sr-users May 2013

sr-users@lists.kamailio.org

95 participants
157 discussions

Need kamailio-3.1.5 configuration
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, After successful install of kamailio-3.1.5 I am trying to start kamailio,but it throws following error loading modules under /usr/local/lib/kamailio/ modules_k/:/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_s/ 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 Listening on udp: 192.168.90.210:5035 udp: 192.168.90.210:5036 udp: 127.0.0.1:5036 tcp: 192.168.90.210:5035 tcp: 192.168.90.210:5036 tcp: 127.0.0.1:5036 Aliases: tcp: localhost:5036 udp: localhost:5036 0(2580) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2582) INFO: usrloc [hslot.c:53]: locks array size 512 0(2582) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2582) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 Please suggest what can be done ? -- Thanks Priyaranjan

2 1

Siremis v4.0.0 Released
by Elena-Ramona Modroiu 09 May '13

09 May '13

Hi, a new release of Siremis is out - v4.0.0 - web management interface that is compatible out of the box with Kamailio v4.0.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ Regards, Ramona

2 2

Do we need rtpproxy for all kind of NAT?
by John Chen 08 May '13

08 May '13

Hi, I can configure kamailio + rtpproxy to enable calling between user behind the NAT. Thanks for the example config, it is very easy to did it. Do we need rtprpoxy for all kind of NAT? From what i have read, only symmetric NAT that hard to be traversal-ed. While other type of NAT can be traversaled using STUN. But my device that behind openwrt router can't work without rtpproxy. Is it expected behaviour? Or i'm doing something wrong in the config? Thanks

4 6

Call routing
by mark＠brightvoip.co.uk 08 May '13

08 May '13

Hi all, Posted a similar query a few weeks ago, without much interest - any advice appreciated. I have two sites and will send calls between them. I have Kamailio at each site which will route the calls out/in. There are multiple distinct network routes between the sites, accessible via different IP addresses. Each Kamailio has multiple IP's, one for each route. The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable. When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%). I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability. My main headache is that RTP must follow the same route as SIP for each call. After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route). Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio. Any comments or alternate solutions/suggestions would be of interest. Many thanks, Mark

2 1

Problem with Realtime Kamailio-Asterisk integration
by aaaa aaaa 08 May '13

08 May '13

Hello, I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error: [Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'. Kamailio and Asterisk are running in the same machine. Any idea about the cause of this problem? Best Regards, Theo

5 6

sip peering
by johnc 07 May '13

07 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have built a kamailio based phone system along the lines of the tutorial below. I have updated it for version 4.0 and it is running successfully on Debian 7.0. I have set my system up with a Gandi SSL cert rather than a self signed one. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I want to federate/peer with another proxy running Kamailio with TLS. What do I need to do to make this work? Thanks. Regards, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRiDBUAAoJELy1jPQ1KER7sBoP/1G7I5rlDWcy2eC5GaN7aIa6 xjtKSB8/2cHIyws9d5Zb1mJFKbKgRPJRlSODxUD4rqLdUwq3oIP8buVBAp4rx7UT Xg8POBwRTPqccF7+AxelhkplnWsh11rVhfCT/2L0FLgKSelGGCaAl5SNI5rwj9HO On9ACQeCarpP4UbyZK15iSkreMe4eMfErQgqhfyJwjmlIoEYlBiT4QrUPR3I0MbA jWJq2IWl2Jz/ZeDe2Z7vX6vDY6iidYL+kfFqyXOzoajw+/HTpzok79pGwPSMWZVx 2UHximuxyzJKqGXH1G1Ai4sRrWurM/02jH5iaddb3K2k/fGI8NxvDVgS4p7hyUTA JTmXetcYhTFstgQlnaD0U+FO268uPtGbVQjzuxnYy/bwHRnRV8sxKsjihpRTJrUY R6/QU2xMuaKigJn2FTzSodLMeHtSifUc3Od/9THoh4r6RlQOwMMxBGH+xSdTIXOo wmzLqly3sFPEsSp/30ncs33xT+P7SvMGzu+XHx16UZHCvrFhG3ME10a4C06dn53H mOvXPQ/xm0a47uytp859lbhG3PiyuiF5OkBmuaxndKon2w+HisEZ6+FqteYL3Xjq K3a2z8aoDGFzkYgqqyh5Fpn/hGzd7/E8KkI6Ldrffo2QY3uQx8CWnwXZnYsqFzXy dFS7DTGFXWeKZSAwDO+I =Iezv -----END PGP SIGNATURE-----

2 2

Max concurrent TCP connection that Kamailio support ?
by Khoa Pham 07 May '13

07 May '13

Hi, What is the maximum number of concurrent TCP connection that Kamailio can handle ? I see the max_tcp_connections (which is set to 2048), is that the answer ? How to test for this ? Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

Question about relaying
by Leo Brown 07 May '13

07 May '13

Hi My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers. MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch | |---> Our internal routing switch The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other. When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be). What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way. Leo

3 7

Presence and Cisco 79xx phones BLF
by Andrzej 07 May '13

07 May '13

Is it possible to run on phones Cisco 79xx series BLFs? So to show the status of the shared lines. Phones registered with Kamailio. Andy

3 2

rfc: restructuring source code tree
by Daniel-Constantin Mierla 07 May '13

07 May '13

Hello, among the topics discussed just before the last major release series (4.0.x) was one about restructuring the source code tree. It started mainly as a proposal to move source code belonging to core in a dedicated folder, but there could be more variants. It's time start discussing if we do it, and if yes, how. Here is what I could collect so far. a) no change, keep like it is b) move source code of core in a dedicated folder named 'core' c) move all source code in a folder named src, with following sub-structure src/core - for core src/lib - for internal libraries src/modules - for module Feel free to propose other variants or express your opinion regarding the proposed options. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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sr-users May 2013