similar to: Anybody successfully using vISDN on A@H?

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

I forgot to pay this month's phone bill, and never noticed until family (the in-laws, who are too cheap to try the cell phone if landline fails, because it is 'more expensive') told me they were unable to reach us... As it turns out, the phone company disconnected us, but because Asterisk routes all outgoing calls in the Netherlands over VoipBuster, I never noticed anything! ;-) If

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

/etc/conf.d/local.start -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Wednesday, January 11, 2006 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel modules load, but Asterisk fails at startup On Wed, January 11, 2006 21:36,

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]

I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it

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

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 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

I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do

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 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

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, 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) --

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

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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