similar to: Best $300 VoIP phone for asterisk?

Dear Friends Greetings, i wish to setup SPF look up f0r Postfix on CentOS 6.2. I could not find any document on the net describing the procedure. is anyone have documentation i can follow to setup the SPF please? Thanks / Regards Prabhpal S. Mavi

Hello all asterisk users! Question: Does anybody know about any good USB handset that would understand SIP and Asterisk and will run with Linux? I have found tons of them, but they are mainly only supported in Windows environment. I would like to set up new phone system in our company that would be based on asterisk acting as PBX and SIP. If you have any suggestions, please let me know. Any

Hey all, I was playing w/ Thomas Fuch''s Localization module (thanks Thomas!), and just realized how hard is to localize an application in Rails :-) The Localization module works nice, but the problem is the error messages. Its not that good to keep the application in [your idiom] and the error messages in english. I saw the other threads about it, specially the ones that recommends

hey all, So, we''ve been trying to get a nice stable config running for deployment with SwitchTower, spinner spawner and reaper, and I have to say, they are, as it says in the title, teh suxx0r. <warning>I''m annoyed, and wrote this while annoyed</warning> ;) The problems are rife: 1. Because lighttpd has no control over the fcgis, the restart process is

Hello, i'm tring to use a cox's model for a survival analysis. I have a dataset, this is a part: VOD SESSO fonte_sct donor RT_CGY STATOBMT TEMPO morto 1 0 F mid related 1200 CP 65 1 2 0 M mid 1200 2RC 5281 0 3 0 M mid unrelated 1200

hm ., as i was typing this msg, i got a reply fr0m the forums . ogg is now supported in winamp3 ?! i hadn't realized ., great great , but sorry for the added posts ., a:/, On Mon, 3 Mar 2003, Adon Irani wrote: > > i was just lo0king at the winamp3 forums . there's a wish list , and i > was the very first pers0n to request OGG ! > > but there's h0pe .. > > --

i was just lo0king at the winamp3 forums . there's a wish list , and i was the very first pers0n to request OGG ! but there's h0pe .. -- We're Counting We are now counting individual requests so see what should be high on the priority list. So, if you have a feature you want, add it here. Better yet though, use the Search feature at the bottom. If you see a thread that matches

Heya folks First of all, sorry if this isn't the correct list, but yet, I think spam is a kind of network attack and should be treated as a security issue.. I run a working mail server using Postfix, MySQL, Courier-IMAP, SpamAssassin and ClamAV (amavisd-new) .. I've checked the configuration file for SpamAssassin, but yet I havn't find any good solution for spam.. Sure, spam will

Now that we can gather diagnostic info, I think I know why our P2Vs kept failing last week. Another one just died right in front of my eyes. I think either the Ethernet or NFS server at this site occasionally "blips" offline when it gets busy and that messes up P2V migrations. The RHEV export domain is an NFS share offered by an old Storagetek NAS, connected over a 10/100 Ethernet.

We have some x86_64 systems that have DisplayPort and USB ports for a monitor & keyboard and when I use these to PXE boot, everything works fine. However, these systems will eventually become headless and in the production process access to the console will be via serial port. Using the serial port mostly works in that once we get to the PXE boot: prompt we can type in the name of the

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

Can someone give me some direction on how to bridge two calls on an asterisk server. I'm originating two calls using asterisk java manager and after some processing in each of the individual dialplans, I want to connect the calls together. Jason Wolfe jasonwolfe@bellsouth.net c (770) 561-6956 This e-mail transmission may contain information that is proprietary, privileged and/or

Hello, If I use an IAX termination service to connect outgoing VoIP calls to a PSTN will I have answer supervision so that my script won't initiate too early? Jason Wolfe jasonwolfe@bellsouth.net c (770) 561-6956 This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use,

Hello, One of our client company is providing hosted contact center solutions with Cisco IPCC. To keep the Call Recording cost at low, they are planning to use Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for call recording ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email:

Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan

Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments?

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

Hi, I am trying to read data from Microsoft SQL. I tried to use odbcConnect (in RODBC) but failed. What should the "dsn" be? Does anyone know how to do it? I will appreciate it. Thank you! Xiaorong ________________________________________________________________________ The preceding e-mail message (including any attachments) contains information that may be confidential, be