similar to: exten=>h and ResetCDR

Hi there, I would like to communicate H323 IP phones with SIP phones. My H323 phones are registered to a gnugk GK, and the SIP phones are registered to a asterisk SIP proxy. I could not create a dialplan that works. Inside my extensions.conf file I created the following two entrances: exten => 4,1,Dial(SIP/4) exten => 5,1,Dial(SIP/5) This allows SIP phones call each other.

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 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

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

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

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

Hello - I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with recording CDRs using MySQL. Unlike all of the other postings and web pages I have found on this issue, my installation successfully stores the -first- CDR, but nothing after that. As background info, I will note that I don't use CDRs for billing, but more in a logging fashion, to record how a given call

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hello, Asterisk party, If block the call before dialing (Hangup()), CDR's don't write to MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write normally. Here is the dialplan: ; we skipped dial, because the number is "blocked" exten => _X.,n(Finish),Hangup() exten => h,1,NoOP("hangup") exten => h,2,ResetCDR(w) exten => h,n,NoCDR() exten =>

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

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

hi.. nope I do know about x100p being the fxo and the usb device being the fxs ....fxo using fxs signalling and fxs using fxo signalling and it is reflected suitably in my zapata.conf and zaptel.conf... The problem is that the S100U usb device is not identified upon zap show channels ... only the 100xp wildcard is recognized....i get a sound when pressing the keypad of the phone connected to

hi, I got a digit networks x100p card and instaled asterisk. everything went fine and upon calling the phone asterisk issues a notification. Now i plan to turn it into an ivr and modified extensions.conf to first record some messages , problem is 1-)I am not able to understand how extensions refer to in my case ( a single analog phone line plugged into line jack and a phone into the phone

I have the standard parking.conf but extension 700 doesn't show up in my dialplan.... Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise.... Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos => 701-705

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,

Hi All My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n,