<--- SIP read from 64.158.177.99:5060 ---> INVITE sip:2482467007@200.xx.36.42 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: From: ;tag=3416263034-861359 Remote-Party-Id: ;privacy=off Call-ID: 1747302-3416263034-861223@is02.broadvox.net CSeq: 1 INVITE Via: SIP/2.0/UDP 64.158.177.99:5060;branch=e9f4d0bea6c5385e4e4d8a41c3f5b20a Contact: sip:5868735246@64.158.177.99:5060 Content-Type: application/sdp Content-Length: 324 v=0 o=Version 1234 17934 IN IP4 64.158.162.74 s=sip call c=IN IP4 64.158.162.71 t=0 0 m=audio 26350 RTP/AVP 0 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no;bitrate=6.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 <-------------> --- (13 headers 15 lines) --- Sending to 64.158.177.99 : 5060 (no NAT) Using INVITE request as basis request - 1747302-3416263034-861223@is02.broadvox.net Found peer 'broadvox' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 64.158.162.71:26350 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format G723 for ID 4 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.158.162.71:26350 Looking for 2482467007 in hosted_in (domain 200.xx.36.42) list_route: hop: sip1*CLI> <--- Transmitting (no NAT) to 64.158.177.99:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.158.177.99:5060;branch=e9f4d0bea6c5385e4e4d8a41c3f5b20a;received=64.158.177.99 From: ;tag=3416263034-861359 To: Call-ID: 1747302-3416263034-861223@is02.broadvox.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2482467007@hosted_in:1] Dial("SIP/broadvox-0086ce20", "SIP/02482957753") in new stack [2008-04-03 22:00:43] WARNING[6472]: chan_sip.c:3008 sip_call: No audio format found to offer. Cancelling call to 02482957753 -- Couldn't call 02482957753