search for: naija

I''ve tried moving the code around in different ways, but so far I haven''t gotten remote_function to work for me (I''m using Rails 2.3.2 on my Mac). I want to make an AJAX call when I double-click a word on my page, and the Javascript function is being called when the ondblclick event is fired, but nothing happens when it gets to my the "new Ajax.Request" part

What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. -Chris Sent from my BlackBerry? wireless handheld

...Re: Load balance Asterisk servers? (Aaron Daniel) > 17. Re: "Username/auth name mismatch" + SIP phone can't connect? > (Fred) > 18. Re: DUNDi Asterisk Cluster (Aaron Daniel) > 19. How do I change the rtp packet size in a Cisco 7490 from 10ms > to 20ms (Naija Man) > 20. Re: Problem with FXS ports of TDM400P (Gustavo Felisberto) > 21. Voice mail transfer between 2 asterisk servers (Naija Man) > 22. RE: Problem with FXS ports of TDM400P (Robert Jenkins) > 23. RE: In the beginning-The first question. (James R. Stevens) > 24. Re: In...

...n using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki (Naija Man) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070107/bb338628/attachment.htm

We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

> > ---------- Forwarded message ---------- > From: Doug Lytle <support@drdos.info> > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Date: Tue, 10 Oct 2006 16:25:11 -0400 > Subject: Re: [asterisk-users] How big is *your* dialplan?? > Steve Murphy wrote: > > Hello! > > > > In my relentless

> > ---------- Forwarded message ---------- > From: "Alexander Lopez" <Alex.Lopez@OpSys.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > Date: Sat, 14 Oct 2006 13:04:08 -0400 > Subject: RE: [asterisk-users] Re: Generate Random Numbers in dialplan > Use the AGI I sent. It looks like