similar to: automatic call with call files

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

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 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:

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

Thanks for replying. So as python-starpy requires 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

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

I'be been having problems with the WaitForSilence application in my dialplan. The behavior I've noticed, and hope to demonstrated with the fallowing examples, is that WaitForSilence (WFS for short) does not work until Background is called. Note that the lenghts of the sounds are approximately 5 seconds for bring, and 10 for presses. [Testing] exten => s,1,Answer() exten =>

Guus, Below a segment of a log file. I am trying to analyse why the satellite link goes down and up all the time (sometimes 10 to 15 times an hour, sometimes not for a day. My guess is that this indicates packet loss on the satelllite link and tinc not really recovering from that. Where I inserted an empty line the log basically stopped and was silent for I guess for 14 minutes? Would setting

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The "other end" (gateway) is a

Hi all. I have a customer who does some automatic messaging.? Back when we were running Asterisk 1.4.x, they used waitforsilence after amd, to wait for an answering machine greeting to finish before leaving their message.? We've upgraded to 1.6, and made no other changes and things don't work. What we're seeing it that a call will get hung up, waiting for silence, well after any sane

Sir, Please help me to decide where is the best place/topologi to put TC with the best network performance. My problem is, I want to manage/limited bandwith for C1,C2 & C3 to internet (ONLY their access to internet, but NOT to X, Y, & Z). NOTE: VSAT = Sattelite connection X = Mail Server Y = www server Z = Proxy Server Q = WaveLAN router C1, C2, & C3 = WaveLAN client Any help

Folks, We have a setup where each mobile node connects with 1 or more tinc instances (over different links) to a central node. tinc is running in switch mode. The link is chosen by setting the IP address on the active link's interface, and the central node sees this after the first packet on the link, and moves the MAC address to a different 'ethernet port' (link). This works really

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

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

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have