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...resh every 3 to 4 seconds, it does actually work. The only loss of course would be if someone hung up on first ring - the browser might not catch it in time. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rusty Shackleford Sent: Wednesday, May 25, 2005 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists...

OS: RHE 3, kernal 2.4.21-32.0.1EL Samba version: samba-3.0.20 Removed RHE version of samba and compiled/installed samba-3.0.20. After running about 30-60 minutes, I get the following syslog msg: "winbindd: Exceeding 200 client connections, no idle connection found" Eventually this grinds the whole system to a halt, and remote & console logins become impossible while samba is

...Unfortunately Trac seems to be pretty much down at this time so no ''New Features"" goodies in this post. Rails Core Weekly July 2 - July 16 ================================== [ thread: http://www.ruby-forum.com/topic/69703#new ] Somehow in the last RCW I missed Johm-Mason P. Shackleford proposal to: 1. allow Rails to load a plugin from gem 2. permit the plugin script to install them 3. give the plugin generator''s default Rake file the ability to package and publish a gem 4. introduce freeze/unfreeze for plugins Jeremy Kemper agrees gems are a good fit. Exciting stuff....

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050609/9eacbae2/attachment.htm

Hello - I have a non-PRI T1 setup and have been making outgoing calls for several months no problem. I have Zapata.conf setup for fxs_ks on these channels. How do I take incoming calls on these same channels? Do I need to change the signaling? Any help is appreciated. Thank you, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

Hello, sorry if these questions have been answered before but please excuse me, I''m new to xVM :) Anyway I have been using Linux/Xen for quite a while and am very familiar with it. I am very interesting in changing to OpenSolaris/xVM for the primary reason of being able to easily leverage the ZFS filesystem and thin provisioning for better storage efficiency. 1) Is this ZFS thin