sr-users April 2017

sr-users@lists.kamailio.org

79 participants
114 discussions

Warnings at apply_changes()
by Diego Nadares 25 Apr '17

25 Apr '17

Hi list, I'm converting body message to multipart. Calls establishes ok and Kamailio it's not logging errors but I'm seeing some info and warnings that I don't know it they are correct. Maybe you could tell me if it's normal or if I'm doing something wrong in my script. kamailio[13428]: INFO: <script>: Original content type is application/sdp kamailio[13428]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> kamailio[13428]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio[13428]: INFO: <script>: Appending GTD to body I use the following sintax: route{ ... route(MANAGE_MULTIPART); ... } route[MANAGE_MULTIPART]{ .... if (is_method("INVITE") && !has_totag() && has_body()){ ... xlog("L_INFO", "Original content type is $cT"); set_body_multipart(); *msg_apply_changes();* xlog("L_INFO", "Appending GTD to body"); ... } ... } Invite example: INVITE sip:222222222@172.16.205.122:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-28085-1-0 From: "111111111" <sip:1111111111@172.16.213.21:5060;user=phone>;tag=1 To: <sip:2222222222@172.16.205.122:5060> Call-ID: 1-28085(a)172.16.213.21 CSeq: 1 INVITE Contact: "1111111111" <sip:111111111@172.16.213.21:5060> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 137 v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.21 s=- c=IN IP4 172.16.213.21 t=0 0 m=audio 6000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 As you can see the log says that "Content-Type hdr has no params <application/sdp>" but in the line before says "Original content type is application/sdp". I'm running kamailio 4.4 by the way. Thanks in advance. Diego.

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Re: [SR-Users] New developer: Guillaume Bour
by Siwek, Cezary 24 Apr '17

24 Apr '17

Nice work Guillaume! On 24 April 2017 at 15:21, Siwek, Cezary <cezary.siwek(a)vonage.com> wrote: > Nice work Guillaume! > > On 21 April 2017 at 09:55, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I would like to announce that Guillaume Bour (https://github.com/gbour) >> has now developer privileges on Kamailio's gihub project. He has >> contributed recently a new module named keepalive: >> >> - http://kamailio.org/docs/modules/devel/modules/keepalive.html >> >> There is also a pull request from him waiting to be merge to drouting >> module, adding capability of detecting active/inactive gateways used by >> the module. >> >> Thanks for the contributions so far and looking forward to collaborate >> in the future! >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > >

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Video recording with RTPENGINE
by Alberto Llamas 24 Apr '17

24 Apr '17

Hi guys, Any of you have experience using the '*start_recording()*' function of the RTPENGINE in a WebRTC scenario?. At this point, I am able to record the stream but I am not able to play them. The output is a .pcap file in a raw format. I have tried the conversion: raw -> H264 -> mp4/avi (with different tools: videosnarf, x264, ffmepg) and not luck. Do you know the right way to convert those stream in raw format?. Any hint will be appreciated. Regards,

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Memory leak after enabling topos module
by Sergey Basov 24 Apr '17

24 Apr '17

Hi All. During loadtest of kamailio 5.0.1 with topos module I got memory leak in "udp receiver" process After 1600 sip calls I got 10 Mbytes of free private memory. I set limit to 32 Mbytes output of kamcmd corex.pkg_summary pid xxx command at begin test and before eom are in attached files. If you need more info please let me know. Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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SIREMIS
by Emil Morganti 24 Apr '17

24 Apr '17

Hello, I have installed kamailio 5.1.0-dev and would like to know if it is compatible with SIREMIS v4.3.0? Has anyone tried it?

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MSRP Connection is not working
by Jegadeesan 24 Apr '17

24 Apr '17

Hi , I'm very new to Kamailio application and trying to setup the test environment for Android RCS client (https://github.com/android-rcs/rcsjta). Parallelly we have deployed the Kamailio on ECS2 and able to make a call/text using any other sip clients. Apparently, we had enabled the MSRP modules over TCP and tried to test with Android RCS. But getting stuck at the connecting screen. Although we could see SIP exchange logs between two clients. Can anyone help me in fixing this issue? -- Jegadeesan M

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tm forward
by Slava Bendersky 24 Apr '17

24 Apr '17

Hello Everyone, What possible cause for messsage /usr/sbin/kamailio[4037]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): no branches for forwarding Thank you.

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Kamailio not forwarding call
by Gonzalo Gasca Meza 22 Apr '17

22 Apr '17

Hi all, I worked in an prototype recently where I *had* the following setup working. Kamailio is using Postgres to forward the call to Twilio domain. *Topology:* SIP client (tls) > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR Right now when I call from SIP Client to IVR, call is not forwarded. Nothing has changed on Kamailio side. *Troubleshooting:* a) *SIP client (tls) > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR [Fails]* - Case described above. b) *Call from PSTN > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR [Works]* I debug and call arrives to Kamailio but is not being forwarded to Twilio. This used to work before. When call works Kamailio seems to find remote domain, when fails we just sent a *502 Bad Gateway* message Twilio does not support DNS SRV records, just A host resolution. *Logs* While looking at debugs I found that this DNS logs: Pastebin below: *Call fails <https://pastebin.com/ttkK1Kkd> * Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (_sip._ udp.parzee.sip.twilio.com(31), 33), h=816 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:741]: get_record(): lookup(_sip._ udp.parzee.sip.twilio.com, 33) failed Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:864]: dns_cache_mk_bad_entry(): (_sip._ udp.parzee.sip.twilio.com, 33, 60, 1) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:798]: dns_cache_add(): adding _sip._ udp.parzee.sip.twilio.com(31) 33 (flags=1) at 816 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (parzee.sip.twilio.com(21), 1), h=267 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:937]: get_record(): skipping 0 NS (p=0xa1d2e7, end=0xa1d2e7) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:952]: get_record(): parsing 0 ARs (p=0xa1d2e7, end=0xa1d2e7) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:1741]: dns_get_related(): (0x7f1b5c227920 ( parzee.sip.twilio.com, 1), 1, *(nil)) (0) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:840]: dns_cache_add_unsafe(): adding parzee.sip.twilio.com(21) 1 (flags=0) at 267 *Call works <https://pastebin.com/AkWsvFVA>* Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (_sip._ udp.parzee.sip.twilio.com(31), 33), h=816 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [resolve.c:741]: get_record(): lookup(_sip._ udp.parzee.sip.twilio.com, 33) failed Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:864]: dns_cache_mk_bad_entry(): (_sip._ udp.parzee.sip.twilio.com, 33, 60, 1) Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:798]: dns_cache_add(): adding _sip._ udp.parzee.sip.twilio.com(31) 33 (flags=1) at 816 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (parzee.sip.twilio.com(21), 1), h=267 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [msg_translator.c:2876]: create_via_hf(): id added: <;i=7>, rcv proto=3 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (54.172.60.0, 54.172.60.0, 0) Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: tm [t_funcs.c:368]: t_relay_to(): SER: new transaction fwd'ed Thank you!

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integration kamilio with open IMS
by Med Amine Hajji 21 Apr '17

21 Apr '17

Hi i need ur help i want to integrate my Open IMS WITH KAMILIO wich configuration i must do ? thank you for helping me

1 0

Kamailio rtpengine sdp
by gerry kernan 21 Apr '17

21 Apr '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk VM's. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext. I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address. I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

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sr-users April 2017