similar to: Asterisk server

Hello, all I apologize for all these questions. But I have so many running through my head for this. What brand of server is a good one to use for running an asterisk box? I did an install the other day on a Compaq Proliant ML370 G2 / Debian Sarge and it is currently working great. But HP/Compaq costs bukooo bucks. I have heard Super Micro is what alot of you are using. Any problem running

I have a table called Post (not the actual name) where users input various piece of information, Most of the input is done by dynamic select''s (drop downs). As an example: User would choose Field: Input: Values that go into Post table Category Full Time 2 Length 45 Days 4 State

Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

Hi there ! I try to figure out how to do something like this in Rails : SELECT * FROM my_table WHERE name LIKE ''GOGONAM%'' My problem is the % and the quotes. >From now, I do like this : searched_name = params[:name]+''%'' user = MyTable.find(:all, :conditions => ["name LIKE ?", searched_name]) But is there a more gracefull way to do a LIKE

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

Hi, Few days ago I installed Email2fax application on my Asterisk box. I works but not in 100 %. Sometimes (to be certain quite often) I don't receive the whole fax. My fax machine cuts off transmission in 1/2 or 1/3 of the page. I read about it on a wiki and some user lists and people say that this behaviour could be cause be the Ghostscript and the conversion to the tiff format, but when I

Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX at Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below).

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]XXXXX -- _06[069]XXXXXXX that are in the database and number 0699123123 comes in asterisk will always choose exten

Hi I am using asterisk-1.6.0.6 and I have noticed strange behaviour lately. When a user ends his call asterisk executes twice the h extensions (in my case this is the AGI script) and writes this to the logs: cdr.c: CDR on channel 'SIP/xxxxxx-b6623038' already posted. and after that it crashes immediately. This had happened twice so far. Does anyone know what is causing this.? Cheers

Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes with IAX. Is it a good solution? Does anyone have other proposals? Cheers Andrew

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary.

Hi, I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this: --NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 --NOTICE [1197]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 And when I want to call a ZAP channel I get

Hi, I have small issue with Asterisk. My customers complaining that sometimes (not always) the outgoing voice (the voice which can be heard by the user a the other end) quality is very low (stutter and sudden clicks). The problem exist in only-IP configuration and in IPtoTDM connections as well. I use alaw codecs. I know that they consume a lot of bandwidth, but the upload and download stream is

Hi I am trying to build reliable fax solution with asterisk, iaxmodem and hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3 1.2GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After installing the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give me good results: 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

Hi I am having problems with Asterisk 1.4.18 and realtime architecture. I use Postgresql-8.3 as the database. Everything works OK; all sip phones (their configs are in the database) are able to register to the server and I can make calls (dialplan is in the database), but each time Asterisk reads the information from the database it shows me this on the console: [Feb 17 12:32:50] WARNING[620]:

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would