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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
username=6128XXXXXXX@sipXX.XXXXXXX.XXXXX.XXX.XX
Incoming Settings:
User Context: 6128XXXXXXX@sipXX.XXXXXXX.XXXXX.XXX.XX
User Details:
canreinvite=no
context=from-trunk
fromuser=6128XXXXXXX@sipXX.XXXXXXX.XXXXX.XXX.XX
qualify=no
secret=9md8RdSv
type=user
username=6128XXXXXXX@sipXX.XXXXXXX.XXXXX.XXX.XX
Registration String:
6128XXXXXXX:password@sipXX.XXXXXXX.XXXXX.XXX.XX
-----------------------
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 vc-rpms@voipconsulting.nl 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 [8XXXXXXX@from-internal:1] Macro("SIP/2001-09 a2af70", "user-callerid|SKIPTTL|") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:1] NoOp("SIP/2001-09a2 af70", "user-callerid: device 2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:2] Set("SIP/2001-09a2a f70", "AMPUSER=2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:3] GotoIf("SIP/2001-09 a2af70", "0?report") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:4] ExecIf("SIP/2001-09 a2af70", "1|Set|REALCALLERIDNUM=2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:5] NoOp("SIP/2001-09a2 af70", "REALCALLERIDNUM is 2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:6] Set("SIP/2001-09a2a f70", "AMPUSER=2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:7] Set("SIP/2001-09a2a f70", "AMPUSERCIDNAME=User Name") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:8] GotoIf("SIP/2001-09 a2af70", "0?report") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:9] Set("SIP/2001-09a2a f70", "AMPUSERCID=2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:10] Set("SIP/2001-09a2 af70", "CALLERID(all)="User Name" <2001>") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:11] Set("SIP/2001-09a2 af70", "REALCALLERIDNUM=2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:12] ExecIf("SIP/2001-0 9a2af70", "0|Set|CHANNEL(language)=") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid:13] NoOp("SIP/2001-09a 2af70", "TTL: ARG1: SKIPTTL") in new stack
[May 8 18:02:38] -- Executing [s@macro-user-callerid: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 [s@macro-user-callerid:23] NoOp("SIP/2001-09a 2af70", "Using CallerID "User Name" <2001>") in new stack
[May 8 18:02:38] -- Executing [8XXXXXXX@from-internal:2] Set("SIP/2001-09a2 af70", "_NODEST=") in new stack
[May 8 18:02:38] -- Executing [8XXXXXXX@from-internal:3] Macro("SIP/2001-09 a2af70", "record-enable|2001|OUT|") in new stack
[May 8 18:02:38] -- Executing [s@macro-record-enable: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 [s@macro-record-enable: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 [s@macro-record-enable:5] NoOp("SIP/2001-09a2 af70", "No recording needed") in new stack
[May 8 18:02:38] -- Executing [8XXXXXXX@from-internal:4] Macro("SIP/2001-09 a2af70", "dialout-trunk|2|8XXXXXXX||") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:1] Set("SIP/2001-09a2a f70", "DIAL_TRUNK=2") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/2001-09 a2af70", "0|Authenticate|") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2001-09 a2af70", "0?disabletrunk|1") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:4] Set("SIP/2001-09a2a f70", "DIAL_NUMBER=8XXXXXXX") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:5] Set("SIP/2001-09a2a f70", "DIAL_TRUNK_OPTIONS=tr") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:6] Set("SIP/2001-09a2a f70", "GROUP()=OUT_2") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk: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 [s@macro-dialout-trunk:9] GotoIf("SIP/2001-09 a2af70", "0?skipoutcid") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:10] Set("SIP/2001-09a2 af70", "DIAL_TRUNK_OPTIONS=") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:11] Macro("SIP/2001-09 a2af70", "outbound-callerid|2") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid: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 [s@macro-outbound-callerid:3] NoOp("SIP/2001- 09a2af70", "REALCALLERIDNUM is 2001") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid: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 [s@macro-outbound-callerid:9] Set("SIP/2001-0 9a2af70", "USEROUTCID=") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid:10] Set("SIP/2001- 09a2af70", "EMERGENCYCID=") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid:11] Set("SIP/2001- 09a2af70", "TRUNKOUTCID="Co. Name" <6128XXXXXXX>") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid: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 [s@macro-outbound-callerid:16] GotoIf("SIP/20 01-09a2af70", "0?usercid") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid:17] Set("SIP/2001- 09a2af70", "CALLERID(all)=Co. Name <6128XXXXXXX>") in new stack
[May 8 18:02:38] -- Executing [s@macro-outbound-callerid: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 [s@macro-outbound-callerid:22] NoOp("SIP/2001 -09a2af70", "CallerID set to "Co. Name" <6128XXXXXXX>") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk: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 [s@macro-dialout-trunk:13] Set("SIP/2001-09a2 af70", "OUTNUM=6128XXXXXXX") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:14] Set("SIP/2001-09a2 af70", "custom=SIP/OXXXXXXX5") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk: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 [s@macro-dialout-trunk:17] Macro("SIP/2001-09 a2af70", "dialout-trunk-predial-hook|") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2001-0 9a2af70", "0?bypass|1") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/2001-0 9a2af70", "0?customtrunk") in new stack
[May 8 18:02:38] -- Executing [s@macro-dialout-trunk: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 [s@macro-dialout-trunk: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 [s-CHANUNAVAIL@macro-dialout-trunk: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 [s-CHANUNAVAIL@macro-dialout-trunk: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 [8XXXXXXX@from-internal:5] Macro("SIP/2001-09 a2af70", "outisbusy|") in new stack
[May 8 18:02:38] -- Executing [s@macro-outisbusy: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 [s@macro-outisbusy: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 [s@macro-outisbusy:3] Macro("SIP/2001-09a2af7 0", "hangupcall") in new stack
[May 8 18:02:42] -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2001-09a 2af70", "w") in new stack
[May 8 18:02:42] -- Executing [s@macro-hangupcall:2] NoCDR("SIP/2001-09a2af 70", "") in new stack
[May 8 18:02:42] -- Executing [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@macro-hangupcall: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 :P
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 :P
Same problem overall
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
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 I will post results!
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
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.
Thanks - SIP Debug Logs
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>;tag=as1a296d76
<--- SIP read from 2XX.XX.XX.3:5060 --->
SIP/2.0 400 SIP Parser Error : Missing '@', line 3, column 26
From: "No CallID"
To:
Call-ID: 7546602d3be8adbb40b078de39cd8187@10.1.1.7
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 '7546602d3be8adbb40b078de39cd8187@10.1.1.7' Method:
OPTIONS;tag=as4ea6ff0f
Reliably Transmitting (NAT) to 10.1.1.4:5062:
OPTIONS sip:2000@10.1.1.4:5062 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK5ad762a7;rport
From: "No CallID"
To:
Contact:
Call-ID: 51b0b9d80d250dd76ac5b5413ada14a4@10.1.1.7
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
---;tag=63250519385dcc3ai2;tag=as4ea6ff0f
trixbox1*CLI>
<--- SIP read from 10.1.1.4:5062 --->
SIP/2.0 200 OK
To:
From: "No CallID"
Call-ID: 51b0b9d80d250dd76ac5b5413ada14a4@10.1.1.7
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 '51b0b9d80d250dd76ac5b5413ada14a4@10.1.1.7' Method:
OPTIONS;tag=c2478d56524c56o2
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"
To: "name"
Call-ID: 1bb3a49d-a1021286@10.1.1.4
CSeq: 14011 PING
Max-Forwards: 70
Contact: "name"
User-Agent: Linksys/SPA942-5.1.10
Proxy-Require: com.nortelnetworks.firewall
Content-Length: 0
<------------->;tag=c2478d56524c56o2;tag=as6964b36f
--- (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"
To: "name"
Call-ID: 1bb3a49d-a1021286@10.1.1.4
CSeq: 14011 PING
User-Agent: oxxxs
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>;tag=as574f8c48
Reliably Transmitting (NAT) to 10.1.1.2:10176:
OPTIONS sip:2001@10.1.1.2:10176;rinstance=24c8d9eb22f9ad8d SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK7c400e32;rport
From: "No CallID"
To:
Contact:
Call-ID: 5e2246bd68e263bf5026901f3c409215@10.1.1.7
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
---;tag=ea7dc56f;tag=as574f8c48
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:
To:
From: "No CallID"
Call-ID: 5e2246bd68e263bf5026901f3c409215@10.1.1.7
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 '5e2246bd68e263bf5026901f3c409215@10.1.1.7' Method:
OPTIONS
trixbox1*CLI>
<--- SIP read from 10.1.1.2:10176 --->
<------------->;tag=as27030364
--- (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:
To:
Call-ID: 42742538060f7656300c790c12832175@127.0.0.1
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:
Event: registration
Content-Length: 0
---;tag=as27030364
trixbox1*CLI>
<--- SIP read from 2XX.XX.XX.3:5060 --->
SIP/2.0 100 Trying
From:
To:
Call-ID: 42742538060f7656300c790c12832175@127.0.0.1
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";tag=as27030364;tag=156589077
To:
Call-ID: 42742538060f7656300c790c12832175@127.0.0.1
CSeq: 108 REGISTER
Via: SIP/2.0/UDP
10.1.1.7:5060;received=10.204.79.9;rport=50020;branch=z9hG4bK7ffccf8b;expires=111
contact:
supported: com.nortelnetworks.firewall,p-
3rdpartycontrol,nosec,join,com.nortelnetworks.im.encryption
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '42742538060f7656300c790c12832175@127.0.0.1'
in 32000 ms (Method: REGISTER);tag=c2478d56524c56o2
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"
To: "name"
Call-ID: 1bb3a49d-a1021286@10.1.1.4
CSeq: 14012 PING
Max-Forwards: 70
Contact: "name"
User-Agent: Linksys/SPA942-5.1.10
Proxy-Require: com.nortelnetworks.firewall
Content-Length: 0
<------------->;tag=c2478d56524c56o2;tag=as2b42e68d
--- (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"
To: "name"
Call-ID: 1bb3a49d-a1021286@10.1.1.4
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
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.
OK here it is
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
I don't think that's the whole SIP conversation. It looks like just the end of it.
Will try again
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
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.
Here it is
Hi managed to paste it all here:
http://pastebin.com/m776f2b5b