similar to: H323 - SIP Interoperability

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

Can anyone please help, I can't tell why it will not connect. I do not know how to read this debug file to were it is wrong. Thanks Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:9082 From: <sip:2203@192.168.1.101>;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: <sip:2203@192.168.1.101> Call-ID: ea352d6f-a879-4db6-a361-365487a20d4a@192.168.1.100 CSeq: 1

Hi friends, I have the entry exten => h,Hangup in my extensions.conf, and I am trying to record the call details for billing. From the wiki i found out that the use of "exten=>h,..." is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=>h? Regards... Girish

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

Hello, >From: "Hermann Wecke" <hermann@wecke.com> >Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern => >extensions.conf >Date: 8 May 2004 22:03:57 +0000 > >Is it possible to strip some numbers from the *end* of a number? > >I know that ${EXTEN:1} will remove 1 position from the beggining... but >how to remove N numbers from

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

Hi, I have received some replies for my previous mail (* configuration), asking for my goals in configuring Asterisk. So here they are: We are planning to host an Inter continental virtual PBX service that will enable our users to register for an account which give them a toll-free # or a DID. Once registered using a web based interface, that user can add as manay extensions he/she wants

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

I ma new to Asterisk. I'd like to setup * as voicemail system for SER. Let's say I have an phone number registered in ser as 5554321. When somebody dial to ser for this number and nobody answer, the ser will forward the call to asterisk and get into voicemail box 5554321. I already have asterisk up and running with mysql setup for asterisk voicemail. Can somebody show me how to do it? Or

I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening: localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtest May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hi. I'm trying to start a MGCP configuration in Asterisk but i have some basic problems. I hope that someone can help me. First ..how do set two call agents in the configuration files? How is the extensions.conf for MGCP?! I'm trying to start the Asterisk, and obtain this: [root@server3 asterisk]# ./asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Not found (No such

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

Buenas, Alguno usa alguno de los paquetes de Microsoft R para la conexion a SQL Server? De ser asi, que paquete y que comandos usais? Yo hasta ahora he usado odbc, de Rstudio, pero me da siempre problemas con el tipo de datos que tiene SQLServer ... Un saludo Jes?s [[alternative HTML version deleted]]

Hi there.... I'd like to know how to use WINE with applications developed in Windows that use net components (such as ODBC clients)......For example: I developed an application in VB6, and it uses an Oracle ODBC conexion (previously I installed the Oracle client). Hot do I use it in Wine? - Do I need to emulate the Oracle ODBC Client for Windows? - If I got a Native Linux ODBC conexion,

Buenas, Tengo un problema y es que intentando conectarme a una base de datos SQL Server, tras cosneguirme conectarme usando el paquete odbc, me deja acceder al contenido de algunas tablas (mediante un select), pero sin embargo en otras me pone lo siguiente: Error in new_result(connection en ptr, statement) : std::bad_alloc No entiendo muy bien porque me salta ese error, ya que desde SQL si