similar to: Call files or AMI originate for mass outbound call

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

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 to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

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

I am trying to compile a dahdi module from checkout: svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.6.1+2.6.1 with ubuntu 3.5.0-17-generic and gcc 4.7.2 Error on compile is: oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47:\ error: 'NULL' undeclared (first use in this function) This is identical to the error reported in this patch fix:

Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!!Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormirandaru -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetwork at gmail.com if you can help. -------------- next part --------------

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

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

Hi I am trying to rsync some ghost images from a windows client running Windows XP to my Linux server. The problem is that rsync sends the complete files again even if nothing changed on the client side. The only way to avoid this is to use the "-c"-option but this takes nearly as long as uploading the files would. The server is running rsync-2.6.9, /etc/rsyncd.conf looks as follows.

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

Hello, I received for X-mas an APC UPS system form my computer. I'm looking for how I can integrate it into the system so that the system will shut down either after the UPS power is low enough or a timed event after the power is out will automatically shutdown. Would also like it to be smart enough to stop the shut down process if power is restored before the shutdown starts. Anyone

I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk

Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

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