similar to: Re: how to fetch a call? (Tony Mountifield)

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

On Tuesday 03 August 2004 12:07, Chris Shaw wrote: >Are you using the double ## transfer patch or just the regular >single # that comes with CVS? Hi Chris, Where to get the 'double ##' transfer patch? We are having the same problem and I was thinking of a similar patch. Thanks, -- sudhir

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->

Dear all, Can any body help m on this to resolve my issue permanently. I m almost done, but one bug is creating problem & unable to resolve it as per mail reply from one of our colgn niranjan.ashok. Following command is not working on my system ie $ svn co http://svn.easysw.com/public/cups/branches/branch-1.3/. Its showing error as " could not resolve hostname, host not found".

Hi team, Anybody have idea on below issue. Pls suggest.. Regards Amit Sudhir Anjarlekar Asst. Systems Engr. Tata Consultancy Services Mailto: amit.anjarlekar@tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________

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Hi, I have created a heatmap using heatmap.2 having 7 clusters. I would like to extract the list of genes that are in these 7 clusters. Is there any function that can be used to extract genes for each cluster? Cheers, Sudhir -- __________________________________________________________ SAVE PAPER - Please do not print this e-mail unless absolutely necessary Being happy doesn't mean

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

There's been a lot of discussion in the past on these somewhat-mysterious "bit already cleared" messages appearing in the logs when using Ext3 file systems. Unfortunately, I didn't see a conclusion to any of these threads. We're still seeing these messages pop up occasionally and fsck's of the file system reveal lot of orphaned inodes and such. Anybody else see these

I have configured my GS HT-486 for "send dtmf" in audio, and on the asterisk box, sip.conf has dtmfmode set to inband. Everything seems to be working fine, however, I see my console get flooded with the following warning: "dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 256 frames" Should I be cautious about them or just ignore? Better still, what should I do

One of our clients has Asterisk with single T1 card (T100P). T100P gets only incoming calls. Is there a way to block specific channels (say channel 9 and 24) so that the Telco does not send calls on those channels? Thanks, -- sudhir

I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement "please check the number and dial again" although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten =>

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

Hi, I wish to know if GNUGk can work with * running as a gateway with the Digium FXO card. Kindly share your experiences in case there are some issues which one must know before going in for such a setup. Also, I've been reading about the DialTone detection capability by the hardware in different countries. What are the issues with it? Thanks & Regards, Deepak ----- Original Message

Hi, package tiff-v3.5.7 contains the currently recommended version of libtiff in order to run spandsp (fax support for asterisk). Imho tiff-v3.5.7 is not very easy to find in the internet, and maybe will almost disappear, because it is an "old" version, I put it on our little asterisk download page. Maybe it'll help someone. It works fine together with the other asterisk stuff

> > Message: 2 > Subject: RE: [Asterisk-Users] Astricon costs... > From: Steven Critchfield <critch@basesys.com> > To: asterisk-users@lists.digium.com > Date: Wed, 21 Jul 2004 21:30:26 -0500 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-07-21 at 19:34, Steven Sokol wrote: > > > Has anyone really looked at the costs for Astricon. But the

Hi Mark, Are you still having audio problems between outside SIP channels? Make sure that you have set the following for all SIP channels in your sip.conf canreinvite=no -- sudhir > Message: 2 > Date: Mon, 26 Jul 2004 22:46:22 -0400 > From: Leif Madsen <leif.madsen@gmail.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Nat...again.... >