On 10/17/2005, "James Sizemore" <[EMAIL PROTECTED]> wrote:
> Doubling the value to 500 did not seem to effect the length of the
> tone played at allhm. Back to the drawing board for me.
> Anyone know what this value is supposed to effect?
>
>>> I have a gateway using a Digium card to conver
Doubling the value to 500 did not seem to effect the length of the
tone played at allhm. Back to the drawing board for me.
Anyone know what this value is supposed to effect?
James Sizemore wrote:
I did a bit of searching around and found this class in chan_sip.c:
I am going to test the
I did a bit of searching around and found this class in chan_sip.c:
I am going to test the Duration at 500, and see how this effect
things. If anyone has already played with these values, and had any
bad gotchas please let me know.
==
static int add_digit(struct sip_request *req,
On Behalf Of James Sizemore
> Sent: Monday, 17 October 2005 2:27 PM
> To: Asterisk Developers Mailing List
> Subject: [Asterisk-Dev] INFO and Duration=250
>
> I have a gateway using a Digium card to convert a PRI
> call to a sip call then I transport the sip call to a Cisco
>
I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD where it is converted back to a PRI. This all works
well except DTMF is sent with a duration of .25sec.
PRI specs says this should be .25sec to .5sec so this
is with in spec, howev
