similar to: Anyone got latest SIP image for Cisco 7960?

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

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

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, 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

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

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 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 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

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?

So I've been fighting to get the X100P working. A battle which I've kinda won but not without a cost. Before I won the Zaptel battle I was able to hear all of the messages that asterisk plays. For example, when I'm accessing VoiceMail 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

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

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

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

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

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

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

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

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

Does anyone know how to get any of these VOIP phones to allow me to do menu selections through asterisk, like when accessing voicemail and such. Thanks :P --