search for: chan_spy

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 i...

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 cur...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050208/9f8e957c/attachment.htm

...rom 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 MoH class, there is no exit. But might there be some trick involved merged or bridged calls, or chan_spy or something, so that callers could quickly switch between 3 streams with a keypress? Thank you!

...sed 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, please see: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a summary of the changes in this release, please see the release summary: http://svn.asterisk.org/svn/asterisk/tags/1.2.33/asterisk-1.2.33-summary.txt http://svn.asterisk...

...sed 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, please see: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a summary of the changes in this release, please see the release summary: http://svn.asterisk.org/svn/asterisk/tags/1.2.33/asterisk-1.2.33-summary.txt http://svn.asterisk...

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] app_chanspy.c: Attaching SIP/209-00001715 to SIP/210-00001714 [Mar 4 17:31:06] NOTICE[25812] app_chansp...

...l, 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 MoH class, there is > no exit. > > But might there be some trick involved merged or bridged calls, or > chan_spy or something, so that callers could quickly switch between 3 > streams with a keypress? > > Thank you! > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Jo...

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.