- sr-users - kamailio.org

OpenSIPS balcklist/whitelist related question
by eyatsko 16 Jun '16

16 Jun '16

Hello! Could you, please, help me to make the above?.. I already have rough understanding about blst, permission modules etc. But I am about to compose CFG to count wrong (erroneous) attempts to REGISTER or INVITE from the same IP through several SIP dialogs. For example, some malicious user attempes to brute-force password. I would like to move it permanently into permissions' database after several (predefined) attempts. Kind regards, Ellad

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uac.reg_refresh not removing records from memory
by DanB 16 Jun '16

16 Jun '16

Hey Guys, I was wondering if there is any dynamic way today to remove a single registration out of memory (and implicitly to force un-REGISTER) Eg: I add one record using uac.reg_refresh (which works fine) but after removing the record from the database and executing uac.reg_refresh again, the record stays in memory until expire. Is there a better way to enforce un-register on remote side? Thanks in advance! DanB

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Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

Hello I am trying to make a test setup to test the RTPENGINE. Case 1: Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) Case 2: When A calls to B - the sound both ways, so it is ok. Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) WITH RTPENGINE: When A calls to B - the sound is absent, no sound. I compared both dumps the difference only in addressesAnd what I read in nathelper module: - 1 - Contact header field is searched for occurrence of RFC1918 addresses. - 2 - the "received" test is used: address in Via is compared against source IP address of signaling - 4 - Top Most VIA is searched for occurrence of RFC1918 addresses - 8 - SDP is searched for occurrence of RFC1918 addresses - 16 - test if the source port is different from the port in Via - 32 - test if the source IP address of signaling is a RFC1918 address All flags can be bitwise combined, the test returns true if any of the tests identified a NAT. in nat-traversal-role.cfgI see: if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } As I understand - both addresses are RFC1918but in case 2 the RTPENGINE is not kernalized Does the RTPENGINE distinguish the ip address? why it was not kernalized in case 2? Any help is appresiated.

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RTPENGINE not kernalizing in some cases
by Dmitry 16 Jun '16

16 Jun '16

Hi I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104 i use rtpengine_manage(). When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media 2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice. The Asterisk configs are absolutely the same. Kamailio log shows no errors In both cases the INVITE's SDP connection is 10.34.101.104 - the address of the Kamailioin the ' case 1' I see all RTP streams : 2 rtp streams - between asterisk and kamailio and vice versa, 2 -rtp streams between kamailio/rtpengine - Freeswitch and vice versa in the second case I see the following RTP streams: 1 rtp stream - from Kamailio to Asterisk, 1 rtp stream - from Freeswitch to Kamailio How to debug such cases? Any help is appreciated.

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HTTP query at the end of a phone call
by Jay Li 15 Jun '16

15 Jun '16

Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks. Regards,Jay

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404", "Not Found location"
by Sébastien Brice 15 Jun '16

15 Jun '16

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found location"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } thats weird i did follow ubuntu realtime trying to get sip account registered and then uac_req_send to asterisk but the first step already failed. :-( i noticed there is no Username in my RURI (whatever UAC i am using) maybe this is the root of evil please find attache my kamailio.cfg i am using kamailio v 4.4 Any idea very welcome

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Kamailio + ToS
by Marcelo Bonin 15 Jun '16

15 Jun '16

Hello, I need to know what are the available parameters to set ToS on Kamailio. I couldn't found much information about that, only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos but there is no reference for the available parameters. Thanks, Marcelo.

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question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

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IP TOS
by Daniel W. Graham 14 Jun '16

14 Jun '16

Could I get some clarification on tos global setting as referenced by documentation below? Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level? I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio. tos The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP). Example of usage: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY Thanks! -Dan

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Fwd: [Free-RTC] GSoC progress, more mentors?
by Daniel Pocock 14 Jun '16

14 Jun '16

This report was sent to the Free-RTC list https://lists.fsfe.org/mailman/listinfo/free-rtc Most of the projects have an emphasis on SIP If anybody wants to facilitate testing the new Telepathy/Empathy SIP connection manager code against Kamailio or other SIP proxies, registrars and presence servers that would be very welcome. -------- Forwarded Message -------- Subject: [Free-RTC] GSoC progress, more mentors? Date: Mon, 13 Jun 2016 22:53:46 +0200 From: Daniel Pocock <daniel(a)pocock.pro> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free-RTC(a)lists.fsfe.org Hi all, The GSoC students have officially been working for 3 weeks now. I created a summary of all the students here: https://wiki.debian.org/SummerOfCode2016/RTCProjects They've been publishing weekly reports on the debian-outreach list: https://lists.debian.org/debian-outreach/ It would be really helpful to increase the size of the mentoring team for the students working on RTC projects: - we probably need at least one more Python developer and one more C++ developer to help - other people can also help test the projects, for example, if you are using Wordpress, please try Mesut's new WPCall module: https://github.com/mesutcang/wpcall As a bonus, some mentors are selected every year for a fully paid trip to California for Google's GSoC mentor summit. If anybody has more questions about it or would like to be more involved please contact me privately. Regards, Daniel _______________________________________________ Free-RTC mailing list Free-RTC(a)lists.fsfe.org https://lists.fsfe.org/mailman/listinfo/free-rtc

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Kamilio as Scalable Registrar + Proxy
by Colin Morelli 13 Jun '16

13 Jun '16

Hello, I currently have Freeswitch acting as a B2BUA, handling registrations, routing, etc for a prototype voice application I have built. Now I'm at the point where it's time to actually scale it out, and I'm looking at a few different options for the SIP proxy + registration later. Essentially, what I would like to have (I think), is: 1) Stateless SIP proxy with consistent hashing to route to: 2) Transaction-stateful SIP proxy handling registrations and initial routing Ideally, the only thing that's stored across the cluster (if possible) would be registrations, in something like memcache/redis. Then, of course, each node might have its own internal state of transactions that it's tracking. I'd prefer the only thing to care about dialog state to be Freeswitch. Finally, the decisions about which FS instance to route to would be made by my application as well, based on runtime rules. Is this something people here have done before? Are there other suggestions on how to scale out kamailio to multiple instances. Thanks in advance, Colin

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BLF lights
by Bill 13 Jun '16

13 Jun '16

I have spent a couple days beating my head against the Kamilio BLF feature. But cannot seem to get presence signals from one ua to another. I am testing with a Zoiper softphone and a Yealink hardware phone, both registering to Kamailio and able to make and receive calls. But getting absoultely nowhere with BLF. Have gone through all the suggestions in whitepaper 43, but still no results. Anybody had success with BLF, please steer me in the ritght direction. Thanks, Bill

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Kamailio benchmark app_python + OPTIONS
by Anonim Stefan 13 Jun '16

13 Jun '16

Hi, I am doing some basic benchmarks with Kamailio. I am using a VM with 1 CPU i7 2,50 GHz and Kamailio with children=1. This is what I measure for each request, inside the benchmark block: - app_python processing (some euclidian distance based on a dlg_var input) -> same for all requests - uac_req_send() OPTIONS -> same for all requests I am running SIPp tests for 1000 calls with 1,10,50,100,150,200 calls per second. What I see is that the Global average processing time per request decreases with the increasing rate; I expected it to increase with the increasing rate. Is there an explanation for this, or am I doing something wrong ?! Thanks, Stefan

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BYE message to an individual client
by kai.ohnacker＠cbc.de 13 Jun '16

13 Jun '16

Hello Community, is it possible to send a BYE message to an individual client via mi command line (siremis)? Cheers, Kai

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TM Module t_suspend question
by gmele 10 Jun '16

10 Jun '16

Hello, we are using the TM Module* t_suspend *function in our kamailio to suspend transactions (mainly INVITE requests). We do this as we send some push notification to the callee that will force it to re-REGISTER. Once we get the REGISTER, we resume the transaction with a *t_continue()*. If the callee never re-REGISTER, the FR Timer will terminate the INVITE transaction. This is working fine for us when the callee has a phone that can receive the push notification. Now, we want to be able to contact users having multiple devices: phone and desktop VoIP clients. The problem with current implementation is that if we wait for the re-REGISTER from the mobile phone, we may never send the INVITE to the desktop VoIP client if the mobile phone is shutdown. So, I'd like to now if it is possible to set a timer to the t_suspend command which will automatically resume the suspended transaction (without issuing an ERROR like the FR timer is doing)? I didn't see such possibility in the TM module, but is there another way to implement this? Hope I was clear enough Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TM-Module-t-suspend-question-tp1493… Sent from the Users mailing list archive at Nabble.com.

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Re: [SR-Users] Push notification question
by Vasiliy Ganchev 10 Jun '16

10 Jun '16

Hi! To cope with your specific cases try connection-oriented transport (tcp/tls). e.g. if app is closed - the connection is closed as well and the usrloc record will be removed. After this use standard approach (with late forking - tsilo module) imho, it is not a good idea to suspend the call if there is records in location. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/Push-notification-question-tp149025… Sent from the Users mailing list archive at Nabble.com.

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Handling ReINVITEs
by Grant Bagdasarian 09 Jun '16

09 Jun '16

Hello, How should reINVITEs be handled in the kamailio script and also in combination with rtpengine? I was reading the RR docs and loose_route function pretty much handles this if I'm correct. We're currently receiving reINVITEs for checking if a session is still active, not for session modifications like SDP changes. Does kamailio handle reINVITEs correctly out of the box or should extra logic be implemented? Also, what happens when rtpengine_manage is called for a reINVITE inside WITHNDLG or in onreply_route for 200 OK? Is the module smart enough to ignore port (de-)allocations when the SDP is still the same as the initial INVITE or 200 OK? Regards, Grant

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forward calls from kamailio(RTPengine) to FREESWITCH
by Ankitt Sharma 09 Jun '16

09 Jun '16

hello i am using using kamailio + RTPengine in my setup and i want to forward every single call to freeswitch(for recording purpose), all three are running on the same machine (testing in the local network) my requirement for media flow is: UA1-->RTPproxy-->>freeswitch-->RTPproxy-->UA2 my user agents are using webrtc media profile so my rtp engine flags are : rtpengine_manage("trust-address replace-origin replace-session-connection RTP/SAVPF"); i've attached my configuration file i've configured freeswitch for media_webrtc=true ,still when i try calling the freeswitch send 488 without sending call to other end should media-address flag be used to forward media through, if yes how it should be done ??

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ACK does not match transaction after stripping Route header
by Francisco Valentin Vinagrero 09 Jun '16

09 Jun '16

Hi there, I'm having an issue in a SBC (ACME) -> KAMAILIO -> Asterisk scenario with an ACK that gets ignored in Kamailio because it does not match any transaction. The INVITE coming from the SBC looks like this (only relevant headers and hidden numbers for simplicity - SBC has IP .12 , Kamailio .30 and Asterisk .34) INVITE sip:mynumber@10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip: mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip: a-number @10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:41754112601@10.15.1.12:5060;transport=udp> And its forwarded to Asterisk with the Record-Route header: INVITE sip: mynumber @10.15.1.30:5060 SIP/2.0 *Record-Route: <sip:10.15.1.30;lr=on;ftag=SDkbo9901-42090> *Via: SIP/2.0/UDP 10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip:a-number@10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:a-number@10.15.1.12:5060;transport=udp> Then, 200 OK from Asterisk: SIP/2.0 200 OK *Via: SIP/2.0/UDP 10.15.1.30;rport=5060;received=10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> Which is sent to the SBC like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> And finally the SBC sends the ACK: ACK sip:10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdt7p9k00dounet8ic600.1 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 ACK Contact: <sip:a-number@10.15.1.12:5060;transport=udp> *Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090 The problem: this ACK gets not retransmitted to Asterisk At first, I thought it was some sanity check but after disabling that I realized that it was in the WITHINDLG route. For the incoming ACK I get in the logs: Jun 8 11:56:47 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53240]: ALERT: <script>: Inside LOOSE route for ACK proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route=sip:10.15.1.30;lr;ftag=SDkbo9901-42090 src_ip=10.15.1.12 And once the ACK is ready to be sent to Asterisk, the Route header has been removed and no Record-Route has been added so it fails. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: INFO: rr [rr_mod.c:402]: pv_get_route_uri_f(): No route header present. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: ALERT: <script>: ACK does not match transaction!! proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route= src_ip=10.15.1.30 My WITHINDLG route looks like this: # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { xlog("L_ALERT","Inside LOOSE route\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_ALERT","Inside LOOSE route for ACK proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_ALERT","ACK does not match transaction!! proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Thanks for reading this :) Any idea about how to validate the transaction? t_check_trans is not being validated... Cheers, Francisco.

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rtpproxy question
by Albert Petit 09 Jun '16

09 Jun '16

Hi list I have installed a newer version of rtpproxy (2.0) in our development server , i was happy because it brings lot of performance improvements However i have found an issue when clients use TLS+SRTP In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic as my B2BUA does that or in some scenarios we do plan to add end to end encryption. RTPProxy 1.X was doing that perfectly Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted by rtpproxy and i receive it clear in the B2BUA How can i disable this new feature and that rtpproxy just keeps sending the received SRTP/SRTCP stream without decrypting it?

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IPv6 Only client-rtp proxy configuration
by aman jeff 09 Jun '16

09 Jun '16

Hi team, How to do the Configuration of Kamailio+RTP-Proxy for SER in IPv6, Currently I can establish SIP Session,while rtp start call are not audible.So Some configuration need to change,Am bit confused on it,Please help to resolve the issue,For my proper VoIP-Communication in IPv6 Surveillance. Regards Arnab

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kamailio mysql+ldap+radius authentication
by wanmon 08 Jun '16

08 Jun '16

Dear list, I have Kamailio 4.4.1 running with local mysql authentication (WITH_MYSQL/WITH_AUTH). i have RADIUS and LDAP services running. I would like to use all of them (MYSQL+RADIUS+LDAP) for Authentication. How is the best "kamailio.cfg" approach to use all defined users on RADIUS and LDAP? so, when a user try to login, kamailio will try MYSQL, them LDAP, them RADIUS. If user/password are not define on any of those 3 system, authentication will fail. Do i need modify my current LDAP tree? Im trying to using standart "userPassword" attribute. This is my reference: http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth i would like to "add" route[LDAPAUTH], not to "replace" route [AUTH]. Kind Regards

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A: README files for modules generated automatically
by Daniel-Constantin Mierla 08 Jun '16

08 Jun '16

Hello, following past discussions about different formats resulted when generating readme files for modules on various systems, which typically led to conflicts when backporting, I put in place a cron job running on kamailio.org to fetch from github the docbook files, generate the readme and push them back. The cron job is running every 15min. >From now on it is no longer needed to generate the readme yourself. Just commit the changes to xml files and push them to git repository. The readme will be refreshed automatically in a bit. The commits in this case are done by a generic git account named kamailio-dev. The commit message looks like: modules: readme files regenerated - app_mono ... Only the first module that has the readme updated is listed (in this example being app_mono). The readmes are automatically generated for master and 4.4 branches. If you do consistent changes to the docbook file, you can generate the readme locally to be sure there is no error, then restore the old version before committing - relevant commands: make modules-readme modules=modules/async rm modules/async/README git checkout modules/async/README Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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4.4 compiler warnings on debian jessie
by Juha Heinanen 08 Jun '16

08 Jun '16

I'll come back to mem/tlsf_malloc.c issue. Camille asked if the compiler warnings on mem/tlsf_malloc.c were occurring on x86 32 bit machine. I now tried and, yes, they do not appear on i386 jessie, but not on amd64 jessie. So looks like this is the same issue that Ovidiu has reported. -- Juha

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4.4 fails to start on debian jessie due to tls issue
by Juha Heinanen 08 Jun '16

08 Jun '16

Was this fixed on master? Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: : tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.1t 3 May 2016" (0x1000114f), compiled "OpenSSL 1.0.1k 8 Jan 2015" (0x100010bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: CRITICAL: <core> [main.c:2514]: main(): could not initialize tls, exiting... If so, please backport to 4.4 since it is now impossible to start kamailio on debian jessie. -- Juha

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ack with cscall-id header
by Rawad Khalil 07 Jun '16

07 Jun '16

Dear kamailio community, I have a problem that seems really weird. i have the same problem using kamailio 4.3 and 4.4.1. The scenario is the following: I make a call from a sip phone to kamailio. kamailio routes the Invite to an asterisk server. After a while asterisk sends a Bye request to kamailio. Now instead of routing the Bye to the sip phone, I want to create a re-invite request from kamailio and send it to the sip phone. The purpose of the re-invite is to connect the sip phone to another sip phone that already has an active call. This part is working fine but the problem occurs after the sip phone sends the ‘200 OK’ message for the re-invite. Kamailio sends an ACK for the 200 message but this ACK has a ‘CSCall-id’ header instead of the ‘Call-id’ header. To create the re-invite I am using the UAC module with one change to the source code. In the uac.c file found in the tm module folder, I added 2 lines of code to the request function in order to stop the UAC module from creating a from tag(attached). I also attached a trace that shows the ack message with cscall-id. Thank you, Rawad

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Re: [SR-Users] Regarding kamailio-4.4.0 msrp module
by Daniel-Constantin Mierla 07 Jun '16

07 Jun '16

Hello, what is missing? What is highlighted seems to be in the code. Can you provide a patch or make a pull request on github project? That will make it simpler to understand what you want to change. Cheers, Daniel On 07/06/16 11:51, Surendra.P wrote: > > Hello Daniel, > > > > Switch case control is missing while parsing MSRP > pseudo variables. Please find the code snippet below: > > > > int pv_parse_msrp_name(pv_spec_t *sp, str *in) > > { > > if(sp==NULL || in==NULL || in->len<=0) > > return -1; > > > > switch(in->len) > > { > > case 3: > > if(strncmp(in->s, "buf", 3)==0) > > sp->pvp.pvn.u.isname.name.n = 1; > > else goto error; > > break; > > case 4: > > if(strncmp(in->s, "body", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 2; > > else if(strncmp(in->s, "code", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 3; > > else if(strncmp(in->s, "hdrs", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 4; > > else goto error; > > break; > > case 5: > > if(strncmp(in->s, "msgid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 5; > > else if(strncmp(in->s, "conid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 21; > > else goto error; > > case 6: > > if(strncmp(in->s, "method", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 6; > > else if(strncmp(in->s, "buflen", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 7; > > else if(strncmp(in->s, "sessid", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 8; > > else if(strncmp(in->s, "reason", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 9; > > else if(strncmp(in->s, "crthop", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 12; > > else goto error; > > break; > > > > > > Regards > > Surendra > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Regarding kamailio-4.4.0 msrp module
by Surendra.P 07 Jun '16

07 Jun '16

Hello Daniel, Switch case control is missing while parsing MSRP pseudo variables. Please find the code snippet below: int pv_parse_msrp_name(pv_spec_t *sp, str *in) { if(sp==NULL || in==NULL || in->len<=0) return -1; switch(in->len) { case 3: if(strncmp(in->s, "buf", 3)==0) sp->pvp.pvn.u.isname.name.n = 1; else goto error; break; case 4: if(strncmp(in->s, "body", 4)==0) sp->pvp.pvn.u.isname.name.n = 2; else if(strncmp(in->s, "code", 4)==0) sp->pvp.pvn.u.isname.name.n = 3; else if(strncmp(in->s, "hdrs", 4)==0) sp->pvp.pvn.u.isname.name.n = 4; else goto error; break; case 5: if(strncmp(in->s, "msgid", 5)==0) sp->pvp.pvn.u.isname.name.n = 5; else if(strncmp(in->s, "conid", 5)==0) sp->pvp.pvn.u.isname.name.n = 21; else goto error; case 6: if(strncmp(in->s, "method", 6)==0) sp->pvp.pvn.u.isname.name.n = 6; else if(strncmp(in->s, "buflen", 6)==0) sp->pvp.pvn.u.isname.name.n = 7; else if(strncmp(in->s, "sessid", 6)==0) sp->pvp.pvn.u.isname.name.n = 8; else if(strncmp(in->s, "reason", 6)==0) sp->pvp.pvn.u.isname.name.n = 9; else if(strncmp(in->s, "crthop", 6)==0) sp->pvp.pvn.u.isname.name.n = 12; else goto error; break; Regards Surendra

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Admin Arnab

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, How to configure rtp proxy for IPv6 Only client,now we are not able to get the audio, Please help Regards Arnab

2 1

Kamailio Sorce code Documents
by aman jeff 07 Jun '16

07 Jun '16

Hi team, I like to know the source code documention of kamailio, I like to customize textops.c,like to get the variables and structure documents gparam_p, etc.. Regards Jeff

2 1

Yealink BLF
by Bill 06 Jun '16

06 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

2 1

RTPENGINE not kernalizing in some cases
by Dmitry 06 Jun '16

06 Jun '16

is it possible that RTPENGINE failes because 2 rtpengine_offer (2 invites during autharization) and the 183 Reply as I see in the log - is seen as an ANSWER Both INVITES are with SDP RTPENGINE_MANAGEThis call is not answered Received command 'offer' from 127.0.0.1:54490 Creating new call] offer time = 0.000134 sec Replying to 'offer' from 127.0.0.1:54490 Received command 'offer' from 127.0.0.1:53879 offer time = 0.000060 secReplying to 'offer' from 127.0.0.1:53879 Received command 'answer' from 127.0.0.1:40099 answer time = 0.000169 sec Replying to 'anser' from 127.0.0.1:40099 ICE negotiated: local interface 10.34.101.104Confirmed peer address as 10.34.101.104:7458Confirmed peer address as 10.34.101.103:62608

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 06 Jun '16

06 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 1

Wrong Publish status
by Aleš Šturm 06 Jun '16

06 Jun '16

Hello, Kamailio 4.3: I am testing Presence module with more complex scenario. I have 2 SIP clients (presentity) with equal identities (3918) and one watcher (2025). All my clients are Monster 0.9.25. My Kamailio Presence module is configured with "retrieve_order=1" (presentity records are retrieve by priority order). I have situation when wrong status is send to the watcher (2025). 1. First all three clients registers and publish its own status. Below is print from my MySQL database (presentity table) mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340687 | 1464337087 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 2. Then first 3918 client publishes his mood >11111<. The status in >Presentity< table is refreshed under the id=84480. NOTIFY is sent to 2025 with status with mood >11111<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 3. After that the second 3918 client publishes his mood >22222<. The status in >Presentity< table is refreshed under the right id=84481. NOTIFY is sent to 2025 with status with mood >22222<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 4. Then finally, the first 3918 changes his mood to "33333". The status in >Presentity< table is refreshed under the right id=84480. BUT, to 2025 is sent NOTIFY with mood "22222". "received_time" timestamp of status with mood "33333" is higher than timestamp of status with mood "22222". BUT the "id" attribute of "presentity" table of status with mood "33333" is smaller. I think that statuses with equal "priority" are retrieved by "id" and not by "received_time"!? mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340729 | 1464337129 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>33333</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | The Wireshark trace is attached. Can anyone help. Please? We have a big installation in the Gazprom's (Rusia) network, where Kamailio is used as a Presence server only. Thanks Ales

2 8

Yealink BLF
by Bill 05 Jun '16

05 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

1 0

rtpproxy ipv6 problem
by aman jeff 03 Jun '16

03 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 0

Kamailio 4.4.1 REGISTER Parse Error in PCSCF
by Mihail Dakov 02 Jun '16

02 Jun '16

Hi Folks, I have the strangest problem just after installing latest stable version. I had tried both ways using the repo (http://deb.kamailio.org/kamailio44) and downloading code with local compile (https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/) The initial scenario is to get M2M with commercial IP and subsequently use the Rx interface to do the same. We are blocked because we cannot even achieve clean register for the UEs. UE1(galaxy s4 zoiper) -------- Kamailio IMS aio | UE2(galaxy s4 zoiper) -------- | What happens is that the first register message triggers some quasi-endless loop in the PCSCF. Hence it cannot be forwarded to SCSCF, ICSCF and HSS. But this doesn't always happen and between all the parse problems there are some register messages which get passed to the scscf icscf and hss. Somehow the parser feeds itself from the previous parse run and thus loops quite intensively. The message I get is SIP/2.0 400 CSeq method does not match request method. I must say that reiterating the registration process sometimes i get the register and sometimes i don't. Has anyone encounter similar issue for similar use case and can share some thoughs about a fix? I attached relevant log file, pcaps and configuration files. Let me know if more config files are needed.

3 15

Nonce extract from SIP Meesage
by Sergio Serrano 02 Jun '16

02 Jun '16

Hi all, I just try integrate Kamailio with external software and We want taht kamailio doesn't authenticate users. I want that the autehtication process would be done by external software. I want To receive a REGISTER for a user. Then I send a 407 and then when I receive all paramters for autehtincation(nonce, response, digestURI, etc, etc etc) send that data to external software and receive if data is correct, then sent 200 OK and save location in external software. Bu t I can see any pseudovariable or enything else to obtain that dato to provide to external software. Anyone could think that this process could be done? Regards, Sergio

3 2

NSQ libevbuffsock dependency
by smititelu 02 Jun '16

02 Jun '16

Hi, Upon trying to compile latest Kamailio from sources I got an: " nsq.h:9:24: fatal error: evbuffsock.h: No such file or directory #include <evbuffsock.h> ^ " Looking on the new NSQ module documentation I see the libevbuffsock dependency. Is there other recommended way, to install the library besides compiling from sources at [1]? Thx, Stefan [1] https://github.com/mreiferson/libevbuffsock

2 2

sip trace
by Jeremy Betts 01 Jun '16

01 Jun '16

I have a new carrier that wants sip traces of certain scenarios. They also require IPSEC tunnelling for all SIP messages. So when I use tcpdump or any other similar tool I can see the incoming SIP messages but outgoing SIP messages are ESP encoded. I've played with the siptace module but can't seem to get it to log replies. Does anyone have any solutions?

2 2

Dynamic advertised ip for rtpengine/rtpproxy
by Serge Yuriev 01 Jun '16

01 Jun '16

Hi, We have several exits from our network and all of them natted. Could we set adv ip for rtpproxy/rtpengine from db in realtime or we must define them on daemon run time only? -- Wbr, Serge via mobile -- Serge via mobile

4 3

Send Register from Kamailio
by Grant Bagdasarian 01 Jun '16

01 Jun '16

Hello, Which module do I need to have Kamailio send REGISTER messages to our carrier? Is it the UAC module or REGISTRAR module? It just needs to register with our carrier in order to receive inbound calls. Regards, Grant

2 4

advertise xxx.xxx.xxx.xxx:5060 is weird
by Sébastien Brice 01 Jun '16

01 Jun '16

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to this email.) I want to force the signalling and sessions through this local proxy (some kind of outbound proxy) I have jitsi and xlite behind (ip 192.168.3.39 and 192.168.3.16) that tries to call themselves. user agents keep retransmitting 200 OK because they never get the ACK following the 200 OK pickup. If i remove the advertise THEPUBLICIPOFMYBOX of the kamailio.cfg the three handshake is working ok and the call is no longer dropped after a few second! i have a true bidirectionnal session and that's working ok. The bad thing is RTP flows straight through router (thanks to rtpproxy i guess) but i want incoming and outgoing RTP traffic to go on central local node 192.168.3.97 How can i achieve that? thanks. PS: i am not using path header because i am not skilled enough and kamailio is really hard horse to train

2 5

DHCP renewal conflicts with kamailio?
by Joonas Harjumäki 01 Jun '16

01 Jun '16

Hi! The attachment contains a log trace of kamailio 3.3.2. There is some errors in the log and I wonder if it is normal operation? The errors appeared in the middle of DHCP lease renewal and I wonder if the configuration (of dhclient/kamailio) could be improved to prevent such errors? Or to make sure kamailio retries udp_send sufficiently many times. The kamailio operated normally before renewal procedure began and after the procedure finished. Joonas Harjumäki

2 1

how to set kamctl to find ctl anf fifo that are not at /var/run
by Uri Shacked 01 Jun '16

01 Jun '16

Hi, I am using centos 7. By default on centos 7, the var/run file system is tmpfs. So, when kamailio runs with user and group other than root, it can not create the fifo file. I set the kamailio service and config files to work on usr/local/var/run/kamailio. Now, how do i set the kamctl to work with the fifo on /usr/local/var/run/kamailio ? Uri

3 3

Want dialog.default_timeout NOT to reset on subseq. req
by Andrey Utkin 01 Jun '16

01 Jun '16

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is processed." So this option works for us when the useragents don't send sequential UPDATEs or reINVITEs. I am not sure if it is adequate to ask for this feature in dialog module. I am aware of possibility to terminate any call by management command, and of cnxcc module which has call duration control. I would love to find something that simple - a single option for max call duration, without external mechanism (cronjob etc.) or full-blown billing. Any suggestions are very welcome. Thanks in advance. Please reply to all recipients, I am not subscribed to this maillist.

2 1

513 Message too big for VIDEO calls
by Varghese Paul 31 May '16

31 May '16

Hi all, I am trying to setup WSS on opensips-2.2. I am getting "513 Message too big " when i am trying to make video calls. Audio calls and Registration is working fine.I tried to change the message_chunk_size in WSS module but still i am getting the message. Client : SIPML5 and JSSIP Opensip version : 2.2 I am attaching the sipml5 console logs. Please let me know is there any settings i can apply to resole this. Thanks in Advance Varghese Paul

2 1

Extending has_body() with additional mime types
by Giacomo Vacca 31 May '16

31 May '16

Hi all, apologies if this is an old topic but I've noticed that has_body() can provide false positives and would like to discuss a solution. As a practical example, has_body("application/vq-rtcpxr") was returning 1 for Content-Type "application/dialog-info+xml". The interpretation is that they are both unknown and so they match. Using 'if ($cT == "application/vq-rtcpxr)" provides a work around, but I wonder what's the suggested solution for this? a. Add the missing mime types (from RFC 6035 and 4235 respectively) to the list b. Instead of returning 1, return another code, e.g. 2, when the mime type is unknown c. Switch to string comparison when the mime type is unknown d. Something else? Thanks, Giacomo

2 1

Accept INVITE with transport=tcp
by Nahum Nir 31 May '16

31 May '16

Hi, I have a client that sends INVITE with transport=tcp. How can I make Kamailio to work with that? Thanks and regards, Nahum

2 1

Issue with LCR an failover routers
by Alberto Sagredo 30 May '16

30 May '16

Hi Im using LCR module and trying to handle 503/408 errors on provider to route to a second one. Actually im using if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Gateways no disponibles"); sl_send_reply("500", "Server Internal Error - No gateway"); exit; } else { xlog("GW Seleccionado '$avp(i:709)'\n"); xlog("Dominio de destino: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("No hay GW de Backup"); sl_send_reply("503", "Service not available - No gateway"); exit; } else { xlog("Enviado por primer GW\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); t_on_failure("FAILBACK"); } failure_route[FAILBACK] { if (t_is_canceled()) { exit; } if (t_check_status("408|50[34]")) { route(LCR_OUT_OP2); } } And using for OP2 same routing block with id=3... This second providers use AUTH so using: t_on_failure("TRUNKAUTH"); failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { if (to_gw("3")) { xlog("Autenticación "); $avp(auser) = "user"; $avp(apass) = "pass"; } uac_auth(); t_relay(); exit; } But getting this error: May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [t_fwd.c:1704]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [tm.c:1466]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Any idea how to resolve that?

2 2

converting integer to string
by Henry Fernandes 30 May '16

30 May '16

In Kamailio 4.2, I’m having problems converting an AVP integer value into a string. My Kamailio config stores the return code from auth_check() and outputs it. $avp(auth_code) = auth_check("$fd", "subscriber", "1"); xlog("L_NOTICE", “response = $avp(auth_code)”); When I check the AVP against an integer, it behaves normally. However, I cannot convert it to a string. The return value -3 (invalid user) gets converted to the string “18446744073709551613” when outputting it. My hunch is that Kamailio is treating the AVP as an unsigned integer (rather than a signed integer) when it converts it to a string. If I try this with an AVP with a positive integer, it works as expected. Can someone tell me how to convert a negative integer into a string in Kamailio? -H

3 3

Variable in dialog event routes
by Minh Phan 30 May '16

30 May '16

Hi List, I'm trying to have an external application to write CDR to a database by doing this: event_route[dialog:start] { evapi_relay("{'cdr_id':* $dlg_var(cdr_id),* 'event': 'answered'}"); } event_route[dialog:end] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'ended'}"); } event_route[dialog:failed] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'failed'}"); } $dlg_var(cdr_id) is set inside a request route. However, in these event routes, its value is always null. Its seems that $dlg_var variables are not available in those event routes. What are the variables available in dialog event route? Any suggestion on how to achieve what I'm trying to do? Best regards, Minh

2 3

http_async_client segfault
by Jason Penton 30 May '16

30 May '16

Hi, Has anyone experienced segfault using http_async_query from http_async_client module? Here is BT: #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 53 iofgets_u.c: No such file or directory. (gdb) bt #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 #1 0x00007fca68f08afa in get_contents (stream=<optimized out>, len=1032, linebuf=0x7fca23ffe660 "127.0.0.1") at nss_files/files-XXX.c:201 #2 internal_getent (result=0x7fca23ffe5c0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, af=af@entry=0, flags=flags@entry=0) at nss_files/files-XXX.c:246 #3 0x00007fca68f09ca3 in _nss_files_gethostbyname4_r (name=name@entry=0x7fca2e790fa0 "localhost", pat=pat@entry=0x7fca23ffebe0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, ttlp=ttlp@entry=0x0) at nss_files/files-hosts.c:402 #4 0x00007fca79eb7757 in gaih_inet (name=<optimized out>, name@entry=0x7fca2e790fa0 "localhost", service=<optimized out>, req=req@entry=0x7fca341c7490, pai=pai@entry=0x7fca23ffed10, naddrs=naddrs@entry=0x7fca23ffed00) at ../sysdeps/posix/getaddrinfo.c:850 #5 0x00007fca79ebae0d in __GI_getaddrinfo (name=0x7fca2e790fa0 "localhost", service=0x7fca23ffeec0 "18000", hints=0x7fca341c7490, pai=0x7fca23ffee78) at ../sysdeps/posix/getaddrinfo.c:2406 #6 0x00007fca6e0f80f4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #7 0x00007fca6e104fa4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #8 0x00007fca6e10284b in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #9 0x00007fca723c9182 in start_thread (arg=0x7fca23fff700) at pthread_create.c:312 #10 0x00007fca79ee547d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 Camille, are you maintaining this module now?

2 2

Multiple async operations in single transaction fail (using http_async_client)
by Thom Seddon 30 May '16

30 May '16

Hi All, I've been doing some work this week with the new http_async_client module, it seems that with it's default behaviour it's not possible to execute another async query in the block immediately following the response, for example the following config: request_route { if (!t_newtran()) { sl_reply_error(); exit(); } http_async_query("http://localhost:8090/req_1", '{"user":"$fU"}', "REQ_1"); } route[REQ_1] { xlog("L_INFO", "route[REQ_1]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_1]: status $http_rs\n"); xlog("L_INFO", "route[REQ_1]: Sending REQ_2\n"); http_async_query("http://localhost:8090/req_2", '{"user":"$fU"}', "REQ_2"); } else { xlog("L_INFO", "route[REQ_1]: error $http_err)\n"); } } route[REQ_2] { xlog("L_INFO", "route[REQ_2]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_2]: status $http_rs\n"); } else { xlog("L_INFO", "route[REQ_2]: error $http_err)\n"); } } Generates the following error: 10(11832) INFO: <script>: route[REQ_1]: INIT 10(11832) INFO: <script>: route[REQ_1]: status 200 10(11832) INFO: <script>: route[REQ_1]: Sending REQ_2 10(11832) WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [53195:560306836] Looking at the source, http_async_client calls t_suspend, then goes away and performs the request in the background. On response it calls t_continue and then this block <https://github.com/kamailio/kamailio/blob/4.4/modules/tm/t_suspend.c#L270-L…> in t_continue will continue execution, the second query will cause http_async_client to call t_suspend whilst sill executing from the previous t_continue. Once they query is sent, the first t_continue will complete, removing the T_ASYNC_SUSPENDED flag. When the second query returns it doesn't find the flag and so fails as shown. I'm far from an expect on this so would really appreciate some pointers, it seems to fundamental to be a race condition in tm but the usage in http_async_client seems to match usage in other modules so I'm not sure on the best tactic. I've tried peppering append_branch and async_route call's in various places to no avail. Full debug log here: http://pastebin.com/raw/rFmm0fmf Any help appreciated, *Thom Seddon*

2 1

username of Contact header
by Al S 27 May '16

27 May '16

Hi, I am trying to read username portion of Contact header: My Contact header content has a URI value such as: sip:8131110000@10.10.10.10.:5060 and I am trying to read : 8131110000 I tried the following perl similar RE and it didn't work: ($var(main_number2)) = $ct =~ /sip:(.*)@/; Thanks,AS

2 1

4.4: Kamailio crashes at start - sometimes
by Sebastian Damm 27 May '16

27 May '16

Hi, we have a machine running 16 Kamailio instances, and while upgrading to 4.4.1 (from 4.3.5), 8 of them wouldn't start. When downgrading to 4.3.5, they all start again. All of them have pretty identical configuration files, except IPs, ports and some code enabled or disabled via defines. After comparing a working and non-working configuration and adjusting setting by setting, we finally ended up with a working configuration. The difference is, that it won't start when a part of the code DOES NOT get included. If it gets included, it will start. This is the mentioned part of our main route: #!ifdef ENABLE_INV_RATELIMIT # Check for INVITE limit if (is_method("INVITE") && $au == $null && !($ua =~ "sipgate") ) { $var(invcount) = $shtcn(invcount=>%~$fU); xlog("L_INFO", "INVITE Requests from $fU in last 30 seconds: $var(invcount)\n"); if ($var(invcount) < 12) { $var(uniqcid) = $ci + $Ts + $ft; $var(tkey) = $fU + '-' + $(var(uniqcid){s.md5}{s.substr,0,10}); $sht(invcount=>$(var(tkey))) = 1; $var(uniqcid) = $null; $var(tkey) = $null; } if ($var(invcount) > 10) { if ($var(invcount) == 11 ) { xlog("L_NOTICE", "User $fU ($var(domain2use)) over ratelimit for new calls, rejecting.\n"); } # Enable this only after evaluating the impact! append_to_reply("Retry-After: 30\r\n"); sl_send_reply("503", "Call Rate Limit Exceeded"); exit; } } #!endif If we put this line at the top of the configuration file, everything works: #!define ENABLE_INV_RATELIMIT If we delete this line, startup does not work. It just sits in ps for one minute without forking, and then gets terminated. We enabled a bit of debugging, and this is apparently the error causing Kamailio to shutdown: May 25 14:50:15 kammel /usr/sbin/kamailio[24989]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 53: nathelper May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0x9f0428 and 0xa34428 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9f0240, 82, 1, (nil)), fd_no=0 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Bad file descriptor [9] May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1747]: tcp_receive_loop(): failed to add tcp main socket to the fd list May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1815]: tcp_receive_loop(): exiting... I have no idea, how this part of the code could lead to this error, but it is reproducable, that at least on this system setting or disabling this define fixes or breaks the startup. Does anybody have an idea, what's happening there? Best Regards, Sebastian

2 4

Multihomed kamailio - one IP for each account
by Attila Megyeri 27 May '16

27 May '16

Hi Folks, I am looking for a solution to do the following, basically an SBC-like solution. I have an internal asterisk system that should make outbound calls towards a provider, but that provider authenticates the customer based on IP address. And I have many such customers. The provider is not willing to make changes to his practice... Asterisk is bound to a single IP, so there is not much I can do. I was thinking about creating several outbound accounts for asterisk, and then a multi-homed Kamailio would do the following: If request comes from user1, Kamailio would send the request to the provider from IP1. If request comes from user2, Kamailio would send the request to the provider from IP2. And so on, for as many IPs as many customers I want to handle. Is there a more elegant way to do this? If not, how should I implement this? Thanks in advance, Attila

2 4

Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is not so much due to the number of entered Via: records but, by the fact that they contain a branch parameter extremely long. Here the call flow example from CARRIER A —> Kamailio1 —> Kamailio2 —> CARRIER B Extract from SIP message sent from CARRIER A to K1 INVITE sip:999912341234@192.168.158.42 SIP/2.0. Via: SIP/2.0/UDP 192.168.220.141:5060;branch=z9hG4bK49e4301c;rport. Extract from SIP message sent from K1 to K2 INVITE sip:999912341234@192.168.127.244 SIP/2.0. Record-Route: <sip:192.168.158.42;lr;did=194.a0f1;nat=yes>. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bK0635.7c1e97c2d5c01ea98e8d0e7fa23a3822.0. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.cuOBV4OBVLMGZfWxF-W.y6Mx1LgRWIC9gIgx4fzxj7MBP7MBVAOBF4M.1jmxuqC93X3heroEWvH9vsCFN43qd4zRj4Mlyf3l1LNSQLpx4uMx3A. Extract from SIP message sent from K2 to CARRIER B INVITE sip:999912341234@192.168.65.248 SIP/2.0. Record-Route: <sip:192.168.127.244;lr;did=194.5122;nat=yes>. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bK0635.f843472880b380acde3a33732975fde9.0. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1LNSQLpx4uMx3AzuaVHRaYpB1JNEt736cQkBIvalaJmly6Mlj7W6Mfg.qw3leqWRMAMRKrz.rIzSPAg.pE3.Vl3.MZMBV7My**. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1JNEt736cQkBIvalaJmwWLORv4mK0Hot3w.jpam6t4kRWWOEWu.4eRWFQGKqfWauYEKwrSOKN7k.gWkxtMgue9mjMsg4IhkxaEkhrbW4uGjxpWPIg0.4eSWF47MRY1kUKfPlqlRxWvg9e5aKp6mxI6N4WS.BPlNRP4kIebWFudkR3loxtM.IWa.U0ZWUKWkxW0. Is there anyone here with the idea how to solve this problem ? Best regards

3 8

rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp udp flow between the phones isn't routed properly by the rtpproxies, ending in the communication drop (all the SIP nego is working well, and the SDP are correctly patched with the rtp proxy address and port). Analyzing the RTP proxy packets, we have found that the Kamailio registrar gives the Kamailio proxy ip address in the RTP proxy create session command, but keeps the original sdp port. command looks like this: Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet> PZU5OITCW;1 We are using rtpproxy_manage() without any flags. It seems to us that this ip and port are used as default forward route as long as the callee hasnt connected to the rtpproxy. *Is it correct? * If its true, Its seems to us that this cant work as we are mixing the proxy sip address with a udp port open on the phone ? *Is our analysis correct ?* Can we use some option in rtpproxy_manage to replace the proxy ip by the phone ip as seen in the via route ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.h… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC>

2 2

webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen.

2 3

SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/sdp Content-Disposition: session Content-Length: 1393 v=0 o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP> s=SBC c=IN IP4 <MyPublicIP> t=0 0 m=audio 30454 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30455 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:iglHZqry a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host m=audio 30484 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30485 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:gM1viWqu a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host My setup : incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy (Kamailio + Rtpengine) -> webRTC clients Is there anybody who could help me with this issue or explain me how to remove the second line of m=audio RTP/SAVPF ? -- Jan

1 0

Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

4 5

nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/rtpengine_conf" loadmodule "db_text.so" modparam("rtpengine", "db_url", DBURLRTP) modparam("rtpengine", "table_name", "rtpproxy") and the database is like root@test:/var/log/rtpengine_conf# ls rtpproxy version root@test:/var/log/rtpengine_conf# cat version table_name(str) table_version(int) rtpproxy:1 root@test:/var/log/rtpengine_conf# cat rtpproxy set_id(int) url(str) weight(int) disable(int) 1:udp\:x.x.x.x\:x:100:0 2:udp\:x.x.x.x\:x:100:0 is there something i missed here? do i must use mysql or postgres drivers for this? (i use a test machine to old to install a driver of those with kam 4.4 maybe?) best regards david

1 1

doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i have a quick question about using module ratelimit on kamailio 4.2 > or 4.4. > on tests made before running it on production, i set the RED > algorithm with a limit of 10 calls per second, but i saw the call > limitation was not done until we got more than 20 calls per second or > so on the queue. we didnt get any INVITE discarded. > after surpassing around 20 INV/sec, it's true the algorithm starts to > work fine and only 10INV/sec are passed more or less > > on production i have no easy way to check this behaviour, besides it's > a feature barely used. > > so i only want to know if this behaviour is known, or if i'm missing > soemthing > > my config is like this > > modparam("ratelimit", "timer_interval", 5) > modparam("ratelimit", "queue", "4:INVITE") > modparam("ratelimit", "pipe", "4:RED:10") > > route[RATE_LIMIT] { > $var(limitation)="4"; > if (!rl_check("$var(limitation)")) { > send_reply("505","Limiting"); > xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, > rejecting with 505\n"); > route(CLEAR); > exit; > } > } > > > thanks a lot and regards > david escartin > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - | CB1 |-----------------| SIPPHONE | ---------- | -------------- | -------- ---------------- | | ---------- | | -------- | SBC2 | - - - - - | | - - - - - | CB2 | ---------- | | -------- | -------------- | ---------- | - - - -| kamailio2 |- - - - -| -------- | SBCn |- - - - - | -------------- | - - - - - | CBn | ---------- | | -------- Kamailio high availabality will be done with loopback address and BGP in two datacenters. SIP carriers will be connected to SBC and customer to one CB. I have made some tests with kamailio, and i use LCR to route call from CB to SBC, using wheigt and prorities to send call to carriers. Then, when a call arrived from carriers, i use DNS NATPER lookup to find the right CB box. CB HA is done using DNS SRV. So user is registred at one location, and only one. No location sharing beetwen boxes. I have some question concerning my architecture, and technologies used to routing call. Is LCR is the best option for routing call between multiple carriers ? DNS NAPTER is a modern solution to find an endpoint on this architecture ? Existing a better way to route call with kamailio, for exemple rt-json and curl ? In my point of view, kamailio should be a routing engine, using internal technologies, to maximise performances. Less work with kamailio and external app, best performances in sip routing. Is it the good vision ? Thanks in advance.

2 4

Kamailio routing to different carriers based on prefix
by José Seabra 25 May '16

25 May '16

Hi there, I'm looking for a solution to route calls based on prefix, my routing table might reach a long list of prefixs. I looked to carrierroute and it seems to be nice for this purpose but i have some doubts to how to handle correctly with carrierroute. IfI understood correctly, if I set the parameter avoid_failed_destinations to 1, can i use the same carrier id and domain on failure route to select another destination that matches the prefix? Can I have the same carrier ID with diferent domains sharing the same prefixs but mapping it to different destinations? Thank you for your help. Best regards -- José Seabra

2 1

call count / and work with despatcher
by jaflong jaflong 24 May '16

24 May '16

Hi Everyone, What is a accurate way to get current call count. I am only relaying calls. Initialy I tried the code below. It seemed to work, it reported the correct number of calls I had dialed. I get the result through kamcmd htable get command. ------------------------------------------ under route[RELAY] # # COUNT # if (is_method("BYE")) { $var(n) = $shtdec(callcount=>active); sht_rm_name_re("callcount=>$ft"); } -------------------------------------------- # Manage incoming replies onreply_route[MANAGE_REPLY] { # # COUNT # if (status=="200") { if (is_method("INVITE")) { # On start if($sht(callcount=>active) == $null) { $sht(callcount=>active) = 0; } # New call if ($sht(callcount=>$ft::from_tag)== $null) { $sht(callcount=>$ft::from_tag) = $ft; $var(n) = $shtinc(callcount=>active); } } xlog("L_INFO", "This is a 200 OK reply\n"); } xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } ------------------------------------------------------- However I wanted to distribute calls to several servers I further added the despatcher code below # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "-----RELAYING going to <$ru> via <$du> time [$Tf] -----\n"); This makes my counting code above give incorrect call count. How can I make call count work with dispatcher ? Do advise how I can do this Regards

2 1

Create "Contac Groups" on the server Kamailio
by ALFONSO MARTINEZ GOMEZ 24 May '16

24 May '16

Hi, I'm working on a VoIP communication application with Kamailio as SIP server and using the SIREMIS interface. As an administrator, I can add "Subscribers" which later will be registered on the server so that they can make the necessary negotiations when they want to make a call. My question is, how you can make groups of "contacts" so that such calls can be made only with each other "contacts" in the same group? that is, we can have many "subscribers" on the server and belong to two different groups and that they can only talk to each other contact each group. Thank you so much and greetings!

1 0

doubt using rate limit
by david escartin 24 May '16

24 May '16

Hello all i have a quick question about using module ratelimit on kamailio 4.2 or 4.4. on tests made before running it on production, i set the RED algorithm with a limit of 10 calls per second, but i saw the call limitation was not done until we got more than 20 calls per second or so on the queue. we didnt get any INVITE discarded. after surpassing around 20 INV/sec, it's true the algorithm starts to work fine and only 10INV/sec are passed more or less on production i have no easy way to check this behaviour, besides it's a feature barely used. so i only want to know if this behaviour is known, or if i'm missing soemthing my config is like this modparam("ratelimit", "timer_interval", 5) modparam("ratelimit", "queue", "4:INVITE") modparam("ratelimit", "pipe", "4:RED:10") route[RATE_LIMIT] { $var(limitation)="4"; if (!rl_check("$var(limitation)")) { send_reply("505","Limiting"); xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, rejecting with 505\n"); route(CLEAR); exit; } } thanks a lot and regards david escartin

2 1

Please help me setting up Kamailio for android application
by Kunal Dhanda 24 May '16

24 May '16

I wanted to create my own voip application using kamailio . I am not able to find any rest integration with kamailio for my server to use for user registration and other task.Please help in developing my appplication. -- Thanks & Regards, Kunal Dhanda, Software Developer E-mail : < "kunaliitkgp09(a)gmail.com" >, Ph. No. (M) : +91 - 8884818805

2 1

Found bug at ACC
by Pavel Eremin 24 May '16

24 May '16

Hi all, when we set syslog facility for ACC to LOCAL3, log_write_cdr() will write at LOG_DAEMON anyway. You have to recompile module acc to user another default syslog facility to save cdr with another syslog facility.

2 1

kamctl stats shmem error changing shared memory
by Mr.Gab 23 May '16

23 May '16

Hi, i have a problem with my kamailio installation. Actually i set the shared memory to 128MB and it's all fine. When i increase the shared memory to 256Mb and more i get the following error while checking kamctl commands (like kamctl stats shmem) ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. If i revert to 128Mb kamailio is able to create the kamailio_fifo file and all works fine. Can somebody help me to solve? Thanks

3 3

Re: [SR-Users] Change CANCEL reason when fr_timer expires
by Bruno Emer 22 May '16

22 May '16

Hello, Alex! Fisrt of all, thanks for you help with this problem! I don't think that this will help me. I can see that this parameter is enabled by default and in fact it is working in some cases. One case where this works is when the INVITE is sended for more than one endpoint (when the same user is registered in many devices). When someone answers the INVITE, Kamailio sends a CANCEL message to the other endpoints and it has the Reason header showing that the call was answered elsewhere. I want to add the CANCEL Reason header when the timer expires, so kamilio will not receive a final replay to that transaction. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-20 15:42 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a situation here where when I send a INVITE to some specific > user-agents in my environment I must change the fr_timer to a value lower > than the default. > > I can acomplish this using the "t_set_fr" commands in my cfg file, and it > works like a charm. > > The point is that when the timer reaches its limit, the kamailio server > sends a CANCEL message to the user-agent. I know that this is the normal > behavior, but I want to add the Reason field on it, so the user-agent > application can know that the call wasn't answered on time. > > I was looking the TM module documentation to understand how to do this, > but I couldn't figure out a way to get this working. > > Is there a way to add the Reason in CANCEL messages generated by the > fr_timer? > > > Thanks. > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >

2 1

Change CANCEL reason when fr_timer expires
by Bruno Emer 21 May '16

21 May '16

Hello all! I have a situation here where when I send a INVITE to some specific user-agents in my environment I must change the fr_timer to a value lower than the default. I can acomplish this using the "t_set_fr" commands in my cfg file, and it works like a charm. The point is that when the timer reaches its limit, the kamailio server sends a CANCEL message to the user-agent. I know that this is the normal behavior, but I want to add the Reason field on it, so the user-agent application can know that the call wasn't answered on time. I was looking the TM module documentation to understand how to do this, but I couldn't figure out a way to get this working. Is there a way to add the Reason in CANCEL messages generated by the fr_timer? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 1

extra characters in re-invite "TO" field
by Al S 20 May '16

20 May '16

Hi, Kamailio adds extra characters in the re-invite "TO" field during session timer. There is no header manipulation in our route[WITHINDLG] block. Re-INVITE received by KamailioTo: <sip:8131111111@?.?.?.?;otg=1>;tag=1c1048632956 Re-INVITE forwarded by KamailioTo: <sip:8132222222@?.?.?.?;T\033\023Z\f1>;tag=1c1048632956 Your help would be greatly appreciated. Thanks,AS

2 2

Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to make such setup that calls without the RTPENGINE are impossible.

1 0

WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

4 12

Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 pejman(a)italkip.com

2 1

Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

2 3

[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne

4 3

Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 <mailto:pejman@italkip.com> pejman(a)italkip.com

1 0

invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

2 2

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> >

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Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config file, same results. Thanks in advance. Regards.

2 3

MySQL database for kamailio as S-CSCF
by Christoph Valentin 17 May '16

17 May '16

Hi Community, I'm doing very first steps to use kamailio as an S-CSCF. Now I'm confused about some defines in the examples/scscf/scscf.cfg file. #!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" Tutorials/ims/Installation-howto tell me only I-CSCF needs a database, I do not find anything about database for S-CSCF. However, when I comment out the defines ##!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" then the kamctl start results in a segmentation fault. I have installed kamailio-4.4.0 from RPMs. Is there any hint about database for S-CSCF? Thanks in advance Christoph

1 1

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by M R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Debian, dnssec.so, and missing dnsval.conf
by David Holl 17 May '16

17 May '16

In the interest of sharing and promoting DNSSEC, I was recently stumped on a problem where I noticed I could not call sip:test.echo@sip5060.net without disabling the dnssec.so module in my kamailio.cfg. The last time I had tried the sip5060 echo test was several months ago, and in the intervening time, I had applied the numerous debian package updates. Thus, I could not isolate the cause of the problem to one of the many package updates such as kamailio 4.4.x, libval, my own DNSSEC dns server, etc... My initial only clue was kamailio's syslog messages: May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve "sip5060.net" :unresolvable A or AAAA request (-7) May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) To summarize the problem, I found that kamailio's dnssec.so was failing queries to domains that were not DNSSEC protected such as sip5060.net. This behavior is a bug, since sip5060.net is in this case, provably not participating in DNSSEC from the toplevel "net." zone [1]. The fix was to provide a default configuration file to libval, which was somehow not included in Debian's libval15 package. I created the following configuration from the default dnsval.conf in the source distribution: https://dnssec-tools.org/download/dnsval-2.2.tar.gz https://dnssec-tools.org/download/dnsval-2.2.tar.gz.asc as well as their wiki: https://www.dnssec-tools.org/wiki/index.php/Dnsval.conf cat << 'EOF' > /etc/dnssec-tools/dnsval.conf ################################## # Global Options ################################## global-options trust-oob-answers yes # I'm running a DNSSEC nameserver locally, # so 4 KiB queries to localhost should be OK: edns0-size 4096 env-policy disable app-policy disable rec-fallback no ; : trust-anchor . DS 19036 8 2 49AAC11D7B6F6446702E54A1607371607A1A41855200FD2CE1CDDE32F24E8FB5 ; : zone-security-expectation . validate ; : provably-insecure-status . trusted ; : clock-skew . 0 ; EOF In summary: I find it odd that a default dnsval.conf was not included with my system's normal installation of the "libval15:amd64" package, version 2.2-4. After providing dnsval.conf, kamailio's dnssec.so was happy again. Side notes: * To identify the directory for dnsval.conf, run libval-config --dnsval-config On Debian, libval-config is in the libval-dev pacakge, and this package includes a man page for dnsval.conf as well. * To test your libval and dnsval.conf, try dt-validate -s On Debian, dt-validate is in the libval-bin package. - David [1] -- Authenticated denial of existence: https://tools.ietf.org/html/rfc5155 https://tools.ietf.org/html/rfc7129 And for sip5060.net, there's an NSEC3 record in the top level net. zone proving non-existance of sip5060.net DS records: http://dnsviz.net/d/sip5060.net/dnssec/ Translation: sip5060.net provably not enabled DNSEC.

2 1

kamailio not authenticate a subscribed username
by Al S 16 May '16

16 May '16

Hi,1) A generic subscriber is added to kamailio database:kamctl add pbx password2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call:UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it)The correct password is being sent to kamailio.Your assistance is greatly appreciatedThanks,AR

1 0

kamailio not authenticate a call with subscribed username
by Al S 16 May '16

16 May '16

Hi, 1) A generic subscriber is added to kamailio database:kamctl add pbx password 2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call: UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it) The correct password is being sent to kamailio. Your assistance is greatly appreciated Thanks,AR

1 0

Re: [SR-Users] Another question about RTPengine module
by Dmitry 16 May '16

16 May '16

the problem is: rtpengine was compiled with USE_SCTP,but kamailio that was installed via RPM is without this flag. How to tell compiler NOT TO " USE USE_SCTP" flag? +++++++++++++++++++++++ Makefile.defs:SCTP ?= 0 - and it compiles without sctp. Now kamailio.log:ERROR: rtpengine [rtpengine.c:1113]: child_init(): Name or service not knownERROR: <core> [sr_module.c:929]: init_mod_child(): Error while initializing module rtpengine (/usr/lib64/kamailio/modules/rtpengine.so) View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users On Sunday, May 15, 2016 6:21 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: yes, I wrote that modparam are recognized.MODPARAM must be placed i the sae place where the RTPPROXY resides in the config.+++++++++++++++++++++++++++++++ Another question:kazoo kamailio KAMAILIO-V :4.3.5:kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 To add the RTPENGINE - I found Kamailio 4.3.5 sources https://www.kamailio.org/pub/kamailio/4.3.5/src/ When I start kamailio I get the following error:ERROR: <core> [sr_module.c:431]: version_control(): module compile flags mismatch for /usr/lib64/kamailio/modules/rtpengine.so I did the following to add missing modules: #make include_modules="rtpengine topoh" cfg #make all cp /usr/src/kamailio-4.3.5/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so How to fix it? The sources version seems to be the same On Saturday, May 14, 2016 1:21 AM, ycaner <yasin.caner(a)netgsm.com.tr> wrote: Hello; i am not sure. i downloaded from git kamailio. add rtpengine instead of rtpproxy . it works fine. it is hard to solve problem without log. Cheers. try to run with kamailio -E -dd , maybe it gives information. check kamailio.cfg is that rtpengine is loaded like loadmodule "rtpengine.so" 2016-05-13 11:35 tarihinde Dmitry [via SIP Router] yazdı: one more question about rtpengine. I installed Kamailio (centos 6.7) from source as You recommended make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg no rtpengine daemon (it is not installed) but the picture is the same: modparam ("rtpengine,....) kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(19523) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found it means that it does not depend upon rtpengine daemon. # single rtproxy modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") I copied it from http://www.kamailio.net/docs/modules/4.2.x/modules/rtpengine.html STRACE kamctl start -says nothing anout rtpengine What else is it possible to do to find the cause of the problem? On Friday, May 13, 2016 11:45 AM, ycaner <[hidden email]> wrote: Hello; rtpengine module isnt default. you need to add with "make include_modules". after make && make install , automatically it will be copied to lib/modules. Cheers. 2016-05-13 09:42 tarihinde Dmitry [via SIP Router] yazdı: And one more question on it: If the Kamailio was installed from RPM and rtpengine.so is missing - May I add rtpengine.so the following way: make include_modules="rtpengine" cfg make all and then copy rtpengine.so to /modules/ directory Is this way (adding missing modules) is correct? On Friday, May 13, 2016 11:25 AM, Dmitry <[hidden email]> wrote: hello I tested on Centos 6.7 as well as Ubuntu server - the problem is the same. RTPENGINE was installed both from dpkg and from Source (manual compilation) So I need a clarification on it. RTPENGINE daemon is started. The Kamailio module itself is taken from /usr/src/kamailio, where the source was compiled. mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so sudo /etc/init.d/kamailio restart | grep rtp ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <rtpengine> found version: kamailio 5.0.0-dev4 (x86_64/linux) Any help is appreciated. On Thursday, May 12, 2016 6:04 PM, Dmitry <[hidden email]> wrote: Hi, I installed Kamailio from GIT. I did make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg Actually loadmodule "rtpengine.so" does not affect kamailio start/restart. modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") causes this failure: [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found On Thursday, May 12, 2016 12:26 PM, ycaner <[hidden email]> wrote: Hello; on bash , kamailio -V gives Version , kamailio -c gives debug result. you can ask from here https://groups.google.com/forum/#!forum/2600hz-dev for kazoo rpm. if there is something wrong , they will fix it. you can install from source code from here => https://www.kamailio.org/w/ and "make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg " in kamailio folder before 'make all' , helps you add modules. if there is a dependecies is missing , gives error. Cheers. 2016-05-11 19:51 tarihinde Dmitry [via SIP Router] yazdı: I installed Kamailio : yum install -y kazoo-kamailio https://2600hz.atlassian.net/wiki/display/Dedicated/via+RPM git clone https://github.com/sipwise/rtpengine.git the latest version of rtpengine and I do not know how to view the version of Kazoo-Kamailio On Wednesday, May 11, 2016 6:31 PM, ycaner <[hidden email]> wrote: hello ; maybe some devel are missing. which version using?i will try. cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML --

1 4

Another question about RTPengine module
by Dmitry 15 May '16

15 May '16

I will clarify the question in my previous post.I installed Kamailio from 2600hz repo via yum (it is for kazoo). But when cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so the kamailio does not see this moduleNo module matching <rtpengine> found Is there something like modules.list (it is used when I compile Kamailio manually) when the kamailio was installed via RPM?

2 12

WebRTC integration
by Moacir Ferreira 15 May '16

15 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

1 0

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 13 May '16

13 May '16

Hello all! After some time I was able to get this working here, and now I am coming back to share my configuration with you guys (in case someone needs to get this working). Here is what we have done: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } xlog("L_INFO", "checking $tu registered with User-Agent:$hdr(User-Agent)\n"); if(reg_fetch_contacts("location", "$tu", "caller")) { xlog("L_INFO", "$(ulc(caller=>count)) records found for $tu\n"); $var(i) = 0; while($var(i) < $(ulc(caller=>count))) { xlog("L_INFO", "record $var(i) with ruid:$(ulc(caller=>ruid)[$var(i)]) - User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); if ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) { $var(ruid) = $(ulc(caller=>ruid)[$var(i)]); xlog("L_INFO", "unregistering $tu ruid:$var(ruid)"); unregister("location", "$tu", "$var(ruid)"); } $var(i) = $var(i) + 1; } } if (!save("location", "0x00")) sl_reply_error(); exit; } } There are some logs over there because we used it to see what was going on while testing the configurations. With this we can get just one register per User-Agent. And again, thanks Olle for your help in this problem *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Sure it is! > > This is exactly what I was looking for! Actually I wasn't able to see that > there was a way to perform this action! > > Thanks for your help, Olle! > > I will try to create this, and when I have it done, I will post my script > here, so if someone else need the same problem I can help. > > Now, I have just one more question about my scenario: when I register a > user, is there a way to create something like a "custom field" on the > location? To explain better: If possible, I can add something like a custom > header that says if the user is registered from a phone device or the web > interface, then I can perform the loop and replace only the oldest contact > using that device... > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > > 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> This is my problem... I want to have 2 contacts for AOR, and not only >> one. >> >> When the 3rd arrives, it must remove the oldest and continue with 2... >> >> *If I am using the 0x04 parameter, it will save just one contact, and the >> user will not be able to get the Invite both, in the mobile device and web >> interface.* >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello, *Olle!* >>> >>> *Thanks for your help in this case, and don't worry about the time >>> (actually I wasn't even expecting to get an answer today).* >>> >>> >>> *I know the max_contacts parameter. But the problem is that if I set the >>> max_contacts to 2, when the user tries to register again, he gets an error >>> message 503. The point is that I want to get the user registered, and >>> replace the oldest location entry for the newest one.* >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello all. >>>> >>>> I have problem here and I tried to find a solution and search over >>>> internet, but without success. >>>> >>>> My scenario is the following: I have an application that must be >>>> registered in Kamailio when a user logs in the web interface, so he can get >>>> calls (something like a web softphone using webrtc). At this point, we are >>>> OK, and everything is working fine. >>>> >>>> To get these register functions I am using the parameter >>>> "save("location", "0x04")" as described in the REGISTRAR module >>>> documentation, so if a user logs in another web browser or computer, only >>>> the last one will continue registered and all calls will be forwarded to >>>> him. >>>> >>>> The point is that now we are creating a phone app that will do almost >>>> the same thing as the web interface, allowing users to receive calls using >>>> the mobile device, and here is my problem: I want to allow my users to be >>>> registered on two devices at the same time, but if a user logs into another >>>> device, I don't want to reply with a 503. I want to allow the user to >>>> register again, deregistering the oldest contact. >>>> >>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>> description it says "implemented overwrite oldest contact behaviour", >>>> but I couldn't find any documentation about it. >>>> >>>> So, is there a way to get this working today? >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>> >>> >> >

2 1

Changing the SDP connection-address to a value in a pseudovariable
by Kevin McAllister 13 May '16

13 May '16

counter to the docs for nathelper fix_nated_sdp the config parser rejects any pseudovariables in the second parameter claiming they need to be constant. Does anyone know another way to change the "c=" line in an SDP? Or do I have to patch the nathelper module to include a free_fixup_function for the fix_nated_sdp stuff? This is on kamailio 4.2 Thanks, Kevin

1 1

Kamailio Crash
by Sunil More 13 May '16

13 May '16

Hello All, Kamailio instance just crashed with the following logs May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12675]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12674]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12682]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12684]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12667]: INFO: <core> [mem/f_malloc.c:598]: fm_free(): freeing a free fragment (0x7f23ff39cb88/0x7f23ff39cbd0) - ignore to add i have collected a gdb backtrace as below, root@IP:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/corefiles/core.kamailio.sig11.12671 The version of Kamilio used by me is as below;. version: *kamailio 4.4.1 (x86_64/linux) 90be8b* 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_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b Thanking You, Sunil More Ph : 9503338275

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Blacklists and DB_Cassandra
by SamyGo 13 May '16

13 May '16

Hi, I am tasked to make use of blacklist module for about 4 to 6 million numbers. I am thinking of using Cassandra for the purpose but reading through the documentatiom of module and recent mailing list discussion made me a bit hesitant. I am looking for advise on this whether this is going to perform as expected or is not even going to work with the BlackListing module ! I am using Cassandra 33x and Kamailio 4.4 over Ubuntu 14.04. Looking for suggestions here. Regards, Sammy

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