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jan.kubr at gmail.com Guest
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Posted: Mon Dec 15, 2008 4:37 am Post subject: [Freeswitch-users] Interrupting read application with DTMF |
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Hi,
I have been having some troubles with the read application for quite a
while which I haven't been able to solve yet.
I have Freeswitch connected to a SIP gateway to accept calls from a
landline-like number. For the incoming calls I have a simple testing
dialplan:
<action application="read" data="1 1
en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>
The behavior I have a problem with is that the read app should
terminate when I press a digit and the execution should jump to the
next action - meaning the playback of the file should be interrupted.
The problem is that when I call the public number from my cell phone
this works only about 50% of the time. In the other cases I need to
wait for the wav file to be played (or press the digit two or three
times). When using a SIP phone it always works.
Today I tried to convert the wav file the read app plays to the GSM
format and found out it fixed the problem! Now I can almost always
interrupt the read app with DTMF from my cell phone. Doing the same
from my SIP phone doesn't work well though when the file is GSM.
Can someone explain me what is going on here and what is the right
approach? I'm on revision 10751. I've tried to set a few configuration
variables based on suggestions from this list, but it didn't make any
difference.
Thanks,
Jan Kubr
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anthony.minessale at g... Guest
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Posted: Mon Dec 15, 2008 11:15 am Post subject: [Freeswitch-users] Interrupting read application with DTMF |
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I think your solution is most likely superstition and that your real problem is related
to your cell phone and the PSTN to SIP translation somewhere along the way.
I bet if you called in to the same extension with a SIP desk phone, that it would work every
time no matter what format your file is.
On Mon, Dec 15, 2008 at 3:27 AM, Jan Kubr <jan.kubr@gmail.com (jan.kubr@gmail.com)> wrote:
Quote: | Hi,
I have been having some troubles with the read application for quite a
while which I haven't been able to solve yet.
I have Freeswitch connected to a SIP gateway to accept calls from a
landline-like number. For the incoming calls I have a simple testing
dialplan:
<action application="read" data="1 1
en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>
The behavior I have a problem with is that the read app should
terminate when I press a digit and the execution should jump to the
next action - meaning the playback of the file should be interrupted.
The problem is that when I call the public number from my cell phone
this works only about 50% of the time. In the other cases I need to
wait for the wav file to be played (or press the digit two or three
times). When using a SIP phone it always works.
Today I tried to convert the wav file the read app plays to the GSM
format and found out it fixed the problem! Now I can almost always
interrupt the read app with DTMF from my cell phone. Doing the same
from my SIP phone doesn't work well though when the file is GSM.
Can someone explain me what is going on here and what is the right
approach? I'm on revision 10751. I've tried to set a few configuration
variables based on suggestions from this list, but it didn't make any
difference.
Thanks,
Jan Kubr
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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