Outgoing Calls Ring 3 Times then Reorder

I am having a curious issue with my Asterisk setup. I am sort of new to Asterisk and seem to have everything working except for this:

Whenever I dial an external number from my extension, the number will ring 3 times and then goes to a reorder (fast busy) tone. If the number that is dialed is answered before the reorder tone, everything is fine. But after 3 rings the reorder tone and the call can not be completed.

My setup is as follows:
Asterisk 1.4.24.1
FreePBX 2.5.1.2
2 Cisco 7970 phones
BroadVoice SIP Trunk for outgoing/incoming

If anyone has any suggestions to try I am open for ideas.

UPDATE: Interestingly if I use ada on my desktop to make a call (it calls my extension and then dials out) I do not have this issue. Just when I dial out from the phone I seem to have the issue. I am going to try to install a soft phone and see if I can dial out.

Hi

without letting anyone see the dialplan for the call, other than telling us you cant call what do you want us to do ???

Ian

This is something has to do with zapata.conf so please post your zapata.conf here

I do not have a zapata.conf

I have zapata_additional.conf and zapata.conf.template but no zapata.conf.

I copied zapata.conf.template to zapata.conf just to try it and it does not work.

Here is what it is now:

[code];# Flash Operator Panel will parse this file for zap trunk buttons
;# AMPLABEL will be used for the display labels on the buttons

;# %c Zap Channel number
;# %n Line number
;# %N Line number, but restart counter
;# Example:
;# ;AMPLABEL:Channel %c - Button %n

;# For Zap/* buttons use the following
;# (where x=number of buttons to dislpay)
;# ;AMPWILDCARDLABEL(x):MyLabel

[channels]
language=en

; include zap extensions defined in AMP
#include zapata_additional.conf

; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2
[/code]

As for the dialplan. Which dialplan do you need to see? The dialplan.xml that is loaded into the phone is as follows.

<DIALTEMPLATE> <TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else --> </DIALTEMPLATE>

Let me know what else could help diagnose this. As I said before I am fairly new to Asterisk.

would you please submit some of the CLI logs when you make any external call!

Here is my CLI while making an outgoing call.

-- Executing [7125726@from-internal:1] Macro("SIP/400-022b9d20", "user-callerid|SKIPTTL|") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/400-022b9d20", "AMPUSER=400") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/400-022b9d20", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/400-022b9d20", "1|Set|REALCALLERIDNUM=400") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/400-022b9d20", "AMPUSER=400") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/400-022b9d20", "AMPUSERCIDNAME=Scott Dickens") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/400-022b9d20", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/400-022b9d20", "AMPUSERCID=400") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/400-022b9d20", "CALLERID(all)="Scott Dickens" <400>") in new stack -- Executing [s@macro-user-callerid:9] Set("SIP/400-022b9d20", "REALCALLERIDNUM=400") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/400-022b9d20", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("SIP/400-022b9d20", "Using CallerID "Scott Dickens" <400>") in new stack -- Executing [7125726@from-internal:2] Set("SIP/400-022b9d20", "_NODEST=") in new stack -- Executing [7125726@from-internal:3] Macro("SIP/400-022b9d20", "record-enable|400|OUT|") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/400-022b9d20", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/400-022b9d20", "recordingcheck|20090506-090011|1241618411.7") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20090506-090011|1241618411.7: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:5] MacroExit("SIP/400-022b9d20", "") in new stack -- Executing [7125726@from-internal:4] Macro("SIP/400-022b9d20", "dialout-trunk|2|7125726||") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/400-022b9d20", "DIAL_TRUNK=2") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/400-022b9d20", "0?sub-pincheck|s|1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/400-022b9d20", "0?disabletrunk|1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/400-022b9d20", "DIAL_NUMBER=7125726") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/400-022b9d20", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/400-022b9d20", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/400-022b9d20", "0?nomax") in new stack -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/400-022b9d20", "0?chanfull") in new stack -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/400-022b9d20", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/400-022b9d20", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/400-022b9d20", "outbound-callerid|2") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/400-022b9d20", "0|SetCallerPres|") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/400-022b9d20", "0|Set|REALCALLERIDNUM=400") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/400-022b9d20", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/400-022b9d20", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/400-022b9d20", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/400-022b9d20", "TRUNKOUTCID=OrderCounter.com <8504177821>") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/400-022b9d20", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/400-022b9d20", "1|Set|CALLERID(all)=OrderCounter.com <8504177821>") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/400-022b9d20", "0|Set|CALLERID(all)=") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/400-022b9d20", "0|SetCallerPres|prohib_passed_screen") in new stack -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/400-022b9d20", "1|AGI|fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Using pattern 1850|NXXXXXX -- AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:13] Set("SIP/400-022b9d20", "OUTNUM=7125726") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/400-022b9d20", "custom=SIP/Broadvoice") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/400-022b9d20", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/400-022b9d20", "dialout-trunk-predial-hook|") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/400-022b9d20", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/400-022b9d20", "0?bypass|1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/400-022b9d20", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/400-022b9d20", "SIP/Broadvoice/7125726|300|") in new stack -- Called Broadvoice/7125726 -- SIP/Broadvoice-02225350 is making progress passing it to SIP/400-022b9d20 == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/400-022b9d20' in macro 'dialout-trunk' == Spawn extension (from-internal, 7125726, 4) exited non-zero on 'SIP/400-022b9d20' -- Executing [h@macro-dialout-trunk:1] Macro("SIP/400-022b9d20", "hangupcall|") in new stack -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/400-022b9d20", "vw") in new stack -- Executing [s@macro-hangupcall:2] NoCDR("SIP/400-022b9d20", "") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/400-022b9d20", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] GotoIf("SIP/400-022b9d20", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] GotoIf("SIP/400-022b9d20", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [s@macro-hangupcall:11] Hangup("SIP/400-022b9d20", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/400-022b9d20' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/400-022b9d20'

In an Asterisk context, “dialplan” means extensions.conf or extensions.ael.

You seem to have in band call progress from the SIP provider. That could mean that the fast busy is coming from them, rather than Asterisk. The call then drops. I don’t know if it drops immediately or after a few seconds. Especially in the former case, using sip history or sip set debug might identify the exact reason they rejected the call.

This is what I get with set sip debug.

[code]<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to 147.135.32.221:5060:
ACK sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK743cb1eb;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as045bc70e
To: sip:7125726@sip.broadvoice.com;tag=4567
Contact: sip:8504177821@192.168.1.50
Call-ID: 5c6d857f0bdfef3a0081c3a7798aa770@sip.broadvoice.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Really destroying SIP dialog ‘5c6d857f0bdfef3a0081c3a7798aa770@sip.broadvoice.co m’ Method: INVITE
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK1e56d47f
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:10:16 GMT
CSeq: 4720 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=u dp>;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model .ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK1e56d47f;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4720 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK1e56d47f;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4720 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0d233426"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.16 8.1.75’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK2c46615c
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:10:16 GMT
CSeq: 4721 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=u dp>;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model .ccm.cisco.com="30006"
Authorization: Digest username=“400”,realm=“asterisk”,uri=“sip:192.168.1.50”,res ponse=“742a7731d4315834bc579bb59e2a52ba”,nonce=“0d233426”,algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (14 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK2c46615c;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4721 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK2c46615c;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce2094d4c18a44e-a4767299
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4721 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=u dp>;expires=60
Date: Wed, 06 May 2009 14:10:25 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.16 8.1.75’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘030915bb03e1454f072073d305ab342d@127.0.1.1’ Method : REGISTER
Scheduling destruction of SIP dialog ‘1fed893e137e0a0e5938c9e966f47d64@192.168.1 .50’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.75:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK323a56ee;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5119879f
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Contact: sip:Unknown@192.168.1.50
Call-ID: 1fed893e137e0a0e5938c9e966f47d64@192.168.1.50
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 81

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.50
Voice-Message: 0/0


Server-01*CLI>
<— SIP read from 192.168.1.75:51724 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK323a56ee;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5119879f
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Call-ID: 1fed893e137e0a0e5938c9e966f47d64@192.168.1.50
Date: Wed, 06 May 2009 14:10:21 GMT
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘0b9492c74e8d72285743ddb02b63098b@192.168.1.50’ Met hod: NOTIFY
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6b86b587;rport
From: sip:8504177821@sip.broadvoice.com;tag=as14961c33
To: sip:8504177821@sip.broadvoice.com
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfue3wjwlT1mcs6mBW”, response=“e01334f60ac4b443dcfc81c38460dcf7”, qop=auth, cnonce=“0a8e8180”, nc=00000002
Expires: 120
Contact: sip:400@192.168.1.50
Event: registration
Content-Length: 0


Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 200 OK
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 109 REGISTER
From: sip:8504177821@sip.broadvoice.com;tag=as14961c33
To: sip:8504177821@sip.broadvoice.com
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6b86b587;received=70.167.229.87;rport=5060
Contact: sip:400@192.168.1.50
Expires: 30
Event: registration
Content-Length: 0
[/code]

I think the interesting bit was just before the bit you quoted. And think you’ve started with the ACK to the final call rejection.

I tried to get it all this time.

[code]<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘030915bb03e1454f072073d305ab342d@127.0.1.1’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
INVITE sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Expires: 20LI>
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 289 0 IN IP4 192.168.1.75
s=SIP Call
t=0 0
m=audio 23236 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.75
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (18 headers 13 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)
Using INVITE request as basis request - 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75

<— Reliably Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as06d9944f
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3d5d199d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75’ in 32000 ms (Method: INVITE)
Found user '400’
Server-01*CLI>
<— SIP read from 192.168.1.75:51809 —>
ACK sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as06d9944f
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 101 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
INVITE sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“db4dbe0ce8a5a774a16a860f984e08ea”,nonce=“3d5d199d”,algorithm=MD5
Expires: 20
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 289 0 IN IP4 192.168.1.75
s=SIP Call
t=0 0
m=audio 23236 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.75
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (19 headers 13 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)
Using INVITE request as basis request - 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Found user '400’
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 192.168.1.75:23236
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 - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/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 192.168.1.75:23236
Looking for 7125726 in from-internal (domain 192.168.1.50)
list_route: hop: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7125726@192.168.1.50
Content-Length: 0

<------------>
– Executing [7125726@from-internal:1] Macro(“SIP/400-90028250”, “user-callerid|SKIPTTL|”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/400-90028250”, “AMPUSER=400”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/400-90028250”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/400-90028250”, “1|Set|REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/400-90028250”, “AMPUSER=400”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/400-90028250”, “AMPUSERCIDNAME=Scott Dickens”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/400-90028250”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/400-90028250”, “AMPUSERCID=400”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/400-90028250”, “CALLERID(all)=“Scott Dickens” <400>”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/400-90028250”, “REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/400-90028250”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/400-90028250”, “Using CallerID “Scott Dickens” <400>”) in new stack
– Executing [7125726@from-internal:2] Set(“SIP/400-90028250”, “_NODEST=”) in new stack
– Executing [7125726@from-internal:3] Macro(“SIP/400-90028250”, “record-enable|400|OUT|”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/400-90028250”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/400-90028250”, “recordingcheck|20090506-094455|1241621095.6”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090506-094455|1241621095.6: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/400-90028250”, “”) in new stack
– Executing [7125726@from-internal:4] Macro(“SIP/400-90028250”, “dialout-trunk|2|7125726||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/400-90028250”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/400-90028250”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/400-90028250”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/400-90028250”, “DIAL_NUMBER=7125726”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/400-90028250”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/400-90028250”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/400-90028250”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/400-90028250”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/400-90028250”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/400-90028250”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/400-90028250”, “outbound-callerid|2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/400-90028250”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/400-90028250”, “0|Set|REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/400-90028250”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/400-90028250”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/400-90028250”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/400-90028250”, “TRUNKOUTCID=OrderCounter.com <8504177821>”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/400-90028250”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/400-90028250”, “1|Set|CALLERID(all)=OrderCounter.com <8504177821>”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/400-90028250”, “0|Set|CALLERID(all)=”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/400-90028250”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/400-90028250”, “1|AGI|fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 1850|NXXXXXX
– AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/400-90028250”, “OUTNUM=7125726”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/400-90028250”, “custom=SIP/Broadvoice”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/400-90028250”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/400-90028250”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/400-90028250”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/400-90028250”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/400-90028250”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/400-90028250”, “SIP/Broadvoice/7125726|300|”) in new stack
Audio is at 192.168.1.50 port 19056
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.32.221:5060:
INVITE sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 18549 18549 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 19056 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called Broadvoice/7125726

Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 100 Trying
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;received=70.167.229.87;rport=5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=ghij
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;received=70.167.229.87;rport=5060
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXfue552qrTz7eynoBW"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 147.135.32.221:5060:
ACK sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=ghij
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Audio is at 192.168.1.50 port 19056
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.32.221:5060:
INVITE sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri="sip:7125726@sip.broadvoice.com", nonce=“BroadWorksXfue552qrTz7eynoBW”, response=“deca2fc65d0cc8ed176d5f6910ae14af”, qop=auth, cnonce=“6b695575”, nc=00000001
Date: Wed, 06 May 2009 14:44:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 18549 18550 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 19056 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 100 Trying
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 183 Session Progress
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Contact: sip:7125726@147.135.32.221
Content-Length: 170
Content-Type: application/sdp

v=0
o=2475106551 10 10 IN IP4 147.135.32.247
s=-
c=IN IP4 147.135.32.247
t=0 0
m=audio 36276 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (9 headers 8 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 147.135.32.247:36276
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 147.135.32.247:36276
– SIP/Broadvoice-00b70c60 is making progress passing it to SIP/400-90028250
Audio is at 192.168.1.50 port 14632
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7125726@192.168.1.50
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 18549 18549 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 14632 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

<------------>
Really destroying SIP dialog ‘030915bb03e1454f072073d305ab342d@127.0.1.1’ Method: REGISTER
Server-01*CLI>
<— SIP read from 192.168.1.76:49160 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:54 GMT
CSeq: 5483 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79cb2f9”;+u.sip!model.ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.76 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5483 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50;tag=as79139fa8
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5483 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="54499ea5"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.76:49160 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:54 GMT
CSeq: 5484 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79cb2f9”;+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest username=“337”,realm=“asterisk”,uri=“sip:192.168.1.50”,response=“0929bfac132e5c01d39861527ff6cf69”,nonce=“54499ea5”,algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (14 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.76 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5484 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50;tag=as79139fa8
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5484 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;expires=60
Date: Wed, 06 May 2009 14:45:03 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
CSeq: 4886 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model.ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4886 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4886 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0f86f375"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
CSeq: 4887 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest username=“400”,realm=“asterisk”,uri=“sip:192.168.1.50”,response=“5f5a7bc5af19f61c85eaa8ac89f74569”,nonce=“0f86f375”,algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30

<------------->
— (14 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4887 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4887 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;expires=60
Date: Wed, 06 May 2009 14:45:04 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75’ in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog ‘792c1f223157143e197b1e795a56920f@192.168.1.50’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.75:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK374d8bc9;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as2b89eccd
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Contact: sip:Unknown@192.168.1.50
Call-ID: 792c1f223157143e197b1e795a56920f@192.168.1.50
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 81

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.50
Voice-Message: 0/0


Server-01*CLI>
<— SIP read from 192.168.1.75:51810 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK374d8bc9;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as2b89eccd
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Call-ID: 792c1f223157143e197b1e795a56920f@192.168.1.50
Date: Wed, 06 May 2009 14:44:59 GMT
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘792c1f223157143e197b1e795a56920f@192.168.1.50’ Method: NOTIFY
Scheduling destruction of SIP dialog ‘4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.76:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2b3a005e;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5eb7ca8a
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp
Contact: sip:Unknown@192.168.1.50
Call-ID: 4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 81

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.50
Voice-Message: 0/0


Server-01*CLI>
<— SIP read from 192.168.1.76:51561 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2b3a005e;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5eb7ca8a
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp
Call-ID: 4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50
Date: Wed, 06 May 2009 14:45:02 GMT
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50’ Method: NOTIFY
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
CANCEL sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:45:06 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP7970G/8.3.0
Content-Length: 0
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“d3e7327ba70cb5ea39ac0aecddb44184”,nonce=“3d5d199d”,algorithm=MD5

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)

<— Reliably Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 147.135.32.221:5060:
CANCEL sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Scheduling destruction of SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ in 32000 ms (Method: INVITE)
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/400-90028250’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 7125726, 4) exited non-zero on ‘SIP/400-90028250’
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/400-90028250”, “hangupcall|”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/400-90028250”, “vw”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/400-90028250”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/400-90028250”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/400-90028250”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/400-90028250”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/400-90028250”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/400-90028250’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/400-90028250’
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
ACK sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKb516ab71
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:45:06 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7970G/8.3.0
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“d3e7327ba70cb5ea39ac0aecddb44184”,nonce=“3d5d199d”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75’ Method: ACK
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 200 OK
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 CANCEL
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 487 Request Terminated
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to 147.135.32.221:5060:
ACK sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.comsip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Really destroying SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ Method: INVITE
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK20184940;rport
From: sip:8504177821@sip.broadvoice.com;tag=as05667d6a
To: sip:8504177821@sip.broadvoice.com
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 195 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfue3wjwlT1mcs6mBW”, response=“eb001b625a8e8d608c2f0d691c88ac22”, qop=auth, cnonce=“1830fa28”, nc=00000058
Expires: 120
Contact: sip:400@192.168.1.50
Event: registration
Content-Length: 0


Retransmitting #1 (no NAT) to 147.135.32.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK20184940;rport
From: sip:8504177821@sip.broadvoice.com;tag=as05667d6a
To: sip:8504177821@sip.broadvoice.com
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 195 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfue3wjwlT1mcs6mBW”, response=“eb001b625a8e8d608c2f0d691c88ac22”, qop=auth, cnonce=“1830fa28”, nc=00000058
Expires: 120
Contact: sip:400@192.168.1.50
Event: registration
Content-Length: 0
[/code]

The call ended because you cleared it. The fast busy was coming from the service provider, not from Asterisk.

It looks as though you’ve used local number dialled format. I would have thought the SIP gateway would have required national number, at least. I’m assuming NANP.

I have tried both, with the same result.

You may need to ask the service provider why they are rejecting your call. They should be able to tell from their logs.