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
Hello,> 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.You can do something like this with the Flash Operator Panel ( http://www.asternic.org ). chan_spy would be a better option because you can use it from the dialplan. As a workaraound, FOP lets you drag your phone to a bridged call and put the three in a meetme room, with the option to start the 3rd led muted so the other won't notice the interruption. Regards, -- Nicol?s Gudi?o Buenos Aires - Argentina
On 08/02/2005 19:23 slohmann@ifad.de said the following:> and SIP clients. I was looking for chan_spy application but it seems to be > no longer available.oddly, ChanSpy seems to be removed from mantis. any idea why this was done ? -- Regards, /\_/\ "All dogs go to heaven." dinesh@alphaque.com (0 0) http://www.alphaque.com/ +==========================----oOO--(_)--OOo----==========================+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=========================================================================+
Sven Lohmann
2005-Feb-08 05:17 UTC
Antwort: Re: Antwort: Re: [Asterisk-Users] live monitoring (SIP only)
I am one of these unhappy people using the wrong USB chip and building my own kernel (RTC is activated) is no option due to company policies. asterisk-users-bounces@lists.digium.com schrieb am 08.02.2005 13:11:44:> On Tue, 8 Feb 2005, Sven Lohmann wrote: > > > Yes, that would work - but I have no Zap and therefor no meetme - oris> > there > > a way to start meetme with SIP interfaces only ? > > Use ztdummy or zaprtc. All that is needed is the zaptel timing. Another > option may be to use app_conference (use google). > > Peter > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050208/751029b9/attachment.htm