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Listonians, Anyone get * to work together with Vonage? Thanks, Jerry

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Is anyone successfully using the AGI script calleridnamelookup.agi (or anything similar) ? I get both name and number caller ID from my POTS line, but I'd save money if I had them deliver ANI only. I've downloaded and installed the AGI script calleridnamelookup.agi, but I always get -- Executing AGI("SIP/9525485560-5359", "calleridnamelookup.agi") in new stack

Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I have something I'd like to try, but don't know if enough information would be delivered to Asterisk to make it work. I have a cell phone that I would like to forward on "Don't Answer" to an iConnect phone number that is then delivered SIP to my Asterisk PBX. I'd like to be able to route the call when it is received by Asterisk based on the number that the caller

In May, I posted an inquiry to the list concerning my desire to configure my own CLASS codes in extensions.conf rather than having them hard coded into the channel drivers. I have a number of old rotary dial telephones that (obviously) can't dial *. Traditionally in the US, "11" can be dialed in place of "*" as the first digit dialed. Many people mentioned that this

I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur

I have set up my asterisk PBX to provide a double-ring for outside calls, and a single ring for station-to-station. (I'm talking about ZAP stations in this email). I had to go into one of the .c files and tell it to expect the Caller ID between the 2nd and 3rd rings in order to get the double-ring scenario to work. My problem is that, in making this change, I now don't see Caller ID on

I have a user who wants to receive an ANI spitback in DTMF. Right now, the "SayDigits(${CALLERIDNUM})" command works fine with voice. But I'd like to end up doing both. Something along the lines of: exten => 34,1,Answer exten => 34,2,Wait(1) exten => 34,3,Playback(vm-extension) exten => 34,4,SayDigits(${CALLERIDNUM}) exten => 34,5,Wait(2) exten =>

I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was

Vonage doesn't sell just a softphone account -- or at least they didn't about six months ago when I was a Vonage customer. But they do allow a softphone as an add-on to an ATA-based account. Because the softphone account works with openly available soft clients, it also works with asterisk. The big "secret" is that they use port 5061, rather than port 5060. > > I

I've been puttering around with extensions.ael, and had a question. (Well, 2 questions, but they're related). First, would asterisk recognize any other .ael files as asterisk extension language? Second, is there a way to #include another file from extensions.ael like there is from extensions.conf? TIA Greg -------------- next part -------------- An HTML attachment was

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Tip side open on the analog line? Have you taken a butt set or normal phone and attached it directly to the outside line to see if you get dial tone? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Jon Gabrielson > Sent: Friday, January 28, 2005 11:09 PM > To: Asterisk Users

Folks, I have to admit that I MAY have changed something (at someone's advice) on a previous CVS head (May 28), but I'm not sure. I think that it had to do with changing "digest realm," but that may be a different issue. At any rate, I had both incoming and outgoing with iConnectHere. Now, I made exactly ONE change: I upgraded to the CVS head dated 7/30. I

Searching through the archives, I couldn't find an answer for this... I have two asterisk systems, (system A and system B), and would like to use a single voicemail system. Phones on system B are SIP phones, registered at system B. Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? If so, how? Thanks, folks.

I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT