similar to: Problem with meetme monitor (recording)

I have benn having trouble with the Monitor Command. Basically any time that I send a call into a MeetMe room I am only able to monitor half of the conversation. File-in is recorded with the incoming voice but file-out does NOT record anything. I have tried this with both the b and m option as well as without any options to the MeetMe command. Also the Monitor correctly records both sides of the

Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no "Proxy-Authorization" information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package)

On 7/27/07, dragoran <drago01 at gmail.com> wrote: > On 7/27/07, Matthias Clasen <mclasen at redhat.com> wrote: > > Given that test1 is around the corner, I thought it might be a good idea > > to give a little status update on the features that the desktop team has > > been working on for F8: > > what happend to compiz-fusion? I've been punting this

I write a small function and test it under clang and gcc, filet test.c: double X[100]; double Y[100]; double DA = 0.3; int f() { int i; for (i = 0; i < 100; i++) Y[i] = Y[i] - DA * X[i]; return 0; } clang -S -O3 -o test.s test.c -march=native -ccc-echo result: "D:/work/trunk/bin/Release/clang.exe" -cc1 -triple i686-pc-win32 -S -disable-fr e -disable-llvm-verifier

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer! Which codec is it talking abt here. How can i resolve this. My dialplan is as

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently

I have TDM40B, TDM04B cards, 4 analog and digital phones. First I want to use 4 analog phones with my TDM40B card. I would like to dial between 4 analog phones. The dialing numbers for 4 analog phone will be 800,801,802 and 803. These are my conf files. /etc/zaptel.conf fxsks=1-4 fxoks=5-8 loadzone = us defaultzone=us ;;;;;;;;;;; /etc/asterisk/zapata.conf [channels] relaxdtmf=yes

I've seen alot of posts lately on Queue and Agent functionality, and alot of hacks to make them do different things that most call center managers want. In the sake of doing this one time, I'd like to develop a single list of request so we can consolidate a feature request for the Queue/Agent system. Here are the ones that I run into the most: 1. Queue should know the status of agents

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"

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

I just installed the pre-compiled binaries for Clang 3.4.1, which was the latest version I could find to download. Starting a new 'blank' project in MSVC I was easily able to change the tool set from MS Visual Studio 2013 (v120) to LLVM-vs2013. However, trying to compile a simple 'hello world' program resulted in the following compiler errors. Is there something simple I am