Elastix 2.0 with ISDN - no Inbound and Outbound

deggial

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Oct 5, 2010
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#1
Hello,
I have a problem with my installation. First I have to say that I am a begginner in Elastix and PBX systems. My problem is that I can make internal calls between the clients but I wasn't able to establish neither outbound calls nor incoming calls to my system.

My network is like this:
- ISDN Line from German Telekom
- a Router (7170 FritzBox)
- Elastix 2.0 installed on a Server PC with an ISDN Card with HFC-S for Asterisk Trixbox Elastix
- 3 Client PCs with softphones on the same network

I have done lots of changes in the Trunk Settings and Outbound & Inbound Routes with no success. These settings are now like that:

Trunk Settings:
Code:
Custom Trunk
Trunk Description: CustomTrunk
Outbound Caller ID: "Twin Brain" <0211 758467-00>
CID Options: Allow Any CID
Maximum Channels: 
Dial Rules: 9|X.
Custom Dial String: mISDN/g:ta/$OUTNUM$
Outbound Routes:
Code:
Route Name: MyOutbound
Route CID: (empty)
Dial Patterns: 9|X.
Trunk Sequence: 0 mISDN/g:ta/$OUTNUM$
I have not configured any inbound routes yet.

I have added a new context at the end of misdn.conf:
Code:
[ta]
ports=1
context=from-pstn
msns=*
echocancel=no
overlapdial=no
senddtmf=yes
far_alerting=yes
te_choose_channel=no
nationalprefix=90
internationalprefix=900
rxgain=0
txgain=0
The dahdi_scan output:
Code:
[root@localhost ~]# dahdi_scan
[1]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 0 [TE]
name=ZTHFC1
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 01 Slot 01
basechan=1
totchans=3
irq=82
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS
[root@localhost ~]#
Under Hardware Detector I have this entry:
Code:
Misdn Card

Port 1: TE-mode BRI S/T interface line (for phone lines)
-> Protocol: DSS1 (Euro ISDN)
-> Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib.
-> childcnt: 2
* Port NOT useable for PBX (maybe there is already a PBX running?)
--------
Whit this configuration, I got it ONLY FOR ONE TIME work. I could have establish a call to my home number. I answered the call, all went allright. But then I gave a break, then I would like to add also Inbound Routes to be able to receive calls as well. And it didn't work for the first try. And I would like to test one more time to call my home number, it didn't work again. I'll go mad with that :/

What did I change, or what went wrong? I don't know. Any ideas??

Any help with be appreciated.
Thanks in advance.

With my best regards,
Yordanov.
 

trymes

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#2
Well, I cannot help with the specific configuration of you ISDN card, but do you get any activity on the Command line when you attempt to make an inbound or outbound call?

In other words, do the following:

1.) log into your server.
2.) Run the command "asterisk -r"
3.) run the command "core set verbose 3"
4.) Attempt to call your system from an outside phone. Do you see anything scroll by on the screen?
5.) Do the same thing while trying to make an outbound call. What do you see?

If you don't already have inbound routes for other trunks, start by creating an inbound route that will match all CID and all DID (leave both fields blank when creating the route).

Let us know what you find.

Tom
 

deggial

Joined
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#3
Hello Tom.

Thank you very much for your reply. There was a new problem today, something with kernel.
The server went down a few times, with a message like:
"Kernel panic: Fatal exception".

I have installed the complete elastix from the beginning. And everything went well and I can make outbound and inbound calls :))

I think it was a problem with the installation.

Thank you very much again for your interest.

Best Regards,
Yordanov.
 

Oosthuizen

Joined
Jun 22, 2010
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#4
Re: Elastix 2.0 with siemens gigaset no Outbound

hi there
i installed elastix 2
i can make outbound calls but no inbound is possible
i have run elastix -r and got the following results
i hope it makes sense for you
please i need this up and running urgently, ant=y help would be highly apprecianted
[root@localhost /]# asterisk -r
Asterisk 1.6.2.10, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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 1.6.2.10 currently running on localhost (pid = 2898)
Verbosity is at least 3
-- Got SIP response 503 "SERVICE UNAVAILABLE" back from 196.41.212.66
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0878084014@from-internal:1] Macro("SIP/201-0000003f", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/201-0000003f", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-0000003f", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-0000003f", "1?Set(REALCALLERIDNUM=201)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/201-0000003f", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/201-0000003f", "AMPUSERCIDNAME=201") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-0000003f", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/201-0000003f", "AMPUSERCID=201") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/201-0000003f", "CALLERID(all)="201" <201>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-0000003f", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-0000003f", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/201-0000003f", "Using CallerID "201" <201>") in new stack
-- Executing [0878084014@from-internal:2] Set("SIP/201-0000003f", "EMERGENCYROUTE=YES") in new stack
-- Executing [0878084014@from-internal:3] Set("SIP/201-0000003f", "INTRACOMPANYROUTE=YES") in new stack
-- Executing [0878084014@from-internal:4] ExecIf("SIP/201-0000003f", "1?Set(TRUNKCIDOVERRIDE=27878084014)") in new stack
-- Executing [0878084014@from-internal:5] Set("SIP/201-0000003f", "_NODEST=") in new stack
-- Executing [0878084014@from-internal:6] Macro("SIP/201-0000003f", "record-enable,201,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/201-0000003f", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/201-0000003f", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/201-0000003f", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/201-0000003f", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/201-0000003f", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:17] NoOp("SIP/201-0000003f", "Recording enable for 201") in new stack
-- Executing [s@macro-record-enable:18] Set("SIP/201-0000003f", "CALLFILENAME=OUT201-20101008-213253-1286566373.63") in new stack
-- Executing [s@macro-record-enable:19] Goto("SIP/201-0000003f", "record") in new stack
-- Goto (macro-record-enable,s,23)
-- Executing [s@macro-record-enable:23] MixMonitor("SIP/201-0000003f", "OUT201-20101008-213253-1286566373.63.wav,,") in new stack
-- Executing [s@macro-record-enable:24] Set("SIP/201-0000003f", "CDR(userfield)=audio:OUT201-20101008-213253-1286566373.63.wav") in new stack
-- Executing [s@macro-record-enable:25] MacroExit("SIP/201-0000003f", "") in new stack
-- Executing [0878084014@from-internal:7] Macro("SIP/201-0000003f", "dialout-trunk,2,0878084014,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/201-0000003f", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/201-0000003f", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/201-0000003f", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/201-0000003f", "DIAL_NUMBER=0878084014") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/201-0000003f", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/201-0000003f", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/201-0000003f", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/201-0000003f", "1?skipoutcid") in new stack
-- Goto (macro-dialout-trunk,s,12)
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/201-0000003f", "1?AGI(fixlocalprefix)") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== Begin MixMonitor Recording SIP/201-0000003f
== fixlocalprefix: Dialpattern . matched. 0878084014 -> 0878084014
-- <SIP/201-0000003f>AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/201-0000003f", "OUTNUM=0878084014") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/201-0000003f", "custom=SIP/vox out") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/201-0000003f", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^)tr)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/201-0000003f", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/201-0000003f", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/201-0000003f", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/201-0000003f", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/201-0000003f", "SIP/vox out/0878084014,300,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called vox out/0878084014
-- Got SIP response 400 "Normal Release" back from 196.41.212.66
-- SIP/vox out-00000040 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/201-0000003f", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 16") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/201-0000003f", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/201-0000003f", "RC=16") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/201-0000003f", "16,1") in new stack
-- Goto (macro-dialout-trunk,16,1)
-- Executing [16@macro-dialout-trunk:1] Goto("SIP/201-0000003f", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/201-0000003f", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/201-0000003f", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 16 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/201-0000003f", "CALLERID(number)=201") in new stack
-- Executing [0878084014@from-internal:8] Macro("SIP/201-0000003f", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/201-0000003f", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/201-0000003f", "1?emergency,1") in new stack
-- Goto (macro-outisbusy,emergency,1)
-- Executing [emergency@macro-outisbusy:1] Playback("SIP/201-0000003f", "all-circuits-busy-now&pls-try-call-later") in new stack
-- <SIP/201-0000003f> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/201-0000003f> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [emergency@macro-outisbusy:2] Congestion("SIP/201-0000003f", "20") in new stack
-- Got SIP response 503 "SERVICE UNAVAILABLE" back from 196.41.212.66
== Spawn extension (macro-outisbusy, emergency, 2) exited non-zero on 'SIP/201-0000003f' in macro 'outisbusy'
== Spawn extension (from-internal, 0878084014, 8) exited non-zero on 'SIP/201-0000003f'
-- Executing [h@from-internal:1] Macro("SIP/201-0000003f", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000003f", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/201-0000003f", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000003f", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000003f", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/201-0000003f", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/201-0000003f", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/201-0000003f' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000003f'
== MixMonitor close filestream
== End MixMonitor Recording SIP/201-0000003f
-- Got SIP response 503 "SERVICE UNAVAILABLE" back from 196.41.212.66
 

Oosthuizen

Joined
Jun 22, 2010
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#5
Re: Elastix 2.0 with siemens gigaset no Outbound

Hello All

just to clear this
my problems are solved, regarding ZAP in and out problems

follow elastix without tears june 2010 to the letter

my problem regarding unable to make outbound calls was that i have a 8 port Digium card
on installation i added a 4 port PSTN card on ports 5 to 8

when i created my trunks i had the ZAP Trunk Identifier on 1 but my PSTN line on port 8
as soon as i changed my Zap Trunk Identifier to 8 it works
i have also successfully added a Linksys SPA3102 again following Elastic Without Tears June 2010 to the letter

Regards
Kobus
 

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