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...al(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I will check out the queue system you referenced. Thanks! Wiley -----Original Message----- From: Seth Remington [mailto:sremington@saberlogic.com] Sent: Saturday, July 17, 2004 7:45 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Using a group variable for a groupofextension to dial It would ring all three at the same time. You are probably looking for a roundrobin call queue -> http://www.voip-info.o...

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

...: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Foster Sent: Thursday, July 29, 2004 9:13 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington <sremington@saberlogic.com> wrote: > On Wed, 2004-07-28 at 21:00, James Gardiner wrote: > > How do I get Asterisk to recognise the # key from the granstream phone for > > doing transfers? > Make sure the Grandstream is configured to send DTMF via SIP INFO > instead of in-audio. > &...

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

...ntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause it to ring through each of these extensions or would that result in these three extensions all ringing togeher? Thanks! Wiley -----Original Message----- From: Seth Remington [mailto:sremington@saberlogic.com] Sent: Saturday, July 17, 2004 6:35 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Using a group variable for a group ofextension to dial Maybe I am misunderstanding your question but are you looking for the '&' operator? Dial(type1/identifier1&am...

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? Appreciate pointers.. Thx a lot, J --------------------------------- Do you

I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch < Makefile.patch that fails with

...t = intern signalling = fxo_ks callwaiting = yes usecallerid = yes echotraining = no ;; yes echocancel = 16 channel => 2 ------------------------------ Message: 23 Date: Fri, 18 Mar 2005 11:22:44 -0500 From: Seth Remington <sremington@saberlogic.com> Subject: Re: [Asterisk-Users] Group Ring after Timeout To: info@compex.edu.mt, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <1111162964.4715.4.camel@localhost.localdomain> Content-Type: text/plain On Fri, 2005...

Hello All, I saw an add in my latest Linux Journal advertising Sangoma's new "AA series" of FXO/FXS analog cards with on-board echo cancellation, but I can't find any information at all on them. Even the link given in the advertisement is a dead end as far as I can tell. Anybody else seen/heard anything about this? -Seth