similar to: Problem with PRI channels, restarted after every call.

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

Just throwing out this question. integrating with Altiware server. PRI appears to be okay. It keeps trying to move my call to a different channel...usually channel 1. This is the deal here: Moving call from channel 23 to channel 1 Then the following errors after no audio then hanging up manually: Mar 24 17:46:17 WARNING[1315]: chan_zap.c:7792 pri_fixup_principle: Call specified, but not

Hello, I have a situation where a remote worker dials in to the asterisk server, enters the "secret code", then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] --

I set up a TE405P to go T1---*---Toshiba. I have the channels configured, and can place calls from the Toshiba, through * to the t1. Incoming calls work great to *, but if they go to the Toshiba, I get a hangup. I think the * is sending the call to the wrong span. I have 2 spans, span 1 from the T1, span 2 to the Toshiba. The bchannels show as 0/1 through 0/23 on both spans in * when it starts.

Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten => 999,1,Dial(Zap/g1/02601591) exten => 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 I am able to

Hi, i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think. marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Donnerstag, 6. April 2006 11:50 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Dial out on Zap Hi,

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

I'm not sure if it's the same problem but your error message likely the same. after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany) marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Dienstag, 11. April 2006 16:33 An:

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hello all, I've got an intermittent issue with my asterisk set-up, and I'm pulling my hair out! Mostly everything works fine, but I get an error once every few days that sometimes (but not always) causes asterisk to stop accepting new calls and also stop responding to commands on the console (asterisk -r). The usual errors look something like this: [Feb 19 13:09:52] ERROR[18728]

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 +

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound

Hi fellow asterisk users, I am running Asterisk 1.4.29 with an Digium TE121 card (wcte12xp kernel module) an approx. 100 snom320. The whole installation is running without issues since 5 months. Without having changed anything on the asterisk server for at least 2 months, i noticed "clicking sounds" at external calls over the PRI. Asterisk constantly throws the following messages: ...

Hi all, Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI> pri intense debug span 1 Enabled debugging on span 1 [Dec 16

I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on

Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.