search for: chan_spi

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

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

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

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'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm wrong, but from previous similar questions a few years ago it seems like once you've entered a

The Asterisk Development Team has released Asterisk versions 1.2.33, 1.4.25.1, 1.6.0.10, and 1.6.1.1. The released versions are available at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes a REGAUTH loop related to security issue AST-2009-001. Asterisk release 1.2.33 also addresses a small compile time error in chan_spy. For more information about the security issue,

The Asterisk Development Team has released Asterisk versions 1.2.33, 1.4.25.1, 1.6.0.10, and 1.6.1.1. The released versions are available at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes a REGAUTH loop related to security issue AST-2009-001. Asterisk release 1.2.33 also addresses a small compile time error in chan_spy. For more information about the security issue,

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790]

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.