similar to: Meetme Recording

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

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

How do I use the Authenticate application in my IVR menu, where do I put the password? here is my menu. I need to ask for a password before I let users log into my conference room. [conf1] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Answer exten => s,4,Authenticate(1234) exten => s,5,Hangup exten => a,1,Meetme,1251 I also can not figure out what "Unknown RTP

Hi all: Very disappointed, finally I left the attended call transfer with ATA 186 using SIP. With image 2.16-1, ATA sens '486 - Busy Here' when trying to transfer the call.. I consulted with Cisco guys and accepts that some problems with this service exist. Soon as I can I will try using MGCP. My doubt now is if somebody proved the DynExtenDB application. I read some commentaries but

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

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

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.

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 folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hi !! I know that a conference room can be made infinitely. but, I think that there is actually a limit. For example, how many conference rooms can be made from CPU 866 [MHz] and RAM 256 [MB]? Is there any person who tried someone? I am studying MeetMe now. Please tell me a hint!!

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: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the "old" way? Thanks in advance, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/6dd07c4b/attachment.htm

Here here, a bit of documentation online would cut down considerably the traffic on this list asking how to questions. Cmon sokal, put the questions and training material online. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Monday, December 20, 2004 4:18 PM To: Asterisk Developers

Hello List, Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of 'Monitor' application affect the file name generated through 'automon' feature? I initialized this variable with a value as follows: Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) a. Should I use 'fname_base' in uppercase (FNAME_BASE)?

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

OK let’s start out with this. I’m not a pro GUI designer… ? Now that that’s done. Welcome to OpenConf. At least that what we call it now. To config an * file click on the filename to the left. For my example use extension.conf. Now you’ll have a FULL text editor and a parsed list of all the [sections] in the extensions.conf file on your left. On the right you will find any numbered var’s