similar to: can't hear vm audio

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

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

Hi Folks, Yesterday I had need to record a phone conversation. This is not something I'd ordinarily have to do and so I have not configured my * server to do any recordings. When looking for example dialplan stuff I found many examples where the calls are always recorded when the phone is picked up but none that could be done dynamicly. What I'd like to be able to do is press a button

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi All, In theory if I do this; exten => s,1,Zapateller(nocallerid) exten => s,2,Privacymanager exten => s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event of no CID the caller gets dumped into the

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

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

Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm

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

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?

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 thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes 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

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

Hi Folks, I'm trying to get DIAX to work as an IAX soft phone and I'm having some issues. Outbound calls from the phone to the rest of the world are fine. It's just that I can't get the inbound to work. I have a stanza in my extensions.conf that reads; exten=>_3410,1,Dial(IAX2/3444@default|20) And a stanza in my iax.conf that reads [3444] type=friend context=default

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

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

So I go into a new Apple store on Sat to buy some stuff for my Mini, and I notice some Snom 360's on the sales counter. Venturing a question, I ask, are they using Asterisk? Guys says yes. Cool! I said: What kind of box are you using. He points to a Mini sitting on the counter! 2 X cool! He's using a SIP-FX0 converter. Plug: http://www.mymacdealer.com great store in Alberta. Anyone else

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: