similar to: Selecting a specific line from Zap/g And secondary dial tone

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Hi List; In the extensions.conf file, at the [global] context, there is a variable configured as: CONSOLE=Console/dsp What does it mean that? What dsp mean and it is shortcut for what? How can I use the core to get some data about such thing ambiguous for me? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Be a

Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Regards Bilal

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -------------- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

I''m trying to get Puppet to run on ESX v3.0.1. Being that ESX doesn''t come with ruby, I installed v1.8.6 under /opt/ruby/ruby-1.8.6 and linked the bin to /usr/bin. facter installed and runs without issue. However, when I try to install puppet, I get: - Could not load openssl; cannot install Is this due to the way I installed ruby or something else? Thanks, Clif

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Dear I am using ser + asterisk, for setting up land line calling. only probelm, each unregistered soft phone can places the call only with callerid, this is critical problem, because any number(soft phone) , has a limit time to use this system, best Mani ____________________________________________________________________________________Be a better Globetrotter. Get better travel

Hi list; While I am writing my configuration on the .conf files, I would like to know if I wrote the command in write syntax (form), there is not any way to check if I am writing correct or not (other than checking my documentation)? Also, is there any method for searching on specific topic about asterisk (a command details and usage), from my computer (like help and so on)? Regards Bilal

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Josh Coalson wrote: > > > --- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > > > > > Hi all, > > > > > > Is seeking working for OggFlac files? I keep on getting a > > > FLAC__STREAM_DECODER_SEEK_ERROR. > > > > yes, it should work fine. in

Hi List; Can someone advise me which IP Phone model that has buttons that can be assigned to do specific functionalities (call pickup, call formward, call appearance) and a transfer button and hold button? Which is the best of the following (that has buttons can be assigned to specific functions): Cisco 7970 or 7960 Polycom 501 Grandsream IP Phone Budge Tone 1001 or 1002 Linksys SPA 942 or 922

Hello, I am using rsync at Mac OS X for synchronizing pictures for our two offices. Unfortunatelly yesterday the script stops working. Here is little workarround. The script select every file from folder A and write it to PENDING-FILES file. Than RSYNC take from PENDING-FILES every line (file) and transfer to folder B on different machine. Unfortunately some Mac user created folder started with

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it

Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without