similar to: live monitoring of SIP calls chan_spy

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

Hi all, I read on serveral pages that chan_spy is not part of asterisk anymore as on http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy on the bottom of this page. I have a testing server with debian-testing and debian packages for asterisk installed. In the modules directory /usr/lib/asterisk/modules is a app_chanspy.so already there. The currently installed version is 1.4.13. So, it is

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

I have created a call file which has been moved into the outgoing directory. However the log file displays the following message: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting I have executed chmod 777 1.call on the file prior to moving it to the outgoing directory but is there something else I need to do before the file can be used by Asterisk? Any help

Hi List, As most know, Chan_Spy and consequently, the MOH patch that used Chan_Spy disappeared around version 1.0.2 (or so). I know the native MOH patch works well and doesn't require the mpg123, which as proved problematic, at least for me. However, I know of no method to "listen in" or supervise a conversation in real-time. I'm getting requests from users where the support

Hi, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. Bye, Sven -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm asking because I'm planning to install multiple machines from the same image and I need to know what file(s) I need to backup/restore to make sure I don't lose my licences in the process. The only options I can think of are: - There's a config file, though I've seen no mention of it - The actual binary shared library is modified - The system contacts Digium every time you

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN

Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi all, I'm researching random call drops on our Asterisk and would like to make sure whether it's something wrong with our VoIP provider or with the Asterisk. I sniffed traffic between Asterisk and our VoIP provider's SIP gateway, and observed that in the middle of the conversation an RTP stream originating from Asterisk gets an ICMP port unreachable from provider's SIP gateway

I know this is a really stupid question, but I just have to ask... Where would I start if I wanted to try and develop my own firmware for a particular phone. Namely, I want to try and 're-write' the SIP firmware for Cisco 7940's. Any ideas? -Chris PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [*

Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050303/facb2a2d/attachment.htm

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. I have mysql and mysql-devel packages installed I think!? pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/