similar to: Migrating from CVS HEAD to Stable 1.0.3?

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a

I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a "Playback(welcome)") results in complete silence. Worse, the Playback command will hang

Hi, I'd like to know which version of Asterisk performs best and most stable with TE110P. I don't need any other features (it'll just terminate interasterisk calls without any other feature - so there is no need for CVS Head features or ? ). Any info on setting up secure interasterisk IAX connections (only one way) ? With IAX authentication by certificates ? Thanks in advance,

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands

Hi all! Was anybody able to install kapejod's zaphfc drivers together with RealTime application? I'm in big trouble because bristuff relay on STABLE version, while RealTime is included in the CVS-HEAD. I found this hint, "Installing zaphfc with CVS-Head" at http://voip-info.org/wiki-Asterisk+zaphfc+install, but it was written many months ago: may it be still useful? TIA, Alex

First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a "preackannounce" option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would

Hi Steve; Do u mean by the Iaxy2 is that IAX digium gateway adaptor? If yes, then it has a codec limitation and it does not take ddns name (it needs IP address), also it is gateway and not IP Phone. Or u mean something else? Do u have a link for it so I can see it? Regards Bilal > >> > >>> Anyone knows an IAX IP Phone works fine and > tested? > >> >

Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an "iax2 debug" output from a DIAX call to a local * server, dialing the extension that goes directly to the "demo" application. AsteriskHouse*CLI> iax2

Ok, I'm quite fond of CVS-Stable 10-26-04 as it's always been fine. One thing I noticed with this version and all versions prior, when I did a "sip show channels" it always displayed info in all caps. But sometime between 10-26-04 and 12-8-04 they changed this to all lower case. I believe this MAY be related to the latest problem I just fixed. My provider was sending me

*** SIP Channel fixed in CVS stable ----------------------------------- During a few days there's been a buggy SIP channel in CVS STABLE, but not in the 1.0.3 release tarballs on the FTP server and mirrors. We have now removed the patch that was integrated by mistake so CVS should be ok again. As far as I know, this was the first error introduced in Asterisk STABLE since we forked into a

We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they all are running just fine in production environments each handling thousands of calls a day. I suppose reliability depends upon what you are using, but for our purposes they all are very stable. I could do without the memory leaks though. MATT--- -----Original Message----- From: Florian Lefeuvre

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

Hi. I can't figure this one out. Hope someone can help me. root@pbx:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName=XXXX Password=XXXX Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings=

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init

I've been experiencing some crashes in Asterisk in the past few weeks. I haven't been able to find out why as gdb shows it's in a different function every time. But, in the meantime, I've been using safe_asterisk hoping that it would simply restart Asterisk by itself. It doesn't seem to do that. Whenever Asterisk crashed, the list of processes doesn't show asterisk or