Kamlesh Kumar wrote: > > SIP.conf > [100] > username=100 > secret=password > type=friend > host=dynamic > nat=yes > canreinvite=no > insecure=port > disallow=all > allow=ulaw > allow=alaw > allow=g729 > context=asterisk > qualify=no
Is there also an "allow=g729" line in sip.conf for the ITSP's SIP peer? > SIP Trace: > 201.xxx.xxx.xxx = SIP Softphone which originates the call > xxx.xxx.xxx.xxx = Asterisk server > yyy.yyy.yyy.yyy = ITSP > > ... > > <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---> > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 > From: "100" <sip:[email protected]>;tag=as643c20b1 > To: <sip:[email protected]>;tag=gK029aaa8c > Call-ID: [email protected] > CSeq: 102 INVITE > Contact: <sip:[email protected]:5060> > Allow: > INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS > Content-Length: 234 > Content-Disposition: session; handling=required > Content-Type: application/sdp > v=0 > o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy > s=SIP Media Capabilities > c=IN IP4 zzz.zzz.zzz.zzz > t=0 0 > m=audio 21996 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=maxptime:20 > <-------------> > [Jun 3 13:11:31] --- (11 headers 11 lines) --- > [Jun 3 13:11:31] Found RTP audio format 0 > [Jun 3 13:11:31] Found RTP audio format 101 > [Jun 3 13:11:31] Found audio description format PCMU for ID 0 > [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 > [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 > (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) > [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), > peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) > [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 > [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress > passing it to SIP/100-000034d8 > [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 > [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP > [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
