similar to: Asterisk 1.2.24 simultaneous call limits.

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

Hello list I am having an issue with Chanspy/SIP that I?m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The

Hi, We?re going to build an IVR system with a TE405P and 4 E1. We?re sure that the 120 channels will be filled by 120 simultaneous calls during peak, so we want to have the good server to manage this. We wonder a lot of things and maybe you could help us. - Are you ever build a similar system ? - What hardware is the most adapted for this type of use ? - What is the most important ?

Dear Louis, Just to check... Would it be possible to have more than 2 DC using Unison to sync? I was trying to make this to the samba wiki. But when reading the list I see Rowland talking about the SID and RID issue Because of built-in group SID is not sync across domain. Which I think samba should have their own way of dealing this or it will just be a mess in a long run. Did we have any

Hello Sébastien Le Ray, The PC reply the following... The processing of Group Policy failed. Windows could not resolve the user name. This could be caused by one or more of the following: a) Name Resolution failure on the current domain controller. b) Active Directory Replication Latency (an account created on another domain controller has not replicated to the current domain controller). The

Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser ..... Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy

Hi All, In previous mail lists, people talked about a solution to record large amount of simultaneous calls. And then it seems that RAM disk solution was the best choice due to the I/O bottleneck of Hard disk (System). Please find the previous discussion as follows: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.htm l

Just good old monitor with no mixing onto the scsi drive. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality-critical! Hi, how do

Hi All, I'm conducting a meta-analysis and have taken a data.frame with multiple rows per study (for each effect size) and performed a weighted average of effect size for each study. This results in a reduced # of rows. I am particularly interested in simply reducing the additional variables in the data.frame to the first row of the corresponding id variable. For example:

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

Dear Sébastien, Sorry for the delay, Please check on the log below. As for the word "存取被拒。" it should translate to Access Deny... Please help. - <Event xmlns="*http://schemas.microsoft.com/win/2004/08/events/event <http://schemas.microsoft.com/win/2004/08/events/event>*"> - <System> <Provider Name="*Microsoft-Windows-GroupPolicy*"

Dear R Community, I am attempting to transpose a dataset from rows to columns but am stuck. I have tried using reshape() with little luck, possibly due to the categorical nature of the data. For example: id<-c(1,2,2,3,3,3) author<-c("j","k","k","l","l","l")

Hi, As it has been brought to my attention (by several parties), my current MB does not support VT-d. Since I don''t feel like purchasing a brand new system, I have searched for replacement MB-s that would fit into my existing uATX case, support my C2D CPU, and support at least 4x2GB ram. I have narrowed it down to the following four models: SUPERMICRO MBD-C2SBM-Q-O (Q35 chipset)

Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,