similar to: Texas User Group

In an effort to bring Asterisk Users from across the state of Texas together, the Texas Asterisk Users Group has been formed. The goal is to help Asterisk users meet other is their area and to help spread the word about the Asterisk community. I anticipate regional meetings of members and look forward to all of our members being able to attend the Astricon Event in Dallas in October. I invite

Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/d3bb612a/attachment.htm

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again

I have seen this product before and wondered, has anyone connected this to Asterisk? http://www.i-2-eye.com/index.html As far as that goes has anyone seen a set top box video phones that work with Asterisk? -- Bruce Reeves Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellphone to compile, but bluez-utils required an update to D-sub which in turn required python 2.4 or better. That apparently in not

Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P +

Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I

I am trying to test out my PRI configuration on a server by conection another asterisk server to it. I have the T-1 cards up and both set to bchan=1-23 and dchan=24 in zaptel.conf I am getting lost on the signalling in zapata.conf. using pri_cpe makes sense for my target box, but I am getting the following errors on my source box if I use pri_net or pri_cpe signalling. May 31 17:15:06

I am looking for input on wether the 4 port T-1 foneBridge is a useful device in a asterisk deployment. Would it be better to loadup a extra asterisk server to trunk the T-1's via IAX or is TDMoE safe in production enviroments. We are only looking at 2 T-1's, one for pstn and the other to a channel bank. Any advice? Bruce Reeves Nortex Networks -------------- next part -------------- An

Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: "Got SIP response 400 "In alert-info header: Empty value expected" Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and

I am looking at the two Digium cards mentioned and am not sure if the echo cancelation is a must. The sever is a 2.8Ghz P4 with 2 GB of ram, and is only for routing PSTN connections to another * server. I would like the advice of the list before making my final decision. -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI: