Problem with Outgoing and Incoming lines

Hello,

I have a 2.2.0 running fine and now I am setting up a secondary server on 2.4.0
I am having problems with getting the trunks running.
My first problem is that I have 1 incoming/outgoing trunk from my provider that I have copied from the 2.2.0 config which I know works with 2.2.0.

Here is the config below.
trunk name: provider
allow=ulaw
canreinvite=no
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=x.x.x.138
insecure=very
nat=no
qualify=10000
sendrpid=yes
trustrpid=yes
type=peer

user context: provider-in
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=x.x.x.138
insecure=very
nat=no
qualify=10000
sendrpid=yes
trustrpid=yes
type=peer

Problem I am having is trying to dial out to our provider.
Attached below is the logs I have for trying to dial out.
The problem is that I never see a SIP connection to my provider and
FreePBX seems to bail out with a "All circuits are busy now, please try again later"
I am about to reformat and reinstall unless someone can help me out here.

Thanks,

Asterisk 1.4.18-3 RPM by [email protected], Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
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.

Connected to Asterisk 1.4.18-3 RPM by [email protected] currently running on PBX (pid = 2671)

Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.102:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK06c20d08;rport
From: “Unknown” sip:[email protected];tag=as3691f495
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK06c20d08;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as3691f495
To: sip:[email protected]:5060;transport=udp;tag=363472204
Call-ID: [email protected]
CSeq: 102 NOTIFY
Contact: sip:[email protected]:5060;transport=udp
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: NOTIFY
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Connect attempt from ‘127.0.0.1’ unable to authenticate
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060
Call-ID: 3c773c73ed2390ef
CSeq: 31610 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1 sip:[email protected]:5060;transport=udp
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 25 lines) —
Sending to 192.168.1.102 : 5060 (no NAT)
Using INVITE request as basis request - 3c773c73ed2390ef

<— Reliably Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060;tag=as3bdaa611
Call-ID: 3c773c73ed2390ef
CSeq: 31610 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="66d55813"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3c773c73ed2390ef’ in 32000 ms (Method: INVITE)
Found user '1101’
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060;tag=as3bdaa611
Call-ID: 3c773c73ed2390ef
CSeq: 31610 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a
Proxy-Authorization: Digest username=“1101”,realm=“asterisk”,nonce=“66d55813”,uri=“sip:[email protected]:5060”,response=“99c68deebd8213900efede80d033f4fd”,algorithm=MD5
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1 sip:[email protected]:5060;transport=udp
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 25 lines) —
Sending to 192.168.1.102 : 5060 (NAT)
Using INVITE request as basis request - 3c773c73ed2390ef
Found user '1101’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.102:3000
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format BV16 for ID 106
Found unknown media description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found unknown media description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x194c (ulaw|alaw|g726|slin|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 192.168.1.102:3000
Looking for 5145555555 in from-internal (domain 192.168.1.252)
list_route: hop: sip:[email protected]:5060;transport=udp

<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/1101-0991f8e8”, “user-callerid|SKIPTTL|”) in new stack
– Executing [[email protected]:1] NoOp(“SIP/1101-0991f8e8”, “user-callerid: device 1101”) in new stack
– Executing [[email protected]:2] Set(“SIP/1101-0991f8e8”, “AMPUSER=1101”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1101-0991f8e8”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/1101-0991f8e8”, “1|Set|REALCALLERIDNUM=1101”) in new stack
– Executing [[email protected]:5] NoOp(“SIP/1101-0991f8e8”, “REALCALLERIDNUM is 1101”) in new stack
– Executing [[email protected]:6] Set(“SIP/1101-0991f8e8”, “AMPUSER=1101”) in new stack
– Executing [[email protected]:7] Set(“SIP/1101-0991f8e8”, “AMPUSERCIDNAME=Phone 1”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/1101-0991f8e8”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/1101-0991f8e8”, “AMPUSERCID=1101”) in new stack
– Executing [[email protected]:10] Set(“SIP/1101-0991f8e8”, “CALLERID(all)=“Phone 1” <1101>”) in new stack
– Executing [[email protected]:11] Set(“SIP/1101-0991f8e8”, “REALCALLERIDNUM=1101”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/1101-0991f8e8”, “1|Set|CHANNEL(language)=en”) in new stack
– Executing [[email protected]:13] NoOp(“SIP/1101-0991f8e8”, “TTL: ARG1: SKIPTTL”) in new stack
– Executing [[email protected]:14] GotoIf(“SIP/1101-0991f8e8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [[email protected]:23] NoOp(“SIP/1101-0991f8e8”, “Using CallerID “Phone 1” <1101>”) in new stack
– Executing [[email protected]ternal:2] Set(“SIP/1101-0991f8e8”, “_NODEST=”) in new stack
– Executing [[email protected]:3] Macro(“SIP/1101-0991f8e8”, “record-enable|1101|OUT|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1101-0991f8e8”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/1101-0991f8e8”, “recordingcheck|20080403-182033|1207261233.6”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080403-182033|1207261233.6: Outbound recording enabled.
recordingcheck|20080403-182033|1207261233.6: CALLFILENAME=OUT1101-20080403-182033-1207261233.6
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:999] MixMonitor(“SIP/1101-0991f8e8”, “OUT1101-20080403-182033-1207261233.6.wav”) in new stack
– Executing [[email protected]:4] Macro(“SIP/1101-0991f8e8”, “dialout-trunk|3|5145555555||”) in new stack
– Executing [[email protected]:1] Set(“SIP/1101-0991f8e8”, “DIAL_TRUNK=3”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/1101-0991f8e8”, “0|Authenticate|”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1101-0991f8e8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/1101-0991f8e8”, “DIAL_NUMBER=5145555555”) in new stack
– Executing [[email protected]:5] Set(“SIP/1101-0991f8e8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/1101-0991f8e8”, “GROUP()=OUT_3”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/1101-0991f8e8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/1101-0991f8e8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/1101-0991f8e8”, “DIAL_TRUNK_OPTIONS=”) in new stack
== Begin MixMonitor Recording SIP/1101-0991f8e8
– Executing [[email protected]:11] Macro(“SIP/1101-0991f8e8”, “outbound-callerid|3”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1101-0991f8e8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing [[email protected]:3] NoOp(“SIP/1101-0991f8e8”, “REALCALLERIDNUM is 1101”) in new stack
– Executing [[email protected]:4] GotoIf(“SIP/1101-0991f8e8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,9)
– Executing [[email protected]:9] Set(“SIP/1101-0991f8e8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:10] Set(“SIP/1101-0991f8e8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:11] Set(“SIP/1101-0991f8e8”, “TRUNKOUTCID=“MYDOTCOM.CA” <5145555555>”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/1101-0991f8e8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,16)
– Executing [[email protected]:16] GotoIf(“SIP/1101-0991f8e8”, “0?usercid”) in new stack
– Executing [[email protected]:17] Set(“SIP/1101-0991f8e8”, “CALLERID(all)=MYDOTCOM.CA <5145555555>”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/1101-0991f8e8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,22)
– Executing [[email protected]:22] NoOp(“SIP/1101-0991f8e8”, “CallerID set to “MYDOTCOM.CA” <5145555555>”) in new stack
– Executing [[email protected]:12] AGI(“SIP/1101-0991f8e8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern .
== fixlocalprefix: Dialpattern . matched. 5145555555 -> 5145555555
– AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/1101-0991f8e8”, “OUTNUM=5145555555”) in new stack
– Executing [[email protected]:14] Set(“SIP/1101-0991f8e8”, “custom=SIP/cgctemp”) in new stack
– Executing [[email protected]:15] GotoIf(“SIP/1101-0991f8e8”, “1?gocall”) in new stack
– Goto (macro-dialout-trunk,s,17)
– Executing [[email protected]:17] Macro(“SIP/1101-0991f8e8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/1101-0991f8e8”, “0?bypass|1”) in new stack
– Executing [[email protected]:19] GotoIf(“SIP/1101-0991f8e8”, “0?customtrunk”) in new stack
– Executing [[email protected]:20] Dial(“SIP/1101-0991f8e8”, “SIP/cgctemp/5145555555|300|”) in new stack
– Couldn’t call cgctemp/5145555555 ***<<< Never even tried to call the connection.
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
– Executing [[email protected]:21] Goto(“SIP/1101-0991f8e8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [[email protected]:1] GotoIf(“SIP/1101-0991f8e8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [[email protected]:3] NoOp(“SIP/1101-0991f8e8”, “TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks”) in new stack
– Executing [[email protected]:5] Macro(“SIP/1101-0991f8e8”, “outisbusy|”) in new stack
– Executing [[email protected]:1] Playback(“SIP/1101-0991f8e8”, “all-circuits-busy-now|noanswer”) in new stack
Audio is at 192.168.1.252 port 15522
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
PBX
CLI>
<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060;tag=as6fa5349b
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 15522 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

<------------>
– <SIP/1101-0991f8e8> Playing ‘all-circuits-busy-now’ (language ‘en’)
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
– Executing [[email protected]:2] Playback(“SIP/1101-0991f8e8”, “pls-try-call-later|noanswer”) in new stack
– <SIP/1101-0991f8e8> Playing ‘pls-try-call-later’ (language ‘en’)
== Manager ‘admin’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
– Executing [[email protected]:3] Macro(“SIP/1101-0991f8e8”, “hangupcall”) in new stack
– Executing [[email protected]acro-hangupcall:1] ResetCDR(“SIP/1101-0991f8e8”, “w”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1101-0991f8e8”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1101-0991f8e8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/1101-0991f8e8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/1101-0991f8e8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/1101-0991f8e8”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1101-0991f8e8’
Scheduling destruction of SIP dialog ‘3c773c73ed2390ef’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060;tag=as6fa5349b
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
== End MixMonitor Recording SIP/1101-0991f8e8
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=0939966e14
To: “5145555555” sip:[email protected]:5060;tag=as6fa5349b
Call-ID: 3c773c73ed2390ef
CSeq: 31611 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
PBX*CLI>
<— SIP read from x.x.x.138:5060 —>
OPTIONS sip:x.x.x.129 SIP/2.0
Via: SIP/2.0/UDP x.x.x.138:5060;branch=z9hG4bK14b334d2;rport
From: “asterisk” sip:[email protected];tag=as32a4fab6
To: sip:x.x.x.129
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:19:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in from-sip-external (domain x.x.x.129)

<— Transmitting (no NAT) to x.x.x.138:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.138:5060;branch=z9hG4bK14b334d2;received=x.x.x.138;rport=5060
From: “asterisk” sip:[email protected];tag=as32a4fab6
To: sip:x.x.x.129;tag=as4796771d
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:x.x.x.129
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060
Call-ID: 332cc09cdeab7e76
CSeq: 24618 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1 sip:[email protected]:5060;transport=udp
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 25 lines) —
Sending to 192.168.1.102 : 5060 (no NAT)
Using INVITE request as basis request - 332cc09cdeab7e76

<— Reliably Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060;tag=as530f2477
Call-ID: 332cc09cdeab7e76
CSeq: 24618 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2c2d7e6b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘332cc09cdeab7e76’ in 32000 ms (Method: INVITE)
Found user '1101’
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060;tag=as530f2477
Call-ID: 332cc09cdeab7e76
CSeq: 24618 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281
Proxy-Authorization: Digest username=“1101”,realm=“asterisk”,nonce=“2c2d7e6b”,uri=“sip:[email protected]:5060”,response=“d843e7060f3057c82d5853cd2ffb43f6”,algorithm=MD5
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1 sip:[email protected]:5060;transport=udp
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596

v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 25 lines) —
Sending to 192.168.1.102 : 5060 (NAT)
Using INVITE request as basis request - 332cc09cdeab7e76
Found user '1101’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.102:3000
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format BV16 for ID 106
Found unknown media description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found unknown media description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x194c (ulaw|alaw|g726|slin|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 192.168.1.102:3000
Looking for 1102 in from-internal (domain 192.168.1.252)
list_route: hop: sip:[email protected]:5060;transport=udp

<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/1101-09935dc8”, “exten-vm|novm|1102”) in new stack
– Executing [[email protected]:1] Macro(“SIP/1101-09935dc8”, “user-callerid”) in new stack
– Executing [[email protected]:1] NoOp(“SIP/1101-09935dc8”, “user-callerid: device 1101”) in new stack
– Executing [[email protected]:2] Set(“SIP/1101-09935dc8”, “AMPUSER=1101”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1101-09935dc8”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/1101-09935dc8”, “1|Set|REALCALLERIDNUM=1101”) in new stack
– Executing [[email protected]:5] NoOp(“SIP/1101-09935dc8”, “REALCALLERIDNUM is 1101”) in new stack
– Executing [[email protected]:6] Set(“SIP/1101-09935dc8”, “AMPUSER=1101”) in new stack
– Executing [[email protected]:7] Set(“SIP/1101-09935dc8”, “AMPUSERCIDNAME=Phone 1”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/1101-09935dc8”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/1101-09935dc8”, “AMPUSERCID=1101”) in new stack
– Executing [[email protected]:10] Set(“SIP/1101-09935dc8”, “CALLERID(all)=“Phone 1” <1101>”) in new stack
– Executing [[email protected]:11] Set(“SIP/1101-09935dc8”, “REALCALLERIDNUM=1101”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/1101-09935dc8”, “1|Set|CHANNEL(language)=en”) in new stack
– Executing [[email protected]:13] NoOp(“SIP/1101-09935dc8”, “TTL: ARG1: novm”) in new stack
– Executing [[email protected]:14] GotoIf(“SIP/1101-09935dc8”, “0?continue”) in new stack
– Executing [[email protected]:15] Set(“SIP/1101-09935dc8”, “__TTL=64”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/1101-09935dc8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [[email protected]:23] NoOp(“SIP/1101-09935dc8”, “Using CallerID “Phone 1” <1101>”) in new stack
– Executing [[email protected]:2] Set(“SIP/1101-09935dc8”, “FROMCONTEXT=exten-vm”) in new stack
– Executing [[email protected]:3] Set(“SIP/1101-09935dc8”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/1101-09935dc8”, “EXTTOCALL=1102”) in new stack
– Executing [[email protected]:5] Set(“SIP/1101-09935dc8”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/1101-09935dc8”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/1101-09935dc8”, “RT=”"") in new stack
– Executing [[email protected]:8] Macro(“SIP/1101-09935dc8”, “record-enable|1102|IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1101-09935dc8”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/1101-09935dc8”, “recordingcheck|20080403-182044|1207261244.8”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
== Manager ‘admin’ logged off from 127.0.0.1
recordingcheck|20080403-182044|1207261244.8: Inbound recording enabled.
recordingcheck|20080403-182044|1207261244.8: CALLFILENAME=20080403-182044-1207261244.8
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:999] MixMonitor(“SIP/1101-09935dc8”, “20080403-182044-1207261244.8.wav”) in new stack
– Executing [[email protected]:9] Macro(“SIP/1101-09935dc8”, “dial||tr|1102”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1101-09935dc8”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/1101-09935dc8”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
== Begin MixMonitor Recording SIP/1101-09935dc8
dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘Phone 1’ number is '1101’
dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 1102 to extension map
– dialparties.agi: Extension 1102 cf is disabled
– dialparties.agi: Extension 1102 do not disturb is disabled
> dialparties.agi: extnum 1102 has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 1102 has ExtensionState: 0
– dialparties.agi: Checking CW and CFB status for extension 1102
– dialparties.agi: dbset CALLTRACE/1102 to 1101
– dialparties.agi: Filtered ARG3: 1102
== Manager ‘admin’ logged off from 127.0.0.1
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/1101-09935dc8”, “SIP/1102||tr”) in new stack
Audio is at 192.168.1.252 port 15432
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.107:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport
From: “Phone 1” sip:[email protected];tag=as3d0208e2
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 15432 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


-- Called 1102

PBX*CLI>
<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060;tag=as382be96c
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport=5060;received=192.168.1.252
From: “Phone 1” sip:[email protected];tag=as3d0208e2
To: sip:[email protected]:5060;transport=udp;tag=694037709
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: sip:192.168.1.252;appearance-index=1
Contact: Phone 2 sip:[email protected]:5060;transport=udp
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (12 headers 0 lines) —
– SIP/1102-09921a30 is ringing
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport=5060;received=192.168.1.252
From: “Phone 1” sip:[email protected];tag=as3d0208e2
To: sip:[email protected]:5060;transport=udp;tag=694037709
Call-ID: 37c3dc34[email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: sip:192.168.1.252;appearance-index=1
Contact: Phone 2 sip:[email protected]:5060;transport=udp
Server: Aastra 51i/2.2.0.166
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 259

v=0
o=MxSIP 0 0 IN IP4 192.168.1.107
s=SIP Call
c=IN IP4 192.168.1.107
t=0 0
m=audio 3000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
— (14 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.107:3000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 192.168.1.107:3000
list_route: hop: sip:[email protected]:5060;transport=udp
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.107, port 5060
Transmitting (NAT) to 192.168.1.107:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK4b5d6115;rport
From: “Phone 1” sip:[email protected];tag=as3d0208e2
To: sip:[email protected]:5060;transport=udp;tag=694037709
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


-- SIP/1102-09921a30 answered SIP/1101-09935dc8

Audio is at 192.168.1.252 port 13104
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060;tag=as382be96c
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 13104 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

<------------>
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8aeb3c338b312514d
Proxy-Authorization: Digest username=“1101”,realm=“asterisk”,nonce=“2c2d7e6b”,uri="sip:[email protected]",response=“c7826eea5ef168f9eb61726fa20989ae”,algorithm=MD5
Max-Forwards: 70
From: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
To: “1102” sip:[email protected]:5060;tag=as382be96c
Call-ID: 332cc09cdeab7e76
CSeq: 24619 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (10 headers 0 lines) —
== Connect attempt from ‘127.0.0.1’ unable to authenticate
Reliably Transmitting (no NAT) to x.x.x.138:5060:
OPTIONS sip:x.x.x.138 SIP/2.0
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK05c1909a;rport
From: “Unknown” sip:[email protected];tag=as014d442b
To: sip:x.x.x.138
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


PBX*CLI>
<— SIP read from x.x.x.138:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK05c1909a;received=x.x.x.129;rport=5060
From: “Unknown” sip:[email protected];tag=as014d442b
To: sip:x.x.x.138;tag=as17ede92d
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
PBX*CLI>
<— SIP read from 192.168.1.108:5060 —>
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9
Max-Forwards: 70
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username=“1103”,realm=“asterisk”,nonce=“7d1b5c9d”,uri=“sip:192.168.1.252:5060”,response=“4c14212ee901a561aafea7a2374b3cc3”,algorithm=MD5
Contact: Phone 3 sip:[email protected]:5060;transport=udp
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.108 : 5060 (no NAT)

<— Transmitting (NAT) to 192.168.1.108:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9;received=192.168.1.108
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>

<— Transmitting (NAT) to 192.168.1.108:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9;received=192.168.1.108
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060;tag=as016f4374
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="44824fc1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘de82517b006e9bbe’ in 32000 ms (Method: REGISTER)
PBX*CLI>
<— SIP read from 192.168.1.108:5060 —>
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5
Max-Forwards: 70
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username=“1103”,realm=“asterisk”,nonce=“44824fc1”,uri=“sip:192.168.1.252:5060”,response=“89ca3c7e669277db0d9cab507f3aa0f3”,algorithm=MD5
Contact: Phone 3 sip:[email protected]:5060;transport=udp
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.108 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.108:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5;received=192.168.1.108
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
PBX*CLI>
<— Transmitting (NAT) to 192.168.1.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5;received=192.168.1.108
From: sip:[email protected]:5060;tag=8f5cb0d8be
To: sip:[email protected]:5060;tag=as016f4374
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: sip:[email protected]:5060;transport=udp;expires=120
Date: Thu, 03 Apr 2008 22:20:49 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘de82517b006e9bbe’ in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 192.168.1.108:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK1e2eecac;rport
From: “Unknown” sip:[email protected];tag=as12760251
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


PBX*CLI>
<— SIP read from 192.168.1.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK1e2eecac;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as12760251
To: sip:[email protected]:5060;transport=udp;tag=126297992
Call-ID: [email protected]
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.107:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK61584f7b;rport
From: “Unknown” sip:[email protected];tag=as57074021
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK61584f7b;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as57074021
To: sip:[email protected]:5060;transport=udp;tag=1156247511
Call-ID: [email protected]
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
Reliably Transmitting (NAT) to 192.168.1.102:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK7a1d9178;rport
From: “Unknown” sip:[email protected];tag=as53697fc6
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK7a1d9178;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as53697fc6
To: sip:[email protected]:5060;transport=udp;tag=2831479275
Call-ID: [email protected]
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d
Max-Forwards: 70
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username=“1102”,realm=“asterisk”,nonce=“6e59c487”,uri=“sip:192.168.1.252:5060”,response=“9d67232af3cbb5912434ca784d5e3288”,algorithm=MD5
Contact: Phone 2 sip:[email protected]:5060;transport=udp
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.107 : 5060 (no NAT)

<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d;received=192.168.1.107
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>

<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d;received=192.168.1.107
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060;tag=as5c71c7c2
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6f42c431"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1849ba88b3e18754’ in 32000 ms (Method: REGISTER)
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3
Max-Forwards: 70
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username=“1102”,realm=“asterisk”,nonce=“6f42c431”,uri=“sip:192.168.1.252:5060”,response=“6941ed63ff9199c6f30c18d6e00abce8”,algorithm=MD5
Contact: Phone 2 sip:[email protected]:5060;transport=udp
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.107 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3;received=192.168.1.107
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
PBX*CLI>
<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3;received=192.168.1.107
From: sip:[email protected]:5060;tag=83ac986e3d
To: sip:[email protected]:5060;tag=as5c71c7c2
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: sip:[email protected]:5060;transport=udp;expires=120
Date: Thu, 03 Apr 2008 22:20:50 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1849ba88b3e18754’ in 32000 ms (Method: REGISTER)
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged off from 127.0.0.1
Reliably Transmitting (no NAT) to x.x.x.138:5060:
OPTIONS sip:x.x.x.138 SIP/2.0
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK782c2204;rport
From: “Unknown” sip:[email protected];tag=as5abea667
To: sip:x.x.x.138
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


PBX*CLI>
<— SIP read from x.x.x.138:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK782c2204;received=x.x.x.129;rport=5060
From: “Unknown” sip:[email protected];tag=as5abea667
To: sip:x.x.x.138;tag=as6aca2b47
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
== Connect attempt from ‘127.0.0.1’ unable to authenticate
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bKd635ca7e5015b3314
Max-Forwards: 70
From: sip:[email protected]:5060;transport=udp;tag=694037709
To: “Phone 1” sip:[email protected];tag=as3d0208e2
Call-ID: [email protected]
CSeq: 1230 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: sip:192.168.1.252;appearance-index=1
Supported: timer
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.107 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bKd635ca7e5015b3314;received=192.168.1.107
From: sip:[email protected]:5060;transport=udp;tag=694037709
To: “Phone 1” sip:[email protected];tag=as3d0208e2
Call-ID: [email protected]
CSeq: 1230 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/1101-09935dc8’
– Executing [[email protected]:1] Macro(“SIP/1101-09935dc8”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/1101-09935dc8”, “w”) in new stack
Really destroying SIP dialog ‘[email protected]’ Method: BYE
– Executing [[email protected]:2] NoCDR(“SIP/1101-09935dc8”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1101-09935dc8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/1101-09935dc8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/1101-09935dc8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/1101-09935dc8”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1101-09935dc8’
Scheduling destruction of SIP dialog ‘332cc09cdeab7e76’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.102, port 5060
Reliably Transmitting (NAT) to 192.168.1.102:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK17d00dcf;rport
From: “1102” sip:[email protected]:5060;tag=as382be96c
To: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
Call-ID: 332cc09cdeab7e76
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


== End MixMonitor Recording SIP/1101-09935dc8
PBX*CLI>
<— SIP read from 192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK17d00dcf;rport=5060;received=192.168.1.252
From: “1102” sip:[email protected]:5060;tag=as382be96c
To: “Phone 1” sip:[email protected]:5060;tag=de2c509eef
Call-ID: 332cc09cdeab7e76
CSeq: 102 BYE
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘332cc09cdeab7e76’ Method: ACK
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.108:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK6f06f743;rport
From: “Unknown” sip:[email protected];tag=as6c9acdaa
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.107:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK688c31a3;rport
From: “Unknown” sip:[email protected];tag=as60de78a0
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


PBX*CLI>
<— SIP read from 192.168.1.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK6f06f743;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as6c9acdaa
To: sip:[email protected]:5060;transport=udp;tag=3744504632
Call-ID: [email protected]
CSeq: 102 NOTIFY
Contact: sip:[email protected]:5060;transport=udp
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: NOTIFY
PBX*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK688c31a3;rport=5060;received=192.168.1.252
From: “Unknown” sip:[email protected];tag=as60de78a0
To: sip:[email protected]:5060;transport=udp;tag=1754018340
Call-ID: [email protected]
CSeq: 102 NOTIFY
Contact: sip:[email protected]:5060;transport=udp
Server: Aastra 51i/2.2.0.166
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: NOTIFY
PBX*CLI> exit