search for: fampeet

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

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

/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, Stephen Bosch...

As I haven't received any posts since yesterday...

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

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 Subject: Re: [Asterisk-Users] Installing the none commercial intel g729codecs into Asterisk@Home 2.2? On Sun, January 22, 2006 13...

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

Its been a few days, I was wondering if anyone else has any ideas on how to get this to work? If not, could I ask a the direct question as to if anyone here has (or knows someone who has) successfully got something like a Cisco Router to successfully establish an ISDN data/Internet connection through the Asterisk PBX?

...> sure what I should be asking them. -->> > -->> > Apologies for this long email. -->> > -->> > Regards -->> > -->> > Lee -->> > -->> > --- On Wed, 14/1/09, Francesco Peeters (linux) -->> > <francesco at fampeeters.com> wrote: -->> > -->> > > Are you using an ISDN cross cable? I don't know -->> > these -->> > > cards, but most cards are wired as a DTE type device -->> > (TE -->> > > port like a router or phone) and not a DCE type d...

...But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line "allow=g729" from sip.conf. And asterisk works fine. 2006/1/22, Guillermo Salas M <gsalas@manta.telconet.net>: > Con fecha 21/1/2006, "Francesco Peeters (Asterisk)" > <francesco@fampeeters.com> escribi?: > > >On Sat, January 21, 2006 23:21, Franz Br?uer said: > >> Hi, > >> > >> MapsAir wrote: > >>> Has anyone successfully Installing the none commercial intel g729 codecs > >>> into Asterisk@Home 2.2? > > I'm...

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

Good Evening, The company I work for is attempting to connect an Cisco ISDN Router to an OpenVOX B200P BRI Card so that we can get it to dial out across an existing ISDN PRI Line also installed in the Asterisk PBX Everything has compiled successfully and the BRI card has been detected when mISDN is started: # misdnportinfo Port 1: NT-mode BRI S/T interface port (for phones) -> Interface

There'll be whiners on all lists, and no list is ever going to be pleasing everybody... (And everyone should decide for themselves whether or not they belong to that group, as I will not be mentioning names! I'm sure there are those who'll think *I* am a whiner...) So at some point just let it be and continue the thread. Keeping on and on about it only increases noise to post

Does anybody know whether somebody ever implemented (linux) drivers or a chan component for the Com-On-Air cards, or is the only way to make it work the use of a Windows box as Dect to SIP gateway for *? (Or does it work at all?) I can still get the cards, which would be great to retire the old DECT based PBX, but it would be of little use if it's not going to work, now is it? ;-) --

I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a different area. 1 of those has two BRI's and a 2 port Nova Compact PABX with DECT First step would be to set up the (*) servers and have them interconnected. When all of that works we'd go on to connect

Is there anybody in this group that is using vISDN on an A@H server? I have a couple of questions, which are quite lengthy, and I do not want to pollute this list of there's no use in asking to begin with! TIA & BRgds -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55

I have been pondering the following... Voipbuster used to announce the cost of the call, but the new SIP servers do NOT. Because there's the choice between free (VoipBuster) and non-free (ADSL), I'd like to let the user know which one is actually being used by announcing it before the actual call gets connected, ie immediately after the channel proceeds from setup to actual

Hi All, Any VoipBuster SIP users on this list that'd be willing to test VoipBuster outbound VoIP to PSTN? All numbers I tried from my (*) server are supposedly being connected, but no phone rings! Also their new WebStart function doesn't cause my phone to ring either... TIA! -- Francesco Peeters

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

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