search for: congratulatori

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the

It appears that my X100P card is only answering after two rings. Ideally, I'd like it to answer on the first ring. Here is the incoming section of my extensions.conf file: [incoming] exten => s,1,Answer exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => 1234,1,Goto(jgunther,1234,1) exten => 4321,1,Goto(mgunther,4321,1) exten =>

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

The extension.conf: exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,5,BackGround,demo-congrats ; Play a congratulatory message When call is coming, asterisk always failed to play the message,

Anyone know of any GUI's that can be used to manage/setup asterisk?

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

I have asterisk configured on a machine connected to Internet by a cable modem with a public ip. The same network card has a private lan address which I'm trying to use to play with an asterisk configuration with X-lite or an softip phone. sip.conf has bindaddr=192.168.0.1 [the private address of the LAN] to which the client [192.168.0.2] is connected. Doing a tcpdump on the register

I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Nihal -----BEGIN PGP PUBLIC KEY

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

I have a problem that is probably so "doh" I will be embarrassed. However, I have spent all evening on this with no success: I have the following setup (asterisk cvshead as of today) 10 Channel EuroISDN<=>Asterisk<=>Meridian What I can do: Call from outside into the asterisk, dial an extension, and pass through to the meridian. WooHoo. What I can't do: Call from

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)