Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.21.2 currently running on pbx-b (pid = 4418) pbx-b*CLI> Verbosity is at least 3 [Kpbx-b*CLI> <--- SIP read from 192.168.50.250:5060 ---> SIP/2.0 200 OK Call-ID: 54a2a33e79e3f7005605096920c7a334@192.168.50.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0372c1f1 To: ;tag=5099e5a7560d7c4 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK090939c6;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Supported: replaces User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.250:5060 ---> SIP/2.0 200 OK Call-ID: 49b233df4f41a6310b6bbd4f13e9df90@192.168.50.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as30c562c4 To: ;tag=68e3bd1fdbe51d0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6819ddc1;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Supported: replaces User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '49b233df4f41a6310b6bbd4f13e9df90@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Really destroying SIP dialog '54a2a33e79e3f7005605096920c7a334@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> INVITE sip:2424@192.168.50.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F From: "Polycom 650" ;tag=1A6A0693-9409C74E To: CSeq: 1 INVITE Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 277 v=0 o=- 1222371374 1222371374 IN IP4 192.168.50.249 s=Polycom IP Phone c=IN IP4 192.168.50.249 t=0 0 m=audio 2234 RTP/AVP 9 0 8 18 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 12 lines) --- Sending to 192.168.50.249 : 5060 (no NAT) Using INVITE request as basis request - 6c7b529a-ba914749-fe3faecc@192.168.50.249 <--- Reliably Transmitting (no NAT) to 192.168.50.249:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F;received=192.168.50.249 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as35c92802 Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 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="09081313" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6c7b529a-ba914749-fe3faecc@192.168.50.249' in 32000 ms (Method: INVITE) Found user '2608' <--- SIP read from 192.168.50.249:5060 ---> ACK sip:2424@192.168.50.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bK42eb0638A096F38F From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as35c92802 CSeq: 1 ACK Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> INVITE sip:2424@192.168.50.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: CSeq: 2 INVITE Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="df84c8550a18f85d5aad62024ea0bfdb", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 277 v=0 o=- 1222371374 1222371374 IN IP4 192.168.50.249 s=Polycom IP Phone c=IN IP4 192.168.50.249 t=0 0 m=audio 2234 RTP/AVP 9 0 8 18 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 12 lines) --- Sending to 192.168.50.249 : 5060 (no NAT) Using INVITE request as basis request - 6c7b529a-ba914749-fe3faecc@192.168.50.249 Found user '2608' Found RTP audio format 9 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 192.168.50.249:2234 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc2e (gsm|ulaw|alaw|g726|adpcm|ilbc), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.50.249:2234 Looking for 2424 in standard (domain 192.168.50.1) list_route: hop: <--- Transmitting (no NAT) to 192.168.50.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2424@standard:1] Queue("SIP/2608-b7d05e10", "group1") in new stack [Kpbx-b*CLI> Audio is at 192.168.50.1 port 16828 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.50.249:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as2c27759c Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 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: 262 v=0 o=root 4418 4418 IN IP4 192.168.50.1 s=session c=IN IP4 192.168.50.1 t=0 0 m=audio 16828 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 - - - - a=ptime:20 a=sendrecv <------------> -- Started music on hold, class 'ring', on SIP/2608-b7d05e10 -- Executing [2604@standard:1] Set("Local/2604@standard-f909,2", "ARGS=dialExt,2604,no") in new stack -- Executing [2604@standard:2] AGI("Local/2604@standard-f909,2", "callRoute3.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- Executing [2605@standard:1] Set("Local/2605@standard-4cfe,2", "ARGS=dialExt,2605,no") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:2] AGI("Local/2605@standard-4cfe,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2604@smartRing:1] Set("Local/2604@standard-f909,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:2] Set("Local/2604@standard-f909,2", "phones=IAX2/2604") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:3] Set("Local/2604@standard-f909,2", "duration=60") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-f909,2", "ALERT-INFO: null") in new stack -- Executing [2604@smartRing:5] Dial("Local/2604@standard-f909,2", "IAX2/2604|60|rwWA()") in new stack -- Called 2604 [Kpbx-b*CLI> -- Local/2604@standard-f909,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.120 (format ulaw) [Kpbx-b*CLI> -- Format for call is ulaw [Kpbx-b*CLI> -- IAX2/2604-14650 is ringing [Kpbx-b*CLI> -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2605@smartRing:1] Set("Local/2605@standard-4cfe,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:2] Set("Local/2605@standard-4cfe,2", "phones=IAX2/2605") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:3] Set("Local/2605@standard-4cfe,2", "duration=60") in new stack -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-4cfe,2", "ALERT-INFO: null") in new stack -- Executing [2605@smartRing:5] Dial("Local/2605@standard-4cfe,2", "IAX2/2605|60|rwWA()") in new stack -- Called 2605 [Kpbx-b*CLI> -- Local/2605@standard-4cfe,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.116 (format ulaw) -- Format for call is ulaw -- IAX2/2605-11114 is ringing [Kpbx-b*CLI> <--- SIP read from 127.0.0.1:50603 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76 Max-Forwards: 70 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78407 REGISTER User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu Contact: Expires: 5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 127.0.0.1 : 50603 (NAT) <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76;received=127.0.0.1;rport=50603 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78407 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx-b*CLI> <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPje0f5e3e0-f3fb-4ff9-a3fe-ac41ce1b6d76;received=127.0.0.1;rport=50603 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: ;tag=as7a50f888 Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78407 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ac1fe83" Content-Length: 0 <------------> [Kpbx-b*CLI> Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER) <--- SIP read from 127.0.0.1:50603 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878 Max-Forwards: 70 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78408 REGISTER User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu Contact: Expires: 5 Authorization: Digest username="SGATE", realm="asterisk", nonce="6ac1fe83", uri="sip:127.0.0.1", response="3352bd3a3d085ef384d6b4c0625745cf", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 127.0.0.1 : 50603 (NAT) <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878;received=127.0.0.1;rport=50603 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78408 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx-b*CLI> <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj5b689857-07f2-4f4f-9bd8-dadb30fc7878;received=127.0.0.1;rport=50603 From: ;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f To: ;tag=as7a50f888 Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78408 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 25 Sep 2008 20:01:50 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER) [Kpbx-b*CLI> -- Nobody picked up in 5000 ms -- Nobody picked up in 5000 ms -- Hungup 'IAX2/2604-14650' -- Hungup 'IAX2/2605-11114' == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-f909,2' [Kpbx-b*CLI> == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-4cfe,2' [Kpbx-b*CLI> Really destroying SIP dialog 'b436e144-f17f4bab-a0a95a86@192.168.50.249' Method: ACK [Kpbx-b*CLI> -- Executing [2604@standard:1] Set("Local/2604@standard-b3cf,2", "ARGS=dialExt,2604,no") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:1] Set("Local/2605@standard-ac07,2", "ARGS=dialExt,2605,no") in new stack -- Executing [2605@standard:2] AGI("Local/2605@standard-ac07,2", "callRoute3.agi") in new stack -- Executing [2604@standard:2] AGI("Local/2604@standard-b3cf,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2605@smartRing:1] Set("Local/2605@standard-ac07,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:2] Set("Local/2605@standard-ac07,2", "phones=IAX2/2605") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:3] Set("Local/2605@standard-ac07,2", "duration=60") in new stack -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-ac07,2", "ALERT-INFO: null") in new stack -- Executing [2605@smartRing:5] Dial("Local/2605@standard-ac07,2", "IAX2/2605|60|rwWA()") in new stack -- Called 2605 [Kpbx-b*CLI> -- Call accepted by 192.168.1.116 (format ulaw) -- Format for call is ulaw -- Local/2605@standard-ac07,1 is ringing [Kpbx-b*CLI> -- IAX2/2605-3281 is ringing -- AGI Script callRoute3.agi completed, returning 0 -- Executing [2604@smartRing:1] Set("Local/2604@standard-b3cf,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:2] Set("Local/2604@standard-b3cf,2", "phones=IAX2/2604") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:3] Set("Local/2604@standard-b3cf,2", "duration=60") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-b3cf,2", "ALERT-INFO: null") in new stack -- Executing [2604@smartRing:5] Dial("Local/2604@standard-b3cf,2", "IAX2/2604|60|rwWA()") in new stack -- Called 2604 [Kpbx-b*CLI> -- Local/2604@standard-b3cf,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.120 (format ulaw) -- Format for call is ulaw -- IAX2/2604-9064 is ringing [Kpbx-b*CLI> -- Remote UNIX connection [Kpbx-b*CLI> -- Nobody picked up in 5000 ms [Kpbx-b*CLI> -- Nobody picked up in 5000 ms [Kpbx-b*CLI> -- Hungup 'IAX2/2604-9064' -- Hungup 'IAX2/2605-3281' == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-ac07,2' == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-b3cf,2' -- Remote UNIX connection disconnected [Kpbx-b*CLI> -- Remote UNIX connection [Kpbx-b*CLI> -- Remote UNIX connection disconnected [Kpbx-b*CLI> -- Remote UNIX connection [Kpbx-b*CLI> -- Remote UNIX connection disconnected [Kpbx-b*CLI> -- Remote UNIX connection [Kpbx-b*CLI> -- Remote UNIX connection disconnected [Kpbx-b*CLI> -- Remote UNIX connection [Kpbx-b*CLI> -- Remote UNIX connection disconnected [Kpbx-b*CLI> == Parsing '/etc/asterisk/manager.conf': Found [Kpbx-b*CLI> == Manager 'admin' logged on from 127.0.0.1 [Kpbx-b*CLI> == Manager 'admin' logged off from 127.0.0.1 [Kpbx-b*CLI> -- Stopped music on hold on SIP/2608-b7d05e10 -- Playing periodic announcement -- Playing 'queue-periodic-announce' (language 'en') -- Started music on hold, class 'ring', on SIP/2608-b7d05e10 [Kpbx-b*CLI> -- Executing [2604@standard:1] Set("Local/2604@standard-34d8,2", "ARGS=dialExt,2604,no") in new stack [Kpbx-b*CLI> -- Executing [2604@standard:2] AGI("Local/2604@standard-34d8,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:1] Set("Local/2605@standard-583e,2", "ARGS=dialExt,2605,no") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:2] AGI("Local/2605@standard-583e,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2605@smartRing:1] Set("Local/2605@standard-583e,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:2] Set("Local/2605@standard-583e,2", "phones=IAX2/2605") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:3] Set("Local/2605@standard-583e,2", "duration=60") in new stack -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-583e,2", "ALERT-INFO: null") in new stack -- Executing [2605@smartRing:5] Dial("Local/2605@standard-583e,2", "IAX2/2605|60|rwWA()") in new stack [Kpbx-b*CLI> -- Called 2605 [Kpbx-b*CLI> -- Local/2605@standard-583e,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.116 (format ulaw) -- Format for call is ulaw -- IAX2/2605-8300 is ringing -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2604@smartRing:1] Set("Local/2604@standard-34d8,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:2] Set("Local/2604@standard-34d8,2", "phones=IAX2/2604") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:3] Set("Local/2604@standard-34d8,2", "duration=60") in new stack -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-34d8,2", "ALERT-INFO: null") in new stack -- Executing [2604@smartRing:5] Dial("Local/2604@standard-34d8,2", "IAX2/2604|60|rwWA()") in new stack -- Called 2604 [Kpbx-b*CLI> -- Local/2604@standard-34d8,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.120 (format ulaw) -- Format for call is ulaw -- IAX2/2604-6615 is ringing [Kpbx-b*CLI> -- Nobody picked up in 5000 ms [Kpbx-b*CLI> -- Nobody picked up in 5000 ms [Kpbx-b*CLI> -- Hungup 'IAX2/2604-6615' [Kpbx-b*CLI> -- Hungup 'IAX2/2605-8300' == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-583e,2' == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-34d8,2' [Kpbx-b*CLI> Really destroying SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' Method: REGISTER [Kpbx-b*CLI> -- Stopped music on hold on SIP/2608-b7d05e10 -- Playing periodic announcement -- Playing 'queue-periodic-announce' (language 'en') -- Started music on hold, class 'ring', on SIP/2608-b7d05e10 [Kpbx-b*CLI> -- Executing [2604@standard:1] Set("Local/2604@standard-1f6c,2", "ARGS=dialExt,2604,no") in new stack [Kpbx-b*CLI> -- Executing [2604@standard:2] AGI("Local/2604@standard-1f6c,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:1] Set("Local/2605@standard-a9b5,2", "ARGS=dialExt,2605,no") in new stack [Kpbx-b*CLI> -- Executing [2605@standard:2] AGI("Local/2605@standard-a9b5,2", "callRoute3.agi") in new stack [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2604@smartRing:1] Set("Local/2604@standard-1f6c,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:2] Set("Local/2604@standard-1f6c,2", "phones=IAX2/2604") in new stack [Kpbx-b*CLI> -- Executing [2604@smartRing:3] Set("Local/2604@standard-1f6c,2", "duration=60") in new stack -- Executing [2604@smartRing:4] SIPAddHeader("Local/2604@standard-1f6c,2", "ALERT-INFO: null") in new stack -- Executing [2604@smartRing:5] Dial("Local/2604@standard-1f6c,2", "IAX2/2604|60|rwWA()") in new stack [Kpbx-b*CLI> -- Called 2604 [Kpbx-b*CLI> -- Local/2604@standard-1f6c,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.120 (format ulaw) -- Format for call is ulaw -- IAX2/2604-10243 is ringing -- AGI Script callRoute3.agi completed, returning 0 [Kpbx-b*CLI> -- Executing [2605@smartRing:1] Set("Local/2605@standard-a9b5,2", "phase=0") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:2] Set("Local/2605@standard-a9b5,2", "phones=IAX2/2605") in new stack [Kpbx-b*CLI> -- Executing [2605@smartRing:3] Set("Local/2605@standard-a9b5,2", "duration=60") in new stack -- Executing [2605@smartRing:4] SIPAddHeader("Local/2605@standard-a9b5,2", "ALERT-INFO: null") in new stack -- Executing [2605@smartRing:5] Dial("Local/2605@standard-a9b5,2", "IAX2/2605|60|rwWA()") in new stack [Kpbx-b*CLI> -- Called 2605 [Kpbx-b*CLI> -- Local/2605@standard-a9b5,1 is ringing [Kpbx-b*CLI> -- Call accepted by 192.168.1.116 (format ulaw) -- Format for call is ulaw -- IAX2/2605-6419 is ringing [Kpbx-b*CLI> -- Nobody picked up in 5000 ms -- Nobody picked up in 5000 ms -- Hungup 'IAX2/2605-6419' -- Hungup 'IAX2/2604-10243' == Spawn extension (smartRing, 2605, 5) exited non-zero on 'Local/2605@standard-a9b5,2' == Spawn extension (smartRing, 2604, 5) exited non-zero on 'Local/2604@standard-1f6c,2' Reliably Transmitting (no NAT) to 192.168.50.249:5060: OPTIONS sip:2625@192.168.50.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK303f90c1;rport From: "asterisk" ;tag=as5108246a To: Contact: Call-ID: 7a3aee0f48e291831d8534c117692bc1@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK303f90c1;rport From: "asterisk" ;tag=as5108246a To: ;tag=25D2A55B-7CAD5A76 CSeq: 102 OPTIONS Call-ID: 7a3aee0f48e291831d8534c117692bc1@192.168.50.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '7a3aee0f48e291831d8534c117692bc1@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.249:5060: OPTIONS sip:2621@192.168.50.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK31e2eb5b;rport From: "asterisk" ;tag=as784262cc To: Contact: Call-ID: 3cdd404a1b779ba56529cf84346b75b5@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK31e2eb5b;rport From: "asterisk" ;tag=as784262cc To: ;tag=4CB55808-105D791F CSeq: 102 OPTIONS Call-ID: 3cdd404a1b779ba56529cf84346b75b5@192.168.50.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '3cdd404a1b779ba56529cf84346b75b5@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.249:5060: OPTIONS sip:2608@192.168.50.249 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK3a984c2e;rport From: "asterisk" ;tag=as61c401bc To: Contact: Call-ID: 734a085e7c521b71339c75533595398e@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK3a984c2e;rport From: "asterisk" ;tag=as61c401bc To: ;tag=D2904F59-CD778D9C CSeq: 102 OPTIONS Call-ID: 734a085e7c521b71339c75533595398e@192.168.50.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '734a085e7c521b71339c75533595398e@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.2.76:50603: OPTIONS sip:SGATE@192.168.2.76:50603;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK2601acc9;rport From: "asterisk" ;tag=as299be91b To: Contact: Call-ID: 4f7b379049cc3019348e2e9c22816ac3@192.168.2.76 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.2.76:50603 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.76:5060;rport=5060;received=192.168.2.76;branch=z9hG4bK2601acc9 Call-ID: 4f7b379049cc3019348e2e9c22816ac3@192.168.2.76 From: "asterisk" ;tag=as299be91b To: CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, norefersub Allow-Events: presence, refer User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu Content-Type: application/sdp Content-Length: 287 v=0 o=- 3431361759 3431361759 IN IP4 192.168.2.76 s=pjmedia c=IN IP4 192.168.2.76 t=0 0 m=audio 4000 RTP/AVP 3 0 8 101 a=rtcp:4001 IN IP4 192.168.2.76 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 13 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '4f7b379049cc3019348e2e9c22816ac3@192.168.2.76' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.200:5060: OPTIONS sip:2601@192.168.50.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6feadd47;rport From: "asterisk" ;tag=as6a633f8d To: Contact: Call-ID: 484e0e5330d466910934e57406ea086e@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK6feadd47;rport From: "asterisk" ;tag=as6a633f8d To: ;tag=830DABD5-59DDE6F0 CSeq: 102 OPTIONS Call-ID: 484e0e5330d466910934e57406ea086e@192.168.50.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.2.0084 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '484e0e5330d466910934e57406ea086e@192.168.50.1' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.50.245:5060: OPTIONS sip:2602@192.168.50.245 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK1fe17f07;rport From: "asterisk" ;tag=as6689a6dd To: Contact: Call-ID: 74cfcd0c066a560c63bbfbf634c42060@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.245:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK1fe17f07;rport From: "asterisk" ;tag=as6689a6dd To: ;tag=86606091-5408CED2 CSeq: 102 OPTIONS Call-ID: 74cfcd0c066a560c63bbfbf634c42060@192.168.50.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> -- Stopped music on hold on SIP/2608-b7d05e10 -- Playing periodic announcement -- Playing 'queue-periodic-announce' (language 'en') Really destroying SIP dialog '74cfcd0c066a560c63bbfbf634c42060@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.247:5060: OPTIONS sip:2609@192.168.50.247:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK5af09efc;rport From: "asterisk" ;tag=as4e3260c7 To: Contact: Call-ID: 285e067815cc3ba3240214021c4476ce@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK5af09efc;rport=5060;received=192.168.50.1 From: "asterisk" ;tag=as4e3260c7 To: ;tag=1592297787 Call-ID: 285e067815cc3ba3240214021c4476ce@192.168.50.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 53i/2.1.0.2145 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '285e067815cc3ba3240214021c4476ce@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.1.220:5064: OPTIONS sip:2603@192.168.1.220:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK0c5a9343;rport From: "asterisk" ;tag=as69b07436 To: Contact: Call-ID: 00c5e755409d1a5a145e2cef76222eab@192.168.2.76 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.1.220:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.76:5060;branch=z9hG4bK0c5a9343;rport From: "asterisk" ;tag=as69b07436 To: ;tag=8597659733de1f50 Call-ID: 00c5e755409d1a5a145e2cef76222eab@192.168.2.76 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.5.15 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '00c5e755409d1a5a145e2cef76222eab@192.168.2.76' Method: OPTIONS [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.250:5060: OPTIONS sip:2620@192.168.50.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK37e5b43f;rport From: "asterisk" ;tag=as065d9faa To: Contact: Call-ID: 3104ff051a5e5dc017d2d140416eba5b@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> Reliably Transmitting (no NAT) to 192.168.50.250:5060: OPTIONS sip:2606@192.168.50.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK257e4467;rport From: "asterisk" ;tag=as743f8842 To: Contact: Call-ID: 3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Sep 2008 20:02:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.250:5060 ---> SIP/2.0 200 OK Call-ID: 3104ff051a5e5dc017d2d140416eba5b@192.168.50.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as065d9faa To: ;tag=32a8fe2eea25b02 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK37e5b43f;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Supported: replaces User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> <--- SIP read from 192.168.50.250:5060 ---> SIP/2.0 200 OK Call-ID: 3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as743f8842 To: ;tag=ad741e34d25cfb0 Via: SIP/2.0/UDP 192.168.50.1:5060;branch=z9hG4bK257e4467;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Supported: replaces User-Agent: Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> --- (10 headers 0 lines) --- [Kpbx-b*CLI> Really destroying SIP dialog '3b19fb95168fdb4b5ed16ed14934ae5e@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> Really destroying SIP dialog '3104ff051a5e5dc017d2d140416eba5b@192.168.50.1' Method: OPTIONS [Kpbx-b*CLI> <--- SIP read from 127.0.0.1:50603 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764 Max-Forwards: 70 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78409 REGISTER User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu Contact: Expires: 5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 127.0.0.1 : 50603 (NAT) <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764;received=127.0.0.1;rport=50603 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78409 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPjb153da99-439f-486e-a502-b5f8e7fa7764;received=127.0.0.1;rport=50603 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: ;tag=as040f4bd0 Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78409 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ad6254b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER) [Kpbx-b*CLI> <--- SIP read from 127.0.0.1:50603 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.76:50603;rport;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a Max-Forwards: 70 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78410 REGISTER User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu Contact: Expires: 5 Authorization: Digest username="SGATE", realm="asterisk", nonce="3ad6254b", uri="sip:127.0.0.1", response="feacdd0352932ef92b607f7b7972f7d7", algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 127.0.0.1 : 50603 (NAT) <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a;received=127.0.0.1;rport=50603 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78410 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx-b*CLI> <--- Transmitting (no NAT) to 192.168.2.76:50603 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.76:50603;branch=z9hG4bKPj46f364e9-731d-4453-8be9-6cd9a82a2d4a;received=127.0.0.1;rport=50603 From: ;tag=9cde16d3-9887-4368-af59-025df7594394 To: ;tag=as040f4bd0 Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa CSeq: 78410 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 25 Sep 2008 20:02:45 GMT Content-Length: 0 <------------> [Kpbx-b*CLI> Scheduling destruction of SIP dialog '0d483bef-eccb-4e6f-80ac-5cd7f430aaaa' in 32000 ms (Method: REGISTER) [Kpbx-b*CLI> <--- SIP read from 192.168.50.249:5060 ---> CANCEL sip:2424@192.168.50.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: CSeq: 2 CANCEL Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="df84c8550a18f85d5aad62024ea0bfdb", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.50.249 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 192.168.50.249:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as2c27759c Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [Kpbx-b*CLI> <------------> <--- Transmitting (no NAT) to 192.168.50.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0;received=192.168.50.249 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as2c27759c Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kpbx-b*CLI> [Sep 25 16:02:46] WARNING[11231]: channel.c:2557 ast_prod: Prodding channel 'SIP/2608-b7d05e10' failed -- Started music on hold, class 'ring', on SIP/2608-b7d05e10 [Kpbx-b*CLI> -- Executing [2604@standard:1] Set("Local/2604@standard-339e,2", "ARGS=dialExt,2604,no") in new stack -- Executing [2604@standard:2] AGI("Local/2604@standard-339e,2", "callRoute3.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- Executing [2605@standard:1] Set("Local/2605@standard-9a30,2", "ARGS=dialExt,2605,no") in new stack -- Executing [2605@standard:2] AGI("Local/2605@standard-9a30,2", "callRoute3.agi") in new stack <--- SIP read from 192.168.50.249:5060 ---> ACK sip:2424@192.168.50.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.249;branch=z9hG4bKbc2723ed296B5CA0 From: "Polycom 650" ;tag=1A6A0693-9409C74E To: ;tag=as2c27759c CSeq: 2 ACK Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.2.2.0084 Proxy-Authorization: Digest username="2608", realm="asterisk", nonce="09081313", uri="sip:2424@192.168.50.1:5060;user=phone", response="d940f8c753f5f514fc5d5a58dbbda37b", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Kpbx-b*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/callRoute3.agi -- Stopped music on hold on SIP/2608-b7d05e10 [Kpbx-b*CLI> == Spawn extension (standard, 2604, 2) exited non-zero on 'Local/2604@standard-339e,2' [Kpbx-b*CLI> == Spawn extension (standard, 2605, 2) exited non-zero on 'Local/2605@standard-9a30,2' [Kpbx-b*CLI> == Spawn extension (standard, 2424, 1) exited non-zero on 'SIP/2608-b7d05e10' [Kpbx-b*CLI> Really destroying SIP dialog '6c7b529a-ba914749-fe3faecc@192.168.50.249' Method: ACK [Kpbx-b*CLI>