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