similar to: New TDM04B 4-port FXO card problems

Hello all, I am having a big issue that i just cant get fixed for a while now and its causing me lots of grief. It seems like since we got FIOS installed (including switching to fios phone lines which are supposed to be the same on our end) i am having massive problems with asterisk not hanging up dead calls for days, even weeks if i dont catch it. It slowly builds up randomly not ending a call

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I setup the FXO with kewlstart signalling, and the outgoing call is never registered as 'Answered' and the dial

Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when

Hello all, I am having a serious problem installing my * with a TDM04B. I made everything work, call are coming in and going out including using a GSM Box in channel Zap/2-1. I did setup voicemail like this on extensions.conf: [incoming] exten => s,1,Dial(SIP/2246,20) exten => s,2,Wait,2 exten => s,3,Voicemail(u${ME}) exten => s,4,Hangup exten => s,102,Wait,2 exten =>

I know, I know, check the archives but I can't find an answer since the new cards are well, NEW! I understand the whole issue of expandability and flexibility of using a T1 card and an Adtran 750. FXO or FXS, you mix and match. With the new card offerings from Digium I can easily put a 4CO by 8 Station system together. Barring the extra interrupts, extra CPU Cycles of extra interfaces, is

Does anyone have disconnect supervision working on their TDM400P or X101P cards (to/from the telco)? Googling "telco disconnect supervision," I found the following description: http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html [cut] Re: How Do I Get "Kewlstart" From my Phone Company? Paul A Lee Tue, 24 Aug 2004 16:58:16 -0400 ... It

FYI... Just installed one of the new Sangoma A200D analog pstn cards with the hardware echo canceller on a trial basis. The card has four fxo interfaces. Excellent audio quality, excellent echo cancelling, and excellent audio levels. The four pstn lines at this location are rather long analog loops that have rather long echo trails. I started with a pair of x100p's a couple of years ago,

I would like to install a TDM04B with Asterisk@Home 2.4 and Asterisk@Home 1.5 but I didn't find documentation about this installation. Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/85ff5252/attachment.htm

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

Is this supported? I have a very simple setup where I have 2 X100P cards and a TDM10B. The TDM10B is connected to a phone that has a digital answering machine built into it. If I make an inbound call on either X100P interface it gets transferred to the TDM10B interface. If I let it ring the TDM10B interface answers the call and the greeting message of the answering machine starts. Then shortly

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Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card,

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank

> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although >