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<body class='hmmessage'><div dir='ltr'>Matthew,<BR> <BR>allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. In case g729 on asterisk box, as I mentioned earlier, call even doesn't go out of the asterisk box. Below extracts from log also indicate the same thing. <BR> <BR>[Jun 5 12:46:49] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)<br>[Jun 5 12:46:49] == Using SIP RTP CoS mark 5<br><strong>[Jun 5 12:46:49] -- Couldn't call yyy.yyy.yyy.yyy/12127773456</strong><br>[Jun 5 12:46:49] == Everyone is busy/congested at this time (0:0/0/0)<br><br>Regards,<BR>Kamlesh <BR> <BR><div>> Date: Tue, 4 Jun 2013 10:27:11 -0500<br>> From: mroth@imminc.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode<br>> <br>> Kamlesh Kumar wrote:<br>> > <br>> > SIP.conf<br>> > [100]<br>> > username=100<br>> > secret=password<br>> > type=friend<br>> > host=dynamic<br>> > nat=yes<br>> > canreinvite=no<br>> > insecure=port<br>> > disallow=all<br>> > allow=ulaw<br>> > allow=alaw<br>> > allow=g729<br>> > context=asterisk<br>> > qualify=no<br>> <br>> Is there also an "allow=g729" line in sip.conf for the ITSP's SIP peer?<br>> <br>> > SIP Trace: <br>> > 201.xxx.xxx.xxx = SIP Softphone which originates the call <br>> > xxx.xxx.xxx.xxx = Asterisk server <br>> > yyy.yyy.yyy.yyy = ITSP <br>> > <br>> > ...<br>> > <br>> > <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---><br>> > SIP/2.0 183 Session Progress<br>> > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060<br>> > From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>> > To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>> > Call-ID: 07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx<br>> > CSeq: 102 INVITE<br>> > Contact: <sip:12127773456@yyy.yyy.yyy.yyy:5060><br>> > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS<br>> > Content-Length: 234<br>> > Content-Disposition: session; handling=required<br>> > Content-Type: application/sdp<br>> > v=0<br>> > o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy<br>> > s=SIP Media Capabilities<br>> > c=IN IP4 zzz.zzz.zzz.zzz<br>> > t=0 0<br>> > m=audio 21996 RTP/AVP 0 101<br>> > a=rtpmap:0 PCMU/8000<br>> > a=rtpmap:101 telephone-event/8000<br>> > a=fmtp:101 0-15<br>> > a=sendrecv<br>> > a=maxptime:20<br>> > <-------------><br>> > [Jun 3 13:11:31] --- (11 headers 11 lines) ---<br>> > [Jun 3 13:11:31] Found RTP audio format 0<br>> > [Jun 3 13:11:31] Found RTP audio format 101<br>> > [Jun 3 13:11:31] Found audio description format PCMU for ID 0<br>> > [Jun 3 13:11:31] Found audio description format telephone-event for ID 101<br>> > [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>> > [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>> > [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996<br>> > [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress passing it to SIP/100-000034d8<br>> > [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042<br>> > [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP<br>> > [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP<br>> <br>> This response from the ITSP says that only u-law may be used for the call.<br>> Please contact the ITSP and confirm that they actually support the G.729 codec.<br>> <br>> Regards,<br>> <br>> Matthew Roth<br>> InterMedia Marketing Solutions<br>> Software Engineer and Systems Developer<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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