RE:[Asterisk-Users] Speeding up the dial of DTMF's in SIP channel
When using dtmfmode=info, Asterisk is hard coded for 250ms
tone duration in chan_sip.c
I've changed this to 100ms without any ill effects.
Chip Schweiss
-Original Message-
From: apalma [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 3:26 PM
To: asterisk-users
Subject: [Asterisk-Users] Speeding up the dial of DTMF's in
SIP channel
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent
from
softphones through Asterisk, and based on them, build the
destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway
user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's
possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking
the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
??lvaro Palma
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speeding up the dial of DTMF's in SIP channel
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
Álvaro Palma
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel
Please Ignore if you cannot edit the code.
You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of 0 it will default to 100millisecs.
The solution is to add an option to dial application for the timeout
which defaults to the current value(250ms) in app_dial which will
provide for custom timeouts through the dialplan.
Also note that too small timeouts like below 100ms will mess up inband
dtmf tones for example in some zap channels.
Imran
On 3/15/06, Álvaro Palma [EMAIL PROTECTED] wrote:
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
Álvaro Palma
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
