Hello, run asterisk in debug mode to understand why is sending BYE.
Cheers, Daniel On 07/08/15 16:40, Loic Chabert wrote: > Hello, > > I have set on the right place "route(RTPPROXY")", and now it works for > internal calls and external calls. > Reason: my request passing througt RTPPROXY twice ... > > One last problem: > - 102 initiate a call to 101 > - 101 refuse call with a 486 response > - as asterisk dialplan said: launch voicemail app > - Sounds files has been read from asterisk, but after 5 secondes, > session has been cut with a BYE request sent by Asterisk. > > Please find in attachement pcap trace file (91.x.x.x is wan kamailio > interface, 10.0.247.197 is lan kamailio interface, facing to asterisk > cluster) > > Why asterisk send this BYE ? Kamailio does not force him to send this > BYE... > > Thanks, > Loic. > > > 2015-08-07 10:34 GMT+02:00 Daniel-Constantin Mierla <mico...@gmail.com > <mailto:mico...@gmail.com>>: > > Hello, > > look at the sip traffic and see what is in SDP, if you don't get > audio, maybe the other ip is advertised. > > Cheers, > Daniel > > > On 07/08/15 09:16, Loic Chabert wrote: >> Hello Daniel, >> >> I have changed my rtpproxy by rtpengine. I have explicitly define >> public and private interfaces, and now it work as expected for >> external calls (througth PSTN). >> But for now, after this change, internal call (like 100 call >> 101), does not work any more. >> >> I need more investigation to see what append on my call flow. >> >> I will update you asap. >> >> Thanks, >> Regards. >> >> >> 2015-08-07 9:04 GMT+02:00 Daniel-Constantin Mierla >> <mico...@gmail.com <mailto:mico...@gmail.com>>: >> >> Hello, >> >> On 30/07/15 17:38, Loic Chabert wrote: >>> Hello everyone, >>> >>> I'm trying put kamailio in front of asterisk server farm. >>> Fow now, 2 asterisk servers are running and i'm trying to >>> make some basic calls between two UACc. >>> >>> All asterisk servers has been ofuscaded from public internet >>> using 10.189.122.0/24 <http://10.189.122.0/24> network. >>> All trafic must be passed throught asterisk so RTPproxy is >>> used to (and used for rtp bridging). >>> Kamailio and rtpproxy is running with public IP address, and >>> private ip address (mhomed=1) >>> >>> But a wired thing append on my SDP body: c line have two >>> rtpproxy public addresses concatenate (see my capture attached). >>> >>> Any reason for this ? Only invite method from my asterisk >>> contains 2 publics IP addresses concatenated. >>> >>> Does it mean than rtp_manage as been executed twice ? >>> >> It could be that it was executed twice. As pointed in another >> response, look at what is received on the network and in the >> logs. >> >> You can enable cfgtrace for debugger module in order to see >> what actions are executed from configuration files -- it is >> good to spot quickly errors in the logic of config file. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - >> http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.com >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users@lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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