similar to: problem with monitor meetme

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) and

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

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:

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

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. > > >

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

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

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

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

Hey there, Just wanted to drop a line and let people know that I'll be heading to San Francisco for O'Reilly's Etel. If you are interested in attending, there are some free passes floating around. If anyone is interested in getting together for a beer, let me know! Info on the conference can be found here: http://conferences.oreillynet.com/etel2006/ I'm looking forward to an

I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now. However, no sound is heard and I get this message from the CLI when accessing MOH: -- Started music on hold, class 'default', on channel

Hello, I am trying to use postfix/dovecot as mail server to be the imap storage for my voicemail system.For that I installed postfix and dovecot and trying to follow the instructions in this post http://etel.wiki.oreilly.com/wiki/index.php?title=Storing_Voicemail_on_an_IMAP_server&printable=yes I should add a master user for Asterisk to your IMAP server that has access to all user's

http://etel.wiki.oreilly.com/wiki/index.php/Main_Page -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' "Wise men talk because they have something to say; fools, because they have to say something." -- Plato -------------- next part -------------- A

Mondada Gabriele <g.mondada at etel.ch> writes: > I just moved to the CMake solution. By the way, the generated libs > haven't the same names. Which ones? The only difference is that we now generate .lib files where .obj were generated on the past, and require a parameter to be passed to the linker for including them on the final executable. > In my opinion, we have to choose

I have RoR running and have configured RAGI (per the new tutorial provided at ETel http://www.snapvine.com/code/ragi) but I get this: --- user# script/server => Booting WEBrick... [2006-01-30 14:20:25] INFO RAGI::CallServer: default-handler= port=4573 [2006-01-30 14:20:25] INFO WEBrick 1.3.1 [2006-01-30 14:20:25] INFO ruby 1.8.2 (2004-12-25) [i586-linux] [2006-01-30 14:20:25] WARN TCPServer

Hi, I am trying to change the dialplan to enable call recording (incoming and outgoing calls) on the "click of a button". Is it possible? All the documentation I found so far, enable recording for 'all calls' to an extension. Does this code look ok? Currently Recording "on" only for 1030 when user presses *44, start recording. *55 to stop recording

Hi, I have some problems and doubts connecting two asterisk servers. I have one asterisk (serverA), with 1 sip client registered (clientA). I have another asterisk (sever B), with another client (clientB). Now I want to call from client A to B and from B to A. Searching in google i find many configuration examples. For instance:

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hi, I am not sure if this is a bug in FOP (Flash Operator Panel), a configuration error on my part or a bug in Asterisk. I am using Asterisk 1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version 2.6.9-22-EL-i686. Kernel updates are excluded and the server has been updated using 'yum -y update'. Okay here is the scenario: I am using AgentCallBackLogin as an extension in my

I have not yet seen this article posted to this list, so I thought many of us would be interested in having a look at this project sponsored by O'Reilly: http://www.oreillynet.com/etel/blog/2007/01/help_create_the_asterisk_cookb.html It seems they are looking for both problems and solutions, and I'm sure we'll have plenty :) l. -- Loway Research - Home of QueueMetrics