search for: ishmael

I finally got my Asterisk all setup and everything seems to be working except for menu interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the SIP phone setup to properly connect when pressing the 'Message' button and that's working perfectly. When the menu starts, it says press 1 to read your messages, but pressing 1 (or any number) fails to send. Does anyone

Dear All, For those who celebrate Eid. I would like to wish you a very Happy Eid Mubarak. For those who do not know what it is, Its a Prayer in memory of Ishmael son of Abraham, when he attempted to sacrifice his beloved son on request by god. Muslim's celebrate it with a sacrifice of a goat every year. I belive Christians & Jew's belive in the same. Peace and Harmony to all. Rehan

Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo "talking". I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001

Hi All, Can anyone please assist with this issue I am facing. There is continuous restart of libvirt, this causes virt-manager to disconnect. syslogs: systemd[1]: libvirt-bin.service: Start operation timed out. Terminating. systemd[1]: Failed to start Virtualization daemon. systemd[1]: libvirt-bin.service: Unit entered failed state. systemd[1]: libvirt-bin.service: Failed with result

G'Day List, Can someone help me out a bit please. I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying to install * I am following http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation After running: cd /usr/src/asterisk make clean make make install make samples The instructions says: Configuring Asterisk - Login to your server as user

On Tue, Aug 02, 2016 at 01:48:55PM +0200, Ishmael Tsoaela wrote: >Hi All, > >Can anyone please assist with this issue I am facing. > >There is continuous restart of libvirt, this causes virt-manager to >disconnect. > >syslogs: > > systemd[1]: libvirt-bin.service: Start operation timed out. Terminating. > systemd[1]...

On Thu, Aug 18, 2016 at 11:51:54AM +0200, Ishmael Tsoaela wrote: > Hi All, > > I am creating snapshot and whilest running blockcommit I am coming acrross > this error: > > virsh blockcommit Node-A vda --verbose --pivot > error: unsupported configuration: active commit not supported with this > QEMU binary This must be...

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

...lt;BR><BR>----- Original Message ----- <BR>> Message: 1 <BR>> Date: Tue, 11 Jan 2005 10:41:42 -0500 <BR>> From: "Ferguson, Michael" <FERGUSON@BRVMLAW.COM><BR>> Subject: RE: [Asterisk-Users] Installing * on fedora 3 <BR>> To: <DISHMAEL@WINDWARDCG.COM>, "Asterisk Users Mailing List - <BR>> Non-Commercial Discussion" <ASTERISK-USERS@LISTS.DIGIUM.COM><BR>> Message-ID: <94E1B25A56D558418D5496ACFC39ACEA037E73@10moses.BWR.COM> <BR>> Content-Type: text/plain; charset="us-ascii&quo...

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Hello, When I listen to a voicemail message, the recorded message is played back at extremely low volume. All the supporting prompts are at the correct volume, it's just the incoming recorded message that is played back almost inaudibly quiet. There's no problem with the volume during normal converstaions so I'm thinking this must be specific to the Voicemail application. I've

Greetings! My little app collects data from the user, persists it in about a dozen tables during the session, writes out / saves an XML file that''s downloaded to the user''s PC when they''re done, then clears the db of all data entered, and deletes the XML file from the server. Now I need to allow the user to upload the same XML file they created earlier and put it back

I'm sure this has been asked more times than anyone cares to count, but I want to make sure I get the right stuff. I'm installing Asterisk at home solely to play with and learn VoIP (plus it sounds pretty cool so it has that going for it too). Eventually all my phones (~4 of them) will go through my Asterisk system. I've ordered my server hardware, here's what I've got:

I'm about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? Thanks, -Dave -------------- next part

I have several contacts that use Vonage and was wondering how I can peer with Vonage (assuming that's possible) so that I can contact these people through the * rather than PSTN. Can that be done? What about other providers (Skype, etc)? Is there something on the Wiki that discusses this? Thanks, Dave

I have been trying to evaluate GoDaddy hosting for RoR. Quite frustrating and painful to date. I have setup a database and uploaded a test app (depot from AEDWR2). Best results to date - ''Internal Server Error''. The log reports ''mod_rewrite: maximum number of internal redirects reached''. ''RewriteOptions MaxRedirects=20'' doesn''t help. I