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Brian, Great job fixing up the list server... postings are happening very quickly now (within minutes)! Thanks, Rich

Hi, Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of Redhat Enterprise 3.1 server.? -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

...n't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

...iklos, I have the same problem here in RH90 - have you found any solution? Or does anybody else know why (safe_)asterisk does not start using rc.local? (normally I start * as root user) Cheers Jeroen --__--__-- Message: 7 Date: Mon, 26 Jan 2004 08:25:47 -0500 From: Frankie Gravato <nanog@cfsdigital.com> Organization: Cfsdigital To: Rich Adamson <asterisk-users@lists.digium.com> Subject: Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller Reply-To: asterisk-users@lists.digium.com Hello Rich, Sunday, January 25, 2004, 8:01:25 PM, you wrote: RA> It...

Hi i was trying to call 17009978275 which is my Fwd line on my notebook from Asterisk and i keep getting this message on the console. -- Executing Dial("Zap/2-1", "IAX2/@iaxtel.com/17009978275@iaxtel") in new stack -- Called @iaxtel.com/17009978275@iaxtel WARNING[1150495040]: File chan_iax2.c, Line 4547 (socket_read): I don't know how to authenticate rob to

Is nufone having problems taking gsm calls today i had some issues dialing overseas to call my folks. here's snip of what the console displayed -- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1

...Read error: Resource temporarily unavailable == Spawn extension (macro-ringall, s, 2) exited non-zero on 'Zap/1-1' in macro 'ringall' == Spawn extension (from-pstn, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1 -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

I recently plugged in Phone to my TDM400P Card to test out something I mostly use sip phones to interface with *. All of sudden I'm getting lot of line static noise coming of the card is there any settings I should look at or anything I need to do on the command line at this point I'm open to any ideas I'm running 0.7.1 on Redhat 9.0 machine. Any insight would be greatly appreciated.

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

Has anyone really looked at the costs for Astricon. But the hotel costs. $111.00 USD per night.. come on guys give me a break. I will not be staying at that hotel. I can rent a car and stay near the air port for almost half that. In addition from what I have been told their will be no shuttle service from the Airport to the hotel. Anyone else have any input on this? bkw_ PS: I'm going

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Is there way to program the keys to transfer calls on analog fones instead of using the pound sign i've notice while i was calling to check voice mail at work that when i hit the pound key i get transfer message i want to use something elese besides pound to transfer calls. any insight would be great -------------- next part -------------- An HTML attachment was scrubbed... URL:

Starting up the asterisk using asterisk -vvvc i get this error is this normal and i purchased license for g729 today? [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) Jan 21 17:31:58 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: Interrupted system call Jan 21 17:31:58 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: