similar to: Streaming channels from Asterisk to the Internet

I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet {~~~~~~~~~} +---------------+ +-----------+ +---------------+ +---------------+ {

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now incorporated my recent patches to support dynamic RTP payload types. As a consequence, its SIP implementation now supports Speex, so if you have a Speex-compatible SIP client, you can use it to make calls using Asterisk. Some caveats: - Only narrowband (8 kHz) Speex is currently supported; not wideband. (Unfortunately,

Hello All, Is it possible to monitor and record a SIP to SIP call? If so, how? I gathered from some previous posts this would not be possible. -- Thanks, Tim

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with "Ttm" flags, when I press # on the SIP phone, the pstn caller hears the "Transfer" and the SIP phone gets the music on

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

HI I want to write my own SIP client (compatible with ASterisk) is there any good API available for this purpose . any help in this regard would be very helpful 4 me Obee _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail

I try to use an ebook reader with wine! It needs to activate itself, so I can open it. When I try to to so I get the error massage from the eBook reader application, that i would be not connected to the internet... witch is silly because I play an MMORPG game with Wine! What could be the problem? Any ideas?

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

I am looking for a function (or package) to plot histograms of directional data such as wind direction. I believe these are called rose diagrams. Is there an R script for this? If not, can it be constructed in a function calling primitive graphic calls (lines, circles, boxes or polygons)? The stars function is not quite right. -- David Finlayson Geomorphogist and GIS Specialist NearPRISM -

I am using the PAM algorithm in the CLUSTER library. When I allow PAM to seed the medoids using the default __build__ algorithm things work well: > pam(stats.table, metric="euclidean", stand=TRUE, k=5) But I have some clusters from a Hierarchical analysis that I would like to use as seeds for the PAM algorithm. I can't figure what the mediod argument wants. When I put in the

I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet {~~~~~~~~~} +--------+ +------+ +---------+ +---------+ {Internet } |Asterisk|--line

I am tangled with a syntax question. I want to calculate basic statistics for a large dataset provided in weights and values and I can't figure out an elegant way to expand the data. For example here are the counts: > counts n4 n3 n2 n1 p0 p1 p2 p3 p4 1 0 0 0 1 1 3 16 55 24 2 0 0 0 0 2 8 28 47 15 3 1 17 17 13 4 5 12 24 8 ... and the values: > values

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Hi There, I have just ported to Linux a program that I wrote years ago in Prolog under DOS to automate our Shortwave radio station. Back in 1990 the original program used ADPCM coded audio files and a special board. Now I can use MP3 files and a Soundblaster compatible card ;-) under Linux, and it runs from a console without any fancy interface. I absolutely need to fade-in and fade-out

Hi all F18 merging has finished so commit access should be back to normal. Thanks Rich > -----Original Message----- > From: llvm-dev <llvm-dev-bounces at lists.llvm.org> On Behalf Of Richard > Barton via llvm-dev > Sent: 9 April, 2020 16:08 > To: llvm-dev at lists.llvm.org > Subject: [llvm-dev] F18 upstreaming Now! > > Hi all > > We are about to merge F18