similar to: RealTime and Macro question?

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, 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

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

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, I would like to make a call and then when I am connected to the destination to transfer the call to my coleague in the office. When we receive the call it is easy using "#". But when I am the originator the "#" doesn't work. Can you give me some suggestions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2)

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:

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

> > 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

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

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

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

No Caller ID comes from the FXO line ( The caller id is on and is working with a standard phone) in zapata.conf everything looks fine usecallerid=yes hidecallerid=no When the call comes in there are some warnings in Asterisk Console -- Starting simple switch on 'Zap/4-1' Jun 22 11:20:24 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17 Jun 22 11:20:24 NOTICE[213006]:

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.

I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? How Can be fixed? message : Jul 1 16:18:29 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'damian <sip:damian@10.1.1.11>' failed for '10.1.1.11' Asterisk and Sip phones are all in one

Hi, How can you use Dial application M(x) option from extensions.ael ? (As a reminder, this M(x) executes macro x when Dial called party answers). It seems to me that asterisk keeps looking for this macro in extensions.conf and not in extensions.ael. I tried both (and variations of those with ^ instead of ,) : Dial(Local/${EXTEN:1},,M(mymacro(${EXTEN}));

Thank you Rui for your code. I basically understood all your suggestions. I am using an old version of R (version 3.6.3, installed in a server I am not allowed to control), and the new pipe operator does not work. I tried to run your code without the "|>" operator, but I get an error when I use apply. Could you please expand your code without the pipe operator? Thank you again