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Outbound route not working
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TOPIC: Outbound route not working
#94344
Outbound route not working 4 Months ago Karma: 0
I have an outbound route that should be sending calls starting with "3xx" to a SIP trunk called callmanager. When I call these extensions I hear the message that all circuits are busy. Wireshark shows me that the call is being routed internally which is what I dont want yet I'm sure the outbound route is setup correctly.
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avenger07
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Last Edit: 2012/01/18 03:45 By avenger07.
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#94346
Re:Outbound route not working 4 Months ago Karma: 181
Outbound route should be setup with the dialplan 3xx. This outbound route should have the trunk (call Manager).

If it is routing internally, have you got other extensions using the 300 range or a ring group/queue using the 300 range??

Regards

Bob
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Bob
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#94351
Re:Outbound route not working 4 Months ago Karma: 0
The ring group is set to 600 and all extensions are in the 100 range. The dial plan has been setup the way you suggested.
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avenger07
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Last Edit: 2012/01/18 04:14 By avenger07.
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#94422
Re:Outbound route not working 4 Months ago Karma: 0
any other suggestions or would showing you some logs help? i am fairly new to elastix
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#94423
Re:Outbound route not working 4 Months ago Karma: 181
Yes if you could show the log /var/log/asterisk/full

Not the whole log, just the part where the call commences and where you get the message "all Circuits are busy"

Regards

Bob
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#94427
Re:Outbound route not working 4 Months ago Karma: 0
How do i navigate to this location as I am using the elastix GUI which also shows the asterisk logs. When I look at the asterisk logs from within the GUI I cannot find any part that has to do with the outgoing call. All I am seeing is some debug messages relating to another SIP trunk and a few extensions.
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#94429
Re:Outbound route not working 4 Months ago Karma: 181
Avenger07,

I recommend accessing the logs via the Linux command line.

The Fault finding document in my signature, will tell you what to access to get the Asterisk logs.....

Note the time of your call, and the logs should show the call commencement starting at that time....

Regards

Bob
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Upgrading Elastix? Read this FAQ Now!!
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Elastix Docs :
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#94445
Re:Outbound route not working 4 Months ago Karma: 0
Code:



<--- Transmitting (no NAT) to 192.168.1.141:61381 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.141:61381;branch=z9hG4bK-d8754z-81453446fb0d5b51-1---d8754z-;rport;received=192.168.1.141
From: "Nexis2"<sip:108@192.168.1.145:5060>;tag=8545d13d
To: <sip:300@192.168.1.145:5060>
Call-ID: ZWM0ZWQwZjg3ZjVkMjYzN2QxMDQ5MWMzMWY1ZDZiYjI.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.6.2.13)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:300@192.168.1.145>
Content-Length: 0


<------------>
    -- Executing [300@from-internal:1] Macro("SIP/108-0000005c", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/108-0000005c", "AMPUSER=108") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/108-0000005c", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/108-0000005c", "1?Set(REALCALLERIDNUM=108)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/108-0000005c", "AMPUSER=108") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/108-0000005c", "AMPUSERCIDNAME=Nexis2") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/108-0000005c", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/108-0000005c", "AMPUSERCID=108") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/108-0000005c", "CALLERID(all)="Nexis2" <108>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/108-0000005c", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/108-0000005c", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/108-0000005c", "CALLERID(number)=108") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/108-0000005c", "CALLERID(name)=Nexis2") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/108-0000005c", "Using CallerID "Nexis2" <108>") in new stack
    -- Executing [300@from-internal:2] NoOp("SIP/108-0000005c", "Calling Out Route: to_tyler") in new stack
    -- Executing [300@from-internal:3] Set("SIP/108-0000005c", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [300@from-internal:4] Set("SIP/108-0000005c", "MOHCLASS=default") in new stack
    -- Executing [300@from-internal:5] Set("SIP/108-0000005c", "_NODEST=") in new stack
    -- Executing [300@from-internal:6] Macro("SIP/108-0000005c", "record-enable,108,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/108-0000005c", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/108-0000005c", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/108-0000005c", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/108-0000005c", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/108-0000005c", "1?MacroExit()") in new stack
    -- Executing [300@from-internal:7] Macro("SIP/108-0000005c", "dialout-trunk,3,300,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/108-0000005c", "DIAL_TRUNK=3") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/108-0000005c", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/108-0000005c", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/108-0000005c", "DIAL_NUMBER=300") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/108-0000005c", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/108-0000005c", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/108-0000005c", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/108-0000005c", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/108-0000005c", "0?sub-flp-3,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/108-0000005c", "OUTNUM=300") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/108-0000005c", "custom=SIP/CallManager") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/108-0000005c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)tr)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/108-0000005c", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/108-0000005c", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/108-0000005c", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/108-0000005c", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/108-0000005c", "SIP/CallManager/300,300,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called CallManager/300

<--- Transmitting (no NAT) to 192.168.1.141:61381 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.141:61381;branch=z9hG4bK-d8754z-81453446fb0d5b51-1---d8754z-;rport;received=192.168.1.141
From: "Nexis2"<sip:108@192.168.1.145:5060>;tag=8545d13d
To: <sip:300@192.168.1.145:5060>;tag=as25506fb6
Call-ID: ZWM0ZWQwZjg3ZjVkMjYzN2QxMDQ5MWMzMWY1ZDZiYjI.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.6.2.13)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:300@192.168.1.145>
Content-Length: 0


<------------>
    -- SIP/CallManager-0000005d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/108-0000005c", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/108-0000005c", "s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/108-0000005c", "RC=21") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/108-0000005c", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/108-0000005c", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/108-0000005c", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/108-0000005c", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/108-0000005c", "CALLERID(number)=108") in new stack
    -- Executing [300@from-internal:8] Macro("SIP/108-0000005c", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/108-0000005c", "") in new stack


Enter code here   
Please note: although no board code and smiley buttons are shown, they are still usable.
avenger07
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#94592
Re:Outbound route not working 4 Months ago Karma: 0
Anyone have a similar issue of the call not being routed out to the proper trunk?
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Please note: although no board code and smiley buttons are shown, they are still usable.
avenger07
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#94627
Re: Re:Outbound route not working 4 Months ago Karma: 1
avenger07:

Normally when you get this message:

("SIP/108-0000005c", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack

means that the callmanager is not accepting the call because this has not available channels to assign to the call.

Hope this Help You.
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