similar to: Recording Calls at the phone

With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says "so and so is on the phone for you", I say "ok put him through", she hangs up and I am connected to the caller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a

Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not

Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local

Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not

I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

Yep I did the same. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, 21 January 2006 5:34 PM To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Sun 1/22/2006 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion

Hi I have two asterisk servers. I just want to connect two asterisk server using IAX2. But the Asterisk Servers are not able to register each other. If some body have done this then Please send me the configuration they have done in iax.conf and extensions.conf. I simply want to connect and call from one sever to another. Thanks Chandan Kumar Mishra Software Engg. -------------- next part

Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel => 1-23 group => 1 channel => 25-47 group => 2 channel => 1-23,25-47 group => 3 I am just curious if anyone has set some thing like this up and how it worked out. Thanks, Patrick

Hi Someone have luck using Sipdiscount service with IAX ? I only can use sipdiscount IAX service using a free account (only 1 minute call) , I have a normal account and with it can login in the IAX server. I using sip1.sipdiscount.com like IAX server but can make free calls (less 1 minute). Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux

When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear "transfer". I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear

Thanks but this is for a test, I didn't buy the first one as it's a non commercial installation. I'm trying to test bandwidth etc so I need to try out how 4 of them handle the link simultaneously, I just don't know how to add a second test license. Dean ________________________________ From: asterisk-users-bounces@lists.digium.com

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |

>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Message: 16 Date: Wed, 18 Oct 2006 16:10:38 +0100 From: "Neil Tancock" <neil@safeharbourit.co.uk> Subject: [asterisk-users] IAX Terminal To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP!

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then

I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers. If I boot the machine without having the wcfxs module autoload, then install the module with modprobe, asterisk works just fine. If I boot the machine and autoload the wcfxs module, the module loads fine: > Jan 11 11:06:55 asterisk Zapata Telephony Interface Registered on major 196 > Jan 11 11:06:55 asterisk ACPI: PCI