the freeswitch version is FreeSWITCH Version 1.0.head (git-c4a2faf 2011-06-10 14-36-35 -0400), and in this version we can put the FS in debug too. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633 which FS version do you use ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 12/06/2011 à 17:01, Gustavo Espeche a écrit :
Hi we found an error in freeswitch
2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0 o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68 s=sip call c=IN IP4 200.49.30.69 t=0 0 m=audio 42410 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 -------------------------------------------- 2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0 o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116 s=SIP Media Capabilities c=IN IP4 38.105.229.114 t=0 0 m=audio 14578 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 -------------------------------------------- 2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0 o=ATSI-B2 188 2 IN IP4 208.37.200.235 s=sip call c=IN IP4 208.37.200.245 t=0 0 m=audio 35538 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 -------------------------------------------------------- 2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0 o=VoipSwitch 10828 10828 IN IP4 50.30.35.13 s=VoipSIP i=Audio Session c=IN IP4 50.30.35.13 t=0 0 m=audio 9828 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-1
the problem is that when this error happens FS send the cdr immediately but the call still connected, and the provider still with all billing, because of the we see a billing difference with our provider. The error happens with different provider or customer equipment. Some one know if we can configure of freeswitch in some way that the call hangup. The problem isn't fix the codec error, for us the real problem is that the FS do a cdr but the call still connected. Thank a lot in advice.
--
Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633
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