rendered paste bodyAsterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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" <sip:asterisk@192.168.50.1>;tag=as0372c1f1
To: <sip:2620@192.168.50.250>;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" <sip:asterisk@192.168.50.1>;tag=as30c562c4
To: <sip:2606@192.168.50.250>;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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>
CSeq: 1 INVITE
Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
Contact: <sip:2608@192.168.50.249>
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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;tag=as35c92802
CSeq: 1 ACK
Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
Contact: <sip:2608@192.168.50.249>
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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>
CSeq: 2 INVITE
Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
Contact: <sip:2608@192.168.50.249>
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: <sip:2608@192.168.50.249>
<--- 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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>
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: <sip:2424@192.168.50.1>
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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;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: <sip:2424@192.168.50.1>
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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>
Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
CSeq: 78407 REGISTER
User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>;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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>
Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
CSeq: 78408 REGISTER
User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>;tag=f164d4b7-b1e5-4bf5-8e79-4f6da84dc93f
To: <sip:SGATE@127.0.0.1>;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: <sip:SGATE@192.168.2.76:50603;transport=UDP>;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
-- <SIP/2608-b7d05e10> 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
-- <SIP/2608-b7d05e10> 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" <sip:asterisk@192.168.50.1>;tag=as5108246a
To: <sip:2625@192.168.50.249>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as5108246a
To: <sip:2625@192.168.50.249>;tag=25D2A55B-7CAD5A76
CSeq: 102 OPTIONS
Call-ID: 7a3aee0f48e291831d8534c117692bc1@192.168.50.1
Contact: <sip:2625@192.168.50.249>
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" <sip:asterisk@192.168.50.1>;tag=as784262cc
To: <sip:2621@192.168.50.249>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as784262cc
To: <sip:2621@192.168.50.249>;tag=4CB55808-105D791F
CSeq: 102 OPTIONS
Call-ID: 3cdd404a1b779ba56529cf84346b75b5@192.168.50.1
Contact: <sip:2621@192.168.50.249>
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" <sip:asterisk@192.168.50.1>;tag=as61c401bc
To: <sip:2608@192.168.50.249>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as61c401bc
To: <sip:2608@192.168.50.249>;tag=D2904F59-CD778D9C
CSeq: 102 OPTIONS
Call-ID: 734a085e7c521b71339c75533595398e@192.168.50.1
Contact: <sip:2608@192.168.50.249>
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" <sip:asterisk@192.168.2.76>;tag=as299be91b
To: <sip:SGATE@192.168.2.76:50603;transport=UDP>
Contact: <sip:asterisk@192.168.2.76>
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" <sip:asterisk@192.168.2.76>;tag=as299be91b
To: <sip:SGATE@192.168.2.76>
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" <sip:asterisk@192.168.50.1>;tag=as6a633f8d
To: <sip:2601@192.168.50.200>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as6a633f8d
To: <sip:2601@192.168.50.200>;tag=830DABD5-59DDE6F0
CSeq: 102 OPTIONS
Call-ID: 484e0e5330d466910934e57406ea086e@192.168.50.1
Contact: <sip:2601@192.168.50.200>
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" <sip:asterisk@192.168.50.1>;tag=as6689a6dd
To: <sip:2602@192.168.50.245>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as6689a6dd
To: <sip:2602@192.168.50.245>;tag=86606091-5408CED2
CSeq: 102 OPTIONS
Call-ID: 74cfcd0c066a560c63bbfbf634c42060@192.168.50.1
Contact: <sip:2602@192.168.50.245>
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
-- <SIP/2608-b7d05e10> 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" <sip:asterisk@192.168.50.1>;tag=as4e3260c7
To: <sip:2609@192.168.50.247:5060;transport=udp>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as4e3260c7
To: <sip:2609@192.168.50.247:5060;transport=udp>;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" <sip:asterisk@192.168.2.76>;tag=as69b07436
To: <sip:2603@192.168.1.220:5064;transport=udp>
Contact: <sip:asterisk@192.168.2.76>
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" <sip:asterisk@192.168.2.76:5060>;tag=as69b07436
To: <sip:2603@192.168.1.220:5064;transport=udp>;tag=8597659733de1f50
Call-ID: 00c5e755409d1a5a145e2cef76222eab@192.168.2.76
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.5.15
Contact: <sip:2603@192.168.1.220:5064;transport=udp>
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" <sip:asterisk@192.168.50.1>;tag=as065d9faa
To: <sip:2620@192.168.50.250>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as743f8842
To: <sip:2606@192.168.50.250>
Contact: <sip:asterisk@192.168.50.1>
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" <sip:asterisk@192.168.50.1>;tag=as065d9faa
To: <sip:2620@192.168.50.250>;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" <sip:asterisk@192.168.50.1>;tag=as743f8842
To: <sip:2606@192.168.50.250>;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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>
Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
CSeq: 78409 REGISTER
User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>;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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>
Call-ID: 0d483bef-eccb-4e6f-80ac-5cd7f430aaaa
CSeq: 78410 REGISTER
User-Agent: PJSUA v0.8.0/i686-pc-linux-gnu
Contact: <sip:SGATE@192.168.2.76:50603;transport=UDP>
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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>
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: <sip:SGATE@127.0.0.1>;tag=9cde16d3-9887-4368-af59-025df7594394
To: <sip:SGATE@127.0.0.1>;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: <sip:SGATE@192.168.2.76:50603;transport=UDP>;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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>
CSeq: 2 CANCEL
Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
Contact: <sip:2608@192.168.50.249>
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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;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: <sip:2424@192.168.50.1>
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" <sip:2608@192.168.50.1>;tag=1A6A0693-9409C74E
To: <sip:2424@192.168.50.1;user=phone>;tag=as2c27759c
CSeq: 2 ACK
Call-ID: 6c7b529a-ba914749-fe3faecc@192.168.50.249
Contact: <sip:2608@192.168.50.249>
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>