On Wed, Apr 16, 2008 at 9:10 AM, broadband Voice <[EMAIL PROTECTED]> wrote: > We have two servers but looks like G729 issues. Works fine on the old server > and not sure if it is T1 related. See SIP Debug. Any experiences to share. > Thanks > > --- > Newark1*CLI> > <--- SIP read from 194.xx.Xx.Xx:5060 ---> > SIP/2.0 183 Session progress > Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=xxxxK784d2637;rport > From: "Cell Phone DC" <sip:[EMAIL PROTECTED]>;tag=as04819ca3 > To: <sip:xx>;tag=xx > Contact: sip:[EMAIL PROTECTED]:5060 > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Type: application/sdp > Content-Length: 198 > > v=0 > o=xxxxxx 12xxxxx 12xxxx IN IP4 62.xx.xx.xx > s=SIP Call > c=IN IP4 62.xx.xx.xxx > t=0 0 > m=audio 8786 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > <-------------> > --- (11 headers 9 lines) --- > Found RTP audio format 0 > Found RTP audio format 101 > Peer audio RTP is at port 62.xx.xx.xx:8786 > Found audio description format PCMU for ID 0 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 > (nothing), combined - 0x4 (ulaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 62.xx.xx.xx:8786 > -- SIP/Voicetrading-08e1ce18 is making progress passing it to Zap/5-1
Looks to be OK to me but you have negotiated Ulaw not G729. Thanks, Steve Totaro _______________________________________________ -- 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
