similar to: asterisk supports VXML?

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string with path in asterisk server, are there any way to do this? --

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in continued evolution of

I'm using i6net's vxml browser in Asterisk. I'm trying to work out how I can return the inputs from a menu or form back into the Asterisk dial plan. Is there a variable? It's not documented if it is. The exit tag apparently can be used to return a value (still trying to work out how to do that), but what about multiple values, such as with a form? Doug. --------------

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Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and

Hello, When I have the following extension: exten => 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

Hello, I have been using IAX to serve clients endpoints for a while with no problem. But recently, to increase the bandwidth to the Asterisk server, I add another network interface card to my Asterisk server which connected to a different service provider that I currently have. Both of my nic is assigned different public ip. the client will actually choose one of these ip and authenticate itself.

hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

Hello all, I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF