similar to: Soft-switch.org is out?

I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks

Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst I'm making the files are there any other files you want? IVR's etc. If so make sure I have a script sent by email. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

I got a call from GV on Monday evening telling me they wanted me to move my Asterisk server over to a new IP address (216.229.127.40) by this saturday. Why the couldn't tell me this in an email is beyond me but anyways .. So I done changed the number and so far its all ok but whilst testing I noticed that I could no longer accept incoming phone calls. I swapped back and still no inbound

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi Folks, OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back to V1.09. Other than downloading the code, how do I do it? I thought someone once said that I have to delete all my modules or something? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more

I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

G'Day All; I am a little unsure on how to get Music On Hold to work. Please critique my extensions.conf. ????? Thanks ; SIP 5001 exten => 5001,1,Dial(SIP/5001) exten => 5001,2,Voicemail(u${EXTEN}) exten => 5001,3,Hangup exten => 5001,102,Voicemail(b${EXTEN}) exten => 5001,103,Hangup Thanks -------------- next part -------------- An HTML attachment was

Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I was wondering if there was a way to deal with "wrap up" time for agents slightly better than we do it at the moment. At the moment, we set a wrap up time of 20 seconds for each of our call queues. The problem with this is that sometimes it's either too long or two short. I would essentially like to have all agents put into wrap up straight after a call, and have to

I've been banging my head with this all day. I today switched from a very old CVS build to AAH1.3 and so far everything has been easy. However I cannot accept calls from a previously working IAX trunk. I've set up an trunk with all the same credentials as before and can call the folks at the other pbx. However whenever they call me I tell them that I don't have an

I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT,