similar to: Hang up detection with TDM400P and Telewest/Virgin Media line

A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time.

Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -rvvvv I see the port pick

Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain

Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -----Original Message----- From: Storer, Darren [mailto:starusers@comgate.tv] Sent: 01 July 2004 09:35 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve,

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

I am on UK Cable (Telewest) will this callerid patch work with the cable caller id? Thanks. Paul.

Hi, got a Sangoma A200 with a bunch of extension cards and having real problems getting it to deal with a normal single BT line Symptoms are that incoming calls are fine. Outgoing calls ring the far end, BUT asterisk never sees that the call is answered (ie no message in the logs files saying so), as a result the remove end can hear the PBX side talking, but there is no audio back from the

what happens when you drop your gains? use /etc/asterisk/zaptel.conf and fiddle with tx and rx values. Works, most of the time. -----Original Message----- From: John McEntee [mailto:john@mcentee87.freeserve.co.uk] Sent: Wednesday, October 11, 2006 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Echo problems on ISDN. (mainly incoming calls) OK I

Hi, I'm trying to connect to Telewest/Virgin Media with a TE110P using asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always appears as PRI span 1/0: Provisioned, Down, Active My zapata.conf is currently ----------------------------------- [channels] echocancel=yes echocancelwhenbridged=no echotraining=yes switchtype=euroisdn contect=from-pri signalling=pri_cpe group=1 channel

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

Hi there, We've got a problem connecting Asterisk with a TE205P to a TeleWest E1 ISDN line in the UK. We get a lot of "HDLC Bad FCS (8) on Primary D-channel" errors, and every so often the Primary D-channel goes down and all the calls got dropped. We've fully tested the card and made sure it's got its own IRQ. I was just wondering if anyone out there has

OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is having calls dropped. Sometimes you can stay on the phone for a long time and sometimes the call is dropped within a minute. There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we replaced did not have this problem. There are 8 SIP phones and 16 analog phones connected to two Astribank-8 units and everyone

On Wed, 13 Dec 2006, Wireless wrote: > Hi Gordon > > I too have this problem with one of my two BT lines, very very annoying. I > was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), > I think the sangoma is very slightly better (less echo) but I might just be > kidding myself. Another think I've found it that using a faster CPU (now on > a 450

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

I have seen mention of limiting a single server to no more than 100 trunks. With Sangoma having an 9 T1 card that kind of blows that limit. How many T1's can we have on a single server assuming dual Xeons? Our application calls for 122 IP Phones and 120 analog ports for faxes. Its an executive suite and we need the fax ports. No way around that requirement. Thanks, Allen.