Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT -> ASTERISK): """ . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. """ The answer coming from asterisk in this case is: """ . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. """ I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090527/4e5373d5/attachment.htm
This is not a problem. Asterisk is under no obligation to offer an audio codec in return. mario staphorst wrote:> Hi Guys, > > Something I have noticed while dealing with T.38 and re-invites in > Asterisk 1.4.22. > > I have a provider who re-invites with the following sdp (message flow > PROVIDER_EQPMT -> ASTERISK): > > """ > . > v=0. > o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. > s=-. > c=IN IP4 CONN_IP_PROVIDER. > t=0 0. > m=audio 0 RTP/AVP 0. > m=image 26858 udptl t38. > a=T38FaxMaxBuffer:288. > a=T38FaxRateManagement:transferredTCF. > a=T38FaxUdpEC:t38UDPRedundancy. > """ > > The answer coming from asterisk in this case is: > > """ > . > v=0. > o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. > s=session. > c=IN IP4 CONN_IP_ASTERISK. > t=0 0. > m=image 4653 udptl t38. > a=T38FaxVersion:0. > a=T38MaxBitRate:9600. > a=T38FaxRateManagement:transferredTCF. > a=T38FaxMaxBuffer:200. > a=T38FaxMaxDatagram:200. > a=T38FaxUdpEC:t38UDPRedundancy. > """ > > I see a problem here since the number of matched media streams from the > offer does not match with the number of matched media streams in reply > from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). > > Please let me know if there are workarounds on this issue, or if this > could be a bug on asterisk side. > > Best regards, > > Mario Staphorst > > ------------------------------------------------------------------------ > Express yourself instantly with MSN Messenger! MSN Messenger > <http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/> > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775
Hi, I think this is not completely right, The scenario is: Carrier ==> Asterisk 1.4 ==> T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA.I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Regards, Mario> Date: Wed, 27 May 2009 09:44:56 -0400 > From: abalashov at evaristesys.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] problem with T.38 media headers > > This is not a problem. Asterisk is under no obligation to offer an > audio codec in return. > > mario staphorst wrote: > >> Hi Guys, >> >> Something I have noticed while dealing with T.38 and re-invites in >> Asterisk 1.4.22. >> >> I have a provider who re-invites with the following sdp (message flow >> PROVIDER_EQPMT -> ASTERISK): >> >> """ >> . >> v=0. >> o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. >> s=-. >> c=IN IP4 CONN_IP_PROVIDER. >> t=0 0. >> m=audio 0 RTP/AVP 0. >> m=image 26858 udptl t38. >> a=T38FaxMaxBuffer:288. >> a=T38FaxRateManagement:transferredTCF. >> a=T38FaxUdpEC:t38UDPRedundancy. >> """ >> >> The answer coming from asterisk in this case is: >> >> """ >> . >> v=0. >> o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. >> s=session. >> c=IN IP4 CONN_IP_ASTERISK. >> t=0 0. >> m=image 4653 udptl t38. >> a=T38FaxVersion:0. >> a=T38MaxBitRate:9600. >> a=T38FaxRateManagement:transferredTCF. >> a=T38FaxMaxBuffer:200. >> a=T38FaxMaxDatagram:200. >> a=T38FaxUdpEC:t38UDPRedundancy. >> """ >> >> I see a problem here since the number of matched media streams from the >> offer does not match with the number of matched media streams in reply >> from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). >> >> Please let me know if there are workarounds on this issue, or if this >> could be a bug on asterisk side. >> >> Best regards, >> >> Mario Staphorst >> >> ------------------------------------------------------------------------ >> Express yourself instantly with MSN Messenger! MSN Messenger >> <http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ What can you do with the new Windows Live? Find out http://www.microsoft.com/windows/windowslive/default.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090527/f7e90f30/attachment-0001.htm
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