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...EN:6) Again, thanks for the help. Jesse Date: Mon, 09 Jun 2003 22:37:01 -0500 From: Steven Critchfield <critch@basesys.com> Subject: Re: [Asterisk-Users] Adding an app To: asterisk-users@lists.digium.com Organization: Reply-To: asterisk-users@lists.digium.com On Mon, 2003-06-09 at 13:01, JKNUTSEN@UP.COM wrote: > I am in the process of testing out the Cisco ATA 186 to provide analog > phone service via VoIP for some of our remote users. I have that working > fine and well, but am struggling with another aspect. We already have a > large centralized voicemail system, which I wou...

I am trying to set up an auto attendant for the first time, and am having trouble getting to the submenu. My extensions.conf file looks like this: [incoming] exten=> s,1,Background,menu1 exten=> s,2,Wait,20 exten=> s,3,Goto,s|1 exten=> 1,1,Playback,option1 exten=> 2,1,Playback,option2 exten=> 3,1,Playback,option3 It is my understanding that asterisk treats the digits entered

...s from callers about it ignoring DTMF at times... I need to play music on hold for a set amount of time but still break on DTMF, how would I do that? Any suggestions? Bill James Hines wrote: > On Thu, 2003-04-24 at 16:00, Tilghman Lesher wrote: > >>On Thursday 24 April 2003 13:49, JKNUTSEN@UP.COM wrote: >>You should instead take advantage of the timeout. Asterisk will not >>receive DTMF during a Wait. >> > > > > Wow, this is not something that I was aware of. I think this might be > causing some of the "DTMF detection problems" my use...

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>I am using some of the sample recordings included with asterisk for my conferencing application.&nbsp; They seem to be rather choppy at times, with the overall quality not being quite where I'd like it.&nbsp; I think I saw in a previous mailing list on here that people suggested we

I am in the process of testing out the Cisco ATA 186 to provide analog phone service via VoIP for some of our remote users. I have that working fine and well, but am struggling with another aspect. We already have a large centralized voicemail system, which I would like to use for these users. I can get the call to roll to new the centralized voicemail no problem, but I'd like to provide

I have a 1.8 GHz, 1 G RAM machine that I would like to use for conferencing. How many users have you all had on a box like that before? I have a Digium quad T1 card in the box right now, so I could have up to 96 users on this system. Has anyone ran into problems with degradation of service as the number of users increased. I haven't tested it out yet, but would be interested to hear any

Greetings, I have a somewhat unique (I think) configuration that I am testing involving MeetMe conferencing and have encountered a problem that I'm not quite sure how to solve. Here is a brief description of my setup for the background. I wanted to offer the ability for users to mute and unmute themselves while in a conference. If they enter a conference as monitor only, they are