similar to: Using Queue - Zap problems (PRI)

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

I have an application where I need to send outbound prerecorded messages. The Asterisk "call file" process works fine if I am sending the call via SIP or IAX, but not via ZAP over a PRI channel. The destination device (my cell phone) never rings. The only unusual thing I see is on the fifth line of the "full" log, below, "channel.c: Don't know what to do with

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hey guys, We have asterisk 1.2.7.1 running and today i got following error message. and users started complain regarding call issue. after reboot everything comes back to normal. I just want to know what happened ? Feb 25 11:27:11 WARNING[10042] app_meetme.c: Error setting conference Feb 25 11:27:11 WARNING[10042] chan_zap.c: Unable to set audio mode on channel 1 to 1 Feb 25 11:27:11

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a day or so, with 100 or so users transfering and three

I am having a problem with the tranfer function since changing our t1 to PRI. I used to be able to answer a forwarded call to my cell phone, then transfer the call back using ## (we have it set to ## in features.conf). Since the update it does not work most of the time. At lunch today I had a call and could not get it to transfer. The logs show asterisk getting the DTMF #, then a bunch of

Today is the first time I've had any show stopping errors with asterisk since I started using it over 8 months ago... While I wait on the telco to check the circuit I thought I would post this here. Not sure yet but I thinks my 4-port PRI card is gone!!! I made a call and right when the person answered - BOOM. This is only a little better than it not working at all.

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is

Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We

I just upgraded my asterisk install from january 2005 CVS HEAD to current CVS HEAD : * zaptel * libpri * asterisk My asterisk have one TE405P with one span (the clock source) pluged into a telco PRI E1, a second span is a PRI E1 to another PBX and the third one is a T1 to a rhino channel bank. Everything went smoothly except that the two PRI span restart every ten seconds after some HDLC

I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes. In the logs I keep seeing Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use Apr 4 15:22:18 WARNING[5054] chan_zap.c: Unable to move channel 2! Apr 4 15:22:55 WARNING[5054] chan_zap.c: Can't fix up channel from 1

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I

hello everyone, Last week I installed asterisk 1.2.24 with digium TE220B card. I have a problem with our PRI and Asterisk: the call be interrupted.It happens either PSTN-to-SIP or SIP-to-SIP,almost every call. After spending several days searching on internet, I found a lot of discussion about this issue and I have tried many, but it still.I am totally new to Asterisk environment and suspect I

Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also