*CLI> <--- SIP read from 10.200.201.2:5061 ---> INVITE sip:6000089@10.156.1.242;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3 Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 From: ;tag=1d567e44 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: 296 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 135538 135538 IN IP4 10.200.201.2 s=Sip Call c=IN IP4 10.200.201.3 t=0 0 m=audio 10776 RTP/AVP 8 0 18 4 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=yes <-------------> --- (12 headers 13 lines) --- Sending to 10.200.201.2 : 5061 (no NAT) Using INVITE request as basis request - 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 Found peer 'soft-x-ngn' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.200.201.3:10776 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format G723 for ID 4 Found description format telephone-event for ID 97 Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/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 10.200.201.3:10776 Looking for 6000089 in ivr-1-in (domain 10.156.1.242) list_route: hop: <--- Transmitting (no NAT) to 10.200.201.2:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3;received=10.200.201.2 From: ;tag=1d567e44 To: Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 10.156.1.242 port 16558 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.200.201.2:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3;received=10.200.201.2 From: ;tag=1d567e44 To: ;tag=as4a040f32 Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 237 v=0 o=root 13496 13496 IN IP4 10.156.1.242 s=session c=IN IP4 10.156.1.242 t=0 0 m=audio 16558 RTP/AVP 0 97 a=rtpmap:0 PCMU/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 10.200.201.2:5061 ---> ACK sip:6000089@10.156.1.242 SIP/2.0 Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK1e7385b15 Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK1e7385b15 Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 From: ;tag=1d567e44 To: ;tag=as4a040f32 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.201.2:5061 ---> BYE sip:6000089@10.156.1.242 SIP/2.0 Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK566bd1b8f Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 From: ;tag=1d567e44 To: ;tag=as4a040f32 CSeq: 2 BYE Reason: Q.850;cause=100;text="unknown" Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Sending to 10.200.201.2 : 5061 (no NAT) <--- Transmitting (no NAT) to 10.200.201.2:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK566bd1b8f;received=10.200.201.2 From: ;tag=1d567e44 To: ;tag=as4a040f32 Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (ivr-1-in, 6000089, 2) exited non-zero on 'SIP/10.200.201.2-007efe80' Really destroying SIP dialog '2297cac624f0358f0e5a61e64517a4ec@10.200.201.2' Method: BYE