All pastes #103781 Raw Edit

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public text v1 · immutable
#103781 ·published 2006-07-28 14:26 UTC
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Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.7.1 currently running on asterisk (pid = 5835)
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
INVITE sip:1306@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK25b5bf2e7
Content-Length: 240
To: sip:1306@10.10.0.23
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738028 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 3591 <sip:3591@10.10.10.24>
Supported: replaces
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1

v=0
o=MxSIP 475483094135019846 520051563523439643 IN IP4 10.10.10.24
s=-
c=IN IP4 10.10.10.24
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 18 4 0 8 13
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 
--- (15 headers 11 lines)---
 Using INVITE request as basis request - d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Reliably Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK25b5bf2e7;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
To: sip:1306@10.10.0.23;tag=as5a8706ef
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738028 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1306@10.10.0.23>
Proxy-Authenticate: Digest realm="asterisk", nonce="28a6227e"
Content-Length: 0


---
 Scheduling destruction of call 'd8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23' in 15000 ms
 Found user '3591'
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
ACK sip:1306@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK25b5bf2e7
Content-Length: 0
To: sip:1306@10.10.0.23;tag=as5a8706ef
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738028 ACK
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (8 headers 0 lines)---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
INVITE sip:1306@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKb4294edad
Content-Length: 240
To: sip:1306@10.10.0.23
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Contact: 3591 <sip:3591@10.10.10.24>
Content-Type: application/sdp
Proxy-Authorization:Digest response="9515cd441006929d47c951c89e0bb388",username="3591",realm="asterisk",nonce="28a6227e",uri="sip:1306@10.10.0.23"
Supported: replaces
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1

v=0
o=MxSIP 475483094135019846 520051563523439643 IN IP4 10.10.10.24
s=-
c=IN IP4 10.10.10.24
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 18 4 0 8 13
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 
--- (16 headers 11 lines)---
 Using INVITE request as basis request - d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Found user '3591'
 Found RTP audio format 18
 Found RTP audio format 4
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 13
 Peer audio RTP is at port 10.10.10.24:5004
 Found description format G729
 Found description format G723
 Found description format PCMU
 Found description format PCMA
 Capabilities: us - 0x100 (g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
asterisk*CLI> 
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing)
 Looking for 1306 in nortel (domain 10.10.0.23)
 list_route: hop: <sip:3591@10.10.10.24>
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKb4294edad;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
To: sip:1306@10.10.0.23
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1306@10.10.0.23>
Content-Length: 0


---
 
asterisk*CLI> 
We're at 10.10.0.23 port 19882
 Adding codec 0x100 (g729) to SDP
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKb4294edad;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
To: sip:1306@10.10.0.23;tag=as1365aab9
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1306@10.10.0.23>
Content-Type: application/sdp
Content-Length: 177

v=0
o=root 5836 5836 IN IP4 10.10.0.23
s=session
c=IN IP4 10.10.0.23
t=0 0
m=audio 19882 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -

---
 
asterisk*CLI> 
Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKb4294edad;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
To: sip:1306@10.10.0.23;tag=as1365aab9
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1306@10.10.0.23>
Content-Length: 0


---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
REGISTER sip:10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKfcf014dc8
Content-Length: 0
To: 3591 <sip:3591@10.10.0.23>
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832131 REGISTER
Contact: 3591 <sip:3591@10.10.10.24>
Authorization:Digest response="00f69e4f7a95df20828f0595808bfece",username="3591",realm="asterisk",nonce="146e14c0",uri="sip:10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (10 headers 0 lines)---
 Using latest REGISTER request as basis request
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKfcf014dc8;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
To: 3591 <sip:3591@10.10.0.23>
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832131 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3591@10.10.0.23>
Content-Length: 0


---
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKfcf014dc8;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
To: 3591 <sip:3591@10.10.0.23>;tag=as1d9b5030
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832131 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

asterisk*CLI> 
Contact: <sip:3591@10.10.0.23>
WWW-Authenticate: Digest realm="asterisk", nonce="680c2447"
Content-Length: 0


---
 Scheduling destruction of call '0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23' in 15000 ms
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
REGISTER sip:10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK5f761ec74
Content-Length: 0
To: 3592 <sip:3592@10.10.0.23>
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154053 REGISTER
Contact: 3592 <sip:3592@10.10.10.24>
Authorization:Digest response="ee2ab8bb73ba4e3348256eb8b76d64a9",username="3592",realm="asterisk",nonce="505fd4d8",uri="sip:10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (10 headers 0 lines)---
 Using latest REGISTER request as basis request
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK5f761ec74;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
To: 3592 <sip:3592@10.10.0.23>
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154053 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3592@10.10.0.23>
Content-Length: 0


---
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK5f761ec74;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
To: 3592 <sip:3592@10.10.0.23>;tag=as4203ec5b
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154053 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

asterisk*CLI> 
Contact: <sip:3592@10.10.0.23>
WWW-Authenticate: Digest realm="asterisk", nonce="5006b160"
Content-Length: 0


---
 Scheduling destruction of call 'c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23' in 15000 ms
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
REGISTER sip:10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKbc0d741db
Content-Length: 0
To: 3591 <sip:3591@10.10.0.23>
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832132 REGISTER
Contact: 3591 <sip:3591@10.10.10.24>
Authorization:Digest response="00dc79344a94e9515eb51a76ae819336",username="3591",realm="asterisk",nonce="680c2447",uri="sip:10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (10 headers 0 lines)---
 Using latest REGISTER request as basis request
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKbc0d741db;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
To: 3591 <sip:3591@10.10.0.23>
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832132 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3591@10.10.0.23>
Content-Length: 0


---
 
asterisk*CLI> 
Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKbc0d741db;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=952f83bbf05acf9
To: 3591 <sip:3591@10.10.0.23>;tag=as1d9b5030
Call-ID: 0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23
CSeq: 1181832132 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:3591@10.10.10.24>;expires=120
Date: Fri, 28 Jul 2006 14:29:51 GMT
Content-Length: 0


---
 Scheduling destruction of call '0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23' in 15000 ms
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
REGISTER sip:10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK08b3ffed3
Content-Length: 0
To: 3592 <sip:3592@10.10.0.23>
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154054 REGISTER
Contact: 3592 <sip:3592@10.10.10.24>
Authorization:Digest response="df6e9b0206c250ecd603bba879d18cab",username="3592",realm="asterisk",nonce="5006b160",uri="sip:10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (10 headers 0 lines)---
 Using latest REGISTER request as basis request
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK08b3ffed3;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
To: 3592 <sip:3592@10.10.0.23>
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154054 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3592@10.10.0.23>
Content-Length: 0


---
 
asterisk*CLI> 
Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK08b3ffed3;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=91030747ac760b5
To: 3592 <sip:3592@10.10.0.23>;tag=as4203ec5b
Call-ID: c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23
CSeq: 1064154054 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:3592@10.10.10.24>;expires=120
Date: Fri, 28 Jul 2006 14:29:51 GMT
Content-Length: 0


---
 Scheduling destruction of call 'c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23' in 15000 ms
 
asterisk*CLI> 
We're at 10.10.0.23 port 19882
 Adding codec 0x100 (g729) to SDP
 Reliably Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bKb4294edad;received=10.10.10.24
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
To: sip:1306@10.10.0.23;tag=as1365aab9
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1306@10.10.0.23>
Content-Type: application/sdp
Content-Length: 177

v=0
o=root 5836 5837 IN IP4 10.10.0.23
s=session
c=IN IP4 10.10.0.23
t=0 0
m=audio 19882 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -

---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
ACK sip:1306@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK69bb72c19
Content-Length: 0
To: sip:1306@10.10.0.23;tag=as1365aab9
From: 3591 <sip:3591@10.10.0.23>;tag=3f8b3295a8caab8
Call-ID: d8734d4d46a15fa68b832b91e5cf6d5e@10.10.0.23
CSeq: 2099738029 ACK
Contact: 3591 <sip:3591@10.10.10.24>
Proxy-Authorization:Digest response="9515cd441006929d47c951c89e0bb388",username="3591",realm="asterisk",nonce="28a6227e",uri="sip:1306@10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (10 headers 0 lines)---
 
asterisk*CLI> 
<-- SIP read from 10.10.0.100:5070: 

 
--- (0 headers 0 lines) Nat keepalive ---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
INVITE sip:1348@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK1389c6ee2
Content-Length: 240
To: sip:1348@10.10.0.23
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242334 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 3592 <sip:3592@10.10.10.24>
Supported: replaces
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1

v=0
o=MxSIP 232863679178397459 442898735492645877 IN IP4 10.10.10.24
s=-
c=IN IP4 10.10.10.24
t=0 0
a=sendrecv
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000

 
--- (15 headers 11 lines)---
 Using INVITE request as basis request - 04c2e35729a0382d786f5a63051b4338@10.10.0.23
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Reliably Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK1389c6ee2;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
To: sip:1348@10.10.0.23;tag=as4d3623d7
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242334 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1348@10.10.0.23>
Proxy-Authenticate: Digest realm="asterisk", nonce="52a4de2f"
Content-Length: 0


---
 Scheduling destruction of call '04c2e35729a0382d786f5a63051b4338@10.10.0.23' in 15000 ms
 Found user '3592'
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
ACK sip:1348@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK1389c6ee2
Content-Length: 0
To: sip:1348@10.10.0.23;tag=as4d3623d7
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242334 ACK
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (8 headers 0 lines)---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
INVITE sip:1348@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK67bd53540
Content-Length: 240
To: sip:1348@10.10.0.23
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242335 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Contact: 3592 <sip:3592@10.10.10.24>
Content-Type: application/sdp
Proxy-Authorization:Digest response="885cdd9acb6c7dd4469fe65d7fdc946b",username="3592",realm="asterisk",nonce="52a4de2f",uri="sip:1348@10.10.0.23"
Supported: replaces
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1

v=0
o=MxSIP 232863679178397459 442898735492645877 IN IP4 10.10.10.24
s=-
c=IN IP4 10.10.10.24
t=0 0
a=sendrecv
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000

 
--- (16 headers 11 lines)---
 Using INVITE request as basis request - 04c2e35729a0382d786f5a63051b4338@10.10.0.23
 Sending to 10.10.10.24 : 5060 (non-NAT)
 Found user '3592'
 Found RTP audio format 0
 Found RTP audio format 18
 Found RTP audio format 4
 Found RTP audio format 8
 Found RTP audio format 13
 Peer audio RTP is at port 10.10.10.24:5006
 Found description format PCMU
 Found description format G729
 Found description format G723
 Found description format PCMA
 Capabilities: us - 0x100 (g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing)
 Looking for 1348 in nortel (domain 10.10.0.23)
 list_route: hop: <sip:3592@10.10.10.24>
 Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK67bd53540;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
To: sip:1348@10.10.0.23
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242335 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1348@10.10.0.23>
Content-Length: 0


---
 Jul 28 17:30:03 NOTICE[18078]: app_dial.c:1029 dial_exec_full:  Unable to create channel of type 'Zap' (cause 0 - Unknown)
 
asterisk*CLI> 
Destroying call '0451bfab4d35afc0a3745ae1a3b1fca9@10.10.0.23'
 
asterisk*CLI> 
Destroying call 'c98863e423c2d1d99603fa4b6fde72ec@10.10.0.23'
 
asterisk*CLI> 
Jul 28 17:30:13 WARNING[18078]: pbx.c:2412 __ast_pbx_run:  Timeout, but no rule 't' in context 'nortel'
 Reliably Transmitting (no NAT) to 10.10.10.24:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK67bd53540;received=10.10.10.24
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
To: sip:1348@10.10.0.23;tag=as3b538030
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242335 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1348@10.10.0.23>
Content-Length: 0


---
 
asterisk*CLI> 
<-- SIP read from 10.10.10.24:5060: 
ACK sip:1348@10.10.0.23 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.24;branch=z9hG4bK67bd53540
Content-Length: 0
To: sip:1348@10.10.0.23;tag=as3b538030
From: 3592 <sip:3592@10.10.0.23>;tag=1becd9fcba6e7bd
Call-ID: 04c2e35729a0382d786f5a63051b4338@10.10.0.23
CSeq: 48242335 ACK
Proxy-Authorization:Digest response="885cdd9acb6c7dd4469fe65d7fdc946b",username="3592",realm="asterisk",nonce="52a4de2f",uri="sip:1348@10.10.0.23"
User-Agent: MxSipApp/5.0.12.78 MxSF/v3.2.1.1


 
--- (9 headers 0 lines)---
 Destroying call '04c2e35729a0382d786f5a63051b4338@10.10.0.23'
 
asterisk*CLI>