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Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . --

Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* --------------

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

...quires asterisk in Debian Wheezy repo, for a Debian setup the alternatives are either : - to install it from source - tto build my own custom package removing this asterisk dependency (is it easy or even possible ?) - to use another solution such as pyst. Regards 2014/1/16 Adolphe Cher-Aime <acheraime at gmail.com> > Yes you can. This what starpy is for. It's build around Python twisted > which allow you to write non blocked socket servers. You can use starpy as > a fastagi server. > Both AMI and FASTAGI can be configured from a .conf file as follow: > > [AMI] > use...

I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how

Hello asterisk gurus, I'm developping a script that create call files dynamicly from a database. Here the scenario script move call file to outgoing dir to place the call call is connected to [extension] which contains a playback app.While line is ringing, playback is triggered I want to start playback file when call is answered to make sure that called

Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Thanks Adolphe Cher-aime From my Iphone

Hello everyone, I'm using redfone fonebridge to have Pstn connectivity on my asterisk box. I can receive in coming calls however outgoing calls don't go to provider. It's seems it's a span config problem. Because in systemconf when I try to config span as follow span=1,0,2,ccs,hdb3,crc4 And start dhadi I have the following message Unable to

Hi Everybody, I'm trying to connect an asterisk box to a provider using Redfone Fonebridge dual E1. Installation seems to run correctly only i can't get the D Channel up and i have the following error displayed. DYN/ SPAN ethmf <mac_address_fo_fbport> Expected seq no 0 , but received 3456 instead . This error keep scrolling until i disconnect the

Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -------------- next part -------------- An HTML attachment was scrubbed... URL: