similar to: Meetme Dumping Call's

Displaying 20 results from an estimated 3000 matches similar to: "Meetme Dumping Call's"

2006 Jan 22
1
Gen. Question
<RANT> Funny your concerned about copyrights and moral issues regarding the work of others. One question you may want to ask YOURSELF is: Why would I use as my email a copyrighted work followed by the name of the Company that owns the copyright??? asteriskdigum@yahoo.com, Come on!! Who are you trying to fool? Are you out for the fast buck, by having someone that thinks you work for
2003 Oct 22
29
Meetme
Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"
2010 Jul 26
2
MeetMe
Hi guys, i'm trying to use the "featuremap" of features.conf inside the app meetme, but it's no working. like: _5XXX => { Set(DYNAMIC_FEATURES=toca_macaco); MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF Hangup(); }; in features.conf: toca_macaco => 123, peer, Playback,tt-monkeys But, if, inside the room, I press *123* the sound file
2006 Feb 06
1
SV: Help on queues
What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've
2006 Mar 07
1
Setting Vaaibles
Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as
2006 Mar 16
7
OT: Unblocking bloced CID
Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam
2006 Mar 11
0
how to check if ztdummy is working properly?
Kernel is 2.6. zaptel and ztdummy load with Linux, so I can check in lsmod that they are loaded. They load without any problem, I've loaded them manually too. Zach A -----Original Message----- From: Dovid Bender [mailto:asteriskdigium@yahoo.com] Sent: Friday, March 10, 2006 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to check if
2006 Mar 22
1
Big Traffic anyway?
Hello, On 3/22/06, Dovid Bender <asteriskdigium@yahoo.com> wrote: > I have a rule in my outlook to delete any email that > in the subject it says "asterisk users mailing list > traffic". Get my drift ? This topic has been around > for a long time as others have mentioned and people > keep replying. This useless topic alone adds several > messages to my inbox
2006 Feb 07
0
Help on queues
Campon, mini-queues, see asterisk tips and tricks on voipinfo... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing
2006 Mar 06
1
Extension 's' in Realtime
Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type
2006 Jan 17
6
OT: DCAP Certification
Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama
2006 Feb 02
1
Re: Contents of Asterisk-Users digest...
-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To
2008 Jun 17
1
looking for help / input with Blind transfer from asterisk to zap
List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is
2005 Mar 16
0
Meetme doesn't react to DTMF keys
Hi, I am playing with conferencing, but might have hit a bug... Any use who wants to hang up or leave the conference should press the # key, after which they get a "goodbye" message and the call gets disconnected. However, this does not happen. whatever keys are pressed by whichever party gets heard on every other party. I am using Zap channels (Digium T405p) My extensions.conf
2007 Mar 15
1
Freepbx Incoming call's configuration
Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody
2010 Jun 09
0
AMI Queue information about incoming call's channel before link
Hi, I'm developing an application using AMI and I need to get information about incoming call _before_ queue member answers it. I'm using static members (queue is pretty simple) and don't want to use chan_agent (I think AgentCalled event will do what I'm looking for). Here is what I'm getting now: 1. Newchannel event for incoming call followed by Newstate and Join. All these
2015 Jan 27
1
Making the bi-weekly VIRT SIG call's IRC only
Hi all, at today's call we decided to make the meetings going forward IRC only. If I don't get any objections by next Tue, I will update invites and wiki Regards Lars
2006 Feb 15
5
Aasterisk large-scale deployment w/analog phones
hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar
2004 Dec 12
0
MeetMe performance
Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the
2006 Mar 27
5
FreePBX & AAH
Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only