sr-users May 2013

sr-users@lists.kamailio.org

95 participants
157 discussions

SER-GettingStarted… update?
by David Wilson 16 May '13

16 May '13

Hi All, One of the most thorough documents I've found for an introduction to Kamailio is the SER-GettingStarted.pdf, but it is now quite dated. Is there an updated version available, or in the works? Cheers, Dave.

1 0

[PATCH] Memory corruption using s.subst transformation
by Juha Heinanen 16 May '13

16 May '13

Daniel wrote: please do the patch to store the new value in _tr_buffer and attach it to mailing list or bug tracker. I haven't looked at code yet, but sounds like there is indeed an issue. I will review the patch and apply it. What was the conclusion about this? Was the patch ever submitted and applied? If not, is the bug still there or has is been fixed by some other means? I'm asking because this looked like a quite serious issue. -- Juha

2 4

Kamailio crash
by kiran bhosale 16 May '13

16 May '13

Hi we have developed the custom module where in we have created shared memory area which is not used by any other module or process and we locked it using locks as recommended. but because of locks the kamailio is getting crashed every now and then . is it really important to protect the shared memory area with locks. if yes then how to avoid the crash ?? so to locate the crash we built kamailio wit h MM_DBG , and other GCC debug flags . but we are getting error "/No symbol table info available/" . how to locate the bugs. the kamailio is built wit h 1 MB of shared memory on MIPS based VOIP gateway which is having * MB of flash and 16 MB or memory (RAM) . thanks in advance .

2 3

kamailio as a proxy
by Giuseppe Carella 16 May '13

16 May '13

Hi all, I'm trying to use kamailio as a proxy between webrtc clients and an openIMSCore. For this reason I'm currently using the dispatcher module to distribute all the requests coming from the clients to the IMS core, and enabled the websocket module for getting the SIP messages in. SO far it works fine for the registration, but it doesn't for the INVITEs. I noticed that kamailio receives the message but it doesn't deliver it to the clients. Please find attached the log and the configuration file that I used. I'm not an expert with the configuration, so any help or feedback would be very appreciated. Thank you in advance. Best regards, Giuseppe Carella

2 1

ERROR messages in DEBUG level
by Alex Solt 16 May '13

16 May '13

Hi, After changing log level from INFO to DEBUG level, I am seeing bunch of ERROR messages which are unknown to me: May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=498 a=26 n=is_methodMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=765 a=26 n=mf_process_maxfwd_headerMay 15 10:26:44 DEBUG: maxfwd [mf_funcs.c:85]: value = 68May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=775 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=27 n=sanity_checkMay 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0May 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:104]: found end of headerMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=snl_0156137698_NSN_CLIENTMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29May 15 10:26:44 DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=508 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=976 a=3 n=returnMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=511 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=833 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=780 a=25 n=has_totagMay 15 10:26:44 DEBUG: siputils [checks.c:106]: totag foundMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=830 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=loose_route The error message are complaining about simple statements in the config file: if (is_method("OPTIONS")) { sl_send_reply("200", "Ok"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } I appreciate your help. Thanks,AS

2 1

home pbx server experience
by u 15 May '13

15 May '13

I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and "scalability". But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage("c") in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u

5 4

log reply message number
by Robert R 15 May '13

15 May '13

Hi, I am trying to add a log when receiving 1xx or 2xx reply message: # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","reply message: $T_reply_code ci:$ci"); route(NATMANAGE); } } Adding the above line does not log the reply message number (180 or 183 or 200, ...). Thanks, R

3 3

Re: [SR-Users] home pbx server experience
by u 15 May '13

15 May '13

Should UAC work with Sipgate and freevoipdeal (dellmont)? Thanks for your answers everyone. 2013/5/15 u <ueberwachungsstaat(a)googlemail.com>: > Should UAC work with Sipgate and freevoipdeal (dellmont)? > > Thanks for your answers everyone. > > 2013/5/15 Stefan Sayer <stefan.sayer(a)googlemail.com>: >> >> if you don't get it all (auth + registrations) working with kamailo, >> you might try sems (iptel.org/sems) with sbc module (auth of outgoing >> calls) and for registrations either reg_agent (cfg file) or >> db_reg_agent (accounts from db) modules. >> >> load_plugins=uac_auth;registrar_client;reg_agent;sbc;xmlrpc2di >> application=sbc >> >> >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_sbc.html >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_reg_agent.html >> http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html >> >> BR >> Stefan >> >> o Edson - Lists on 05/15/2013 06:40 AM: >>> Hi... >>> >>> Use UAC module to manage registrations and play a little with the >>> config (INVITE section) to forward output calls correctly. >>> --- >>> Edson. >>> >>> Em 14/05/2013 11:27, u escreveu: >>>> I would like to share my experience with kamailio and other home pbx >>>> servers. >>>> >>>> Kamailio on my kirkwood home router for my 6 SIP users is perhaps >>>> overkill: I don't really need mysql and "scalability". But at last I >>>> finally managed to make calling between registered users work stable. >>>> My voip clients only work in all NAT scenarios if I work around some >>>> bugs: to use csipsimple on android I had to change rtpproxy_manage() >>>> to rtpproxy_manage("c") in kamailio's default config, so that problems >>>> with conflicting c: entries in the SDP go away. >>>> >>>> I propose kamailio could ship with a special example >>>> kamailio-compatible.cfg that doesn't try to be RFC compliant, but >>>> compatible to the most common voip clients. Right now the only thing I >>>> would change for this is the option for rtpproxy_manage, but I'm sure >>>> others will know more common quirks that could safely be enabled to >>>> increase compatibility. I think this compatibility idea is what yate >>>> sticks to for their defaults. In freeswitch you also have to do it all >>>> manually, and it's much more work to figure things out in their >>>> enormous config files. >>>> >>>> The other SIP proxies I had tried before kamailio officially fit all >>>> my requirements, including support for multihomed dynamic IPs, but >>>> contrary to their claims it didn't work. >>>> Yate was easy to set up, but the default dialplan is more confusing >>>> than powerful and after having made everything work I realised yate >>>> was clogging my CPU and RAM and after some time always randomly >>>> stopped working. This is with only 2 users connected! It also wasn't >>>> possible to fix NAT sdp while leaving the codecs section in the SDP >>>> alone at the same time. I tried to debug the code, but the C++ was so >>>> complex that I had to give up. >>>> Freeswitch was much more difficult to setup, a multihomed setup with >>>> dynamic IP was super buggy and it also didn't help that the >>>> unintuitive configuration is all in complex unreadable XML >>>> configuration files. >>>> >>>> Kamailio and rtpproxy don't officially support dynamic IP address, but >>>> I can just restart both each time my DSL provider forces me to a new >>>> IP address. This happens automatically in the night and is no big >>>> hassle really. The most simple, least-featureful solution works best >>>> it seems. >>>> >>>> Now the last problem I have with kamailio: I don't know how to connect >>>> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). >>>> I would like a simple way to do this, preferably without other >>>> features that always seem to complicate the matters. Is there >>>> something more lightweight and simple than asterisk, freeswitch and >>>> yate, that people use successfully for this task together with >>>> kamailio and rtpproxy? >>>> >>>> u >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> . >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

HELP! I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end.
by Future Lian 15 May '13

15 May '13

I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end. in CRIT.log of siremis/log, I get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all.

2 1

Kamailio 3.3.x is crashing frequently...
by Krishna Kurapati 15 May '13

15 May '13

Hi, When the Server is running it is crashing occasionally. (gdb) bac #0 0x000000000043359e in ?? () #1 0x0000000000000000 in ?? () (gdb) file /usr/local/sbin/kamailio Reading symbols from /usr/local/sbin/kamailio...done. (gdb) bac #0 free_lump_list (lump_list=<value optimized out>) at data_lump.c:504 #1 del_nonshm_lump (lump_list=<value optimized out>) at data_lump.c:661 #2 0x00007f571ff6cd9c in ?? () #3 0x00000000008db480 in mem_pool () #4 0x00007f571ff6c7e1 in ?? () #5 0x000000000139086e in ?? () #6 0x00000039294546e1 in ?? () #7 0x000000000140c7d0 in ?? () #8 0x00007f561f7acb80 in ?? () #9 0x00007f561f3fa6e0 in ?? () #10 0x00007f561f3fa6e0 in ?? () #11 0x0000000000000001 in ?? () #12 0x0000000000000000 in ?? () What could be going wrong here? Krish Kura

2 1

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sr-users May 2013