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

...o 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? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

...ne type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

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

...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 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

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:

...lists.digium.com with esmtp (Exim 4.63) (envelope-from <asterisk-users-bounces at lists.digium.com>) id 1I2zQW-0004ty-N8; Mon, 25 Jun 2007 20:01:04 -0500 Received: from exprod8mx64.postini.com ([64.18.3.164] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from <g7ltt at g7ltt.com>) id 1I2zQL-0004tU-PU for asterisk-users at lists.digium.com; Mon, 25 Jun 2007 20:00:54 -0500 As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. -- Loway...

...l I would have been requested to input my password. This did work but now it doesn't. Asterisk does show that it is playing the file but no audio is heard. I have audio on regular SIP based calls as well as IAX based ones. I'me not getting and audio when I make a ZAP call. Ideas? Mark G7LTT/KC2ENI Mark Phillips G7LTT/KC2ENI Mark Phillips

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Most often the s...

...ncoming phone calls. I swapped back and still no inbound calls. Is this the bug that someone was asking about or is it just me. GV claim that I'm sending them a 404 but I see nothing coming on from them on the trace. In my playing I have been able to get only one call into my system. Ideas? G7LTT/KC2ENI Mark Phillips

...annot 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 extension/context by the name they dialed. Any ideas? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

...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

...ce, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} Mark -- Mark Phillips <g7ltt@g7ltt.com>

...oxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server and then have it play into the MOH? If so how? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

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>

...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

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