[asterisk-users] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
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=K784d2637;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=xx 12x 12 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 ___ -- 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
Re: [asterisk-users] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
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=K784d2637;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=xx 12x 12 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
