sip level and nor on networking level (tcpdump) Actually i don't know where the INVITE packets are going Any advice Regards

1 1

install mongoDB module with kamailio-cli doker image
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i wan to install mongodb module in kamailio/kamailio-ci image pl help me thnaks -- *Regards:* Gaurav Kumar

2 3

Late offer ACKs not passing through branch_route
by Bandiera, Enrico 01 Jul '19

01 Jul '19

Hi, I'm encountering an issue when relaying late offer ACKs, before relaying I set the branch_route with t_on_branch() but kamailio never enters that route after calling t_relay I'm keeping the rtpengine module calls inside branch_routes to avoid issues when re-routing in failure routes I suppose it's something related to ACK packets but I'm not sure what the issue is. Any hints? Thanks, Enrico.

3 2

ATCF/ATGW functionality
by Maryam Baghdadi 01 Jul '19

01 Jul '19

Hello, Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks.

2 3

Kamailio 5.x and Call_control module
by Efelin Novak 30 Jun '19

30 Jun '19

Hi Folks, I'm migrating to Kamailio 5.2.x from 4.4.x. Everything seems to be fine, however I have came to an issue with call_control module as this one is still using old MI interface. Standard situations work nice (maximum debit, prepaid, "CDRs") however when call_control needs to kill a call (credit is gone), it tries to send dlg_end_dlg over MI and it fails. Question are: Is call_control still supported? I haven't found any note about it. Is there any workaround from Kamailio point of view? I'm running last versions of both applications. I'm also rising a ticket at ag-projects side. Thanks Efelin

3 7

Re: [SR-Users] sr-users Digest, Vol 169, Issue 27
by ali mahfouz 29 Jun '19

29 Jun '19

if I send my kamailio.cfg file can you edit it to me please because it is so urgent On Thu, Jun 27, 2019 at 1:06 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 the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, 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: TOPOS uid in contact param instead of username > (Daniel-Constantin Mierla) > 2. Kamailio 5.x and Call_control module (Efelin Novak) > 3. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 4. Statistics for locally generated replies (Duarte Rocha) > 5. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 6. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 7. Re: kamailio as UAC and NAPTR / sendsocket question > (Karsten Horsmann) > 8. MongoDB for Kamailio (Gaurav Bmotra) > 9. Presence publish manipulation (Gertjan Wolzak) > 10. Re: Question about registrar behavior > (Володимир Іванець) > 11. Re: MongoDB for Kamailio (Henning Westerholt) > 12. Re: DMQ: dmq_t_replicate() (Henning Westerholt) > 13. Re: does kamailio support graph database ?? (Henning Westerholt) > 14. Re: Question about registrar behavior (Henning Westerholt) > 15. Need help with "no branches for forwarding" message (Andrew Chen) > 16. Topos / Remove HF_re (Nicolas Breuer) > 17. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 18. Re: Question about registrar behavior > (Володимир Іванець) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jun 2019 12:52:05 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] TOPOS uid in contact param instead of username > Message-ID: > < > CAFRry4VMJd9aaJ_teWd3E-CYg-O9K+qutW699n9GDc9is7wAxQ(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello, > ok, feel free to ask question if something is not clear ... and I think > topoh might help looking at its code, iirc, it uses uri param for contact > masking ... > > Anyhow, there is no better fun that C coding ;-) ... > Cheers, > Daniel > > On Wed, Jun 26, 2019 at 10:12 AM Thomas Weber <thomas.weber(a)pascom.net> > wrote: > > > Hello, > > > > thank you for your opinions. > > > > So i will give it a try throughout the next weeks because we have some > > customer pressure here. > > Didn't do a C project since some 10 years, so it could be fun to "learn > it > > again". > > > > Cheers, > > > > > > Thomas > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >

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Re: [SR-Users] Kamailio next_state_dlg(): bogus event
by Olli Attila 29 Jun '19

29 Jun '19

Hello, The prack is for the initial INVITE establishing the call. No re-INVITE messages are generated in the dialog. --Olli Message: 11 Date: Fri, 28 Jun 2019 08:35:09 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio next_state_dlg(): bogus event Message-ID: <CAFRry4XAF5qYqRxgJA4tzsuUt-56ON1KnzzwvJnxeKOdTUaxSA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, is the PRACK following a re-INVITE or is for the initial INVITE establishing the call? Cheers, Daniel On Fri, Jun 28, 2019 at 8:13 AM Olli Attila <attiolli(a)gmail.com> wrote: > Hello, > > I'm getting this error in Kamailio log every now and then. Can anyone > suggest methods to debug this further for root cause? > > Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog > [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg > 0x7fceeabf8d38 [749:3185] with clid > 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags > 'SDceked01-ecd4281b' '14001358846426' > > Behaviour in the call: > When I look at the SIP flow from this call I can see that a PRACK > message which is routed to Kamailio gets discarded (=the proxy does > not route the PRACK onwards to the UA that requested the usage of the > 100rel in 183 session progress message.) The PRACK message itself > seems valid and when compared to a PRACK from a working call, there is > no difference in the content. > > Cheers, > Olli

2 1

Htable not reloading from db on expiry
by Daniel Greenwald 28 Jun '19

28 Jun '19

I have an htable that loads from mysql. The data loads fine on startup but disappears on expiry. If I issue kamcmd htable.reload it loads again until expiry. It seems as if db_expires is not taking affect. I see them expiring in the debug log but not an errors. Mysql general log does not show any attempt to connect on expiry. Another odd thing is that if I set the expires column in db to anything other than 0 the data doesn't load at all. Any ideas would be appreciated. modparam("htable", "timer_interval", 10) modparam("htable", "htable", "h=>size=20;autoexpire=20;dbtable=validdomains;updateexpire=0;") modparam("htable", "db_url", AUTHDBURL) modparam("htable", "db_expires", 1) create or replace view validdomains as select distinct subscriber.domain as key_name, 1 as key_value, 0 as key_type, 0 as value_type, 0 as expires from subscriber;

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Kamailio next_state_dlg(): bogus event
by Olli Attila 28 Jun '19

28 Jun '19

Hello, I'm getting this error in Kamailio log every now and then. Can anyone suggest methods to debug this further for root cause? Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg 0x7fceeabf8d38 [749:3185] with clid 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags 'SDceked01-ecd4281b' '14001358846426' Behaviour in the call: When I look at the SIP flow from this call I can see that a PRACK message which is routed to Kamailio gets discarded (=the proxy does not route the PRACK onwards to the UA that requested the usage of the 100rel in 183 session progress message.) The PRACK message itself seems valid and when compared to a PRACK from a working call, there is no difference in the content. Cheers, Olli

2 1

Statistics for locally generated replies
by Duarte Rocha 27 Jun '19

27 Jun '19

Greetings, Does the statistics command "kamcmd tm.stats" has an argument or a way to only list locally generated replies? As far as i know, sl.stats only lists local generated replies. Can tm work the same way? Best Regards, Duarte Rocha

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Need help with "no branches for forwarding" message
by Andrew Chen 27 Jun '19

27 Jun '19

Hey guys, I need help to identify the cause of this is error: Jun 26 12:03:53 ashintgtpsg51 /usr/sbin/kamailio[1651]: ERROR: tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding This message was shortly printed after I xlog a message in onsend_route[] block. What else is strange is that the dialog continued on with that same SIP Call-ID. This is what it looks like in the wireshark flow diagram: [image: image.png] The dispatcher entry is an SRV record which lists available hosts. Any thoughts on how I can identify the cause of this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Topos / Remove HF_re
by Nicolas Breuer 27 Jun '19

27 Jun '19

Hello, When using TOPOS module, Kamailio add this header : P-SR-XBranch before executing the topos module. Doing this "remove_hf_re("P-")" in the scripting seems to bypass the addition of the P-SR-XBranch in the sip message. In my mind, remove header function just remove the headers based on the incoming message. What do you think ?

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Question about registrar behavior
by Lars Olsson 27 Jun '19

27 Jun '19

Hi, I have found a behavior in the registrar module that I do have a question about. Is the current behavior correct and wanted? Using the save() method in the script I see the following: * Processing a register request for a user gives return code 1 ( or 2 ) * Processing a unregister request (expires=0) for registered user gives return code 3 * Processing a unregister request (expires=0) for a user which is NOT registered gives return code 1. Why? What is the reason behind this? No database entry is added which is expected. Test performed on 5.1.4, using DB mode 3. For handling a late unregister request ( where registration has already expired) return code does not reflect the the action. I assume that manually checking $expires(max) is the option to go then if I want to detect the unregister request or? Cheers, Lars

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does kamailio support graph database ??
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , does kamailio support graph database ?? thank you -- *Regards:* Gaurav Kumar

2 1

MongoDB for Kamailio
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , on Ubuntu 18.4 and right now i m using Maria DB for kamailio , but i want to migrate my Maria data to MingoDB ,, plz help me how can i use MOngoDB with kamailio thanks -- *Regards:* Gaurav Kumar

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Presence publish manipulation
by Gertjan Wolzak 26 Jun '19

26 Jun '19

Hello Kamailions, I'm running into the following challenge. A kamailio setup where phones register with their mac address as uid and a password. Users have to login to the system by dialing a special number where they have to enter their internal extension and a pin code. Now the system knows which extension(number) is logged in on which phone(macaddress). This is done with from updates, just a type of hotdesking. Now I want to use presence.... When a phone makes a phone call, kamailio generates a publish message where the "local" part of the xml code contains the mac address. How can I manipulate the Publish xml information? I mean the manipulation can be done with textops, but which variables to use and where in the configuration? Im asking because when I disable the presence route and do not call it in my configuration, but with presence enabled, I still get Publish messages.... I have tried adding a Sender header which contains the "extension" of the phone and then call the publish in the PRESENCE route like this: handle_publish("$hdr(Sender)"); But this does not change the xml information. Am I doing things wrong? Below the first publish message that is send, where 10.88.77.172 is the kamailio server. U 2019/06/26 14:54:37.233345 10.88.77.172:5060 -> 10.88.77.172:5060 #7 PUBLISH sip:0015659a9931@10.88.77.172:5060 SIP/2.0. Via: SIP/2.0/UDP 10.88.77.172;branch=z9hG4bK1a87.5510d093000000000000000000000000.0. To: <sip:0015659a9931@10.88.77.172:5060>. From: <sip:0015659a9931@10.88.77.172:5060>;tag=a2652bb825a097f3b7285d4c70edd51a-5c5d. CSeq: 10 PUBLISH. Call-ID: 173009c21f4e3c77-94480(a)10.88.77.172. Content-Length: 579. User-Agent: kamailio (5.1.8 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Expires: 125. Content-Type: application/dialog-info+xml. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:0015659a9931@10.88.77.172:5060"> <dialog id="0_2285997339(a)10.88.77.184" call-id="0_2285997339(a)10.88.77.184" direction="initiator"> <state>Trying</state> <remote> <identity>sip:604113@10.88.77.172:5060</identity> <target uri="sip:604113@10.88.77.172:5060"/> </remote> <local> <identity>sip:0015659a9931@10.88.77.172:5060</identity> <target uri="sip:0015659a9931@10.88.77.172:5060"/> </local> </dialog> </dialog-info> I would like to change the local identity and target uri into the extension assigned to that phone, for example 604114.... So that subscribers to the extension 604114 get informed about the state of the phone with the used uid 0015659a9931. Feedback would be appricated. Pointers to where I am going wrong as well, ... that's feedback too. Rgds, Gertjan

1 0

TOPOS uid in contact param instead of username
by Thomas Weber 26 Jun '19

26 Jun '19

Hello, we're using topos/redis since some time and it's working quite well for most situations. It's used to route asterisk traffic to sip providers. Some providers, however, are validating the contact header strictly against i.e. the accounts username and so we have to disable topos by using event_route[topos:msg-sending]. Now i'm wondering if it would be useful to other people to add a mode to topos where the topos uid is not forwarded as contact username but instead added as a contact uri param. So instead of e.g. Contact: <sip:btpsh-5abcdef5-374-1@x.x.x.x:5060;transport=tcp> it would use Contact: <sip:+49......@x.x.x.x:5060;transport=tcp;line=btpsh-5abcdef5-374-1>. What do you think? Would it make sense? Is it a lot of effort? Cheers, Thomas

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Apache version for FHoSS
by Woscek, Martin W. 26 Jun '19

26 Jun '19

Hello all, What is the version of Apache that FHoSS was built to? Thanks, Martin

2 1

Info: ClueCon Weekly - Kamailio Project Updates
by Daniel-Constantin Mierla 25 Jun '19

25 Jun '19

Hello, later today I will join ClueCon Weekly for a discussion about what is new for Kamailio project and the plans for the future. ClueCon Weekly is a web video conference organized by the FreeSwitch project team. They announced recently a revamp of the format, however, I expect that other people can still join live if they want to listen and ask questions, couple of ways to connect being listed at: * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… If joining live is no longer possible, I will post a link to the recording and if someone has any questions following my presentation, I can answer here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Carrier Route installation Failed
by Amar Tinawi 24 Jun '19

24 Jun '19

Hello i'm facing a dependency error when trying to install kamailio 5.2.3+xenial with carrier route module using apt package installer The following packages have unmet dependencies: kamailio-carrierroute-modules : Depends: kamailio (= 4.3.4-1.1ubuntu2) but 5.2.3+xenial is to be installed E: Unable to correct problems, you have held broken packages. all other modules installed fine any suggestions? thanks in advance

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Kamailio routing BYE back to originator
by Andrew White 24 Jun '19

24 Jun '19

Hi team! I’ve recently found an issue in my Kamailio setup. It appears when the trunk replies to call in progress with a BYE, we relay it immediately back to them: https://i.imgur.com/h5fusau.png <https://i.imgur.com/h5fusau.png> The only differences in the second BYE is the Route header is removed and replaced with a Via header referencing the Kamailio machine, and the From URI is rewritten to show the Kamailio hostname. I’m unsure what steps I can take here, as all of my out of dialog messages (180, 200, ACK) are forwarded along with no issue. It seems only the in dialog are having this issue. Here’s my WITHINDLG and RELAY: def ksr_route_relay() if KSR.is_method_in("IBSU") then if KSR::TM.t_is_set("branch_route") < 0 then KSR::TM.t_on_branch("ksr_branch_manage") end end if KSR.is_method_in("ISU") then if KSR::TM.t_is_set("onreply_route") < 0 then KSR::TM.t_on_reply("ksr_onreply_manage") end #KSR.info("Onreply route: #{KSR::TM.t_is_set("onreply_route")}") end if KSR.is_INVITE() then if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage") end end if KSR::TM.t_relay() < 0 then KSR::SL.sl_reply_error() end exit end def ksr_route_withindlg() return if KSR::SIPUTILS.has_totag() < 0 if KSR::RR.loose_route() > 0 then ksr_route_dlguri() if KSR.is_BYE() then #KSR.setflag($myenv['FLT_ACC'].to_i) #KSR.setflag($myenv['FLT_ACCFAILED'].to_i) elsif KSR.is_ACK() then ksr_route_natmanage() elsif KSR.is_NOTIFY() then KSR::RR.record_route() end ksr_route_relay() exit end if KSR.is_ACK() then #KSR.info("Handling an ACK within dialog") if KSR::TM.t_check_trans() > 0 then ksr_route_relay() exit else exit end end #KSR.info("404ing within dialog") KSR::SL.sl_send_reply(404, "Not here"); exit end I suspect I must be sending this to the wrong branch somehow, but I’m unsure how to correct this. The call gets originally initiated from Kamailio writing a custom RURI and To header before calling t_relay(). I’m wondering if I’m somehow initiating the dialog incorrectly and as such the BYE doesn’t know where to go. Thanks! Andrew

2 3

Kamailio config file flow chart
by Gaurav Bmotra 22 Jun '19

22 Jun '19

hi i m looking for kamailio config file flow chart ,,, if any one have plz provide thanks -- *Regards:* Gaurav Kumar

2 1

5.2.3 - dispatcher module looses setid
by Andrew Chen 21 Jun '19

21 Jun '19

Hi all, For a few times now I've noticed Kamailio 5.2.3 dispatcher module looses view of setid destination and this system been up and running for 13 days: root:~# kamcmd dispatcher.list | grep ash.tpgw.service To work around it, we have to reload the module again: root:~# kamcmd dispatcher.reload root:~# kamcmd dispatcher.list | grep ash.tpgw.service URI: sip:<some host>:5062 URI: sip:<some host>:5060 root:~# Process been running for several days: root:~# ps -elf | grep kam 0 S root 8484 6520 0 80 0 - 3714 pipe_w 16:04 pts/0 00:00:00 grep --color=auto kam 5 S kamailio 32086 1 0 80 0 - 214845 pause Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32164 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32165 32086 0 80 0 - 233342 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32166 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32167 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32168 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32169 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32170 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio Anyone seen this? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Kamailio doesn't start with TLS
by Олег Подгуйко 21 Jun '19

21 Jun '19

Im trying to use kamailio with websocket and tls. It will be use as registrar for websocket clients. I'm using default config with additional websocket, https and tls modules. I created some certs for it and everething looks like good. But when I started kamailio via (systemctl start kamailio) I get some errors июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: <core> [core/tcp_main.c:2753]: tcp_init(): added 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: <core> [core/tcp_main.c:2855]: tcp_init(): bind(9, 0x7fb6ef026f84, 16) on 37.143.9.210:443 : Permission denied июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: tls [tls_init.c:342]: tls_h_init_si(): Error while initializing TCP part of TLS socket 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: rtpengine [rtpengine.c:1986]: mod_destroy(): rtpengine_hash_table_destroy() success! If I start it via /usr/sbin/kamailio - all work. What I do wrong? -- Олег Подгуйко

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kamcmd uac.* commands dont find records
by Karsten Horsmann 21 Jun '19

21 Jun '19

Good morning List! I have an issue with my kamcmd from kamailio 5.2.3 on CentOS 7.6 with kamailio-rpms. The kamcmd util can not find my registrations via command kamcmd uac.reg_enable auth_username FFFFFFFFFFFF kamcmd uac.reg_info l_username +49XXXXXXXX and soone. Only global commands for uac works. Whats going wrong with my kamcmd? Any ideas? Thank for your hints! # kamcmd uac.reg_dump { l_uuid: 49XXXXXXX l_username: +49XXXXXXXX l_domain: sip-trunk.telekom.de r_username: +49XXXXXXXX r_domain: sip-trunk.telekom.de realm: sip-trunk.telekom.de auth_username: FFFFFFFFFFFF auth_password: *************** auth_proxy: sip:reg.sip-trunk.telekom.de expires: 360 flags: 20 diff_expires: 129 timer_expires: 1561103682 reg_init: 1561101919 reg_delay: 3 } # kamcmd uac.reg_enable auth_username FFFFFFFFFFFF error: 404 - Record not found -- Kind regards Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] FHoSS db web interface failing for retrievals -UPDATE
by Woscek, Martin W. 20 Jun '19

20 Jun '19

I can only compile down to JDK 1.6 java 6 but still get same error below from the web interface. But get the following error when pointing to JDK1.5 java 5 which would imply some parts of the code was built with 1.6. BTW I downgraded Ant to 1.9 (Apache Ant(TM) version 1.9.14 compiled on March 12 2019). Buildfile: /opt/OpenIMSCore/FHoSS/build.xml init: [mkdir] Created dir: /opt/OpenIMSCore/FHoSS/bin compile: [javac] /opt/OpenIMSCore/FHoSS/build.xml:48: warning: 'includeantruntime' was not set, defaulting to build.sysclasspath=last; set to false for repeatable builds [javac] Compiling 202 source files to /opt/OpenIMSCore/FHoSS/bin [javac] /opt/OpenIMSCore/FHoSS/src/de/fhg/fokus/hss/auth/AuthenticationVector.java:49: cannot access de.fhg.fokus.diameter.DiameterPeer.data.AVP [javac] bad class file: /opt/OpenIMSCore/FHoSS/lib/jdp.jar(de/fhg/fokus/diameter/DiameterPeer/data/AVP.class) [javac] class file has wrong version 51.0, should be 49.0 [javac] Please remove or make sure it appears in the correct subdirectory of the classpath. [javac] import de.fhg.fokus.diameter.DiameterPeer.data.AVP; [javac] ^ [javac] 1 error BUILD FAILED /opt/OpenIMSCore/FHoSS/build.xml:48: Compile failed; see the compiler error output for details. Any insight would be appreciated. Thanks, Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Woscek, Martin W. Sent: Tuesday, June 18, 2019 2:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] FHoSS db web interface failing for retrievals Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

1 0

XMPP
by David Villasmil 19 Jun '19

19 Jun '19

Hello guys, I'm kind of confused as to what module exactly i need to user, let me explain: I have a service for DIDs and SMS from s.ms, i created a python script that connects to the service using xmpp to their jabber server with a user and a password, i.e.: xmpp_client.py MY-NUMBER(a)s.ms MY-PASSWORD What i do with that script is basically connect and wait to receive SMS, which i then insert in the proxy db (silo) and they get delivered by msilo. I have several numbers and doing this for every number seems stupid, so i started looking for a better solution, since the end user will be receive the message via SIP MESSAGE on they sip client. i came across kamailio/opensips' jabber and xmpp modules, but i'm not sure which (if any) to use, and even HOW to, or the actual architecture, since the documentation keeps talking about a local xmpp server? I would REALLY appreciate some help here. Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 2

Question on Database Version Table
by JR Richardson 18 Jun '19

18 Jun '19

Hey All, I'm doing some upgrades and noticed database table version numbers have increased. database table adjustments: | address | production ver 3 | upgrade 5.2 ver 6 | need to add accountcode and notes column | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add ruri_pattern and priority column I run mysql master/slave so I'm concerned about the master database not having the correct version numbers or can I just update the version number in the table when I do the upgrade of the kamailio versions? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 2

FHoSS db web interface failing for retrievals
by Woscek, Martin W. 18 Jun '19

18 Jun '19

Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

1 0

Kamailio Crashing when using Evapi with CGRateS
by Efelin Novak 18 Jun '19

18 Jun '19

Hi folks, I am doing some experiments with Kamailio and Cgrates. To be precise I am testing evapi module. When in testing environment I do not have any problem. After redirecting some production traffic to this kamailio, It crashes in at random times (few minutes to 20). I didn't find any pattern here. CGRateS doesn't write anything in logs. I'm on the last 4.4 Kamailio (compiled recently). CGRateS is on the last version. Can you please point to how to continue with debugging? Thanks Efelin Here is the problem in syslog: kamailio[9594]: : <core> [mem/f_malloc.c:599]: fm_free(): BUG: fm_free: bad pointer 0x97894818 (out of memory block!), called from tm: h_table.c: free_cell_helper(185) - abortingkamailio[9608]: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 16 kamailio[9588]: ALERT: <core> [main.c:743]: handle_sigs(): child process 9594 exited by a signal 6 kamailio[9588]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated kamailio[9588]: INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD kamailio[9604]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9606]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9603]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received bt full #0 0xb77d1424 in __kernel_vsyscall () No symbol table info available. #1 0xb7673661 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7676a92 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #3 0x082611a3 in fm_free (qmp=0x56779000, p=0x2d4e4f49, file=0x970932ee "tm: h_table.c", func=0x970935b7 "free_cell_helper", line=185, mname=0x97092d20 "tm") at mem/f_malloc.c:601 qm = 0x56779000 f = 0x56df046c __FUNCTION__ = "fm_free" #4 0x96fd0559 in free_cell_helper (dead_cell=0x56d935c4, silent=0, fname=0x970b90ca "timer.c", fline=648) at h_table.c:185 b = 0x56df0490 "CANCEL sip:00444123456@192.168.1.3:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKb6ae.5d03d7e94e5ec25f34930629f434a006.1\r\nMax-Forwards: 66\r\nFrom: \"0044454321\" <sip:0044454321@i"... i = 1 rpl = 0x9797e818 tt = 0xbfe44c68 foo = 0x970ccb14 cbs = 0x0 cbs_tmp = 0x56d12b68 __FUNCTION__ = "free_cell_helper" #5 0x9706884f in wait_handler (ti=31715179, wait_tl=0x56d9360c, data=0x56d935c4) at timer.c:648 p_cell = 0x56d935c4 ret = 0 #6 0x08241e20 in timer_list_expire (t=31715179, h=0x567a3b68, slow_l=0x567a5394, slow_mark=1772) at timer.c:873 tl = 0x56d9360c ret = 0 #7 0x08242211 in timer_handler () at timer.c:938 saved_ticks = 31715179 run_slow_timer = 0 i = 748 __FUNCTION__ = "timer_handler" #8 0x0824266b in timer_main () at timer.c:977 No locals. #9 0x080f0005 in main_loop () at main.c:1699 i = 3 pid = 0 si = 0x0 si_desc = "udp receiver child=2 sock=192.168.1.2:5060\000\270\301\310V\270\301\310V\000\000\000\000\000\000\000\020c\000\000\000\001\000\000\000\000\000\000\000\374Q\344\277HO\344\277\341`&\b\230\071\220\227\004\000\000\000٧5\b$\271\065\bn\000\000\000\324\247\065\b\n\000\000\000\363\320\372\060", '\000' <repeats 11 times> nrprocs = 3 woneinit = 1 __FUNCTION__ = "main_loop" #10 0x080f6b3a in main (argc=13, argv=0xbfe45374) at main.c:2644 cfg_stream = 0x8491008 c = -1 r = 0 tmp = 0xbfe4577c "" tmp_len = 1 port = 0 proto = 0 options = 0x8350e74 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:" ret = -1 seed = 1759582648 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbfe452a0 p = 0x805cc8c "[\201Ä4;" st = {st_dev = 14, __pad1 = 0, st_ino = 6161, st_mode = 16877, st_nlink = 2, st_uid = 109, st_gid = 111, st_rdev = 0, __pad2 = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1560160293, tv_nsec = 983219454}, st_mtim = {tv_sec = 1560165082, tv_nsec = 861178334}, st_ctim = {tv_sec = 1560165082, tv_nsec = 861178334}, __unused4 = 0, __unused5 = 0} __FUNCTION__ = "main"

2 2

Modify DMQ 200OK in on_reply route
by jenus＠cyberchaos.nl 18 Jun '19

18 Jun '19

Hi all, I'm trying to run a kamailio DMQ node inside a docker container. All works fine but i have one issue with DMQ. Since the kamailio process inside the container is behind a nat i have set the advertise address to the external ip, this works fine for the via headers but DMQ keeps sending the internal interface in the KDMQ notification_peer messages. The external ip is in there as well. Example: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.13;branch=z9hG4bK2dcb.c6326de2000000000000000000000000.0 To: <sip:notification_peer@10.10.10.94:5060>;tag=616b7acb681ed5e9cbc485b3faa09d62.0a3d From: <sip:notification_peer@10.10.10.13:5060>;tag=e8f29c06c4b85a188f6533f08d60feb6-ff29 CSeq: 10 KDMQ Call-ID: 17042656602ca532-23367(a)x.x.x.x Content-Type: text/plain Server: kamailio (5.1.4 (x86_64/linux)) Content-Length: 222 sip:10.10.10.13:5060;status=active sip:10.10.10.12:5060;status=active sip:10.10.10.92:5060;status=active sip:10.10.10.93:5060;status=active sip:10.10.10.94:5060;status=active <-- entry for container external ip sip:172.17.0.2:5060;status=disabled <-- entry for container internal ip I tried to strip this entry using the replace_body_atonce in the tm:local-request route. This works fine for the KDMQ request, but when i try to remove the same line from 200OK responses that are send out as a response to incomming KDMQ messages this seems to fail. I use the following code: # Working code (for outgoing KDMQ requests): event_route [tm:local-request] { if(is_method("KDMQ") && $rU == "notification_peer") replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } # Failing code (for 200OK responses): reply_route { xlog("L_ALERT", "REPLY: Sending out reply, tU is $tU rs is $rs\n"); if ($rs == "200" && $tU == "notification_peer") { replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } } I'm running kamailio 5.1.4. Is there a way to strip a line from the body of a local generated 200OK response? Or is there a way to avoid that DMQ inserts the internal container ip in the first place. Thanks, Jan Hazenberg

3 11

Rabbitmq module blocks sip request
by Denys Pozniak 18 Jun '19

18 Jun '19

Hello! I use rabbitmq module to get destination IP for the next call routing, but this module blocks sip request in case of RabbitMQ server is unreachable (timeout_sec/usec does not run out here). - Is there any solution to avoid such behavior and force function rabbitmq_publish_consume() to return an error code and continue script execution? - Is this connected with the nature of rabbitmq-c client? ("Currently librabbitmq offers no async API, but sync API, with a timeout.") -- BR, Denys Pozniak

1 0

Info: Kamailio.org - Infrastructure Maintenance - June 18, 2019
by Daniel-Constantin Mierla 18 Jun '19

18 Jun '19

The infrastructure hosting the main server for Kamailio project will go under scheduled maintenance during June 18, 2019 (tomorrow). The work is expected to start around 7:00am UTC and can last for a few hours. During the maintenance interval, the services related to Kamailio project may become unavailable, among them: * the web server (www.kamailio.org), including the wiki portal (www.kamailio.org/wiki) * the mailing lists and the email contact addresses * the GIT repository mirror git.kamailio.org Once the maintenance work is completed, there will be an update notification sent about it! If there are relevant events to be notified, we are going to publish on our twitter stream: * https://twitter.com/kamailio Also, you can join #kamailio channel on irc.freenode.net in case you want to report some incident related to our infrastructure that you discover while maintenance goes on. For bugs or other issues related to Kamailio code, use the tracker on github.com/kamailio/kamailio – that service is not affected being maintained by GitHub corporation. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

Kamailio 4.4.x CDR db accounting
by Laura 18 Jun '19

18 Jun '19

Dear guys, I'm looking to understand the "info" field inside the CDR written on acc table for Kamailio 4.4.x.. It seem that the field is never written with datas.. Is there a way to write some informations inside it ? Thanks for any reply

3 3

Forward calls between kamailio registrar nodes (dmq + dmq_usrloc)
by Ildar Mukhametzhanov 18 Jun '19

18 Jun '19

Hello! I am developing setup with several kamailio nodes as registrar cluster (dmq + dmq_usrloc). Location sync works, I see locations of both servers on every server. Is there any standard approach to redirect call between nodes except replacing of $ru? Executing of lookup("location") for user registered on another node return negative response. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

1 0

Authenticating xhttp request
by Olli Attila 18 Jun '19

18 Jun '19

Hello, I have this xhttp event_route on Kamailio that I am using to signal the proxy to reload dialplans and htable when necessary: event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Now instead of returning 403 forbidden for requests coming from other src_ip than proxy itsef, I would like to authenticate the http request via proxy database. How can this be done if possible? Cheers, Olli

3 7

dispatcher and TLS targets
by Karsten Horsmann 17 Jun '19

17 Jun '19

Hi all, i try to configure an SBC OS config [1] based kamailio 5.2.3 [2] with dispatcher and rtpengine. I used transport=tcp to see the plain traffic and then switched to TLS (with tls.cfg, valid certificate and stuff). After starting up, the Target is marked as "down". Due the encryption its hard to debug that. Any hints? Did i made an mistake in the configuration? TLS calls from the target to my kamailio proxy works. So its "half broken" :) at the moment. [1] https://github.com/voiceboys/sbcOS/blob/master/SbcOS/configs/voice/kamailio… [2] kamailio -v version: kamailio 5.2.3 (x86_64/linux) c36229 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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_BLACKLIST, 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: c36229 compiled on 11:28:11 May 22 2019 with gcc 4.8.5 -- %< --------------------- kamctl dispatcher dump "SET": { "ID": 1004, "TARGETS": [{ "DEST": { "URI": "sip:sip101.example.de;transport=tls", "FLAGS": "TP", "PRIORITY": 0, "ATTRS": { "BODY": "access=212.xx.xx.xx:5061;socket=tls:212.xx.xx.xx:5061;weight=100;ping_from=sip: mykamailio.example.de", "DUID": "", "MAXLOAD": 0, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "tls:212.xx.xx.xx:5061" }, "LATENCY": { "AVG": 30000, "STD": 0, "EST": 30000, "MAX": 30000, "TIMEOUT": 1 } } }] } }, -- %< --------------------- kamctl dispatcher dump WARNING: <script>: Destination down: OPTIONS ru=sip101.example.de;transport=tls du=<null> -- %< --------------------- tls.cfg [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [server:212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client: 212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [client:default] verify_certificate = no require_certificate = no -- %< --------------------- tls.cfg Cheers Karsten -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

sr -user unsubscribe request
by Martin Lane 17 Jun '19

17 Jun '19

Please unsubscribe me from the sr-users list. V/R, Martin

1 0

Dispatcher's probing malfunction
by Duarte Rocha 17 Jun '19

17 Jun '19

Greetings, I have some questions about dispatcher's probing mechanics : 1 - In my SBC i have 26 gateways being probed. However, if i do a SIP capture no more than 16 gateways are probed each minute. Some gateways are always probed, others only once in a while. 2 - I have two SBC working in a pair with DB being replicated and with HA IPs. That being said i configure one NAP twice, each configuration with one HA socket. Is this configuration supported? I get very weird behaviours when i configure it like this. Best Regards, Duarte Rocha

1 0

restrict calls to only registered users
by Gaurav Bmotra 17 Jun '19

17 Jun '19

hi i m using kamailio 5.1 , with UBUNTU 18.1 i want to restrict call to only registered user ,,, plz help me thank you -- *Regards:* Gaurav Kumar

3 2

Querying in-memory location table (dmq_usrloc)
by Gholamreza Sabery 16 Jun '19

16 Jun '19

I am using dmq_usrloc, to replicate user locations across multiple Kamailio instances. Everything works fine. The problem occurs when a user on Kam1 wants to call a user on Kam2. How can I query the in-memory location table to see where the user is registered? PS: I use a Path header to specify the server that each user is registered on.

2 2

Softswitch
by Mary Honey 16 Jun '19

16 Jun '19

Hello, Is it possible to use Kamailio as a grade 5 Softswitch ? I need to handle 1000 calls/s

2 1

Stateless message forwarding with forward() add extra identical Via header.
by Ali Shirvani 15 Jun '19

15 Jun '19

Hi all, According to this SO question <https://stackoverflow.com/questions/32519551/kamailio-concurrent-calls-afte…> I tried to forward received message to multiple destinations with the following Lua script. But in the second UA I got duplicate Via header. Here is the Lua script: ```Lua KSR.forward_uri("sip:ali@192.168.1.50:7080;transport=UDP"); KSR.forward_uri("sip:tes@192.168.1.50:9080;transport=UDP"); ``` And here is the received Via headers is `tes` UA: ``` Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 10.189.219.4:5060 ;rport=5060;rport=5060;branch=z9hG4bK-4177-1-2 ``` Would you please help me to forward message to multiple destination without duplicate Via headers? Regards, Ali

1 0

403 Forbidden
by Pejic, Miroslav 14 Jun '19

14 Jun '19

Hi :) I'm trying to save a failed call in the kamailio_router.acc_cdrs database after sending the 403 Forbidden by router, using the module acc, if is it possible? https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html Something like this: route[IS_CALLER_BLACKLISTED] { python_exec("check_caller_blacklisted"); if ($var(callerBlacklisted) == 1){ t_send_reply("403", "Forbidden"); IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? exit; } Thanks in advance! Miro

4 6

Re: [SR-Users] 403 Forbidden
by Federico Cabiddu 14 Jun '19

14 Jun '19

I understand, to get the cdr you should have created the dialog before replying. Cheers, Federico On Fri, 14 Jun 2019, 17:05 Pejic, Miroslav, <miroslav.pejic(a)cloud.timenet.it> wrote: > [image: image.png] > > The call doesn't even begin, I immediately block it. :) > > Il giorno ven 14 giu 2019 alle ore 16:59 Pejic, Miroslav < > miroslav.pejic(a)cloud.timenet.it> ha scritto: > >> Hi Federico, >> >> i tryied also with `modparam("acc", "cdr_on_failed", 1)` >> but in my case the call is not saved at all on `acc_cdrs`, >> with the second method i mentioned works :) >> Thanks >> Miro >> >> Il giorno ven 14 giu 2019 alle ore 16:14 Federico Cabiddu < >> federico.cabiddu(a)gmail.com> ha scritto: >> >>> Hi, >>> dialog CDR for failed calls can be generated enabling >>> >>> http://www.kamailio.org/docs/modules/devel/modules/acc.html#acc.p.cdr_on_fa… >>> >>> It has been some time since I last tested it, but it still should work :) >>> >>> Cheers, >>> >>> Federico >>> >>> On Fri, Jun 14, 2019 at 3:25 PM Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> it should be ok also with 2) if it gives what you want. Kamailio is >>>> just writing to acc_cdrs, if you have extra columns, it should not be >>>> affected. >>>> >>>> Cheers, >>>> Daniel >>>> On 14.06.19 14:16, Pejic, Miroslav wrote: >>>> >>>> Hi Daniel :) >>>> >>>> i tryied 2 methods to save it >>>> >>>> 1) i created a new table on kamailio_router called `acc` >>>> >>>> from `utils/kamctl/mysql/acc-create.sql` >>>> >>>> ``` >>>> CREATE TABLE `acc` ( >>>> `id` INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, >>>> `method` VARCHAR(16) DEFAULT '' NOT NULL, >>>> `from_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `to_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `callid` VARCHAR(255) DEFAULT '' NOT NULL, >>>> `sip_code` VARCHAR(3) DEFAULT '' NOT NULL, >>>> `sip_reason` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `time` DATETIME NOT NULL >>>> ); >>>> CREATE INDEX callid_idx ON acc (`callid`); >>>> ``` >>>> then >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> in `acc` i have now calls that were not answered with 200ok :) >>>> >>>> >>>> 2) add new fields on exsisting acc_cdrs table used to store cdrs >>>> >>>> ``` >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `from_tag` varchar(128) DEFAULT NULL >>>> AFTER `dst_domain`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `to_tag` VARCHAR(128) DEFAULT NULL >>>> AFTER `from_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_code` VARCHAR(3) DEFAULT NULL >>>> AFTER `to_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_reason` varchar(128) DEFAULT >>>> NULL AFTER `sip_code`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `time` datetime DEFAULT NULL AFTER >>>> `sip_reason`; >>>> ``` >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc_cdrs"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> In this way i have both stored calls that were not answered with 200ok >>>> and that were answered with 200ok. >>>> The idea is to have all in one table `acc_cdrs`. >>>> >>>> Can be ok also the second solution? >>>> >>>> Thank you very much! >>>> Miro >>>> >>>> Il giorno ven 14 giu 2019 alle ore 09:06 Daniel-Constantin Mierla < >>>> miconda(a)gmail.com> ha scritto: >>>> >>>>> Hello, >>>>> >>>>> I don't think that you can get a dialog-based CDR for calls that were >>>>> not answered with 200ok. But you can get a record in acc table, by storing >>>>> failed transaction record (see the mod params for acc and failed >>>>> transaction flag or use acc_db_request()). >>>>> >>>>> Cheers, >>>>> Daniel >>>>> On 13.06.19 22:45, Pejic, Miroslav wrote: >>>>> >>>>> Hi :) >>>>> >>>>> >>>>> I'm trying to save a failed call in the kamailio_router.acc_cdrs >>>>> database >>>>> >>>>> after sending the 403 Forbidden by router, using the module acc, if is >>>>> it possible? >>>>> >>>>> >>>>> https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html >>>>> >>>>> >>>>> Something like this: >>>>> >>>>> >>>>> route[IS_CALLER_BLACKLISTED] { >>>>> >>>>> >>>>> python_exec("check_caller_blacklisted"); >>>>> >>>>> >>>>> if ($var(callerBlacklisted) == 1){ >>>>> >>>>> >>>>> t_send_reply("403", "Forbidden"); >>>>> >>>>> >>>>> IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? >>>>> >>>>> >>>>> >>>>> exit; >>>>> >>>>> } >>>>> >>>>> >>>>> Thanks in advance! >>>>> >>>>> Miro >>>>> >>>>> _______________________________________________ >>>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>> >>>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>

1 0

Getting which timer has expired
by Igor Olhovskiy 14 Jun '19

14 Jun '19

Hi! Is there any way to get which timer on transaction actually expired? Scenario is following: INVITE - 180 - 200 OK - (restransmissions 250 msec starts) - pause 15 sec - ACK. In INVITE i'm setting t_on_reply("MY_REPLY") But in around 10 sec after first 200 OK (and 12 sec after INVITE) I'm getting in logs DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7fb632d77090 from timer.c:689 and on next 200 OK I'm never getting into onreply_route[MY_REPLY] again. Nowhere in the code or config I have explicit set any of timers with 10 or 12 seconds value. Most of them are default. So, can I somehow get which timer expired exactly? PS: If any, pause in 15 sec for a reason to emulate this issue. Regards, Igor

2 2

Event Route not found when loading XHTTP module
by Dhruva TS 14 Jun '19

14 Jun '19

Hello, I am new to Kamailio and I am using Kamailio 5.2. When trying to the load the XHTTP module, event route not found error is thrown and XHTTP is module is not getting loaded. I am trying to use JSONRPC module through HTTP transport for RPC calls. Using XMLRPC, things work fine. DEBUG: <core> [core/sr_module.c:896]: init_mod(): xhttp ERROR: xhttp [xhttp_mod.c:136]: mod_init(): failed to find event_route[xhttp:request] ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module xhttp (/usr/lib/kamailio/modules/xhttp.so) ERROR: error while initializing modules I have added below to Kamailio.cfg file: loadmodule "xhttp.so" event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Any help in this regard. Thanks, Dhruva

2 1

S-CSCF replies “Forbidden - Domain not served”
by Bao, Yin 13 Jun '19

13 Jun '19

We have configured Kamailio IMS (P, I, S-CSCF) to use a single realm with the same NETWORKNAME and trying to make a call between bob and alice. We are getting S-CSCF replies “Forbidden – Domain not served” in two scenarios: 1. When bob calls alice, the INVITE went from P->S->I->S and the destination S-CSCF responded this error 2. During SIP registration in responding to SUBSCRIBE (endpoint still can register even with this reply, so we proceeded with making call) In quick search we found that in scscf/kamailio.cfg route{ } implementation there is a case when this error is generated: # Terminating if (uri == myself || uri =~ "tel:.*") { if (!term_impu_registered("location")) { xlog("L_DBG", "We need to do an UNREG server SAR assignemnt"); assign_server_unreg("UNREG_SAR_REPLY", "location", "term"); exit; } } else { sl_send_reply("403","Forbidden - Domain not served"); exit(); } route(term); break; What is this error designed for? Also in general are we supposed to rework the routing code in Kamailio.cfg (P, S, I-CSCF)? Thanks for any suggestions in advance. Yin

1 0

Problem with Kamailio 5.2.x Repository
by Joshua Riffle 13 Jun '19

13 Jun '19

Hi, Heads up... I suspect the Kamailio 5.2.x RPM files for CentOS 7 were recently updated June 10th and are now no longer accepted by a standard yum install... *Repository file in /etc/yum.repos.d/:* [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… gpgcheck=1 gpgkey= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… enabled=1 *Operating System Version:* CentOS Linux release 7.6.1810 (Core) (Fresh image -- varying cloud providers) *Command to install:* yum install kamailio *Resulting error message:* kamailio-5.2.3-4.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.3-4.el7.centos.x86_64: [Errno 256] No more mirrors to try. *Troubleshooting:* I verified the IP address for the opensuse URL with three different Internet providers and tested the installation on servers in different cloud providers geolocated in both Canada and United States and received the exact same results in all cases. -- Joshua Riffle *DevOps Lead & Duke of the Dial Tone* We Are Skyetel and You Are Awesome. p: (360) 843-9903 o: (360) 986-5200 s: http://www.skyetel.com e: josh(a)skyetel.com

2 2

CRITICAL ERROR in data_lamp.c!
by Mojtaba 13 Jun '19

13 Jun '19

Hello, What does this CRITICAL error mean? Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: <core> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out of memory block!) called from core: core/data_lump.c: free_lump(466) - aborting I got it when i make new call during debugging new module in Kamailio? In module we have to change some message's body with data_lump. I asked relaed issue (strange behaviour while developing new module) in Kamailio developer forum <sr-dev(a)lists.kamailio.org> With Best Regards. -- --Mojtaba Esfandiari.S

2 10

dialog:end event route
by Igor Olhovskiy 13 Jun '19

13 Jun '19

Hi! Just to confirm, is event route [dialog:end] not called on fr_inv_timer expiry? Cause it's called on call busy (486) and reject (whatever 6xx). So, what's the idea of not calling it on timeout? Or I'm missing something? Kamailio 5.2 Thanks! Regards, Igor.

2 8

Best practises for websocket authentication
by Grant Bagdasarian 12 Jun '19

12 Jun '19

Hello, I was wondering if there are any best practices for authenticating websocket connection establishment inside the xhttp event route? Thanks and regards, Grant

3 3

TCP reset behavior
by R, Rajkumar (Raj) 12 Jun '19

12 Jun '19

Hi All, In our setup running kamailio (alpine image) in stateless mode and have following TCP config set. After client registers to the backend see two TCP connections created - one towards the client and the other towards the backendserver. If there's a abrupt network disconnect either at the client side or at the backend side, do not see TCP reset happening from kamailio as per the config. Per below config expected after 120seconds( = [tcp_keepidle] + ( [tcp_keepintvl * tcp_keepcnt] ) instead it keeps resending TCP keepalive packets at varying intervals for varying duration length(like 6mins, 14mins..) before resetting the TCP connections. Also want to mention that we are not using TCPOPS module here. I am trying to achieve consistent duration for TCP reset in kamailio. Could you please point me if I have missed anything here? Setup: Client ---- Kamailio (stateless mode) ---- Backend server TCP Config: tcp_connection_lifetime = 3600 tcp_accept_aliases = no tcp_async = yes tcp_connect_timeout = 10 tcp_crlf_ping = yes tcp_keepalive = yes tcp_keepcnt = 3 tcp_keepidle = 30 tcp_keepintvl = 30 # Total 2 mins - 30+(30*3)=120 sec tcp_linger2 = 30 tcp_rd_buf_size = 65536 tcp_send_timeout = 10 thanks, raj

2 9

Setting the probing state failed
by Daniel Greenwald 12 Jun '19

12 Jun '19

ERROR: dispatcher [dispatch.c:2635]: ds_options_callback() Setting the probing state failed (sip:10.0.9.133:5064, group 0) I get this message periodically from 3 kamailio dispatchers simultaneously for the same destination host. Any idea what this error means? Dispatcher settings below: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "sock_avp", "$avp(dssocket)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "ds_hash_size", 3) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:pinger@MYPRIVATEIP :INTERNALPORT") #modparam("dispatcher", "ds_ping_method", "INFO") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")

2 1

How reply INVITE request with 200 OK that contains SDP?
by Ali Shirvani 11 Jun '19

11 Jun '19

Hi all, I tried to implement simple announcer by using play_media function in latest rtpengine module. I implemented simple Lua script and I tried to test the implementation with simple SIPp scenario. Here is the Lua function to reply the received INVITE request. ```Lua function ksr_route_announcer() KSR.info("======= Announcer =====\n"); local u = KSR.pv.gete("$rU"); KSR.info("username is: " .. u .. "\n"); if u == "ann" then KSR.info("announcer has been called.\n"); if KSR.textops.has_body_type("application/sdp") then KSR.info("request contains SDP.\n"); KSR.tm.t_newtran(); KSR.rtpengine.rtpengine_manage0(); KSR.tm.t_reply(200, "OK"); KSR.x.exit(); end end end ``` According to kamailio logs rtpengine modified the INVITE SDP, but I want to add SDP to 200 OK response. How should I manage rtpengine to append SDP answer to the 200 OK response? The attached is the SIPp log, as you can see that last 200 OK doesn't contain SDP. Regards, Ali

2 1

Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Daniel, Actually, the HSS is not returning the uri with the +. There are no users int the HSS database with a public ID with a + so the HSS returns the USER_UNKNOWN. Looking at the PCAP, the I-CSCF sends a diameter msg to the HSS asking for the verification the public ID of: Public-Identity: sip:+alice@ims522.irisims.org The diameter msg from HSS is returning : AVP: Experimental-Result-Code(298) l=12 f=-M- val=DIAMETER_ERROR_USER_UNKNOWN (5001) Why do we want the I-CSCF to add a + to all URI's don't start with a +? The I-CSCF is the module that is adding the +, assuming that we should be able to included Kamailio.cfg for respective modules, is there a directive or variable that I need to set? Or do I simply comment out that code? Thanks, Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Tuesday, June 11, 2019 6:22 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Woscek, Martin W. <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2.2 ICSCF changing sip to tel Hello, that prefix operation is done only if the r-uri username does not have +. So if you HSS is returning the r-uri with +, then all is ok, no other + is added. On the other hand, I am not sure what is the error or the issue you try to sort out. The error message is because of the xlog(...) in the config. Cheers, Daniel On 10.06.19 16:17, Woscek, Martin W. wrote: Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org<mailto:alice@ims522.irisims.org> syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>

2 1

Relay NAP Probing
by Duarte Rocha 11 Jun '19

11 Jun '19

Greetings, I have a Kamailio as Sip Proxy between my clients and my Media gateway. I have NAPs (One per client) for clients and NAPs for the MediaGateway (One for each client). The MediaGateway probes Kamailio in order to see if it's alive and Kamailio probes clients. However, i would like for the MediaGateway to be aware of the client status. I thought of a few solutions : - When Kamailio probes a client and it fails, both NAPs (Client Side and MG side) could be set as inactive. - Automatically relay the MG probe to the client - Some scripting that reads the result of dispatcher.list and if it finds an inactive NAP, set the MG NAP as inactive too. Is there a better/simpler way to do this? I'd rather have everything on Kamailio configuration files instead of doing exatra scripting. Any opinions ? How can i achieve this? Best Regards, Duarte Rocha

2 1

Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek

2 1

Operations after relay
by Nicolas Breuer 11 Jun '19

11 Jun '19

Hello, I want to do some sql operations after a 200 reply, after relaying the sip packet Is it possible or sending the sip packet is the last operation permitted ?

3 3

Connect SIP Trunk to PBX
by Chad Kenealy 10 Jun '19

10 Jun '19

Hello, I hope you can help, We are trying to connect our SIP trunk to our new PBX system, however our SIP trunk provider requires a static IP to add to the endpoint. Our new PBX provider uses a range of IPs that are connected to a domain and the IPs change all the time. Is there a way of using Kamailio that we could get our Sip Trunk provider and PBX system to work together? Regards

2 1

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