READ application should do the job for you
[Syntax]Read(variable[,filename[filename2[...]][,maxdigits[,options[,attempts[,timeout])
i normally use first three arguments i.e. variable name, filename to play and
max digits
you might consider using waitexten before read though
Rgds
From:
On Sunday 07 Sep 2014, Anurag Rana wrote:
Hi,
I created a dummy dialplan where I ask the user to enter the age.
[macro-age]
exten = s,1,Background(my/age) ;;Play recorded message to enter age
exten = s,n,WaitExten(10)
exten = _XX,1,Set(AGE=${EXTEN});; this line is not
Thanks for the suggestion.
@Stiles - Look like this may work. Will try this. Thanks.
Anurag Rana
http://newbie42.blogspot.in/
On Mon, Sep 8, 2014 at 1:42 PM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
On Sunday 07 Sep 2014, Anurag Rana wrote:
Hi,
I created a dummy dialplan
@A J Stiles : If you could provide an example as you said, It would be very
nice. Thanks.
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On Monday 08 Sep 2014, Anurag Rana wrote:
@A J Stiles : If you could provide an example as you said, It would be very
nice. Thanks.
This is excerpted from a dialplan application I wrote. It's actually a PIN
entry but should be usable for any general purpose application. Sound files
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all
On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes
the ice-related lines from the sdp when it sends the INVITE out, and the
called webrtc client rejects the INVITE due to the
Hi Matthew,
Here's the debug output:
--- SIP read from UDP:PU.BL.IC.IP:5060 ---
INVITE sip:6...@testers.com SIP/2.0
Record-Route: sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes
Record-Route:
sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes
Via: SIP/2.0/UDP
On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hi Matthew,
Here's the debug output:
--- SIP read from UDP:PU.BL.IC.IP:5060 ---
INVITE sip:6...@testers.com SIP/2.0
Record-Route: sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes
Record-Route:
Hi,
My bad, below is the output for peer 661 and the log output, which does not
tell me much. I guess Asterisk assumes it's working correctly as there are
no errors etc, however for some reason the INVITE leaves Asterisk without
the ice definitions in the sdp. I must assume it's a configuration
We exchange information among call using sipaddheader.
Is there a similiar command in IAX?
Thanks,
Valter
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Thanks. I will try it. Meanwhile I was trying below code.
call goes to 'test' context and from there is passed to macro 'age'.
In 'age' macro when I am using any patter to accept even single digit, its
not working. So instead of using pattern I hardcoded the extension, but
still when I am
Can't we use pattern matching inside a macro?
Because when I am trying to do so call is terminating even for a very
simple dummy dialplan.
[demo3]
exten=98,1,NoOp()
exten=98,2,Macro(testme)
exten=h,1,NoOp(terminating call);
[macro-testme]
exten=s,1,Playback(Digits/2)
exten=s,2,WaitExten(15)
On Tue, 2014-09-02 at 13:18 -0500, Khalid Touati wrote:
so it seems Asterisk Versions does not support video I guess
Used it with jitsi and linphone softphones, works just OK.
Just for testing i did a video-call on the loop-back, great test tool
for showing the influence of (limited-)
There are some issues if you use WaitExten inside a macro.
On Mon, Sep 8, 2014 at 2:48 PM, Anurag Rana anuragrana31...@gmail.com
wrote:
Can't we use pattern matching inside a macro?
Because when I am trying to do so call is terminating even for a very
simple dummy dialplan.
[demo3]
Hi all,
I continue to see the following msg on my Asterisk log:
[Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite:
Failed to authenticate device 9009sip:9...@196.107.xx.xx;tag=8dd48dd2
IP: 196.107.xx.xx is my asterisk server IP address.
I don't know what it means and how to
On Mon, 8 Sep 2014, motty cruz wrote:
I continue to see the following msg on my Asterisk log:
[Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite:
Failed to authenticate device 9009sip:9...@196.107.xx.xx;tag=8dd48dd2
First step is to determine the source -- is it coming
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