similar to: Power alarm on module 1, resetting.

Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren

Hi all! It's great to start with "for dummies" question, but hey, we all have been human infants also =) Problem is, that I can not log on to this channel and I haven't found anything helpful during the past few days either. 1. The irc.freenode.net server gives me "Couldn't look up your hostname" and "No identd (auth) response" followed with

For those that are interested, here is my report back to Sayson on the 480i ---------- Forwarded message ---------- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: xxx@bgcfreedom.com To: xxx@sayson.com Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com

I finally got my ASUSCOM (Cologne chip) ISDNLink card working here in the US. When a call arrives, I get "Unknown IE 42 (Unknown Information Element)" and "Unknown IE 21 (Unknown Information Element)". IE 21 (0x15) is defined as Q931_CALL_STATE_SUSPEND_REQUEST. IE 42 (0x2a) is not defined in the code and I couldn't google it. Is this something perculiar with ISDN in

I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Has anyone implemented HOBIC SMDR output from *? Can someone point me to the Bell HOBIC specification? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Occasionally my /etc/hosts file gets corrupted. The IP address and the host name switch positions with the host name to the left. What this happens, my 7940 phones won't register. Fixing /etc/hosts allows the phones to register. Do any of you Linux gurus know who is corrupting the hosts file? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575

This Proliant (ml330 G3) has an HP CSB-6 (LSI/Megaraid) disk controller that isn't recognised by the installer. A quick search on Google reveals drivers from RH7 - AS2.1 but nothing for 3 or 4 unless anyone knows different!? If not I'll just have to disable it and install a supported RAID card. Anyone!? Nigel

Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an "assembled" package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Thanks in advance Scott Stingel Scott M. Stingel President, Emerging Voice

Hullo over there. i'm trying to link an asterisk box with a legacy PBX system with a four wire trunk line. the legacy PBX has 21 analog phones connected to it and i would like to route calls to another site via the asterisk box. i would like to use E&M signaling over this line. my question is how many simultaneous calls can you have over this line with E&M signaling. is there a better

Greetings, I purchased a WiSIP at the VON conference and am now trying to configure it to work with Asterisk. I have read all of the previous postings regarding the WiSIP and most of the information apparently does not apply to the version of firmware installed on my phone (version WF.00.0F). I cannot get the WiSIP to register with my Asterisk box. It leases an IP from my DHCP server, then

Hello all, My set up is a fractionated t1, with 1-6 voice channels, 21-24 data channels. I have a t100P installed in a amd 1500 with 512meg. Question: In order to set up the data channels, is the following correct? 1) compile new kernel with generic hdlc support 2) compile hdlc from hq.pm.waw.pl/hdlc 3) uncomment KFLAGS+=DConfig_ZAPATA_NET in zaptel makefile 4) compile zaptel, 5 compile rest

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

Hello, I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I am not able to get any lights on the T100P and the Adit 600 only shows red. I have already modprobed the t100p. Has anyone successfully connected these before? I've tried both a cross-over cable and a standard cat 5 cable, but I can't seem to get them to work. Any suggestions? Thanks! Craig

Hi, all Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1 I do not want to upgrade yet. Thanks, Rudolf

Hi all, We have tested both tdm400 and BRI 1 port HFC based with mISDN driver and with both cards we have the same problem... so probably it's due to our misconfiguration, but we couldn't find anything useful on google. With TDM400 we can place and receive calls, but if we receive a call from TDM400 and we reject it from a sip phone , asterisk send the hangup but the line continue to

When we were first installing these SIP phones, the time and date was displayed on the top status line. Firmware version 6.3. Using dhcp, tftp, http, and ntp servers for configuration all on the LAN (* box). Now that we have these working with *, the time/date display is missing. Any ideas? Thanks, Tom