similar to: Errors with bristuff-0.3.0-PRE-1e and asterisk cores

We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. We have recently implemented Call Queuing for our main incoming line with hold music. The call queue type is: Ring all - One call at a time (no position announcement). Since implementing this feature we've been receiving the below error daily in the mornings and lunchtime when the queue will jump to the next available

Update: I can reproduce the error by putting the reception phone (call queue 0) in Do Not Disturb mode, then call in from outside using a mobile, then pick up the call from the 2nd phone in the queue. It will cause the error only if I hang up _before_ the mobile hangs up. The error doesn't seem to happen if the external call hangs up, or if the call is answered by the reception phone (first

I'm running bristuff-0.3.0-PRE-1q The line seems to work but I get these messages on the screen every few minutes : 1 received TEI check request for TEI = 81 Jun 20 20:18:05 WARNING[27571]: chan_zap.c:8503 zt_pri_error: 1 TEI remove TEI = 127 Jun 20 20:18:05 WARNING[27571]: chan_zap.c:8503 zt_pri_error: 1 TEI remove TEI = 127 Any ideas ? Guess this is not normal behaviour? I'm using

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:

I applied the florz patch but my problems remain. Now I get all sorts of weird errors on the console and I cannot make outgoing calls. Incoming calls work as expected. I am using a single HFC-S card with BRI. Any clue what these errors below are? Ri = 44651 TEI msg = 3 TEI = 7f Ri = 3800 TEI msg = 3 TEI = 7f Ri = 42399 TEI msg = 3 TEI = 7f Ri = 42409 TEI msg = 3 TEI = 7f Ri = 22078 TEI msg = 3

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.

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

Hello everyone, I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like "The number is wrong or user is switched off" in some cases, and it's just a silence for the user. Now while

I've got some weird behavior on my HFC-s cards. asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel 20040623 When i pick up my ISDN phone on Zap5-1 ("3987") and call the external number "1901" it will do so, connect me and everything is fine. In the second, where it tries to attempt the native bridge, the audio will disappear. Using another card

Hello all, if I try to call from one phone on the internal S0 to another on the same S0 using zaphfc, the bus is hung up. The called phone is ringing, but I can't talk from one phone to the other. The error I get is: -- Executing Dial("Zap/2-1", "ZAP/1/55|15|tr") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/55 -- Channel 0/1,

Hi, I'm trying to install a Junghanns quadbri for a few days but i stay with an asterisk error. (Everyone is busy/congested ) Asterisk is working with a Fritz PCbut from one year and now i want to add the quadbri. The quadbri card has been configured in NT mode and with no 100 ohms S/T termoination. (I'm not sure if the S/T parameter is correct) I have installed the bristuff package

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

Asterisk 1.8.4-rc2 (and 1.8.3) DAHDI Version: 2.4.1.2 libpri version: 1.4.12-beta3 We are having a problem with getting the nationalprefix option of chan_dahdi.conf to work. National calls do not have a "1" added to them when nationalprefix=1. The PRI debug shows the call coming in as a National Call, but the dialplan sees the call without a 1. chan_dahdi.conf: <snip>