search for: brodmann

...4 to double/quadrouple the loudness of a channel's input/output using chan_sip? All clients come in via chan_sip so using another channel type is no solution in my situation. I use G711ulaw only. Btw: I would be very glad if someone could point me to a solution of this. Kind regards, Andreas Brodmann -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070525/6c1b2f32/attachment.htm

...in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does not send the rfc2833 messages anymore. Does anyone know about this or has seen an open bug case for it (I haven't found any myself)? Thanks for help and feedback. Kind Regards, Andreas Brodmann -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071207/d7e2716a/attachment.htm

...hink). But my current setup is based on asterisk, so I'd rather not move it from there or install new apps. Thanks a bunch! Cesc ---------- Forwarded message ---------- From: Cesc Santa <cesc.santa at gmail.com> Date: Fri, Nov 28, 2008 at 3:26 PM Subject: Asterisk RTP pager To: Andreas.Brodmann at gmail.com Hi, I came across your "RTPpage" application and just made me very happy. If I may, some questions. * With which Asterisk versions has it been tested? is it in the official tree? * What I'd like to do is to link this RTPpage with incoming SIP calls ... so that all RT...

There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload" works only once with a bigger dialplan. If I issue "dialplan reload" again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

Hi I have a strange behaviour; perhaps someone who had a similar issue can help. I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager 6.1 cluster. Two phones/users from the Cisco environment call extensions on the Asterisk. Phone 1 / Call 1 is parked on the asterisk using: exten => xyz,1,Answer() exten => xyz,n,Set(PARKEXTENSION=555) exten => xyz,n,Park() Phone

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.