sr-users February 2018

sr-users@lists.kamailio.org

90 participants
113 discussions

issue with running kamailio
by justin barnes 20 Feb '18

20 Feb '18

I am installing from githum kamailio 5.0 on my ubuntu-16.04-x86_64 VPS. when I start the kamailio server I get this error : /# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. failed! Ilooking up the status, I get this kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled; vendor preset: enabled) Active: failed (Result: exit-code) since Mon 2018-02-19 15:23:41 EST; 4h 10min ago Process: 18969 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GR Feb 19 15:23:41 khynepassion systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Feb 19 15:23:41 khynepassion systemd[18969]: kamailio.service: Failed at step EXEC spawning /usr/sbin/kamailio: No such file or directory Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Control process exited, code=exited status=203 Feb 19 15:23:41 khynepassion systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Unit entered failed state. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Failed with result 'exit-code'.

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LCR options as of 2018
by Francisco Valentin Vinagrero 19 Feb '18

19 Feb '18

Hello, I am configuring a Kamailio node that will interface with 3 carriers and will implement Least Cost Routing for around 900 destinations. I have already implemented a quick prototype with the LCR module and I have been reading the CarrierRoute doc and the CGrates site. I don't really need advanced functionality, just: - Re-routing to a backup carrier when the main one is not available. - Ping with OPTIONS to all carriers so I can detect if any of them is not available (à la Dispatcher) - Translate prices from a central DB to routing priorities. - Get some accounting post-call info about the routing decision (acc module) So far, while testing the LCR module I have had some issues with the rule IDs being integers that auto-increment, so the LCR rules need to be referenced from the rule_target table using this numeric ID. I can see some issues when scaling this to 900 lcr_rules and 2700 lcr_rule_targets with 3 priorities. In addition to that, I think the OPTIONS mechanism is just used by the lcr module when a gateway is marked as defunct, but the gateways are not actively monitored in the normal status. Is this correct? I have been considering CarrierRoute, which has a slightly simpler table structure, but I can't find any other advantage for my use case. Regarding CGrates, it looks great but it looks like it exceeds my current needs. For those who have implemented LCR in your kamailios, am I missing something regarding performance, scalability, etc? Are there any newer modules that can implement this, or maybe just a dispatcher-on-steroids setup will do the trick? Cheers, Francisco.

3 6

User state
by Magnus Löfqvist 19 Feb '18

19 Feb '18

Hi, We are running for a customer a kamailio server for user registration and routing of calls. Also using Asterisk for queue management and IVR. Now we are in the progress of writing a web portal where the customers users should be able to see if there collueges are busy in phone or availible. In asterisk I would use hints and listing in events in AMI. But what way are the best in Kamailio? Best Regards Magnus Sent from Nine<http://www.9folders.com/>

2 1

PR1447: first attempt at app_python3
by Anthony Alba 18 Feb '18

18 Feb '18

Hi Daniel and kamailio contributors, I have submitted a PR for app_python3 here: https://github.com/kamailio/kamailio/pull/1447. It is very rough, but would appreciate your feedback. Notes: 1. Almost a verbatim port of app_python; just changing obvious semantics. String->Unicode is practically the only large change. No change to symbol names and export functions so the two modules cannot coexist. (Don't loadmodule "app_python.so" and loadmodule "app_python3.so) 2. Module initialization semantics in Py3 have changed: add the top level modules to the inittab before calling Py_Initialize(). I have done this for KSR and Router. I.e. PyImport_AppendInittab() is called for Router and KSR *before* Py_Initialize() 3. Makefile looks for a python3 binary (PYTHON3 in the Makefile) to get include paths and link libraries. Here is an example compile line for IUS Python 3.6 on CentOS 7 (no native Python3 from RHEL) ## default: needs to find python3 and python3-config ## needs to find both python3.6 and python3.6-config make -C src/ PYTHON3=python3.6 SKIP_MODULES="${SKIP_MODULES}" quiet=verbose module_group_py="app_python3" group_include=py every-module 4. The semantics for GIL and thread management are too different for one-to-one replacements. I have therefore decided to comment out all threading calls. Fortunately, kamailio does not used threads at the moment. This is more complicated than it seems: even with Py2+thread management: there would be deadlocks if kamailio->python2 script->Cython extension with threads(e.g. grpcio). For an MVP it would be best to get it in without thread management. 5. It should backport to 5.1. 6. Tested locally, with a routing scripting making many python_exec() calls Thank you. Anthony Alba

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PUBLISH and ETag
by Cyrille Demaret 16 Feb '18

16 Feb '18

Hi, I'm using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk : When I make a first call Asterisk sent the following: PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610 82 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Event: dialog Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early.. Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972 3a61082;received=192.168.100.37 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19863.16.0 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 When the call is done, Asterisk sent another PUBLISH telling that the call if terminated : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7 52 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19863.16.0 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 230 <?xml version="1.0" encoding="UTF-8"?> terminated.. And Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97 ab8c752;received=192.168.100.37 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19873.18.1 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 Here, the SIP ETag is a.1518775074.19873.18.1. The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22 f0 From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528 To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10699 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19873.18.1 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early. Kamailio refuse it with this error : "Trying to update an already terminated state. Skipping update." because the call is considered as terminated. The RFC is stating : When updating previously published event state, PUBLISH requests MUST contain a single SIP-If-Match header field identifying the specific event state that the request is refreshing, modifying or removing. This header field MUST contain a single entity-tag that was returned by the ESC in the SIP-ETag header field of the response to a previous publication. Why Kamailio is acting like that? Best regards, Cyrille

2 2

Problem with "transport=tcp" and rtpproxy
by Pau Giralt 16 Feb '18

16 Feb '18

Hi! I'm working with kamailio and rtpproxy. When I configure the clients to send sip messages through TCP adding the ";transport=tcp" in the sip message the rtpproxy is not working anymore. The code is never passing the point pasted below, inside route[NATMANAGE] in kamailio.cfg if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; Therefore, rtpproxy_manage("co"); is never reached. Could it be a parsing problem? ( check_route_param("nat=yes") ) The config file is exactly the same as the one provided in the kamailio repository (the default one) Thanks!

2 7

Raspberry Pi deb packages -> built
by Mititelu Stefan 15 Feb '18

15 Feb '18

Hi people! After a bit of research and lib installations, I've managed to build the full kamailio deb packages for a raspberry pi 3 (version B) running raspbian stretch (lite) that can be found at [1] (with a really few module exceptions that might be used in really specific scenarios -> I think) Just for fun, I'm planning to do some CPS stats tomorrow (if I got the time), with a basic config and also share here, but, if someone else tried it before and is able to share some stats/benchmarks, would be greatly appreciated (and interesting to compare, given my "decent" rpi overclock), because I'm also curious about VoIP stuff on raspberry pi! :D [1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1 Thank you, Stefan

2 3

KEMI
by Aleksandar Sosic 15 Feb '18

15 Feb '18

Hi everyone, we're valuating on rewriting our kamailio routes in python. We're doing a lot of string manipulation and GraphQL API queries so the kamailio scripting language is a little bit limited for this use case. Can someone tell us if the KEMI framework is stable enough for production? Kind regards, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

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Re: [SR-Users] RTPEngine Transcoding
by Social Boh 15 Feb '18

15 Feb '18

Hello, I'm trying to use this configuration: Softphone A audio codec g722 and Opus Softphone B audio codec PCMA and PCMU Kamailio with rtpengine_manage with codec-transcode-PCMU codec-transcode-PCMA in the request Softphone A call Softphone B The RTPEngine DEBUG show many Decoder Error: https://pastebin.com/k1Zhw8HS Thank you Regards --- I'm SoCIaL, MayBe El 13/02/2018 a las 20:05, Richard Fuchs escribió: > On 2018-02-13 07:22 PM, Social Boh wrote: >> Thank you. >> >> which is the best way to implement ir >> >> codec-strip-all and the add codecs? >> >> Is there a document about how work the transcoding? > It's really up to you and depends on what you want to achieve. Doing a > strip-all and then adding codecs you want to offer is definitely one > way. Another is to just add codecs that you know or think the peer may > want to see. > > Cheers >

2 2

Invalid payload location
by Soltanici Ilie 15 Feb '18

15 Feb '18

Hi All, Can someone explain why i'm receiving such errors in syslog: [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:228]: set_sdp_payload_fmtp(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location This is how the SIP Packet is looking: 2018/02/15 10:51:40.916040 46.166.139.7:30189 -> X.X.X.X:5060 INVITE sip:01031203697179@X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP 46.166.139.7:30189;branch=z9hG4bK-5342365299746297160962993236828333618949;rport=30189 From: <sip:88893855391@46.166.139.7:30189>;tag=0vkn019bur To: <sip:01031203697179@X.X.X.X:5060> Contact: <sip:88893855391@46.166.139.7:30189> Call-ID: 0889633580294714035892119863939070195463 CSeq: 101 INVITE User-Agent: Hello Annonymous Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=SIP-Session 950029666 1307322426 IN IP4 46.166.139.7 s=SIP c=IN IP4 46.166.139.7 t=0 0 m=audio 5075 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:104 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Thank You.

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sr-users February 2018