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All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060131/79fbb6dd/attachment.htm

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> > Unnumbered frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000 EA: 1 > M3:

...second Zaptel.conf loadzone = us defaultzone = us span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 bchan=73-95 dchan=96 TIA, Jon Weisman | Sales Engineer International Bell Communications www.ibell.net www.ibellhost.com www.aeronhelpdesk.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070626/a401faa0/attachment.htm

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and...

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

...gitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20080925/04d43106/attachment-0001.pgp > > ------------------------------ > > Message: 15 > Date: Thu, 25 Sep 2008 15:08:19 -0400 > From: "Jon Weisman" <jweisman at ibell.net> > Subject: [asterisk-users] Server Dimensioning > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: <6C86A52999604699998A419BD4F8289C at VaioMatic> > Content-Type: text/plain; form...

...ect: Re: [Asterisk-Users] Anyone know a good ITSP in Canada that supports *? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20060202130614.83736.qmail@web36511.mail.mud.yahoo.com> Content-Type: text/plain; charset=iso-8859-1 iBell just announced termination only to CA for I believe $0.0039 a minute. --- Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: > On Thursday 02 February 2006 07:39, hugolivude > wrote: > > I'm looking for a new Internet Telephony Service > Provider for my company in...

We've got a white route w/ VSNL. $0.09 / min, billing is 1/1 prepaid only. SIP or H323 w/ G729 Codec. E-mail me off-list for testing. Thanks, Jon ----- Original Message ----- From: "Jerry Romney" <Jerry@uspcom.com> To: <daniel.silaro@gmail.com>; <asterisk-biz@lists.digium.com>; <asterisk-dev@lists.digium.com>; <asterisk-users@lists.digium.com>