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Incoming call activates Zap channel

huydd
Posts: 3
 Member Since:
2009-07-04
Submitted by huydd on Sat, 07/04/2009 - 9:30am.

Hi everyone,
I have install TrixBox CE v2.8.0 with TDM400P Digium Card for an IVR system. When I am setting up Zap Trunks for PSTN incoming and outgoing, I have found a funny problem.
1. After restart system, I try to make an outgoing call (from Softphone registered with SIP account) to PSTN through DAHDI/1-1 , trixbox give me a "All circuits are busy..." announcement and "TRUNK Dial failed due to CONGESTION - failing through to other trunks" on asterisk log.
2. Then, I try to check the PSTN trunk by using my cellphone making incoming call to Trixbox. It 's work with "-- Starting simple switch on 'DAHDI/1-1'" on asterisk log and lead me to Trixbox IVR system
3. After that, I try to make outgoing call again (same as 1st step) and it is funny, I can now call to PSTN phone with this DAHDI trunk. The log in asterisk console is:
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/1501-b7dd19d8", "DAHDI/g0/0904368368,300,") in new stack
-- Called g0/0904368368"
4. If I restart dahdi (/etc/init.d/dahdi restart) or reboot Trixbox system, the outgoing call through ZAP Trunk is failed again

Is someone have ideas or suggestions for this problems
Thanks in advances

‹ Cisco telephony integration with trixbox TDM800 Analog Interface Card ›

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pk6tech
Posts: 13
 Member Since:
2008-10-05
Hi, I'm having the same
Mon, 07/06/2009 - 4:29pm

Hi,
I'm having the same problems.... I have openvox a400 on Dell PE2650. I get incoming calls from the zap but I get no outgoing calls at all.... I have reinstalled TB2.8 about 4 time still same problem. Not only that.. when I install TB2.8 gizmo5 is missing too....


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huydd
Posts: 3
 Member Since:
2009-07-04
I have faced with this
Wed, 07/08/2009 - 2:29am

I have faced with this problem before when I try to compile PBX from scratch with asterisk 1.4.23. At that time, I have downgrade asterisk version to 1.4.22 and this problem was fixed.
Is this the same bug in 1.6 version. Can anyone give me a suggestion or post this bug to trixbox or asterisk community

Thanks in advance


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rjefferis
Posts: 111
 Member Since:
2006-05-31
Hi everyone, was there ever
Mon, 08/10/2009 - 5:40pm

Hi everyone, was there ever a solution for this?

I have just installed 2.8 with a TDM800 card and everything seems to be working ok except for this issue.

When dialing out, prior to receiving i get this burried in my log

[Aug 11 01:28:26] WARNING[2587] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
[Aug 11 01:28:26] VERBOSE[2587] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Executing [s@macro-dialout-trunk:20] [1;36;40mGoto [0;37;40m(" [1;35;40mSIP/111-09cd5cc0 [0;37;40m", " [1;35;40ms-CHANUNAVAIL,1 [0;37;40m") in new stack
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] [1;36;40mGotoIf [0;37;40m(" [1;35;40mSIP/111-09cd5cc0 [0;37;40m", " [1;35;40m1?noreport [0;37;40m") in new stack
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] [1;36;40mNoOp [0;37;40m(" [1;35;40mSIP/111-09cd5cc0 [0;37;40m", " [1;35;40mTRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks [0;37;40m") in new stack
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Executing [907967657424@from-internal:5] [1;36;40mMacro [0;37;40m(" [1;35;40mSIP/111-09cd5cc0 [0;37;40m", " [1;35;40moutisbusy, [0;37;40m") in new stack
[Aug 11 01:28:26] VERBOSE[2587] logger.c: -- Executing [s@macro-outisbusy:1] [1;36;40mPlayback [0;37;40m(" [1;35;40mSIP/111-09cd5cc0 [0;37;40m", " [1;35;40mall-circuits-busy-now,noanswer [0;37;40m") in new stack

When dialing in i then get

Aug 11 01:29:33] VERBOSE[2240] logger.c: == Starting post polarity CID detection on channel 1
[Aug 11 01:29:33] VERBOSE[2597] logger.c: -- Starting simple switch on 'DAHDI/1-1'
[Aug 11 01:29:34] NOTICE[2597] chan_dahdi.c: CallerID number: 07967657424, name: (null), flags=4
[Aug 11 01:29:36] VERBOSE[2597] logger.c: -- Executing [s@from-pstn:1] [1;36;40mSet [0;37;40m(" [1;35;40mDAHDI/1-1 [0;37;40m", " [1;35;40m__FROM_DID=s [0;37;40m") in new stack
[Aug 11 01:29:36] VERBOSE[2597] logger.c: -- Executing [s@from-pstn:2] [1;36;40mGosub [0;37;40m(" [1;35;40mDAHDI/1-1 [0;37;40m", " [1;35;40mapp-blacklist-check,s,1 [0;37;40m") in new stack

Dialing out then works. This is trunk independant. ie. I have 2 trunks in my group 0. If trunk 1 has not received an incoming call yet, and trunk 2 has, all outgoing calls will default to trunk 2 until trunk 1 recieves a call.

Weird.

Any help would be much appreciated.


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rjefferis
Posts: 111
 Member Since:
2006-05-31
UPDATE! Ok this seems to be
Mon, 08/10/2009 - 5:54pm

UPDATE!

Ok this seems to be a known problem with Dahdi currently. Look like it has been solved though here.

https://issues.asterisk.org/view.php?id=14577

Could someone more intelligent than me give us a quick walk through of what us trixbox users need to do to actually apply these patches.

Thanks again all.

Rob.


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