asterisk1*CLI> sip debug SIP Debugging re-enabled Destroying call '50f21849787b1a5623e030a903be9e57@127.0.0.1' asterisk1*CLI> <-- SIP read from 192.168.11.111:63318: --- (0 headers 1 lines) --- Destroying call '5f0400ab79c91d7a1ae64fe763e60f9d@192.168.11.112' 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.11.113:5060: OPTIONS sip:64@192.168.11.113:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK339ae831;rport From: "Unknown" ;tag=as36206ca3 To: Contact: Call-ID: 2ce61a537ed698641c2b59d44dc4b934@192.168.11.112 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 13:23:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.11.113:5060: SIP/2.0 200 OK To: ;tag=d8edacfaa725db8ci0 From: "Unknown" ;tag=as36206ca3 Call-ID: 2ce61a537ed698641c2b59d44dc4b934@192.168.11.112 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK339ae831 Server: Linksys/SPA2102-3.3.6 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines) --- Destroying call '2ce61a537ed698641c2b59d44dc4b934@192.168.11.112' 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.11.113:5060: OPTIONS sip:62@192.168.11.113:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK46009d6b;rport From: "Unknown" ;tag=as4900cf24 To: Contact: Call-ID: 7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 13:23:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.11.113:5060: SIP/2.0 200 OK To: ;tag=f45df18a6757e03ci1 From: "Unknown" ;tag=as4900cf24 Call-ID: 7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK46009d6b Server: Linksys/SPA2102-3.3.6 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines) --- Destroying call '7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112' asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: INVITE sip:8*675148048384@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=e6204a9553045f22 To: Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 1 INVITE Contact: 63 Supported: timer Session-Expires: 180 Content-Type: application/sdp Content-Length: 194 v=0 o=- 1 0 IN IP4 192.168.11.114 s=- c=IN IP4 192.168.11.114 t=0 0 m=audio 49152 RTP/AVP 18 0 8 2 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726/8000 --- (12 headers 10 lines) --- Using INVITE request as basis request - 945cbc7a903ed9d4@192.168.11.114 Sending to 192.168.11.114 : 5060 (non-NAT) Reliably Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=e6204a9553045f22 To: ;tag=as67579b5b Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ece17b0" Content-Length: 0 --- Scheduling destruction of call '945cbc7a903ed9d4@192.168.11.114' in 15000 ms Found user '63' asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: ACK sip:8*675148048384@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=e6204a9553045f22 To: ;tag=as67579b5b Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 1 ACK Content-Length: 0 --- (8 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: INVITE sip:8*675148048384@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=e6204a9553045f22 To: Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 INVITE Contact: 63 Supported: timer Session-Expires: 180 Proxy-Authorization: Digest username="63", realm="asterisk", nonce="1ece17b0", opaque="", uri="sip:8*675148048384@192.168.11.112", response="cca8a821db56cf20691e0322216f4782" Content-Type: application/sdp Content-Length: 194 v=0 o=- 1 0 IN IP4 192.168.11.114 s=- c=IN IP4 192.168.11.114 t=0 0 m=audio 49152 RTP/AVP 18 0 8 2 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726/8000 --- (13 headers 10 lines) --- Using INVITE request as basis request - 945cbc7a903ed9d4@192.168.11.114 Sending to 192.168.11.114 : 5060 (NAT) Found user '63' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Peer audio RTP is at port 192.168.11.114:49152 Found description format G729 Found description format PCMU Found description format PCMA Found description format G726 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (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 8*675148048384 in from-internal (domain 192.168.11.112;user=phone) list_route: hop: Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=e6204a9553045f22 To: Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/63-09ab1cc8", "dialout-trunk|3|*675148048384||") in new stack -- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK=3") in new stack -- Executing Set("SIP/63-09ab1cc8", "_NODEST=") in new stack -- Executing Set("SIP/63-09ab1cc8", "DIAL_NUMBER=*675148048384") in new stack -- Executing Set("SIP/63-09ab1cc8", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/63-09ab1cc8", "GROUP()=OUT_3") in new stack -- Executing Macro("SIP/63-09ab1cc8", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/63-09ab1cc8", "user-callerid: device 63") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?report") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?start") in new stack -- Executing Set("SIP/63-09ab1cc8", "REALCALLERIDNUM=63") in new stack -- Executing NoOp("SIP/63-09ab1cc8", "REALCALLERIDNUM is 63") in new stack -- Executing Set("SIP/63-09ab1cc8", "AMPUSER=63") in new stack -- Executing Set("SIP/63-09ab1cc8", "AMPUSERCIDNAME=Catherine Laplante") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?report") in new stack -- Executing Set("SIP/63-09ab1cc8", "CALLERID(all)=Catherine Laplante <63>") in new stack -- Executing Set("SIP/63-09ab1cc8", "REALCALLERIDNUM=63") in new stack -- Executing NoOp("SIP/63-09ab1cc8", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/63-09ab1cc8", "Using CallerID "Catherine Laplante" <63>") in new stack -- Executing Macro("SIP/63-09ab1cc8", "record-enable|63|OUT") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/63-09ab1cc8", "recordingcheck|20070709-092340|1183987420.164") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070709-092340|1183987420.164: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/63-09ab1cc8", "No recording needed") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?skipoutcid") in new stack -- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing Macro("SIP/63-09ab1cc8", "outbound-callerid|3") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/63-09ab1cc8", "REALCALLERIDNUM is 63") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/63-09ab1cc8", "USEROUTCID=") in new stack -- Executing Set("SIP/63-09ab1cc8", "EMERGENCYCID=") in new stack -- Executing Set("SIP/63-09ab1cc8", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/63-09ab1cc8", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf("SIP/63-09ab1cc8", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/63-09ab1cc8", "CallerID set to "Catherine Laplante" <63>") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?nomax") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?chanfull") in new stack -- Executing DeadAGI("SIP/63-09ab1cc8", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/63-09ab1cc8", "OUTNUM=*675148048384") in new stack -- Executing Set("SIP/63-09ab1cc8", "custom=SIP/Acanac2") in new stack -- Executing GotoIf("SIP/63-09ab1cc8", "0?customtrunk") in new stack -- Executing Dial("SIP/63-09ab1cc8", "SIP/Acanac2/*675148048384|300|") in new stack We're at 192.168.11.112 port 11418 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 66.49.255.38:5060: INVITE sip:*675148048384@66.49.255.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK09d899b2;rport From: "Catherine Laplante" ;tag=as4bb6a3eb To: Contact: Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 13:23:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 5118 5118 IN IP4 192.168.11.112 s=session c=IN IP4 192.168.11.112 t=0 0 m=audio 11418 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called Acanac2/*675148048384 asterisk1*CLI> <-- SIP read from 66.49.255.38:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 24.200.182.199:5060;branch=z9hG4bK09d899b2;received=24.200.182.199;rport=5060 From: "Catherine Laplante" ;tag=as4bb6a3eb To: ;tag=as49c94d71 Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62b3a69a" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (no NAT) to 66.49.255.38:5060: ACK sip:*675148048384@66.49.255.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK09d899b2;rport From: "Catherine Laplante" ;tag=as4bb6a3eb To: ;tag=as49c94d71 Contact: Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.11.112 port 11418 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 66.49.255.38:5060: INVITE sip:*675148048384@66.49.255.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK58cfe938;rport From: "Catherine Laplante" ;tag=as4bb6a3eb To: Contact: Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="5146674639", realm="asterisk", algorithm=MD5, uri="sip:*675148048384@66.49.255.38", nonce="62b3a69a", response="863d0ebfd04278bc0a05f05575f37eb9", opaque="" Date: Mon, 09 Jul 2007 13:23:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 5118 5119 IN IP4 192.168.11.112 s=session c=IN IP4 192.168.11.112 t=0 0 m=audio 11418 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 66.49.255.38:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 24.200.182.199:5060;branch=z9hG4bK58cfe938;received=24.200.182.199;rport=5060 From: "Catherine Laplante" ;tag=as4bb6a3eb To: ;tag=as49c94d71 Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (no NAT) to 66.49.255.38:5060: ACK sip:*675148048384@66.49.255.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK58cfe938;rport From: "Catherine Laplante" ;tag=as4bb6a3eb To: ;tag=as49c94d71 Contact: Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/Acanac2-09b2b6d8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/63-09ab1cc8", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/63-09ab1cc8", "Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing Macro("SIP/63-09ab1cc8", "outisbusy|") in new stack -- Executing Playback("SIP/63-09ab1cc8", "all-circuits-busy-now|noanswer") in new stack We're at 192.168.11.112 port 12858 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=e6204a9553045f22 To: ;tag=as496c2560 Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 5118 5118 IN IP4 192.168.11.112 s=session c=IN IP4 192.168.11.112 t=0 0 m=audio 12858 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Playing 'all-circuits-busy-now' (language 'fr') Destroying call '3bb1953f4c35d621612dd8db4031d986@192.168.11.112' asterisk1*CLI> <-- SIP read from 66.49.255.42:5060: OPTIONS sip:5146674473@24.200.182.199:5060 SIP/2.0 Via: SIP/2.0/UDP 66.49.255.42:5060;branch=z9hG4bK5d59fe42;rport From: "asterisk" ;tag=as02b5530a To: Contact: Call-ID: 53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 08 Jul 2007 17:52:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- (13 headers 0 lines) --- Looking for 5146674473 in from-trunk (domain 24.200.182.199) Transmitting (no NAT) to 66.49.255.42:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.49.255.42:5060;branch=z9hG4bK5d59fe42;received=66.49.255.42;rport=5060 From: "asterisk" ;tag=as02b5530a To: ;tag=as787d3d86 Call-ID: 53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42' -- Executing Playback("SIP/63-09ab1cc8", "pls-try-call-later|noanswer") in new stack -- Playing 'pls-try-call-later' (language 'fr') 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.11.114:5060: OPTIONS sip:63@192.168.11.114:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK5253eae1;rport From: "Unknown" ;tag=as2e7eef05 To: Contact: Call-ID: 3233bd1c492363d178281c0e12b16a16@192.168.11.112 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Jul 2007 13:23:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: CANCEL sip:8*675148048384@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=e6204a9553045f22 To: Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 CANCEL Content-Length: 0 --- (8 headers 0 lines) --- Sending to 192.168.11.114 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 487 Request Terminated ia: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=e6204a9553045f22 To: ;tag=as496c2560 Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number --- Transmitting (NAT) to 192.168.11.114:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114 From: 63 ;tag=e6204a9553045f22 To: ;tag=as496c2560 Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number --- == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/63-09ab1cc8' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/63-09ab1cc8' asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK5253eae1;rport= From: "Unknown" ;tag=as2e7eef05 To: ;tag=95f741a83b81a040 Call-ID: 3233bd1c492363d178281c0e12b16a16@192.168.11.112 CSeq: 102 OPTIONS Content-Type: application/sdp Content-Length: 190 v=0 o=- 1 0 IN IP4 192.168.11.114 s=- c=IN IP4 192.168.11.114 t=0 0 m=audio 0 RTP/AVP 18 0 8 2 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726/8000 --- (8 headers 10 lines) --- Destroying call '3233bd1c492363d178281c0e12b16a16@192.168.11.112' asterisk1*CLI> <-- SIP read from 192.168.11.114:5060: ACK sip:8*675148048384@192.168.11.112;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.11.114:5060 Max-Forwards: 70 From: 63 ;tag=e6204a9553045f22 To: ;tag=as496c2560 Call-ID: 945cbc7a903ed9d4@192.168.11.114 CSeq: 2 ACK Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '945cbc7a903ed9d4@192.168.11.114' asterisk1*CLI> quit