sip*CLI> sip set debug peer tassi2 SIP Debugging Enabled for IP: 192.168.2.125:5060 -- Accepting call from '00402341350' to '0399168263' on channel 0/5, span 1 -- Executing [0399168263@main:1] Set("Zap/5-1", "CALLERID(num)=0402341350") in new stack -- Executing [0399168263@main:2] Dial("Zap/5-1", "Local/263@extensions") in new stack -- Called 263@extensions -- Executing [263@extensions:1] Macro("Local/263@extensions-935d,2", "stdexten|263|SIP/tassi&SIP/tassi2") in new stack -- Executing [s@macro-stdexten:1] GotoIf("Local/263@extensions-935d,2", "1?2:4") in new stack -- Goto (macro-stdexten,s,2) -- Executing [s@macro-stdexten:2] GotoIf("Local/263@extensions-935d,2", "0?4:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing [s@macro-stdexten:3] SIPAddHeader("Local/263@extensions-935d,2", "Alert-Info: External") in new stack -- Executing [s@macro-stdexten:4] Dial("Local/263@extensions-935d,2", "SIP/tassi&SIP/tassi2|30") in new stack [Jan 8 16:51:22] WARNING[9721]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Video is at 192.168.2.85 port 19598 Audio is at 192.168.2.85 port 17364 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.2.125:5060: INVITE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK29297e78;rport From: "0402341350" ;tag=as5283f296 To: Contact: Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 08 Jan 2007 05:51:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Alert-Info: External Content-Type: application/sdp Content-Length: 418 v=0 o=root 9309 9309 IN IP4 192.168.2.85 s=session c=IN IP4 192.168.2.85 b=CT:384 t=0 0 m=audio 17364 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 19598 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv --- -- Called tassi2 sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK29297e78;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F CSeq: 102 INVITE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK29297e78;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F CSeq: 102 INVITE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/tassi2-0833b750 is ringing -- Local/263@extensions-935d,1 is ringing sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK29297e78;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F CSeq: 102 INVITE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Type: application/sdp Content-Length: 320 v=0 o=- 1168235492 1168235492 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2252 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 31 34 103 a=inactive a=rtpmap:31 h261/90000 a=rtpmap:34 h263/90000 a=rtpmap:103 h263-1998/90000 <-------------> --- (11 headers 14 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found RTP video format 31 Found RTP video format 34 Found RTP video format 103 Peer audio RTP is at port 192.168.2.125:2252 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Found description format h261 for ID 31 Found description format h263 for ID 34 Found description format h263-1998 for ID 103 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x1c0008 (alaw|h261|h263|h263p)/video=0x1c0000 (h261|h263|h263p), combined - 0x1c0008 (alaw|h261|h26 3|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2252 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Transmitting (no NAT) to 192.168.2.125:5060: ACK sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK2835040e;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F Contact: Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/tassi2-0833b750 answered Local/263@extensions-935d,2 -- Local/263@extensions-935d,1 stopped sounds -- Local/263@extensions-935d,1 answered Zap/5-1 == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Local/263@extensions-935d,2' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Local/263@extensions-935d,2' sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:0402341350@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK71ea96caC3245D81 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 CSeq: 1 INVITE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1168235492 1168235493 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2252 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 9 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.125:2252 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2252 Audio is at 192.168.2.85 port 17364 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK71ea96caC3245D81;received=192.168.2.125 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 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: 238 v=0 o=root 9309 9310 IN IP4 192.168.2.85 s=session c=IN IP4 192.168.2.85 t=0 0 m=audio 17364 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on Zap/5-1 sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> ACK sip:0402341350@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK5bc038a459FB7723 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 CSeq: 1 ACK Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:256@asterisk.starrez.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKecf484ae7B83D0B5 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: CSeq: 1 INVITE Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1168235499 1168235499 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Using INVITE request as basis request - adee168-af0f82b7-88a1c152@192.168.2.125 <--- Reliably Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKecf484ae7B83D0B5;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: ;tag=as6417d182 Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="614589c1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'adee168-af0f82b7-88a1c152@192.168.2.125' in 32000 ms (Method: INVITE) Found user 'tassi2' sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> ACK sip:256@asterisk.starrez.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKecf484ae7B83D0B5 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: ;tag=as6417d182 CSeq: 1 ACK Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:256@asterisk.starrez.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKa97fde0b3496D0B6 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: CSeq: 2 INVITE Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="tassi2", realm="asterisk", nonce="614589c1", uri="sip:256@asterisk.starrez.com:5060;user=phone", response="f28e75eb9e3b382329e1d1a64fcdb687", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1168235499 1168235499 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 =sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Using INVITE request as basis request - adee168-af0f82b7-88a1c152@192.168.2.125 Found user 'tassi2' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.125:2222 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2222 Peer video RTP is at port 192.168.2.125:12296 Looking for 256 in extensions (domain asterisk.starrez.com) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKa97fde0b3496D0B6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [256@extensions:1] Macro("SIP/tassi2-b7706aa8", "stdexten|256|SIP/ashleyc1") in new stack -- Executing [s@macro-stdexten:1] GotoIf("SIP/tassi2-b7706aa8", "0?2:4") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Dial("SIP/tassi2-b7706aa8", "SIP/ashleyc1|30") in new stack -- Called ashleyc1 -- SIP/ashleyc1-08342228 is ringing <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKa97fde0b3496D0B6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: ;tag=as4d4afc79 Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- SIP/ashleyc1-08342228 answered SIP/tassi2-b7706aa8 Audio is at 192.168.2.85 port 17536 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKa97fde0b3496D0B6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: ;tag=as4d4afc79 Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 310 v=0 o=root 9309 9309 IN IP4 192.168.2.149 s=session c=IN IP4 192.168.2.149 t=0 0 m=audio 2250 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/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 - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/tassi2-b7706aa8 and SIP/ashleyc1-08342228 sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> ACK sip:256@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK4924f6da3C2E6CD1 From: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC To: ;tag=as4d4afc79 CSeq: 2 ACK Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Proxy-Authorization: Digest username="tassi2", realm="asterisk", nonce="614589c1", uri="sip:256@asterisk.starrez.com:5060;user=phone", response="f28e75eb9e3b382329e1d1a64fcdb687", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Reliably Transmitting (no NAT) to 192.168.2.125:5060: OPTIONS sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK3de96002;rport From: "asterisk" ;tag=as758bbc3d To: Contact: Call-ID: 74ad1ec903b1c374640b1eb47b79e929@192.168.2.85 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 08 Jan 2007 05:51:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK3de96002;rport From: "asterisk" ;tag=as758bbc3d To: ;tag=B9B213F3-636E37BE CSeq: 102 OPTIONS Call-ID: 74ad1ec903b1c374640b1eb47b79e929@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '74ad1ec903b1c374640b1eb47b79e929@192.168.2.85' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> REFER sip:0402341350@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK7b0e4f05690263F8 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 CSeq: 2 REFER Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 595d29e63429646a6da498061aad81e3@192.168.2.85 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 256@extensions by tassi2@asterisk.starrez.com sip*CLI> <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK7b0e4f05690263F8;received=192.168.2.125 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on Zap/5-1 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: NOTIFY sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK084e8c58;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F Contact: Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- Scheduling destruction of SIP dialog 'adee168-af0f82b7-88a1c152@192.168.2.125' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: BYE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK6f52458e;rport From: ;tag=as4d4afc79 To: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '595d29e63429646a6da498061aad81e3@192.168.2.85' in 6400 ms (Method: REFER) == Spawn extension (main, 0399168263, 2) exited non-zero on 'SIP/tassi2-b7706aa8' sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK084e8c58;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F CSeq: 103 NOTIFY Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> BYE sip:0402341350@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKe85c96877C495762 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 CSeq: 3 BYE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Scheduling destruction of SIP dialog '595d29e63429646a6da498061aad81e3@192.168.2.85' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKe85c96877C495762;received=192.168.2.125 From: ;tag=2D0CA4A0-D99BC74F To: "0402341350" ;tag=as5283f296 Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK6f52458e;rport From: ;tag=as4d4afc79 To: "Tass Iliopoulos" ;tag=F65B21A9-BB4042EC CSeq: 102 BYE Call-ID: adee168-af0f82b7-88a1c152@192.168.2.125 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'adee168-af0f82b7-88a1c152@192.168.2.125' Method: ACK -- Started music on hold, class 'default', on Zap/5-1 == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Zap/5-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Zap/5-1' -- Stopped music on hold on Zap/5-1 -- Hungup 'Zap/5-1' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: BYE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK415bccb7;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '595d29e63429646a6da498061aad81e3@192.168.2.85' in 6400 ms (Method: BYE) sip*CLI> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK415bccb7;rport From: "0402341350" ;tag=as5283f296 To: ;tag=2D0CA4A0-D99BC74F CSeq: 104 BYE Call-ID: 595d29e63429646a6da498061aad81e3@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.125 sip*CLI> sip no debug