similar to: super macro

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

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

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

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

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

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

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

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

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

I was ZapBarging an extension the other day (sorry, CVS Head as of last week) and "forgot" to put down the phone when the conversation ended. To my consternation, I then started to listen to a totally different conversation - I think that the zap channel got re-used. Is there any way to force the zap barge to drop when either party hangs up ? Thanks. Julian

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 --

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

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 have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use in conjunction with asterisk, serving something like 4000+ lines. does anyone have experience with the APX8100 and it's integration with SIP on asterisk ? does the APX8100 handle SS7<->SIP signalling well enough to be used ? any anecdotes would be well appreciated. -- Regards,

I am looking for help in writing the scripts for numeric paging via Asterisk. Here's the general flow: answer lookup D-I-D if "on" prompt with "please enter your telephone number" store in variable prompt "please hangup now" Shell to SNPP executable and send message to server. I have the code in "C" for the command line SNPP client. Three parameters

I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? Thanks John Fullington

Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple register=XXXXXXXXX1@sip.broadvoice.com:passwd:XXXXXXXXX1@sip.broadvoice.com register=XXXXXXXXX2@sip.broadvoice.com:passwd:XXXXXXXXX2@sip.broadvoice.com register=XXXXXXXXX3@sip.broadvoice.com:passwd:XXXXXXXXX3@sip.broadvoice.com