similar to: PCI, PCI-X and PCI-e -- Server / Interface Card Selection

Anyone else taken a look at this? Looks exactly like Asterisk (the random source files I browsed all show Digium copyright) but with autoconf. https://developer.berlios.de/projects/solid-pbx/ From the summary page: Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

This is a follow/find me script that I can't quite get to work, asterisk wont save forward/${calleridnum} to AstDB... any comments or thoughts on how to make this work or change it to work differently are appreciated. The voice prompts to go with all playback/background extensions are commented appropriately. I hope this code is of use to some of you and any help with a perfected

Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas.

I've been working with Asterisk for a little while now, and have been looking recently at my next career opportunity. It seems from searching the various job sites that the predominant VOIP technology is not the applications-based open source approach we took, but Cisco, with a really heavy emphasis on the networking (ie network engineer) aspect. If you do a job search for (VOIP or

Hello, I have installed GlusterFS one month ago, and replication have many issues : First of all, our infrastructure, 2 storage array of 8Tb in replication mode... We have our backups file on this arrays, so 6Tb of datas. I want replicate datas on the second storrage array, so, i use this command : # gluster volume rebalance REP_SVG migrate-data start And gluster start to replicate, in 2 weeks

Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '765e20595817e9897b77cff23f821cc5@10.0.0.254' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '4858cde16223cc0716e325921a8a0654@10.0.0.254' of Request

Hi list! I tried to install Xen 4 in a HP Proliant ML350 G4p, but it seems not compatible, is there another way to run Xen 4 in this server ? I couldn''t find any information on Hp''s site neither Intel''s site ... Thanks ! -- @ntonio _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hi, I have been trying to install CentOS 5.1, 5.2,and 5.3 and the HP server but to no success. It starts loading well the on trying to load the cciss driver, the machine hangs there and refuses to move on. Any help will help quite alot. Thanks

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hi, we''ll start virtualization with Xen in our company and we''re looking for the right distribution. The most important Servers in DomUs are working with a MS OS (Win2000/2003/2008). I''ve read something about to accelerate Windows with paravirtual drivers in some distributions. Other distributions have specific tools to install and support DomUs. And now I''m

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this

Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all. Since half december we are trying to implement * as our primary PBX. We had a test machine running with 2 sip phones and a single ISDN card. Since this was working ok, we installed * on our main Debian server. Some specs: P4 2.8 Ghz HyperThreading 512 MB RAM Debian Sid updated every week. Linux 2.6.9 vanilla kernel (not the debian package) 1 Intel Pro 100 nic for internal network 2

Hi, I'm integrating acts_as_versioned (ar-versioned.rubyforge.org) in my customized CMS rails application to set up revision control for pages. Getting and viewing versions works allright, but when I try to revert to a version in the following way, nothing happens to the actual data in the object. Link from page: <%= link_to image_tag('revert'), :action => 'do_revert',

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with