similar to: Callerid UK patches (from Lusyn)

Folks, I've seen a few reports that people have had problems with hang up detection on UK cable phone lines. I have a TDM400P with two FXO ports, one connected to my BT line and the other connected to my Telewest/Virgin Media cable line. If I ring the BT line and then clear down, Asterisk detects this and acts accordingly. If I ring the Telewest line, the clear down is not detected, hence

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

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,

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.

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep

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

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

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

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

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

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

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

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call: