o=HuaweiSoftX3000 2410258 2410260 IN IP4 200.57.0.85 s=Sip Call c=IN IP4 200.57.0.139 t=0 0 m=image 7328 udptl t38 m=audio 7328 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax --- (10 headers 12 lines)--- Using INVITE request as basis request - 3671d21d4f02b0e37143b82c6f164d01@200.57.30.247 Sending to 200.57.0.85 : 5060 (non-NAT) Apr 3 19:05:49 WARNING[26531]: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 7328 udptl t38 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 200.57.0.139:7328 Found description format PCMA Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 200.57.30.247 port 17596 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 200.57.0.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bK4716a1d21;received=200.57.0.85 From: ;tag=41e3d4b8 To: "voice-1" ;tag=as39a050f5 Call-ID: 3671d21d4f02b0e37143b82c6f164d01@200.57.30.247 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 26524 26525 IN IP4 200.57.30.247 s=session c=IN IP4 200.57.30.247 t=0 0 m=audio 17596 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- unknown*CLI> <-- SIP read from 200.57.0.85:5060: ACK sip:3310861401@200.57.30.247 SIP/2.0 Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bK71a3bc30a Call-ID: 3671d21d4f02b0e37143b82c6f164d01@200.57.30.247 From: ;tag=41e3d4b8 To: "voice-1" ;tag=as39a050f5 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 --- (8 headers 0 lines)--- unknown*CLI> <-- SIP read from 200.57.0.85:5060: BYE sip:3310861400@200.57.30.247 SIP/2.0 Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKfafe340c8 Call-ID: 03f1c8d971fa30090957a3cb28f6fef3@200.57.0.85 From: ;tag=28f6fef3 To: ;tag=as4e778d78 CSeq: 2 BYE Reason: Q.850;cause=16;text="normal call clearing" Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines)--- Sending to 200.57.0.85 : 5060 (non-NAT) Transmitting (no NAT) to 200.57.0.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKfafe340c8;received=200.57.0.85 From: ;tag=28f6fef3 To: ;tag=as4e778d78 Call-ID: 03f1c8d971fa30090957a3cb28f6fef3@200.57.0.85 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Destroying call '03f1c8d971fa30090957a3cb28f6fef3@200.57.0.85' unknown*CLI> <-- SIP read from 200.57.0.85:5060: INVITE sip:3310861400@200.57.30.247;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKb85bb3edc Call-ID: 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 From: ;tag=b85bb3ed To: CSeq: 1 INVITE Contact: Supported: 100rel Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER Content-Length: 244 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 2410484 2410484 IN IP4 200.57.0.85 s=Sip Call c=IN IP4 200.57.0.139 t=0 0 m=audio 13900 RTP/AVP 8 0 18 4 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=fmtp:18 annexb=yes --- (12 headers 11 lines)--- Using INVITE request as basis request - 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 Sending to 200.57.0.85 : 5060 (non-NAT) Found peer 'softswitch' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Peer audio RTP is at port 200.57.0.139:13900 Found description format PCMA Found description format PCMU Found description format G729 Found description format G723 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 3310861400 in sip (domain 200.57.30.247) list_route: hop: Transmitting (no NAT) to 200.57.0.85:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKb85bb3edc;received=200.57.0.85 From: ;tag=b85bb3ed To: Call-ID: 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Macro("SIP/200.57.0.85-0822ff80", "ccm-next|3310861400") in new stack -- Executing Dial("SIP/200.57.0.85-0822ff80", "SIP/3310861400@callmanager|60|tr") in new stack -- Called 3310861400@callmanager Transmitting (no NAT) to 200.57.0.85:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKb85bb3edc;received=200.57.0.85 From: ;tag=b85bb3ed To: ;tag=as02f23bf0 Call-ID: 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 unknown*CLI> --- -- SIP/callmanager-221d is ringing -- SIP/callmanager-221d answered SIP/200.57.0.85-0822ff80 We're at 200.57.30.247 port 17500 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 200.57.0.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bKb85bb3edc;received=200.57.0.85 From: ;tag=b85bb3ed To: ;tag=as02f23bf0 Call-ID: 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 26524 26524 IN IP4 200.57.30.247 s=session c=IN IP4 200.57.30.247 t=0 0 m=audio 17500 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/200.57.0.85-0822ff80 and SIP/callmanager-221d unknown*CLI> <-- SIP read from 200.57.0.85:5060: ACK sip:3310861400@200.57.30.247 SIP/2.0 Via: SIP/2.0/UDP 200.57.0.85:5060;branch=z9hG4bK24669b89e Call-ID: 2f9fb46e8fe6942ba6acf037b85bb3ed@200.57.0.85 From: ;tag=b85bb3ed To: ;tag=as02f23bf0 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 set_destination: Parsing for address/port to send to set_destination: set destination to 200.57.0.85, port 5060 Reliably Transmitting (no NAT) to 200.57.0.85:5060: BYE sip:33294810575843@200.57.0.85;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK64e046af;rport From: "voice-1" ;tag=as39a050f5 To: ;tag=41e3d4b8 Contact: Call-ID: 3671d21d4f02b0e37143b82c6f164d01@200.57.30.247 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- unknown*CLI> <-- SIP read from 200.57.0.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK64e046af;rport=5060 Call-ID: 3671d21d4f02b0e37143b82c6f164d01@200.57.30.247 From: "voice-1" ;tag=as39a050f5 To: ;tag=41e3d4b8 CSeq: 103 BYE Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '3671d21d4f02b0e37143b82c6f164d01@200.57.30.247' -- Executing Dial("SIP/3310861492-99e1", "SIP/332948015555456684@bestel|60|tr") in new stack We're at 200.57.30.247 port 16762 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP 13 headers, 9 lines Reliably Transmitting (no NAT) to 200.57.0.85:5060: INVITE sip:332948015555456684@200.57.0.85 SIP/2.0 Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK198e6cbb;rport From: "3310860501" ;tag=as6824e20e To: Contact: Call-ID: 5181ac5c506e8797273b73ee63c5e040@200.57.30.247 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 04 Apr 2006 00:07:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 186 v=0 o=root 26524 26524 IN IP4 200.57.30.247 s=session c=IN IP4 200.57.30.247 t=0 0 m=audio 16762 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Called 332948015555456684@bestel unknown*CLI> <-- SIP read from 200.57.0.85:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK198e6cbb;rport=5060 Call-ID: 5181ac5c506e8797273b73ee63c5e040@200.57.30.247 From: "3310860501" ;tag=as6824e20e To: CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines)--- unknown*CLI> <-- SIP read from 200.57.0.85:5060: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK198e6cbb;rport=5060 Call-ID: 5181ac5c506e8797273b73ee63c5e040@200.57.30.247 From: "3310860501" ;tag=as6824e20e To: ;tag=de3cc709 CSeq: 102 INVITE Reason: Q.850;cause=17;text="user busy" Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 486 "Busy Here" back from 200.57.0.85 Transmitting (no NAT) to 200.57.0.85:5060: ACK sip:332948015555456684@200.57.0.85 SIP/2.0 Via: SIP/2.0/UDP 200.57.30.247:5060;branch=z9hG4bK198e6cbb;rport From: "3310860501" ;tag=as6824e20e To: ;tag=de3cc709 Contact: Call-ID: 5181ac5c506e8797273b73ee63c5e040@200.57.30.247 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/bestel-17e8 is busy == Everyone is busy/congested at this time (1:1/0/0) Destroying call '5181ac5c506e8797273b73ee63c5e040@200.57.30.247' Apr 3 19:07:19 WARNING[11181]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'casas' unknown*CLI>