TRUNK Dial failed due to CHANUNAVAIL

Hi All, I have been through forums for the last week nearly now, but cant seem to get incoming or out going calls working. I have tried most things that have been suggested online more than once. Im getting the all familiar ‘All Circuits are busy’ I have internal calls working from extension to extension. But cant make or receive external calls.
I am using a SIP Trunk which is registering successful according to the freePBX Status Page.
Just to keep things simple i have set dial patterns for local calls and thats it. Im in Australia, Sydney. a local number would be 8XX1 XXXX.
Thanks for any help, from what I have seen i love the trixbox. Just need to work out what is wrong with my config.
DELL Vostro200
trixbox 2.6.0.0-4
freePBX 2.4.0-2
Asterisk 1.4.19-1
Nortel LG AX-IAD100
Linksys SPA942-EU
SIP TRUNK

Outbound Caller ID: “Co. Name” <6128xxxxxxx>
Never Override CallerID: not ticked
Maximum Channels: left blank
Disable Trunk: not ticked
Monitor Trunk Failures: left blank
Outgoing Dial Rules
Dial Rules:
612+NXXXXXXX
Outbound Dial Prefix: left blank
Outgoing Settings:
Trunk Name: MytelcoXXXX
Peer Details:
allow=alaw&ulaw&gsm
canredirect=no
canreinvite=no
disallow=all
host=sipXX.XXXXXXX.XXXXX.XXX.XX
insecure=very
secret=password
type=peer
[email protected]
Incoming Settings:
User Context: [email protected]
User Details:
canreinvite=no
context=from-trunk
[email protected]
qualify=no
secret=9md8RdSv
type=user
[email protected]
Registration String:
6128XXXXXXX:[email protected]

My Only Outbound:
Route Name: OXXXXXXX5
Route Password: left blank
Pin Set: NONE
Emergency Dialing: Unticked
Intra Company Route: Unticked
Music on Hold? default
Dial Patterns: NXXXXXXX
Trunk Sequence: SIP/OXXXXXXX5

Connected to Asterisk 1.4.19-1 RPM by [email protected] currently runnin g on trixbox1 (pid = 5455)
Verbosity was 0 and is now 9
[May 8 18:02:10] == Parsing ‘/etc/asterisk/manager.conf’: [May 8 18:02:10] F ound
[May 8 18:02:10] == Parsing ‘/etc/asterisk/manager_additional.conf’: [May 8 18:02:10] Found
[May 8 18:02:10] == Parsing ‘/etc/asterisk/manager_custom.conf’: [May 8 18:0 2:10] Found
[May 8 18:02:10] == Manager ‘admin’ logged on from 127.0.0.1
[May 8 18:02:12] == Manager ‘admin’ logged off from 127.0.0.1
[May 8 18:02:18] == Parsing ‘/etc/asterisk/manager.conf’: [May 8 18:02:18] F ound
[May 8 18:02:18] == Parsing ‘/etc/asterisk/manager_additional.conf’: [May 8 18:02:18] Found
[May 8 18:02:18] == Parsing ‘/etc/asterisk/manager_custom.conf’: [May 8 18:0 2:18] Found
[May 8 18:02:18] == Manager ‘admin’ logged on from 127.0.0.1
[May 8 18:02:19] == Manager ‘admin’ logged off from 127.0.0.1
[May 8 18:02:25] == Parsing ‘/etc/asterisk/manager.conf’: [May 8 18:02:25] F ound
[May 8 18:02:25] == Parsing ‘/etc/asterisk/manager_additional.conf’: [May 8 18:02:25] Found
[May 8 18:02:25] == Parsing ‘/etc/asterisk/manager_custom.conf’: [May 8 18:0 2:25] Found
[May 8 18:02:25] == Manager ‘admin’ logged on from 127.0.0.1
[May 8 18:02:26] == Manager ‘admin’ logged off from 127.0.0.1
[May 8 18:02:38] – Executing [[email protected]:1] Macro(“SIP/2001-09 a2af70”, “user-callerid|SKIPTTL|”) in new stack
[May 8 18:02:38] – Executing [[email protected]:1] NoOp(“SIP/2001-09a2 af70”, “user-callerid: device 2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:2] Set(“SIP/2001-09a2a f70”, “AMPUSER=2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:3] GotoIf(“SIP/2001-09 a2af70”, “0?report”) in new stack
[May 8 18:02:38] – Executing [[email protected]:4] ExecIf(“SIP/2001-09 a2af70”, “1|Set|REALCALLERIDNUM=2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:5] NoOp(“SIP/2001-09a2 af70”, “REALCALLERIDNUM is 2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:6] Set(“SIP/2001-09a2a f70”, “AMPUSER=2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:7] Set(“SIP/2001-09a2a f70”, “AMPUSERCIDNAME=User Name”) in new stack
[May 8 18:02:38] – Executing [[email protected]:8] GotoIf(“SIP/2001-09 a2af70”, “0?report”) in new stack
[May 8 18:02:38] – Executing [[email protected]:9] Set(“SIP/2001-09a2a f70”, “AMPUSERCID=2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:10] Set(“SIP/2001-09a2 af70”, “CALLERID(all)=“User Name” <2001>”) in new stack
[May 8 18:02:38] – Executing [[email protected]:11] Set(“SIP/2001-09a2 af70”, “REALCALLERIDNUM=2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:12] ExecIf(“SIP/2001-0 9a2af70”, “0|Set|CHANNEL(language)=”) in new stack
[May 8 18:02:38] – Executing [[email protected]:13] NoOp(“SIP/2001-09a 2af70”, “TTL: ARG1: SKIPTTL”) in new stack
[May 8 18:02:38] – Executing [[email protected]:14] GotoIf(“SIP/2001-0 9a2af70”, “1?continue”) in new stack
[May 8 18:02:38] – Goto (macro-user-callerid,s,23)
[May 8 18:02:38] – Executing [[email protected]:23] NoOp(“SIP/2001-09a 2af70”, “Using CallerID “User Name” <2001>”) in new stack
[May 8 18:02:38] – Executing [[email protected]:2] Set(“SIP/2001-09a2 af70”, “_NODEST=”) in new stack
[May 8 18:02:38] – Executing [[email protected]:3] Macro(“SIP/2001-09 a2af70”, “record-enable|2001|OUT|”) in new stack
[May 8 18:02:38] – Executing [[email protected]:1] GotoIf(“SIP/2001-09 a2af70”, “0?2:4”) in new stack
[May 8 18:02:38] – Goto (macro-record-enable,s,4)
[May 8 18:02:38] – Executing [[email protected]:4] AGI(“SIP/2001-09a2a f70”, “recordingcheck|2001508-180238|1210233758.0”) in new stack
[May 8 18:02:38] – Launched AGI Script /var/lib/asterisk/agi-bin/recording check
[May 8 18:02:38] recordingcheck|20080508-180238|1210233758.0: Outbound record ing not enabled
[May 8 18:02:38] – AGI Script recordingcheck completed, returning 0
[May 8 18:02:38] – Executing [[email protected]:5] NoOp(“SIP/2001-09a2 af70”, “No recording needed”) in new stack
[May 8 18:02:38] – Executing [[email protected]:4] Macro(“SIP/2001-09 a2af70”, “dialout-trunk|2|8XXXXXXX||”) in new stack
[May 8 18:02:38] – Executing [[email protected]:1] Set(“SIP/2001-09a2a f70”, “DIAL_TRUNK=2”) in new stack
[May 8 18:02:38] – Executing [[email protected]:2] ExecIf(“SIP/2001-09 a2af70”, “0|Authenticate|”) in new stack
[May 8 18:02:38] – Executing [[email protected]:3] GotoIf(“SIP/2001-09 a2af70”, “0?disabletrunk|1”) in new stack
[May 8 18:02:38] – Executing [[email protected]:4] Set(“SIP/2001-09a2a f70”, “DIAL_NUMBER=8XXXXXXX”) in new stack
[May 8 18:02:38] – Executing [[email protected]:5] Set(“SIP/2001-09a2a f70”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
[May 8 18:02:38] – Executing [[email protected]:6] Set(“SIP/2001-09a2a f70”, “GROUP()=OUT_2”) in new stack
[May 8 18:02:38] – Executing [[email protected]:7] GotoIf(“SIP/2001-09 a2af70”, “1?nomax”) in new stack
[May 8 18:02:38] – Goto (macro-dialout-trunk,s,9)
[May 8 18:02:38] – Executing [[email protected]:9] GotoIf(“SIP/2001-09 a2af70”, “0?skipoutcid”) in new stack
[May 8 18:02:38] – Executing [[email protected]:10] Set(“SIP/2001-09a2 af70”, “DIAL_TRUNK_OPTIONS=”) in new stack
[May 8 18:02:38] – Executing [[email protected]:11] Macro(“SIP/2001-09 a2af70”, “outbound-callerid|2”) in new stack
[May 8 18:02:38] – Executing [[email protected]:1] GotoIf(“SIP/200 1-09a2af70”, “1?start”) in new stack
[May 8 18:02:38] – Goto (macro-outbound-callerid,s,3)
[May 8 18:02:38] – Executing [[email protected]:3] NoOp(“SIP/2001- 09a2af70”, “REALCALLERIDNUM is 2001”) in new stack
[May 8 18:02:38] – Executing [[email protected]:4] GotoIf(“SIP/200 1-09a2af70”, “1?normcid”) in new stack
[May 8 18:02:38] – Goto (macro-outbound-callerid,s,9)
[May 8 18:02:38] – Executing [[email protected]:9] Set(“SIP/2001-0 9a2af70”, “USEROUTCID=”) in new stack
[May 8 18:02:38] – Executing [[email protected]:10] Set(“SIP/2001- 09a2af70”, “EMERGENCYCID=”) in new stack
[May 8 18:02:38] – Executing [[email protected]:11] Set(“SIP/2001- 09a2af70”, “TRUNKOUTCID=“Co. Name” <6128XXXXXXX>”) in new stack
[May 8 18:02:38] – Executing [[email protected]:12] GotoIf(“SIP/20 01-09a2af70”, “1?trunkcid”) in new stack
[May 8 18:02:38] – Goto (macro-outbound-callerid,s,16)
[May 8 18:02:38] – Executing [[email protected]:16] GotoIf(“SIP/20 01-09a2af70”, “0?usercid”) in new stack
[May 8 18:02:38] – Executing [[email protected]:17] Set(“SIP/2001- 09a2af70”, “CALLERID(all)=Co. Name <6128XXXXXXX>”) in new stack
[May 8 18:02:38] – Executing [[email protected]:18] GotoIf(“SIP/20 01-09a2af70”, “1?report”) in new stack
[May 8 18:02:38] – Goto (macro-outbound-callerid,s,22)
[May 8 18:02:38] – Executing [[email protected]:22] NoOp(“SIP/2001 -09a2af70”, “CallerID set to “Co. Name” <6128XXXXXXX>”) in new stack
[May 8 18:02:38] – Executing [[email protected]:12] AGI(“SIP/2001-09a2 af70”, “fixlocalprefix”) in new stack
[May 8 18:02:38] – Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalp refix
[May 8 18:02:38] > fixlocalprefix: Using pattern 612+NXXXXXXX
[May 8 18:02:38] == fixlocalprefix: Dialpattern 612+NXXXXXXX matched. 8XXXXXXX -> 6128XXXXXXX
[May 8 18:02:38] – AGI Script fixlocalprefix completed, returning 0
[May 8 18:02:38] – Executing [[email protected]:13] Set(“SIP/2001-09a2 af70”, “OUTNUM=6128XXXXXXX”) in new stack
[May 8 18:02:38] – Executing [[email protected]:14] Set(“SIP/2001-09a2 af70”, “custom=SIP/OXXXXXXX5”) in new stack
[May 8 18:02:38] – Executing [[email protected]:15] GotoIf(“SIP/2001-0 9a2af70”, “1?gocall”) in new stack
[May 8 18:02:38] – Goto (macro-dialout-trunk,s,17)
[May 8 18:02:38] – Executing [[email protected]:17] Macro(“SIP/2001-09 a2af70”, “dialout-trunk-predial-hook|”) in new stack
[May 8 18:02:38] – Executing [[email protected]:18] GotoIf(“SIP/2001-0 9a2af70”, “0?bypass|1”) in new stack
[May 8 18:02:38] – Executing [[email protected]:19] GotoIf(“SIP/2001-0 9a2af70”, “0?customtrunk”) in new stack
[May 8 18:02:38] – Executing [[email protected]:20] Dial(“SIP/2001-09a 2af70”, “SIP/OXXXXXXX5/6128XXXXXXX|300|”) in new stack
[May 8 18:02:38] – Couldn’t call OXXXXXXX5/6128XXXXXXX
[May 8 18:02:38] == Everyone is busy/congested at this time (0:0/0/0)
[May 8 18:02:38] – Executing [[email protected]:21] Goto(“SIP/2001-09a 2af70”, “s-CHANUNAVAIL|1”) in new stack
[May 8 18:02:38] – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[May 8 18:02:38] – Executing [[email protected]:1] GotoIf( “SIP/2001-09a2af70”, “1?noreport”) in new stack
[May 8 18:02:38] – Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
[May 8 18:02:38] – Executing [[email protected]:3] NoOp(“S IP/2001-09a2af70”, “TRUNK Dial failed due to CHANUNAVAIL - failing through to ot her trunks”) in new stack
[May 8 18:02:38] – Executing [[email protected]:5] Macro(“SIP/2001-09 a2af70”, “outisbusy|”) in new stack
[May 8 18:02:38] – Executing [[email protected]:1] Playback(“SIP/2001-09a2 af70”, “all-circuits-busy-now|noanswer”) in new stack
[May 8 18:02:38] – Playing ‘all-circuits-busy-now’ (la nguage ‘en’)
[May 8 18:02:40] – Executing [[email protected]:2] Playback(“SIP/2001-09a2 af70”, “pls-try-call-later|noanswer”) in new stack
[May 8 18:02:40] – Playing ‘pls-try-call-later’ (langu age ‘en’)
[May 8 18:02:42] – Executing [[email protected]:3] Macro(“SIP/2001-09a2af7 0”, “hangupcall”) in new stack
[May 8 18:02:42] – Executing [[email protected]:1] ResetCDR(“SIP/2001-09a 2af70”, “w”) in new stack
[May 8 18:02:42] – Executing [[email protected]:2] NoCDR(“SIP/2001-09a2af 70”, “”) in new stack
[May 8 18:02:42] – Executing [[email protected]:3] GotoIf(“SIP/2001-09a2a f70”, “1?skiprg”) in new stack
[May 8 18:02:42] – Goto (macro-hangupcall,s,6)
[May 8 18:02:42] – Executing [[email protected]:6] GotoIf(“SIP/2001-09a2a f70”, “1?skipblkvm”) in new stack
[May 8 18:02:42] – Goto (macro-hangupcall,s,9)
[May 8 18:02:42] – Executing [[email protected]:9] GotoIf(“SIP/2001-09a2a f70”, “1?theend”) in new stack
[May 8 18:02:42] – Goto (macro-hangupcall,s,11)
[May 8 18:02:42] – Executing [[email protected]:11] Hangup(“SIP/2001-09a2 af70”, “”) in new stack
[May 8 18:02:42] == Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/2001-09a2af70’ in macro ‘hangupcall’
[May 8 18:02:42] == Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/2001-09a2af70’ in macro ‘outisbusy’
[May 8 18:02:42] == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2001-09a2af70’
trixbox1*CLI>
[trixbox1.localdomain ~]#

Additional Info:
I have managed to register X-Lite with our VSP, they appear to be the same as 2B in HK using nortel comms manager, successfully receiving and making calls externally with X-Lite. I am a NOOB at this the registering seems fine, obviously struggling with the outgoing and incoming peer and user details. Any help would be really greatly appreciated!!
Thanks so much :stuck_out_tongue:
Oli
Setting used with X-Lite:
Display Name: 'my name’
User Name: 61XXXXXXXX5
Password: ‘password’
Authorization user name: 61XXXXXXXX5
Domain: sip01.XXXXXXX.XXXXX.XXX.au
’ticked’ Register with domain and receive incoming calls
Send outbound via: domain
Dialing plan: #1\a\a.T;match=1;prestrip=2;
NOOB
Slowly going bald :stuck_out_tongue:

I have a trunk that doesnt require registration, all they need is the correct terminal number from us, ie our wan IP, but on my system they see nothing coming back from us on an invite. Have you found out anything about this issue elsewhere?

Turn on SIP debugging. I’ve had this happen for multiple reasons, usually due to codec negotiation. Regardless, SIP debugging should tell you the exact reason the channel is unavailable (Asterisk doesn’t necessarily return the exact SIP error code.)

Turn on SIP debugging by typing the command:

“sip debug”

at the Asterisk command line.

“sip no debug” turns it off.

Thanks Kodak I will run this today and post results, what is the best way to check codecs that are installed and operational?

That’s more difficult to answer than it may seem. You can see the codec modules by typing:

show modules like codec

but that doesn’t mean that you’ve allowed any of them for a particular trunk or endpoint. That’s what the “allow=” and “deny=” directives are for in the various places they exist.

The SIP debugging should show what each side of the SIP conversation thinks it has available, as far as codecs go. But don’t get hung up on codecs yet, as that may not be the reasons, I merely said that a few of the times I’ve had this problem it was a codec issue. Just check the debug messages and see what they say.

Hi Thanks so much for the feedback, Hope this helps!
Not sure what i am looking for, hope this is what you ment.
I have not placed a call just let the SIP Debug mode run.
Thanks again!

trixbox1*CLI>
<— SIP read from 2XX.XX.XX.3:5060 —>
SIP/2.0 400 SIP Parser Error : Missing ‘@’, line 3, column 26
From: “No CallID” <sip:No [email protected]>;tag=as1a296d76
To: sip:sipxx.xxxxxxx.xxxxx.xxx.au
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP

10.1.1.7:5060;received=10.204.79.9;rport=50020;branch=z9hG4bK6e4fbbbb
User-Agent: oxxxs
Max-Forwards: 70
Supported: replaces
Date: Wed, 14 May 2008 03:24:20 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method:

OPTIONS
Reliably Transmitting (NAT) to 10.1.1.4:5062:
OPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK5ad762a7;rport
From: “No CallID” <sip:No [email protected]>;tag=as4ea6ff0f
To: sip:[email protected]:5062
Contact: <sip:No [email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: oxxxs
Max-Forwards: 70
Date: Wed, 14 May 2008 03:24:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


trixbox1*CLI>
<— SIP read from 10.1.1.4:5062 —>
SIP/2.0 200 OK
To: sip:[email protected]:5062;tag=63250519385dcc3ai2
From: “No CallID” <sip:No [email protected]>;tag=as4ea6ff0f
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK5ad762a7
Server: Linksys/SPA942-5.1.10
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method:

OPTIONS
trixbox1*CLI>
<— SIP read from 10.1.1.4:5062 —>
PING sip:10.1.1.7 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.4:5062;branch=z9hG4bK-78ac0cb5
From: “name” sip:[email protected];tag=c2478d56524c56o2
To: “name” sip:[email protected]
Call-ID: [email protected]
CSeq: 14011 PING
Max-Forwards: 70
Contact: “name” sip:[email protected]:5062
User-Agent: Linksys/SPA942-5.1.10
Proxy-Require: com.nortelnetworks.firewall
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.1.1.4 : 5062 (no NAT)
trixbox1*CLI>
<— Transmitting (no NAT) to 10.1.1.4:5062 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 10.1.1.4:5062;branch=z9hG4bK-78ac0cb5;received=10.1.1.4
From: “name” sip:[email protected];tag=c2478d56524c56o2
To: “name” sip:[email protected];tag=as6964b36f
Call-ID: [email protected]
CSeq: 14011 PING
User-Agent: oxxxs
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 10.1.1.2:10176:
OPTIONS sip:[email protected]:10176;rinstance=24c8d9eb22f9ad8d SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK7c400e32;rport
From: “No CallID” <sip:No [email protected]>;tag=as574f8c48
To: sip:[email protected]:10176;rinstance=24c8d9eb22f9ad8d
Contact: <sip:No [email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: oxxxs
Max-Forwards: 70
Date: Wed, 14 May 2008 03:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


trixbox1*CLI>
<— SIP read from 10.1.1.2:10176 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK7c400e32;rport=5060
Contact: sip:10.1.1.2:10176
To: sip:[email protected]:10176;rinstance=24c8d9eb22f9ad8d;tag=ea7dc56f
From: "No CallID"sip:No%[email protected];tag=as574f8c48
Call-ID: [email protected]
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method:

OPTIONS
trixbox1*CLI>
<— SIP read from 10.1.1.2:10176 —>

<------------->
— (0 headers 1 lines) —
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 2XX.XX.XX.3:5060:
REGISTER sip:sipxx.xxxxxxx.xxxxx.xxx.au SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK7ffccf8b;rport
From: sip:[email protected];tag=as27030364
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: oxxxs
Max-Forwards: 70
Authorization: Digest username=“6128XXXXXX5”, realm=“Realm”, algorithm=MD5,

uri=“sip:sipxx.xxxxxxx.xxxxx.xxx.au”,

nonce=“MTIxMDczNTM4NDMyNjZhNjJlMjQ3MjUyMThiMmVjYTRkOWJkZmI4MzU1MjNj”,

response=“22b59682b97af4bbf4512cf454136b02”, opaque="", qop=auth,

cnonce=“72ac69a4”, nc=00000002
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0


trixbox1*CLI>
<— SIP read from 2XX.XX.XX.3:5060 —>
SIP/2.0 100 Trying
From: sip:[email protected];tag=as27030364
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
Via: SIP/2.0/UDP

10.1.1.7:5060;received=10.204.79.9;rport=50020;branch=z9hG4bK7ffccf8b
Content-Length: 0

<------------->
— (7 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 2XX.XX.XX.3:5060 —>
SIP/2.0 200 Registration Successful
From: "SIPLineUser

SIPLineUser"sip:[email protected];tag=as27030364
To: sip:[email protected];tag=156589077
Call-ID: [email protected]
CSeq: 108 REGISTER
Via: SIP/2.0/UDP

10.1.1.7:5060;received=10.204.79.9;rport=50020;branch=z9hG4bK7ffccf8b
contact: sip:[email protected];expires=111
supported: com.nortelnetworks.firewall,p-

3rdpartycontrol,nosec,join,com.nortelnetworks.im.encryption
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]

in 32000 ms (Method: REGISTER)
trixbox1*CLI> e
<— SIP read from 10.1.1.4:5062 —>
PING sip:10.1.1.7 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.4:5062;branch=z9hG4bK-3515d9b7
From: “name” sip:[email protected];tag=c2478d56524c56o2
To: “name” sip:[email protected]
Call-ID: [email protected]
CSeq: 14012 PING
Max-Forwards: 70
Contact: “name” sip:[email protected]:5062
User-Agent: Linksys/SPA942-5.1.10
Proxy-Require: com.nortelnetworks.firewall
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.1.1.4 : 5062 (no NAT)
trixbox1*CLI> e
<— Transmitting (no NAT) to 10.1.1.4:5062 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 10.1.1.4:5062;branch=z9hG4bK-3515d9b7;received=10.1.1.4
From: “name” sip:[email protected];tag=c2478d56524c56o2
To: “name” sip:[email protected];tag=as2b42e68d
Call-ID: [email protected]
CSeq: 14012 PING
User-Agent: oxxxs
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
trixbox1*CLI> exit
[trixbox1.localdomain ~]#

Any thing stand out to be causing the problem?

Yeah, you kinda have to place a call in debugging mode, if you’re wanting to debug something that happens during a call.

Don’t paste the output here, paste it at http://pastebin.ca or http://pastebin.com so it doesn’t take up so much vertical space.

Cool thanks for the tip!

I have placed a call with sip debug enabled. Results here:

http://p.caboo.se/197343

Thanks again

I don’t think that’s the whole SIP conversation. It looks like just the end of it.

Ok sorry bout that, I assumed the debug mode would keep logging so i stopped it after i had place a test call. At what stage is it safe to stop the debug?

This is the end of the SIP conversation. You need to capture everything from the whole call. I honestly don’t know how to make that any more clear.

Hi managed to paste it all here:

http://pastebin.com/m776f2b5b

I could be because I’m tired as hell, but I’m not seeing your server trying to communicate with the provider. That leads me to believe the SIP trunk isn’t set up correctly, or your outbound route isn’t set up correctly.

Do the following and report it back here, from the cli:

sip show peers

sip show registry

Also, have you talked to your provider to ask them if they’re seeing your traffic?

Well, it’s registering. I’m kinda out of ideas. :confused:

Hi I have had not spoken to my provider yet, I will make a call today.
Below are sip peer and registry logs.
Thanks again for taking time to look at my problem!!!
Getting closer i must be :slight_smile:

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.05.20 09:35:36 =~=~=~=~=~=~=~=~=~=~=~=
login as: root
[email protected]'s password: 
Last login: Tue May 20 09:34:34 2008 from 10.1.1.2
Welcome to trixbox CE
-------------------------------------------------

For access to the trixbox web GUI use this URL
eth0: http://10.1.1.7

For help on trixbox commands you can use from this
command shell type help-trixbox.

[trixbox1.localdomain ~]# asterisk -r
Asterisk 1.4.19-1 RPM by [email protected], Copyright (C) 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 1.4.19-1 RPM by [email protected] currently running on trixbox1 (pid = 2692)
trixbox1*CLI> sip show peerstrixbox1*CLI> 
Name/username              Host            Dyn Nat ACL Port     Status       

        
SIPXXXXX/612XXXXXXX5       2xx.xx.xxx.3                5060     OK (24 ms)           
2001/2001                  10.1.1.2         D   N      46688    OK (101 ms)           
2000/2000                  10.1.1.4         D   N      5062     OK (10 ms)           
3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline]

e[Ktrixbox1*CLI> sip show registry
trixbox1*CLI> 
Host                            Username       Refresh State                Reg.Time                 
sipXX.XXXXXXX.XXXXX.XXX.au:506  612XXXXXXX5         99 Registered           Tue, 20 May 2008 09:35:42

e[Ktrixbox1*CLI> exit
[trixbox1.localdomain ~]# exit
logout

I tried with pbxinaflash and elastix