similar to: Inserting access codes as prefixes to CID

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I'm and Asterisk@home user - been so now for almost a year. Lately, I've upgraded to the latest & greatest.. (which is built on 1.2.5) and am unable to install oh323. I've already asked over at the (A@H) Sourceforge forum but no one seems to think it worth answering. The error I get is pretty obvious but I don't know where to go from here. More importantly, I need to have

Ahhh this could be my problem! I just checked which wires on the RJ11 cable had a voltage across them and it was the yellow and green (3 & 4?). From what someone posted the other day it's supposed to be Bumble Bee and Christmas Tree. I did have to get a technician out to fix my line when it was first installed because it was dead. Maybe he wired it up incorrectly or maybe they just do it

Thank you (for spamming) - it was the clue I needed to push this through. Sorry it took me a while (and a google :-) ) to realize you'd addressed my initial query - basically, my loss. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Becker Sent: Wednesday, August 24, 2005 00:36 To: Asterisk Users

Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what

I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant

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.

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

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

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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

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

I recently changed to all IAX providers for my DIDs, and none of them offer incoming caller-names. Back in the days when I did have incoming caller names, I found the names provided by the various phone-companies fairly useless -- "WIRELESS CALLER", "YOURCITYHERE" etc. Last Friday I finally set off to roll my own, in order to meet the following requirements: - uniformely

Hi all, using Skype for Asterisk I have the following problem. In my dialplan I need to have a CID matching (example.skype) over a DID (test.skype) : [example] exten => test.skype/example.skype,1, NoOp(nothing) exten => test.skype/example.skype,n, Hangup() Where test.skype and example.skype are Skype business account. In this case, when I get a : CLI> show dialplan example I get: [

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

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 folks, My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL database, including those handled by the Privacy Manager. Unfortunately, even though I can use the CLI to see the information being submitted by anonymous callers to satisfy the demands of the the Privacy Manager, that information is not recorded in the database. Instead, all that is written to it: clid:

Hi, We have recently implemented asterisk as the office PBX, and as a GUI, we chose Elastix. All seems quite easy and am very pleased with the way Asterisk is running and handling calls. To summarize briefly, our setup looks like this: LAN -> Asterisk PBX (Elastix) -> Asterisk TDM (basic CLI with libSS7) We get a whole 3 digit number series from our telco. So, like, +1 312 765 7xxx

Hi list, My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get this error message on the asterisk console: == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' [Jul 8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable A long time ago my