similar to: Test to see if I'm still on list...

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

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

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

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

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?

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

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

I've tried to find some details on the wiki, but was unable to get a satisfactory result, so I am asking here: I have a Linux (FC3) box with these specs: vendor_id : AuthenticAMD cpu family : 6 model : 3 model name : AMD Duron(tm) Processor stepping : 1 cpu MHz : 797.388 cache size : 64 KB MEM: currently 256, looking to upgrade to 512/768

Hi all, I am struggling with the following and I can't get it work: In the Netherlands where I live it is possible to use special codes (aka vertical service codes) to set special 'behaviour' of phonecalls. So e.g. when I want to dial out with a normal phone and I dial *31*<phonenumber to dial> then it will turn off my numberindication (CID) at the called party. They seem to

Hi All, The wiki seems to be struggling this evening. Anyone else seeing this? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* 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

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]

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 have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

"Unable to login to the MySQL database 'tiki' on 'localhost' as user 'tiki' Go here to begin the installation process, if you haven't done so already. Lost connection to MySQL server during query" Hi, Everyone: What is the URL of the "list archive"? The 'wiki' seems broken. I want to connect to a single home POTS from an internet

Sorry if this not the right place to post this.... BUT... Since May 31st, ALL of these user list messages have been filtered by "spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as spam. - Have no clue why this is happening. Luckily, "spamassassin" sent the messages to the "probably-spam" folder on the Linux box & I was able to

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

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? ;-) --

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit