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...er 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David Gomillion [mailto:dgomillion@eyecarenow.com] > >> > *snipped > > > David, this is completely different from what I am trying to do. > > > > Let's try this a different way. Let's say you have two > companies. When someone calls a number in their own company, > we use their...

Hello, You can use ZapBarge as an extension in your dialplan to listen in on conversations going on in Zap channels(Zaptel device channels) As for recording you can use the Manager interface command StartMonitor to start recording of a Zap channel and StopMonitor to stop it. Zap channels are pretty much the only ones right now that you can directly monitor and record through Asterisk. If

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

> -----Original Message----- > From: David Gomillion [mailto:dgomillion@eyecarenow.com] > Sent: Wednesday, December 20, 2006 10:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > I think you're making it far too difficult. > > What I do is something like this...

...hose Perl/TK because it is very fast and easy to develop in and is cross platform because we have an equal number of Win32 machines and linux machines in our office. If you have any other specific questions please let me know. MATT--- -----Original Message----- From: David Gomillion [mailto:dgomillion@eyecarenow.com] Sent: Thursday, November 06, 2003 10:56 AM To: 'mattf' Subject: RE: [Asterisk-Users] recording calls I think I am keenly interested in this. Let me make sure I have this correct first, though. You are able to monitor which lines are in use and record calls that are i...

...er 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David Gomillion [mailto:dgomillion@eyecarenow.com] > >> > *snipped > > > David, this is completely different from what I am trying to do. > > > > Let's try this a different way. Let's say you have two > companies. When someone calls a number in their own company, > we use their...

I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver

I have a new user question. Sorry I know most of you are Linux experts I am not! I am just getting my feet wet with this. And I am sorry to ask this stupid question. I was following an installation post from Wiki that said when using RH 9 you need to make sure that you have the following installed first and you should check them with the following command. Are there any other items I need to

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

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Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk.

...> _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <critch@basesys.com> --__--__-- Message: 4 From: "David Gomillion" <dgomillion@eyecarenow.com> To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] MP3Player problem Date: Thu, 6 Nov 2003 10:25:23 -0600 Reply-To: asterisk-users@lists.digium.com Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed...

...risk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271200l3536e7cbp9ffe8a269c51b8ca@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You can use FastAGI > > See http://www.asteriskjava.org > > 2006/3/27, David Gomillion <dgomillion@eyecarenow.com>: >> >> Sorry for thread breaking... I'm on digest. >> >>>> I'm curious (ok, well I admit it - it's for perosnal gain) what >>>> methods people are using to manage asterisk config files when they >>>> have multipl...

Some people have been airing dirty laundry on this person, so I thought I'd air some clean laundry! John, Thank you so much for answering my email so quickly, and I also got your voicemail. And thank you for shipping on the same day I ordered. Without these phones, there's no way I would be able to get this demo done! Thanks again, David Gomillion

I have looked and understand how to create an outgoing call by putting a file in the spool. After searching the archives, I came up with stuff about predictive dialers, but nothing right on topic. I would like to be able to detect an answering machine in outgoing calls so that the message will restart with the beep. In other words, I want to send a message to all of my customers, and if they

>Message: 9 >Date: Thu, 02 Dec 2004 11:33:16 -0700 >From: "Kevin P. Fleming" <kpfleming@starnetworks.us> >Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <41AF5FEC.7050406@starnetworks.us> >Content-Type: text/plain;

I am running stable 1.07. I have tweaked the queue app a little, so I'd like verification from someone running stock... When the queue tries to announce the position, I go ahead and read the options for the escape context, which are in the queues.conf as the thank-you. However, pressing any key during the announcement is ineffective. As soon as the announcement is over, I can press the

I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on

Since this issue has raised its ugly head again, and I still don't know a very good solution, I wanted to bounce a few ideas off the gurus on this list. Scenario: You have an administrative assistant who need to be able to take calls for a PHB Desired Behavior: Assistant has a line button that shows status of the boss's phone. Pressing the button, no matter the state of the call, allows