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<body class='hmmessage'><div style="text-align: left;">Thanks, in my case though it looks like the originating party (polycom softphone) is hearing a clipped voicemail prompt because of that; should I look into having that fixed into their firmware? As a workaround, I was thinking to just add a few seconds delay in app_voicemail, or wait through AGI before calling voicemail, makes sense?<br></div><br><br><br><hr id="stopSpelling">> Date: Wed, 1 Oct 2008 15:43:37 +0100<br>> From: greymanvoip@gmail.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)<br>> <br>> ><br>> > CLI output does not show any error that I see. Is there any reason other<br>> > than a SIP 183 that would trigger this and isn't asterisk supposed to<br>> > ACK/answer the channel before playing any prompt?<br>> ><br>> <br>> Asterisk wil start the audio as soon as it sends back the 200 Ok<br>> response it doesn't wait for the ACK. Most SIP servers will work like<br>> that. The matching of ACK requests to a SIP transaction is not a<br>> particulalrly robust mechanism (for instance if a user agent puts its<br>> IP address in the Call-ID and a SIP ALG fiddles with the SIP packet<br>> for INVITEs but ignores ACKs then there will be a mismatch. This<br>> happens more frequently then you would think) so sending RTP after an<br>> OK response is the correct thing to do.<br>> <br>> I think Asterisk will actually cut off the call after 32s if it<br>> doesn't get an ACK which is not such a great idea but that may have<br>> been changed in later versions. The arrival of an RTP packet from the<br>> remote end should be used as the definitive indication of an answered<br>> call not the ACK.<br>> <br>> Regards,<br>> <br>> Greyman.<br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>> Register Now: http://www.astricon.net<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></body>
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