sr-users June 2015

sr-users@lists.kamailio.org

82 participants
135 discussions

Kamailio Server Maintenance - Wed-Thu Night

by Daniel-Constantin Mierla

Hello, during the night between Wed (June 24) and Thu (June 25), planned to start not early than 00:00 GMT+1, there will be some schedule maintenance work to the infrastructure that is hosting some of the kamailio.org servers. The main affected services will be: - main website (www.kamailio.org) - wiki portals - mailing lists (lists.sip-router.org) - git mirror (git.kamailio.org) It is expected to have short downtimes of few minutes. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

8 years, 10 months

Kamailio + RTPEngine + Asterisk

by Javier Aristizábal

Hello folks, I am trying to configure this on the same server: Kamailio v4.2 port 5060 RTPEngine v2.3 port 22222 Asterisk v11.18.0 port 5080 I am follow the Asipto tutorials, for the Kamailio and the real time configuration. Now, I can talk between extension and PSTN without issues. But the trouble is with the Status registration with Asterisk. And I need that Status OK because I am using queues in asterisk for call centre. I got this from with: sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101 (Unspecified) D Yes Yes 0 UNREACHABLE Cached RT the status is UNREACHABLE and the asterisk table sipregs won't update after each registration. To be sure, I tried first all config only with asterisk to check that is not a problem with MySQL or something. On my cfg file I have this for the asterisk registration: route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Any clue? Thanks! -- Javier Aristizábal

8 years, 10 months

Kamailio proxy at Amazon EC2 (behind NAT)

by Alexandru Covalschi

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

8 years, 10 months

Kannot start kamailio 4.3

by Yuriy Gorlichenko

Try to start kamailio on Ubuntu 14.04.02 Get this errors ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) my params for ctl # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 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, F_MALLOC, DBG_F_MALLOC, 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: 1dbd53 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

8 years, 10 months

kamailio fails after start

by Yuriy Gorlichenko

Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual systemOpenVZ. after starting kamailio I see that it runs ok with kamailio start or kamctl start at ps -ax I see working processes. But after one minute of working kamailio fails with kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe I see nothing more at debug file. No errors. (I run with debug=4) my cfg parameters at start mhomed=1 memdbg=5 memlog=5 fork=yes children=6 version: kamailio 4.2.4 (x86_64/linux) ac3682 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, F_MALLOC, DBG_F_MALLOC, 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: ac3682 compiled on 04:27:04 Apr 10 2015 with gcc 4.8.2

8 years, 10 months

Kamailio transformation

by Ali Taher

Hello, Suppose I have $var(a) = 1234#5678 and $var(b)=123455667 I need to get the number before '#' character if exists , meaning that from $var(a) I must get 1234# and from $var(b) I must get empty string. I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole string if # doesn't exists. There is a way to check if # exists by subtracting the length of the string before and after removing # character and check if length changes , but I want a cost-less method. Any help would be appreciated. Thanks Ali

8 years, 10 months

Capture a Match from Regex comparison

by Sebastian Damm

Hi, we have some cases, where we get multiple PCMA lines with different numbers from a peer. One of our backend systems doesn't handle that correctly. So what I want to do is detect whether there is a PCMA line with a non-standard number in the SDP. This is how an SDP looks like: v=0 o=hiQ9200 2990220150519102450 1229717573 IN IP4 1.2.3.4 s=Phone Call c=IN IP4 1.2.3.4 t=0 0 m=audio 40174 RTP/AVP 8 0 18 96 97 13 100 98 99 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:0 PCMU/8000 a=fmtp:0 vad=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=rtpmap:98 PCMA/8000 a=gpmd:98 vbd=yes a=rtpmap:99 PCMU/8000 a=gpmd:99 vbd=yes a=sqn: 0 a=cdsc: 1 image udptl t38 a=sendrecv a=pmft: T38 a=ptime:20 I tried something like that (expecting that the number doesn't change): if (sdp_get_line_startswith("$avp(badcodec)", "a=rtpmap:98 PCMA")) { But this makes Kamailio drop the INVITE when arriving at the line in the config. I guess, "startswith" stops parsing the line at the colon. So I thought I'll make it more generically: sdp_get("$avp(sdp)"); if ($avp(sdp) =~ "a=rtpmap:([1-9][0-9][0-9]?) PCMA") { sdp_remove_codecs_by_id("$missingvariable"); } What I need is to capture the number in the a=rtpmap line so I can delete the codec by ID. But I couldn't find a way to capture the matches from my regex comparison. Am I missing something? (Other approaches for my problem would be welcome, too.) Thanks in advance. Best Regards, Sebastian

8 years, 10 months

Kamailio with Blox SBC (opensips based) - Wrong CSeq

by kai.ohnacker＠cbc.de

Hello, my setup is a kamailio server as registrar and Blox as SBC. Blox is a freeware GUI with Opensips as framework. Kamailio is located in the private network and Blox operates with 2 NICs (private and public) Now the following problem: If I make a call from public (phonerlite) to private (mayah), the connection don’t reach the “framed” state and is “broken” after 30 sec. After evaluating the log files, I found that the CSeq from Kamailio is wrong. The invite comes from CSeq: 9 INVITE The reply is with CSeq 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK38ae.cf317344.0 Max-Forwards: 70 To: <sip:1001@192.168.1.3:8000>;tag=qTKGZvr0ItW From: "PhonerLite" <sip:1000@192.168.1.3:8000>;tag=1005520878 Call-ID: SBCbFgsQn5HcHpEUxJUVF8gQWVZREt6QwJvWVJkXCdAUzdkXl41CUJ4cEdWZV1NQ0tFYQ CSeq: 1 Contact: <sip:1001@192.168.0.33:5060> Record-Route: <sip:192.168.0.1;lr=on> User-Agent: MAYAH 4.9.12.0-2.1.0.45 Content-Length: 0 And here the information from the opensips log files: Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: no method found Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: bad cseq Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:get_hdr_field: bad cseq Perhaps someone of you knows this failure or can tell me how to solve this problem? Cheers, Kai

8 years, 10 months

Redirect with path

by Kelvin Chua

on my registrar server: after calling lookup, and getting AOR with path, how do I send the path info back as a redirect? Kelvin Chua

8 years, 10 months

event_route: what modules/functions are likely to work or not work

by Armen Babikyan

Hello, With respect to *_route blocks in kamailio configs, I gather that event_route is the newest type of route in the config vocabulary. However, many modules have functions that specifically state that they work within certain routes, like REQUEST_ROUTE and FAILURE_ROUTE, with the implication that they won't work in other routes. For example, there are some functions that "work" in EVENT_ROUTE as well, and that this could be a matter of either documentation not being updated yet, or funny/bad behavior I may encounter at some future point in time... Generally speaking, are there any general statements that can be made about what "common" modules/functions are definitely only available within request_route and onreply_routes? event_route blocks, in particular, elucidate many other use cases for several modules' functions that I'm sure the original authors didn't think of, and it would be great to have a high-level commentary/discussion like "module XYZ or modules that touch A, B, and C don't have a snowball's chance of working in an event_route". Comments are welcome. Thanks! Armen

8 years, 10 months

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sr-users June 2015