similar to: Where to best start looking for voicemail/moh sound quality problem?

I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've found the driver source code on https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is this code considered "somewhat ready for prime time use"? Thanks, Frank

I'm trying to set-up H.323 support under Asterisk. I built a recent CVS release and the ooh323c code from the asterisk-addons. Everything built and installed and the H.323 stuff loads OK when asterisk starts. What is the easiest way to check if the H.323 code is working? I've edited the h323.conf and extensions.conf files but I'm sure that things aren't right. I've

I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help?

I'm having a problem with my IAXy not always connecting to my Asterisk box. When I pick-up the phone plugged in to the IAXy I get a busy signal. I have to hang-up the phone and wait a few seconds after the orange LED goes out and then try again. When this happens I don't see any connection attempts in the Asterisk -r output. When I do get the IAXy to connect to Asterisk I get a

I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86de in ast_dynamic_str_thread_build_va (buf=0x8172763, max_len=0, ts=0x81482a0, append=0,

My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and DAHDI doesn't want to load. I've tried building it from the sources, but get this error message: CC [M] /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o In file included from /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/xpd.h:31, from

Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the provider's server that G.729 is an option, so the remote server sends the call

>Message: 11 >Date: Sun, 28 Aug 2005 11:46:29 +0800 >From: "chris" <chris@fivestartel.com> >Subject: [Asterisk-Users] error compiling on solaris 10 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> >Message-ID: <003a01c5ab83$149a9e30$650fa8c0@acid> >Content-Type: text/plain;

I am trying to music on hold but I am having all sorts of problems with it. I am running RH9 and the latest version of Asterisk as of yesterday. Here is what I did to test it: 1. I manually deleted the mpg123 softlink to mpg321. 2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed the the archive and loaded ztdummy.o module. 3. I threw a couple extra files in the mohmp3 directory

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hello list, is it normal that when adding new moh-files to the directory /var/lib/asterisk/moh/, asterisk does not see these new files ?! When I do a "moh reload", then Asterisk is aware of the new files... Is there a solution that does not need a "moh reload" ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there anyway for a sip station to play MoH out of the speaker? I know I can do it by calling the station and putting it on hold. For example: On a samsung pbx with MoH, if you goto one of the workstaions and press a button The moh plays out of the speaker. Is there any way to do this with asterisk? Kevin, Honeycomb Internet Services

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a very strange problem. In the middle of the call the MOH starts for 30 seconds approximately. After this the call run normally. Anybody have an ideia or has some similar problem? Thanks in advance!! -- Atenciosamente, ------------------------------------------------------- Dario Quiroz Analista de Suporte (71) 9275-9080

I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. When I throw two calls into a queue, they are both

Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper

Hi All... I installed the Debian Sarge Asterisk package and in the docs it had the licensing terms for the MOH, explaing that Digium (or someone) had licensed the mucic for distribution as MOH only. That's fine, but I can't find the music! Does anyone know where it can be found? Is there another source of free MOH that sounds good with Asterisk? Thanks...

Hi all, I want to allow different users to have their own unique MOH. Is there anyway to do it? Asterisk does not have a realtime MOH feature but I am wondering if there is anyway to get around it? Thank you for your suggestion. Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL:

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server