similar to: Autoreply: Re: SunRocket / ALLO / etc special offer

If you have been affected by the SunRocket / ALLO folding issue, ChiliTech would like to extend our hand to you to help you in this time. We will transfer your numbers to us for no cost, and will match your SunRocket or ALLO rate. Please contact us at 1-866-678-6858 x 126 or e-mail mhoppes at chilitech.net. We have been around since 2001 serving the Internet community. Matt Hoppes ChiliTech

about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration

Has anyone tried SunRocket with Asterisk? http://www.sunrocket.com/ The $199/yr. plan seems like an excellent value, and most reviews have been favorable. However, I don't know if it is possible to obtain the SIP credentials, so one can bypass their "gizmo". Thanks, Stewart

Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. --

hello, running asterisk 1.4.11 on CentOS 4.5 I am getting no response on function STRPTIME() the system just hangs, STRFTIME() is working fine as seen below. Same thing happens whether I called in from a softphone or via teliax. While executing the following code : ; exten => s,n,Set(v_ts=) exten => s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)}) exten =>

We have been running around than 40 asterisk servers running on Debian Squeeze for last three years, handling traffic of more than few hundred thousand calls per day. Our setup's PRI-banks were using Redfone's Fonebridge. We had PRIs from multiple telephony providers. And Redfone's Fonebridge handled all that easily. But all good things come to end. Redfone's Fonebridge was not

Dear all, I'm searching someone who already installed allo.com gsm card with Asterisk NOW. I installed the hardware in my new server, but when runing dahdi_genconf I always have the "no span" message. I tried to install the driver according to allo.com doc, but it seems not up to date, and maybe done more specificly for an appliance.. >[root at localhost ~]# dahdi_genconf

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix