similar to: Calls Dropping

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

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could point me to a howto / guide or relate their experiences with this, that would be great ! Thanking you all in advance. Callum P.S. Before anyone asks; it's a legacy database that

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

On 02/11/2016 01:48 PM, Frank Cox wrote: > On Thu, 11 Feb 2016 13:42:39 -0500 > ken wrote: > >> Maybe I haven't run the correct yum commands or looked in the right >> repo, but I haven't found a utility called 'gmessage'. Anyone know >> where to find it? > > gmessage has been more-or-less superseded by zenity. > That's nutz. From docs

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == ==

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log

> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:

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

Hi, I have an IBM server running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to

Hi, Trying to trace an asterisk hang on a production (it had to be didn't it) system. The last thing before it crashed was [Mar 16 12:32:42] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Mar 16 12:54:34] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 12:54:35] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED

Hi everyone, I have a PRI conection on an * system running Asterisk 1.0.9, libpri 1.0.9 and zaptel 1.0.9.2 connected to an AXE 10 (APZ 21220 System 64) in the network side. I know the system and the wildcard I´m using are ok because I´ve used them before with other PRI connections (to a Siemens EWSD) without any problem. First the PRI didn´t work (I got the TE110P alarmed with red). After

Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI with my Asterisk using a Digium TE220B card. The Panasonic is master clock and Asterisk is slave, additionally the Panasonic takes the clock from the PSTN E1. This scheme works and I?m able to make and receive calls the problem is that some times the Asterisk lose the synchronism with the Panasonic and the log shows this

Hello All, I have got my TE110P working at the hardware level, turned out to be a dodgy cable causing the Yellow errors in zttool. However, now I am getting yellow errors in asterisk, but zttool shows nothing out of the ordinary. here is my current config, and some asterisk console errors, any suggestions most welcome. zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31

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

Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a "straight" cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same.

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

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'