similar to: How to soft hangup all channels at a time .

Displaying 20 results from an estimated 8000 matches similar to: "How to soft hangup all channels at a time ."

2006 Mar 10
1
Extensions base policy
Dear List, I am new in this world (Asterisk) and facing a problem . i want to make some group, base on extensions .so that certain extensions could call to certain predefine number only. let me give u all a short example extensions 1,2,3,4 will be group A , extensions 5,6 will be group B . so group A only allowed to call one or certain predefine number but group B could call anywhere.
2006 Mar 23
1
Dialling Problem
Dear List, I am facing another strange problem . some of my envisions like to use other prepaid card (whatever they found in market) but when they dial that access number (phone number to put the pin) they get IVR (Please provide your pin number ) but when my user press pin its not going through, that IVR even can't get wrong pin number . just get disconnected as no pin number provided. what
2006 May 31
4
how to decrease answer time !
Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque
2006 Mar 21
1
Asterisk and gateway
Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal
2008 Oct 29
4
Dimensioning a telephony system based on openser!
Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work
2008 Mar 05
6
Asterisk based UNIX
Hi All; Anyone tried to install Asterisk based on UNIX (not linux)? Which UNIX was good to work with Asterisk? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs
2008 Nov 14
4
Looking for a good lightweight Linux softPhone
I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route
2009 Aug 18
2
Channels don't go away with soft hangup
Hello List, our setup: Callcenter IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular providers on the xircom analog port, ~60 agents Debian 5.0.1 (Lenny) Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue segfault fix Zaptel 1.4.11 Debian Package My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a duration of over 4 hours. I am
2008 Apr 06
7
Where is the Digium DS3 card?
Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt
2008 Dec 01
3
OT: What do you guys think of this?
http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
2008 Aug 21
1
DSS1 vs SS7
Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Jul 19
2
OT Astricon/Digium Beach Ball Mailing
Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just
2008 Jun 25
1
AS5400 E1 SS7
Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------
2008 Nov 13
2
CANCEL FORWAR
Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | |
2008 Dec 06
1
Add volume sip accounts
Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Aug 20
1
vicidial mysql problem
I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: > /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi > install_driver(mysql) failed: Can't load > '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' > for module DBD::mysql: libmysqlclient.so.15:
2009 Feb 11
3
Billing and Soft Switch.
Looking for a Free VOIP Billing and Soft Switch. Any suggestions ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090211/9ff6e652/attachment.htm
2009 Nov 02
5
Forward DID to another server
hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs
2008 Oct 05
5
asterisk, phpagi and singleton
Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The