similar to: How to transfer call in case that I am the originator

Hi, I would like to find some way for hardware coding instead software (using the Host CPU). Are there any PCI boards just with codecs (DSP) or other way? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: m_natchev@yahoo.com miro@space-comm.com http://www.space-comm.com

Hi, I would like to ask you for advice how to solve the following case: I have a client (who happened to be my friend) and I have convinced him that the IP PBX solution is much better than the conventional telephone centrals (PBX). At the beginning he wanted to buy PBX Panasonic, but at this moment he is waiting for my decision. Because at the moment we are not so deeply familiar with these

Hi, We have potential client which would like to offer to him VoIP solution for 2000 subscribers (SIP based Phones) and 2 x PRI ISDN interfaces to the PSTN. In the next stage the subscribers will be increased up to 13,000. Because I am not haven't done similar big project I don't know how to calculate the price. The one way is using number of subscribers and the other is using PSTN

Hi, I can't find anywhere on the Asterisk web the license terms for commercial use of Asterisk software. Do I have to pay something (and how much) if I want to use the Asterisk in our IP PBX solutions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: m_natchev@yahoo.com

Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in

Hi, Is it possible to connect more than one Asterisk in cluster to works as one Asterisk IP PBX? I need of this for cases where I need of more FXO/FXS ports which can't be placed in one machine (server) and in the same time all Asterisk have to works as one with one dial plan and etc. -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile:

Email : miro@space-comm.com FirstName : Miroslav LastName : Nachev Company : COSMOS Software Enterprises, Ltd. Phone : (+359-88) 897-31-95 Fax : Address : P. O. Box 941 Address2 : City : Sofia State : Outside the US, Mexico, or Canada Zip/Postal

Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro("SIP/1007-2165", "dialnumber_wvm,1004,SIP/1004") Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

Hi, Are there any optimized codecs for Analog Blackfin DSP? If yes, from where we can download it? We are looking for Speech, Audio and Video codecs. Best Regards, Miroslav Nachev

Hi, I have a question: do I have to worry if I run with wine an executable that may contain a virus (I am running linux as a normal user, not root)? -- Peter Kostov, webdesigner, photographer Sofia, Bulgaria Home sites - www.webdesign.light-bg.com - www.light-bg.com

Hi, We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To do this using Digium products I need of 8 PCI slots. This is not possible to be done in one computer and that's why I try to start using TDMoE. Unfortunately all my tries are without success. The network is crashed everytime. Can you give me some ideas/suggestions how to solve this case? Best Regards,

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Hi, Thank you. I will try it. Do you know some G.72x, GSM, and iLBC optimized for Blackfin ? I mean open source. -- Best regards, Miroslav mailto:miro@space-comm.com Wednesday, March 23, 2005, 9:05:11 PM, you wrote: JMV> Hi, JMV> As far as I understand, the last patch (for TI C5x) I merged in SVN also JMV> allows Blackfin to work, but I haven't

> > On 2022-11-08 15:49, Orion Poplawski wrote: > > On 11/8/22 13:12, Simon Matter wrote: > > Is anyone else experiencing trouble with > kernel-3.10.0-1160.80.1.el7.x86_64? > > I'm seeing a kernel panics in the kvm module on one of our VM hosts with > > it. > > I did notice a new libvirt

Change it to GRUB_DEFAULT=0 (I encountered the same issue week ago with a workstation booted for three month with an older kernel because of https://bugzilla.redhat.com/show_bug.cgi?id=2143438 , and solved it this way) Regards, Petko On 3/14/23 10:51, Rob Kampen wrote: > Can I edit /etc/default/grub and change > > GRUB_DEFAULT=saved > > to something else? -- Petko Alov

I've reading this lists few months. We are small company, that makes some system intregration, development and deployments in VoIP scene. Completely under linux. Today i have 6 machines with asterisk, huge test base - including devices like AS5350, Audiocodes gateways, ATAs, IP phones ... Now is time to make a decition for including * in our future projects. Main goal for us is the Stability.

Dear Jean, The source code for G.729 can be download from ITU for free. Also, some developer can do yourself as open source G.729 codec without any help. In this case each who use this codec which source code is free and open source must pay, but not to the developer. Best Regards, Miroslav Nachev JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit : >>

Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark