carousel*CLI> -- Starting simple switch on 'Zap/1-1' -- Executing SetCallerID("Zap/1-1", "Incoming <>") in new stack -- Executing Macro("Zap/1-1", "stdexten_motors|269|SIP/269|Incoming") in new stack -- Executing NoOp("Zap/1-1", "Ext Cld 269 Ringing SIP/269 CID Incoming ExtCld From ") in new stack -- Executing SetMusicOnHold("Zap/1-1", "motorsMOH") in new stack -- Executing Dial("Zap/1-1", "SIP/269|20||rtT") in new stack We're at 192.168.168.123 port 28810 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (no NAT) to 192.168.168.78:50710: INVITE sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK114b52fd;rport From: "Incoming" ;tag=as07c979fe To: Contact: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 29 Aug 2006 04:04:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 347 v=0 o=root 3550 3550 IN IP4 192.168.168.123 s=session c=IN IP4 192.168.168.123 t=0 0 m=audio 28810 RTP/AVP 0 3 97 111 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 269 carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK114b52fd;rport From: "Incoming" ;tag=as07c979fe To: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Content-Length: 0 --- (8 headers 0 lines)--- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK114b52fd;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/269-09d93338 is ringing carousel*CLI> <-- SIP read from 192.168.168.78:50710: --- (0 headers 0 lines) Nat keepalive --- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK114b52fd;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=269 8000 8000 IN IP4 192.168.168.78 s=SIP Call c=IN IP4 192.168.168.78 t=0 0 m=audio 7898 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.168.78:7898 Found description format PCMU Found description format telephone-event Capabilities: us - 0x516 (gsm|ulaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/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 192.168.168.78, port 50710 Transmitting (no NAT) to 192.168.168.78:50710: ACK sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK5e86780d;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Contact: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/269-09d93338 answered Zap/1-1 carousel*CLI> <-- SIP read from 192.168.168.78:50710: INVITE sip:asterisk@192.168.168.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.78:50710;branch=z9hG4bK1e0ce2ee6cf9a2c4 From: ;tag=8ad686f32f09bfbe To: "Incoming" ;tag=as07c979fe Contact: Supported: replaces, timer Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 40868 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 311 v=0 o=269 8000 8001 IN IP4 192.168.168.78 s=SIP Call c=IN IP4 192.168.168.78 t=0 0 m=audio 7898 RTP/AVP 0 8 4 18 3 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 15 lines)--- Using INVITE request as basis request - 32ad2fe561bc5d02568901572a95c465@192.168.168.123 Sending to 192.168.168.78 : 50710 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.168.78:7898 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Found description format telephone-event Capabilities: us - 0x516 (gsm|ulaw|g726|g729|ilbc), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'motorsMOH', on channel 'Zap/1-1' We're at 192.168.168.123 port 28810 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.78:50710;branch=z9hG4bK1e0ce2ee6cf9a2c4;received=192.168.168.78 From: ;tag=8ad686f32f09bfbe To: "Incoming" ;tag=as07c979fe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 40868 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 3550 3551 IN IP4 192.168.168.123 s=session c=IN IP4 192.168.168.123 t=0 0 m=audio 28810 RTP/AVP 0 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- carousel*CLI> <-- SIP read from 192.168.168.78:50710: ACK sip:asterisk@192.168.168.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.78:50710;branch=z9hG4bK8bba70fb3228a632 From: ;tag=8ad686f32f09bfbe To: "Incoming" ;tag=as07c979fe Contact: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 40868 ACK User-Agent: Grandstream GXP2000 1.1.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- carousel*CLI> <-- SIP read from 192.168.168.78:50710: REFER sip:asterisk@192.168.168.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.78:50710;branch=z9hG4bK1ab6c90b4c21de9c From: ;tag=8ad686f32f09bfbe To: "Incoming" ;tag=as07c979fe Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 40869 REFER User-Agent: Grandstream GXP2000 1.1.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (14 headers 0 lines)--- Transfer to *700 in motors Transfer from 269 in motors -- Stopped music on hold on Zap/1-1 Transmitting (no NAT) to 192.168.168.78:50710: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.78:50710;branch=z9hG4bK1ab6c90b4c21de9c;received=192.168.168.78 From: ;tag=8ad686f32f09bfbe To: "Incoming" ;tag=as07c979fe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 40869 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.168.78, port 50710 Reliably Transmitting (no NAT) to 192.168.168.78:50710: NOTIFY sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK440b2c53;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Contact: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=40869 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.168.78, port 50710 Reliably Transmitting (no NAT) to 192.168.168.78:50710: BYE sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK0a36bba1;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Contact: Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- == Spawn extension (motors, *700, 0) exited non-zero on 'Zap/1-1' in macro 'stdexten_motors' == Spawn extension (motors, *700, 0) exited non-zero on 'Zap/1-1' -- Executing ParkAndAnnounce("Zap/1-1", "pbx-transfer:PARKED|9|SIP/269|parkedcallstimeout|s|1") in new stack -- Dial Tech,String: (SIP,269) -- Return Context: (parkedcallstimeout,s,1) ID: (null) -- Started music on hold, class 'motorsMOH', on channel 'Zap/1-1' == Parked Zap/1-1 on 701. Will timeout back to extension [parkedcallstimeout] s, 1 in 9 seconds -- Added extension '701' priority 1 to parkedcalls -- Call Parking Called, lot: 701, timeout: 9000, context: parkedcallstimeout We're at 192.168.168.123 port 30132 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (no NAT) to 192.168.168.78:50710: INVITE sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK4bf6feac;rport From: "asterisk" ;tag=as06f4a40b To: Contact: Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 29 Aug 2006 04:04:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 347 v=0 o=root 3550 3550 IN IP4 192.168.168.123 s=session c=IN IP4 192.168.168.123 t=0 0 m=audio 30132 RTP/AVP 0 3 97 111 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK440b2c53;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK0a36bba1;rport From: "Incoming" ;tag=as07c979fe To: ;tag=8ad686f32f09bfbe Call-ID: 32ad2fe561bc5d02568901572a95c465@192.168.168.123 CSeq: 104 BYE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '32ad2fe561bc5d02568901572a95c465@192.168.168.123' carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK4bf6feac;rport From: "asterisk" ;tag=as06f4a40b To: Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Content-Length: 0 --- (8 headers 0 lines)--- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK4bf6feac;rport From: "asterisk" ;tag=as06f4a40b To: ;tag=9de3c08fdd6b1816 Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (10 headers 0 lines)--- carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK4bf6feac;rport From: "asterisk" ;tag=as06f4a40b To: ;tag=9de3c08fdd6b1816 Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=269 8000 8000 IN IP4 192.168.168.78 s=SIP Call c=IN IP4 192.168.168.78 t=0 0 m=audio 7898 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.168.78:7898 Found description format PCMU Found description format telephone-event Capabilities: us - 0x516 (gsm|ulaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/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 192.168.168.78, port 50710 Transmitting (no NAT) to 192.168.168.78:50710: ACK sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK7c32a7ce;rport From: "asterisk" ;tag=as06f4a40b To: ;tag=9de3c08fdd6b1816 Contact: Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- > Channel SIP/269-09cea1b8 was answered. > Announce Template:pbx-transfer:PARKED > Announce:pbx-transfer -- Playing 'pbx-transfer' (language 'en') > Announce:PARKED -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.168.78, port 50710 Reliably Transmitting (no NAT) to 192.168.168.78:50710: BYE sip:269@192.168.168.78:50710;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK422ed04f;rport From: "asterisk" ;tag=as06f4a40b To: ;tag=9de3c08fdd6b1816 Contact: all-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (parkedcallstimeout, s, 1) exited non-zero on 'Parked/Zap/1-1' carousel*CLI> <-- SIP read from 192.168.168.78:50710: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.123:5060;branch=z9hG4bK422ed04f;rport From: "asterisk" ;tag=as06f4a40b To: ;tag=9de3c08fdd6b1816 Call-ID: 07a5fc796842832b44d982c472e850eb@192.168.168.123 CSeq: 103 BYE User-Agent: Grandstream GXP2000 1.1.1.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '07a5fc796842832b44d982c472e850eb@192.168.168.123' -- Stopped music on hold on Zap/1-1 == Timeout for Zap/1-1 parked on 701. Returning to parkedcallstimeout,s,1 -- Executing NoOp("Zap/1-1", "user was parked on parkingslot #") in new stack -- Executing Playback("Zap/1-1", "tt-monkeys") in new stack -- Playing 'tt-monkeys' (language 'en') == Spawn extension (parkedcallstimeout, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' carousel*CLI>