2009/4/15 John covici cov...@ccs.covici.com
Its not there and the link you gave me says its for sip originating
rather than calls to a sip channel.
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
It's been around awhile. I've used it in 1.4 Check out this link for
Hi,
On Mon, Apr 13, 2009 at 05:32:45PM -0400, John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like
Well, this solution seemed not to work for me, maybe because I did not
set the minimum and also if I am using a sip phone or ATA, the
solution would not apply -- correct me if I am wrong on either of
these.
on Wednesday 04/15/2009 Mark G. Thomas(m...@misty.com) wrote
Hi,
On Mon, Apr 13,
Hi,
Setting the minimum was necessary in my case, and did affect
tones to and from SIP devices as well as the SIP provider, though
this was some time ago and you may have different results with
your setup than I did with mine.
Mark
On Wed, Apr 15, 2009 at 11:31:06AM -0400, John covici wrote:
OK, thanks. If I could convince them to use info, would that be
better as far as the duration is concerned?
on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to
To the best of my knowledge, the only way for you to control the
duration sent to the PSTN lines is for you to be directly connected to
the lines so you can set the tone duration in zapata.conf / dahdi.conf
or to use inband signalling.
One thing you might try is researching the SipDtmfMode
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
To the best of my knowledge, the only way for you to control the
duration sent to the PSTN lines is for you to be directly
It's been around awhile. I've used it in 1.4 Check out this link for
basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode
John covici wrote:
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent
Its not there and the link you gave me says its for sip originating
rather than calls to a sip channel.
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
It's been around awhile. I've used it in 1.4 Check out this link for
basic info:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp packets of some sort, so
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp
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