similar to: Help Getting Local Exchange Dialtone on PRI

Hi There, I have an asterisk an a Digium 4 Port E1 Card On E1 Port No. 1 I have the Telekom PRI On E1 Port No. 2 I have an Alcatel PBX that cannot be changed So I have setup my asterisk between Alcatel and Telekom In extension.conf i configured: [telekom] exten => _9149.,1,Dial,ZAP/g2/${EXTEN}; exten => _9149.,2,Hangup This works great, all incoming calls are directly routed to alcatel

Hi there, I have an asterisk server with cvs-code from May 13,2004 and a Quad-PRI Card. 1 Port of the Quad-Pri is connected to the Telekom-PSTN, the other Port is connected to an Alcatel PBX. What I want is to make Asterisk Bridge the Calls from PBX to PSTN and to add some VoIP Functionality and Logging. Everything worked fine while testing but suddenly (maybe under higher load ?) the system

How can I get external (telecom local exchange) dialtone on HFC ISDN BRI with bristuff/zaphfc driver? with capi, voip-info say that it should be something like: Dial(CAPI/MSN:b) But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 -

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

extensions.conf has ignorepat => 9 exten => _9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it asked if instead of keeping the same dialtone after pressing 9, if I could play a different dialtone. Can this be done? I'm running asterisk 1.0.0 in case that matters.

1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hi there, I have a customer with domain a at tld and domain b at tld. b.tld is an alias for a.tld The Server is using virtual users based on mysql, dovecot for delivery and Maildir for Mail-Storage. If someone writes an e-mail to a.tld with a cc: to b.tld the customer receives the E-Mail twice. Is there a way to deliver the same Mail in the same Mailbox only once ? -- Christoph Adomeit

Hi there, I have to migrate data out of a very old (debian 4) and buggy cyrus mailserver. About 40 Users with 200000 Messages each. Doveadm backup works quite well and gives me most of the mails. About every 100.000 Mail cannot be exported by cyrus. I do not want to bother about these errors. Maybe it is filesystem, maybe it is memory leak, or cyrus bug or whatever. My problem is that doveadm

Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 "database put number name" via a shellscript and "asterisk -rx" I only have 600 entries later in my asterisk- database. The asterisk

Hi, I just recieved a new TE401P card from digium, I put it into a PC with asterisk and did a fresh CVS update and compile. However, when I load the driver (modprobe wct4xxp) for this card it detects the card and loads the driver but then I keep getting all these warning messages, such as: "Tried to load 00000000 into 00000004, but got ffffffff instead Not prepped yet" Every few

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? </edg>

Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?

A question: is there any way to get * to answer certain DTMF sequences entered on an extension with a stutter tone? Long version: I would like to add features to my dialplan like "Caller ID Unblock" which work in the same way that the PSTN works: I pick up the phone, get a regular dialtone, press *82, and get a short stutter dialtone which confirms acceptance of the request, and then

I have a problem with voicemail recording dialtone on hangups. From time to time, callers hang up without leaving a massage, so Asterisk records around 100 seconds worth of dialtone instead. I've seen this off and on for almost 9 months now, it's currently happening with 1.0.2 on Linux 2.6.9. My Asterisk box has a X100P connected to Verizon and IAX service through NuFone. I have

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my