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48 participants
37555 discussions

Kamilio verify online servers
by Marino Mileti 10 Nov '14

10 Nov '14

Hi guys, I've an architecture with multiple Kamailio. I would like to implement some check that allow me to verify if a server is online or not so for example i can use msilo to save chat message and relay it when the server come's up. Is it possible to use UAC module to send a REGISTER message from one server to another so i can "dump" all message in very simple way? Is there any example ? Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamilio-verify-online-servers-tp132… Sent from the Users mailing list archive at Nabble.com.

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htable locking
by Alex Balashov 10 Nov '14

10 Nov '14

The documentation for the htable module gives no guidance on whether sht_lock()/unlock() should be used in all situations when writing or reading from the htable. Should it be? I was under the impression (not supported by anything said in the documentation) that read and write access to the htable is fundamentally thread-safe, i.e. it is safe to both read and write to the htable even if the potential exists for another worker process to do the same concurrently. Is that not the case? If not, should there not be strong admonitions to use sht_lock/sht_unlock around concurrent operations--and it sounds like nearly all useful applications of the htable are essentially concurrent? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Generate REFER msg
by Agiftel 10 Nov '14

10 Nov '14

Hi all, i need some advice on how to solve a problem. I've a scenario like following: [SIP PBX] <---->CUCM (Call Manager v 10.5 ) User A: belong to CUCM User B: belong to SIP PBX User C: belong to SIP PBX User A calls User B. After some conversation, User A wants to transfer call to User C. CUCM transfer the call with re-INVITE; in this case on SIP PBX 2 channel SIP are used. CUCM does not support transfer, on Trunk, using REFER method. My idea is using Kamailio beetwen SIP-PBX and CUCM and handles call generating REFER msg in some way. Is this idea a good one? Or there is a better way? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Generate-REFER-msg-tp132317.html Sent from the Users mailing list archive at Nabble.com.

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rr problem: sips instead of sip
by Jon Bonilla (Manwe) 07 Nov '14

07 Nov '14

Hi all I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based on kamailio 4.1.6) The problem is as follows: A custom mobile client B blink for linux The path of a call from B to A would be this one: B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and registrar and sems acts as B2BUA. Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the inside. Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080). My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the Kamailio LB to the proxy. The problem is that the kamailio proxy added itself as sips in the record-route header: U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080 INVITE sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0' Record-Route: <sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->' Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Record-Route: <sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route: <sip:lb@127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>' Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0' Via: SIP/2.0/UDP 10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x' When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy: NOTICE: <script>: New request on lb - M=ACK R=sip:ngcp-lb@XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 NOTICE: <script>: Relaying request, du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--', fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1 WARNING: <core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062) I think the problem might be in the way the client registers? This is the registration info where it uses sips in the contact: contact: sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp If I try the same calling blink, which has the following contact in the registration it works without issues as the proxy doesn't insert sips in the Record-Route header: contact: sip:91807432@10.10.10.4:57160;transport=tls Any thoughts? cheers, Jon

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Kamailio v4.0.7 Released
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, Kamailio SIP Server v4.0.7 stable release is out. This is a maintenance release of the 3rd old stable branch, 4.0, that includes fixes since release of v4.0.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.7, or even better, to 4.1.x or 4.2.x, as the focus from now on will be on maintenance of these release series. For more details about version 4.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-0-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Releasing v4.0.7
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, I am planning to release v4.0.7 based on latest source code in branch 4.0. This branch is going out of official maintenance, the current latest stable branches being 4.2 and 4.1. If you have any remarks regarding it, write back to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

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cdp.so segment fault
by Tom Secula 06 Nov '14

06 Nov '14

I am just wondering if anyone has seen this type of segfault on 4.2 ? I'm trying to run kamailio as an scscf on centos 6.6 ? Nov 6 14:17:20 scscf kernel: kamailio[6246]: segfault at 0 ip 00f3f425 sp bfa0e9f4 error 6 in cdp.so[f08000+96000]

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Re: [SR-Users] RPC uac.reg_info hangs second time
by Dan Christian Bogos 06 Nov '14

06 Nov '14

Hey Daniel, Thanks for your fast reply! I am using the 4.2.0 out of debian packages (not nightly). Will shift on nightly and retest. Not a big deal regarding 150 seconds, maybe it should be a good reason for async reloads (reply with the delay number of seconds when reload will happen). Reloads can be always implemented outside of kamailio but since everybody would need it I assume a more centralized approach would be useful. Anyway, thanks again! DanB On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Hello, > > are you using branch 4.2 or some tgz/package from 4.2.0 release? There > was a fix related to such issue, so 4.2 branch must be used (or debian > nightly builds for 4.2). > > I will have to check the code how flexible that is, but afaik, the limit > on reload interval was thought to avoid sending again registration > requests while retransmissions from previous one can occur. > > Cheers, > Danel

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Kamailio v4.1.7 Released
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, Kamailio SIP Server v4.1.7 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.7. For more details about version 4.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-1-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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how to check if reply is retransmission?
by Juha Heinanen 06 Nov '14

06 Nov '14

is there a means to check if reply to t_relay()ed request is retransmission? there is t_is_retr_async_reply(), but that does not apply to regular transactions. i would like to avoid double calling of rtpengine_answer(). -- juha

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