similar to: Registrations, how many is too many?

Hello, we are setting up another system that will run either 1.2.4, the latest version of 1.2 or 1.4. We have not yet decided on the version. Anyhow, this is a higher volume system (dual processor) which will handle 30-50 simultaneous calls with 60 to 100 simultaneous channels lit up. Most calls are g711 with very little g729 and a little gsm mixed in. We have a similar system doing exactly

Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in

Why are there separate registrations for every tab in the Wine HQ? Shouldn't there only be ONE registration for HQ, AppDB, Bugzilla and Forums? Having separate registrations for different parts of a site is just silly and definitely not user friendly.

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

We are looking to be able to put a device in front of an array of Asterisk systems which would do the job of load balancing them. We would store all the particulars on one or more MySQL servers. What want to accomplish is to have all calls sent to/from a single IP, then push the calls off to another Asterisk server in the array. If one server goes out, we are hoping there will be no effect other

So a few people just reported that they couldn't make any calls. I logged into asterisk and at first everything on the console looked normal, then I got swamped with messages about too many open files. This is from my asterisk/messages log file: [Oct 2 16:46:00] WARNING[19429] rtp.c: Unable to allocate RTCP socket: Too many open files [Oct 2 16:46:00] WARNING[19429] udptl.c: Unable to

Hi, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, ------------------------------------- Abid

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx

I have a decent sized asterisk 1.4 installation of about 5000 registered users spread across various servers. I have noticed that when keeping the SIP users in the Realtime database, the Asterisk servers become completely unstable at around 600 - 800 users. MySQL is running on a different set of servers, but the bottleneck is definately the number of registered users. We can buy some room by