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Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? Gene Kochanowsky Solution

I'm having a problem building Asterisk 1.2.22. It fails in codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4. Here's the error. Can anyone help me with this? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o

Is there a way to configure voicemail to do reminder paging? I would like to configure some voicemail boxes to send an e-mail message to a pager every 10 minutes until the message is retrieved.

Should Asterisk run under it's own user id, or the web server user id, or root, or what?

I started using Festival for the first time today and am having a problem with zombies left behind after every time that it speaks. I'm using Festival 1.4.3 with today's CVS of Asterisk. Everything seems to work. The only obvious problem is that a defunct process is left behind every call to Festival. Is this a known problem? Does anyone know how I can fix this?

I'm trying to debug a problem with robbed bit signalling on a T1 coming into an Asterisk box on a T100P card. Specifically, I need to look at the signalling timing. Is there a way to turn on this kind of debugging in Asterisk, similar to what 'pri debug' does?

Brad's recent list of enhancements look good, but I haven't looked at the code yet. If the code looks good, I hope it will be committed to the project CVS. Here's a partial list of enhancements that I would like to see in Comedian Mail. I am probably interested in helping to fund the enhancement of Asterisk voicemail. Is anyone else interested? -Address message to multiple

I'm having a problem with intermittent lookup of Caller ID Name info using LookupCIDName. The same problem occurs when doing: asterisk -rx "database show cidname" No data is returned on every fourth or fifth query. No errors are being logged. I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the problem a few weeks ago. Is anyone seeing a similar problem?

It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: > Hi folks, > > I used to have some constructions like > > exten => number/callerid,1,Goto(somewhere) > > After updating to 1.0.8 those does not work any more. > Any hints? > > Regards, > Stefan >

I've been using the following macro to ring SIP and IAX devices for a few seconds, and then add on a cell phone if there is no answer on the SIP or IAX device. Periodic problems began a few versions ago and now the problem happens every time with 1.2.9 and 1.2.9.1. The problem is that when a call from the PRI falls through to voicemail, the call is dropped before the voicemail greeting

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

I'm running a pretty successful Asterisk system and recently moved our PRI to a T100P board. The PRI was previously connected to a Cisco 2600 that was serving as a voice gateway. We are having a frequent problem with inbound and outbound calls being disconnected shortly after they are answered since moving the PRI directly to the Asterisk box. Most calls work fine, but approx 3 out 10 are