similar to: CONSOLE=Console/dsp

Displaying 20 results from an estimated 10000 matches similar to: "CONSOLE=Console/dsp"

2007 Oct 02
0
Selecting a specific line from Zap/g And secondary dial tone
Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP
2007 Jul 27
4
Asterisk Wiki
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
2007 Sep 28
1
How can I know if I wrote the configuration like correctly
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
2007 Jul 23
2
Upgrade and keep the configuration
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
2007 Jun 14
11
Asterisk GUI
Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469
2007 Aug 27
4
Prepaid Billing: A2Billing, AstBill, ASTCC
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
2007 Mar 30
1
Which IP Phones have buttons can be assigned to functions with Asterisk
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
2007 Jul 01
4
Not able to find the file zaptel.conf after compiling asterisk and zaptel
Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.
2007 Jul 01
1
How can we block the calls for specific code
Hi List; What is the command and where I can write it to block specific code from calls (then no one will be able to place call for any number start by that code)? --------------- Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: + (965) 9849460 Yahoo ID: bilmar_gh at yahoo.com MSN ID: bghayad at hotmail.com
2007 Oct 11
4
Buying Polycom
Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal ____________________________________________________________________________________ Be a better
2007 Aug 23
3
Asterisk Prompt
Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad
2007 May 01
10
Digital Phones
Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around
2007 Jul 11
3
Could not load openssl; cannot install
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
2007 May 25
0
Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not
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",
2007 May 12
0
ser problem
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
2007 Sep 11
0
Is FLAC__stream_decoder_seek_absolute working for OggFlac?
--- 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
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I
2007 Aug 14
1
Rsync on Mac OS X
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
2007 Oct 09
2
Asterisk Realtime woes
I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk
2007 Sep 09
3
canreinvite
Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send