search for: jonr800

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever

...tack - tried the sipura 3000 last night, has > > anyone got this working with faxes. Set all the fax detection tones to > > yes, but it still treats a fax call just like any other voice call :-( > > > > Thanks, > > > > Sam. > > > > Jon Radon <jonr800@gmail.com> wrote on 28/09/2004 16:52:32: > > > > > After you changed the zapata.conf per the wiki... make sure you stop > > > asterisk, unload your zap drivers, reload them, then start asterisk > > > again. > > > > > > ----- Original Mess...

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

...l me.Iam searching the asterisk server site in google but i dint get this server u please tell me the site for me Is only for sip to sip or sip to h.323 please tell me Thank u Bye Sailatha ------------------------------ Message: 14 Date: Tue, 11 Jan 2005 09:27:06 -0500 From: Jon Radon <jonr800@gmail.com> Subject: Re: [Asterisk-Users] Vmail.cgi - "Hrm, can't seem to open /var/spool/asterisk/voicemail .... To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <a7b1310005011106276f2718ae@mail.gmail.com> Content-Ty...