search for: wesnarat

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

>recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6 >but I just couldn't complie the app_rxfax and txfax application. The >SpanDSP 0.0.3 was successfully complied though. .3 is for developers only it is not intended for enduser use.

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

Type "swift" at the command line so you can see the -options. Then modify the line to use the correct switch and specify the name of the voice you want to use. Thanks, Steve -----Original Message----- From: Pimjai Wesnarat [mailto:pw@nummerndirekt.de] Sent: Fri 4/21/2006 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swiftapplication?? Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swi...

Hi All, I'm trying to add this function to my find-me application: when all available numbers are dialed in parallel , if any number is busy, take it at busy and go to voice mail. I read the Dial() Application but there's nothing written about this. My question is, is it possible to do this with Asterisk? Thank you, Pim

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like "A new user has joined the conference" and that need not to record user's name. Is there a way to do this?? Pim

hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk and

Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten => 999,1,Dial(Zap/g1/02601591) exten => 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 I am able to

Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My

Hi all, I would like to ask for some help about the queue here. I want to implement a call Queue that when there's no agent logged in, they should execute the next extension. eg. if I do it like this exten => 700,1,Answer exten => 700,2,Queue(TestQueue) exten => 700,3,Playback(noagent) exten => 700,4,Hangup When there's no agent present in TestQueue, it should tell the

Hi all, I am trying to find a way to set the maximum wait time for a user in MeetMe. I am creating a dynamic conference. If there's only 1 user in the conference room, I would like to allow him to wait only until the certain time. eg. may 1 min. If there's no other user joining the conference, I'll hang up his call. Is there any technique to enable this?? Pim

Hi, I've tired to forward a Fax from Asterisk to Hylafax. It works so far until I tried with a Fax machine. I just got error shown in the log below. I'm not sure why. I've tested it with other 6 machines and they all work fine. Do you have any idea why? Pim Hylafax Session log: Apr 12 11:16:48.82: [ 5933]: SESSION BEGIN 000000078 492212601860 Apr 12 11:16:48.82: [ 5933]:

Hi, I am able to use SpanDSP 0.0.2 (all pre version) with Asterisk 1.2.6 to recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6 but I just couldn't complie the app_rxfax and txfax application. The SpanDSP 0.0.3 was successfully complied though. I'm running it on Debian. I've searched the space for the answer, there're so many people asked this but I just

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk