-- Executing Dial("SIP/cftest-5bd6", "Sip/5520/199") in new stack We're at x.x.x.x port 6876 Answering/Requesting with root capability 0x100 (g729) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting (NAT) to y.y.y.y:1030: INVITE sip:199@y.y.y.y:1030 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a;rport From: "cftest" ;tag=as17a89309 To: Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 08 Sep 2005 23:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 241 v=0 o=root 30921 30921 IN IP4 x.x.x.x s=session c=IN IP4 x.x.x.x t=0 0 m=audio 6876 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 5520/199 pbx*CLI> <-- SIP read from y.y.y.y:1030: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a From: "cftest" ;tag=as17a89309 To: ;tag=as30a3da9f Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5f3ba829" Content-Length: 0 --- (11 headers 0 lines)--- Transmitting (NAT) to y.y.y.y:1030: ACK sip:199@y.y.y.y:1030 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a;rport From: "cftest" ;tag=as17a89309 To: ;tag=as30a3da9f Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- We're at x.x.x.x port 6876 Answering/Requesting with root capability 0x100 (g729) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT) to y.y.y.y:1030: INVITE sip:199@192.168.24.50 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK25ffcf5d;rport From: "cftest" ;tag=as17a89309 To: Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="5520", realm="asterisk", algorithm=MD5, uri="sip:199@192.168.24.50", nonce="5f3ba829", response="c6b73a7db1051fc24eaf14c3ab0a67db", opaque="" Date: Thu, 08 Sep 2005 23:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 241 v=0 o=root 30921 30922 IN IP4 x.x.x.x s=session c=IN IP4 x.x.x.x t=0 0 m=audio 6876 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from y.y.y.y:1030: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK25ffcf5d From: "cftest" ;tag=as17a89309 To: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/5520-6e12 is ringing pbx*CLI> <-- SIP read from y.y.y.y:1030: SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK25ffcf5d From: "cftest" ;tag=as17a89309 To: ;tag=as0f21485d Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 31882 31882 IN IP4 192.168.24.50 s=session c=IN IP4 y.y.y.y t=0 0 m=audio 14702 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 12 lines)--- Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port y.y.y.y:14702 Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.24.50, port 5060 Transmitting (NAT) to y.y.y.y:1030: ACK sip:199@192.168.24.50 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK5f99d059;rport From: "cftest" ;tag=as17a89309 To: ;tag=as0f21485d Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/5520-6e12 answered SIP/cftest-5bd6 -- Attempting native bridge of SIP/cftest-5bd6 and SIP/5520-6e12