similar to: SIP to SIP monitor and record?

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,

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave

Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, There have been some reports of person(s) checking out the latest version of Asterisk using CVS, and having serious difficulties compiling and installing it. I thought the person(s) who are concerned about having these problems might like to know that I checked out Asterisk Version: Asterisk CVS-09/14/03-08:48:21 compiled and installed it with no problems of any kind. Everything I

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've spent some time on the Wiki adding documentation on all asterisk applications from the cli 'show application xxxx' commands. I've also added some cross references and pointers. http://www.voip-info.org/tiki-index.php?page=Asterisk If you find this useful, please go there and help us build a reference database. A Wiki is a wonderful collaboration and documentation tool -

Hello All, I am a newbie looking to learn about Asterisk. I'm new to IVR and all that goes with it. I would like to know if it is possible to grab the number of an incoming call, have Asterisk, or third party software return the call with an automated voice message allowing the original placer of the call to select another person to call (eg "Select 1 for Bob") then have

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

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

At 07:55 PM 3/11/03, Jean-Marc Valin wrote: > > - Only narrowband (8 kHz) Speex is currently supported; not > > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > > is riddled throughout the Asterisk code.) > >Perhaps it's still possible to send wideband, while telling Asterisk >it's narrowband (the bit-stream is such that you can decode

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

Is it possible in vim to do the following: Search for this block of data: # Catalog Service 2.0 for uat03 <LocationMatch "^/Services/?"> PathPrepend /inquiryservices Cluster 172.21.1.1:999 </LocationMatch> And change Cluster 172.21.1.1:9999 to Cluster 172.21.1.2:7000 It needs to have uat03 (or 02, 01) on the line and this line

New Sun workstation with Solaris 8. installed the gnu compilers and libraries off the Solaris CD and compiled R successfully. I don't have a tex installation or makeinfo (sp?) so the manuals failed to compile, but otherwise R appears to be working. Unfortunately the on-line help (text-only) isn't working. Is this related to the failed manuals? Example: >?help Error in

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin.

Just saw this from eWeek. "IBM, which paid roughly $500,000 for the testing, and SuSE (pronounced "SOOS-ah") were announcing the certification jointly. " The article is here: http://www.eweek.com/article2/0,3959,1212529,00.asp --- Darren Reed <avalon@caligula.anu.edu.au> wrote: > In some mail from twig les, sie said: > > > > I actually just asked

Dear all, i try to plot with ggplot2. Therefor I have an matrix with 3 colums. With cbind I add an additional column called "col". I need this column "col" because in a later step and want to specify here some plot details which I will get from another analysis If I want to plot with this code, I have the problem that the legend is wrong. Blue changed to green and green to

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen