similar to: Help with Weird Setup

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can Asterisk(*) be installed on this and if so are the setup instructions any different? I have a client that's asking. Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS - -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, What are some common reasons why a phone would report not registered even when the extension has been setup through Asterisk(*) AND phone username/password is correct? Joshua - -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Here's my public PGP Key - -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 jabbott@SuccessfulHosting.com ================================= The Success behind your web site!

Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/ Feedback? Is there something as good as it or better ? Recommendations? -- ================================= Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 jabbott@SuccessfulHosting.com

I have been trying to optimise my ADSL connections for VOIP. Funny things were happening - for example increasing the ping packet size by 50% had no effect, but then adding one byte had a major effect. It took me a while to figure out that I was seeing the effects of the fixed ATM cell size. This is probably obvious to some of you. For the rest: ADSL uses ATM as its transport. An ATM

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/attachment.htm

Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/ac580fa1/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Currently we have a server setup for Asterisk(*) and a TFTP server. My extension has been setup with Asterisk(*) and downloads the information from the TFTP server correctly (I've tried to erase my phone using *468 to see if that will do the trick). I have a Polycom SoundPoint 600 phone. Attached in zip format are the .cfg and .log files

I can''t decide between a2hosting.com, successfulhosting.com, site5.com (which offers 11gb of disk space for $8.77/month -- I''ll never use it), planetargon.com (which offers 1 gb for $11/month), or railsplayground.com (which offers a gigabyte practically for free). Any experiences you''d like to share? Only a2 and planetargon list their gems on their web sites. All of

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

Here''s one I haven''t seen before I have a model "section.rb" as follows: class Section < ActiveRecord::Base ###Lot of common stuff end class Normal < Section end class Faq < Section has_many :faq_categories, :order => :position end class Rate < Section has_many :rate_tables end Now I''ve just added the Rates class and it works just

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

I am planning to sign up for a VoIP service in the U.S. Can anyone recommend anything cheap, reliable and good quality? I want to use it for my primary house phone (I also own a cell phone). I also want the service to be asterisk friendly so I can play with it :-) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Well, I have been busy getting child #4 off on her way to college (joining #2). So now it is quite with just the 2 youngest and I can get work done again!. tar stops as follows: /root/.nautilus/metafiles/x-nautilus-desktop:%2F%2F52F.xml tar: Error exit delayed from previous errors. I thought perhaps this is some file locked becuase I was running from a terminal session in gnome. So I

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I am looking at installing logcheck on a openbsd box. It doesn't have lockfile-progs or lockfile.h. Am I going to have too much of an uphill battle to get this going? --ja --

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use

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Hi I currently have a 3072kbps line that I'm splitting in half for 5 of my phones. That's 307.2kbps +/- a couple of kpbs. What is the minimum kbps for a phone to maintain clarity and volume? Joshua