Displaying 20 results from an estimated 20000 matches similar to: "Return Vars to Dial Plan from VXML()"
2008 Jul 03
2
Asterisk VXML... Help.
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
2008 Jun 27
1
Asterisk 1.2 app_vxml
I just downloaded the app_vxml for Asterisk 1.2 from i6net.
Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I grabbed the 1.4 version of the module hoping in vain that would work, but it fails with invalid symbols, which isn't surprising.
Any ideas on how I can get this to work?
2007 Aug 01
0
Announcing free (GPL) VXML for Asterisk - Voiceglue
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
2005 Feb 24
2
asterisk supports VXML?
Hello,
Does asterisk supports VXML?
Couldn't find much resource on that on google and wiki.
Thanks
Foong
2012 Dec 25
2
Vxml record voice parameter
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?
--
2007 Oct 29
5
A Leg Control on Asterisk Callback
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface.
Lets say our originate commands looks like this:
ACTION: Originate
Async: yes
Timeout: 60000
Exten: callback
Channel: SIP/5551212 at provider
Variable: destination=SIP/8675309 at provider
Callerid: 5551212
Context: default
ActionID: 849120
2006 Dec 22
0
VXML in Asterisk HELP!
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2007 Nov 28
3
Multiple Return Values from func_odbc
Is there any way to return multiple values from functions defined in func_odbc.conf?
It appears that you can only return one value.
True? Hope not....
Doug.
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2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten => s,n,AGI(agi://localhost/url=${ENCODED})
exten => s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s at VXML:1]
2007 Oct 25
3
Getting SIP Response Code from HANGUPCAUSE
I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead?
Doug.
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An HTML
2007 Dec 06
3
CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup extensions?
Asterisk 1.4.13
Thanks, Doug.
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2006 Mar 21
1
'Click to Dial'
Oooo I think I am gonna poo my pants.
Using the microbrowser on a Polycom 601, I was able to get it to execute a cgi script upon selection of an item. The cgi script used Net::Telnet connect to the manager interface on another Asterisk system, call the user back at their phone and then bridge the call to the number associated with the microbrowser menu option. That's pretty schweet.
:)
2007 May 02
2
Large dial plans and variables
I have a large dial plan here with over 3000 lines, and several dozen
macros. As it grew, it became apparent that there was some problems.
1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc,
if that macro calls another macro, and passes arguments like this as
well, you lose the original values.
2. When the macro's 'return' some value, it has to set a channel
2008 Apr 18
1
REGISTER Outboundproxy
Is it possible to set an outboundproxy for REGISTER from Asterisk?
register => xxx:yyy at sip99.foobar.com
[foobar]
type=peer
host=sip99.foobar.com
disallow=all
allow=g729
canreinvite=no
secret=yyy
fromuser=xxx
port=5099
outboundproxy=xxx.42.149.69
However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69.
Thanks,
Doug.
2006 Jun 14
1
dial plan return values
Is there a method for detecting return values of applications in the
dial plan?
Thanks
Mark Price
UNETA
2006 Jun 09
2
Dial Plan rules
Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or
mix them? Not that I would do that as a rule but sometimes you make
mistakes!
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* doug@crompton.com *
* http://www.crompton.com *
****************************
2006 Dec 08
1
Douglas Garstang <dgarstang@oneeighty.com>
On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote:
> Hi Steve.
>
> Thanks, but unfortunately, I can't be involved in that. We are
> running Asterisk in a production environment and we're using
> 1.2, not 1.4. I don't have the resources to work with 1.4.
> Last time I filed a bug against 1.2 I got told off.
>
2005 Jun 01
1
FW: TellMe pay-as-you-go? - UPDATE
As some of you know I've been trying to facilitate an involvement with
www.tellme.com <http://www.tellme.com/> speech recognition tools and
Asterisk. See www.studio.tellme.com <http://www.studio.tellme.com/>
There have been a number of people who are already integrating the two
and utilizing Tellme as an ASP to deliver speech recognition to their
asterisk applications.
2008 Jan 09
1
Help! channel_find_deadlocked: Avoided initial deadlock for ...
Hope someone can help.
I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.
Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial