similar to: 2 NuFone lines- which one to dial out on

hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. "ztcfg -vv" shows me a this info: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03:

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Hello There, I am trying to get Asterisk to work with Billion ISDN Adaptor, But i couldnt get isdn4linux to work. I am pretty new with isdn card but this is the only available option here right now , I've looked at this post and found that the author has been successful in installation of the same card. Can someone please help me out by giving the author's email address , so i can

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

Hello. New to the list. We're in the process of deploying Asterisk. Actually, we're going live tomorrow, and I just found out that my Zaptel cards have been mis-configured. I'll preface this by saying that I have looked in the wiki, read through the samples, and attempted to call Digium (they're closed.) So I'm praying that someone on the list can help me out! I have two

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

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error

Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it can't find channel 35 but 35 belongs to my wctdm.

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously).

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try 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

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.