similar to: 911 Notices

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 did a quick google search of the lists site and couldn't find a definitive answer, so if it's there, I apologize for asking again. Starting about noon yesterday, I am no longer able to send/receive calls via Broadvoice. When calling in, I get a fast busy, and when calling out I get the following error: -- Executing Dial("SIP/112-572a",

Good Day All I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings. But on external network with PROXY setting ASTERISK DID NOT WORK. My question are 1 Can ASTERISK work in a PROXY setting . 2 If it can work how can i implement it . Expecting your reply Thanks Paul --------------------------------- Yahoo! Messenger

I am planning to sign up for a VoIP service in the U.S. Can anyone recommend anything cheap, reliable and good quality? I want to use it for my primary house phone (I also own a cell phone). I also want the service to be asterisk friendly so I can play with it :-) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

Who is everyone contracting with for 911 services with the upcoming FCC deadline? I've got a few feelers out there working on this issue, but no real solid leads yet.

Hi all!....i'm in course to implement Faxing in my asterisk box and for that I've installed all succesfully like libtiff and spandsp and next rebuild and reinstall asterisk modules, but when i call to Rxfax from dialplan nothing happens and i get some errors like "XCN with final frame tag In state 9" or a paper copy of a transmision report from the fax machine with COMMUNICATION

FYI there is a way to do 911 its called E-911 enhanced 911 the user has to set it up with the local emergency services to it and you setup your pbx to xmit the data. Here is the fcc rule about it http://www.fcc.gov/911/enhanced/

Hi folks, It seems that BV has messed it up yet again. I noted this weekend that any call going in or out had no incoming audio. All my other SIP providers seem to be OK. Is anyone else having this problem? Perhaps it's time to move on. What providers do you recommend that provide unlimited US/Canada and Western Europe? Thanks Mark

To anyone seeking 911 services Highly recommended to everyone to stay away from this issue I do not have a name for the company right off hand, but they got sued really bad when they tried 911 via VOIP and the 911 drop kept occurring in different areas and someone died! Implement 1 single hard wire for this service and cover your tush! Brad -------------- next part -------------- An HTML

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

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

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,

I've been trying to get a new asterisk box setup with Broadvoice for over a week now. I have it connecting and registering with them according to 'sip show registry', I can't dial out through it, but it does dial out through my regular phone line. I'd like to set it only to dial 911 through that line and have all other calls go over voip. I've checked out a bunch of

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?

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

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/

Dear List, I am a little unsure how to structure my model and was after some advice. I am a little unsure if this question is appropriate for this list, if it is not please just delete and accept my apologise. I have 10 factors that are categorical variables and 5 levels of response variables - A B C D - Factors RESPONSE? 2 2 2 2 1 2 4 2 2 2 2

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