similar to: alpha/numeric paging

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Greetings, I'm ready to start and setup Asterisk. Any preference on which Linux to use? Windows & FreeBSD in use here. I'd like to get up and running as quickly and easily as possible. Thanks -- James Taylor jltaylor@metrotel.net 903-793-1953 --

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Just installed ASTCC, got it working. I've noticed that only part of the sounds come from Allison. Someone (male voice) has recorded the necessary balance,call cost, etc. So, there's this mix of male/female announcements. Is this new or am I missing some sound files? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

I've read the WIKI. Any additional information on the TDD mode? Anyone done a TDD to email app? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Anyone out there have Verizon or SBC as the local exchange company in your area? I'm in the process of negotiations for additional interconnect agreements with these carriers (we're a wireless carrier). I'm looking for Asterisk folks who want a T1 (or several) for a PSTN gateway and would like to "network" their Asterisk boxes. This can take place at little or no cost

What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. What

Hi There I am trying to get the following setup going. Two PCs (gateways) running Linux FC2 both have aDSL connections running. And both run the Asterisk PBX software, I defined the IAX trunks in the iax.conf on both sides, the systems seem to call eachother but never get the handshaking completed ... The consoles keep reporting errors on both sides. Is there a example for doing this ?? --

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

I'm still having problems. Festival works from command line and I can make the speakers talk. But when I dial my weather extension: -- Executing Answer("SIP/3000-a844", "") in new stack -- Executing AGI("SIP/3000-a844", "weather.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi -- AGI Script weather.agi

Somewhere, last week, I saw a reference to a "super macro" for for outbound calls. I can't find it now. Anyone seen it? -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Hi- Can someone help me conceptualize how I could setup an * box which features both English and Spanish prompts? I know it's possible to configure the box to use one or the other, but how could I setup a multi-lingual box? Any suggestions? Thanks in advance, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine........ As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i s E a r

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal