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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:

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...

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

...This is a digitally 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/pla...

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>

All, I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy errors. I rewired this thing three times, then I connected the same cable from the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events and frame sync errors went away, so the cable is good. STX --> DSX