o that spot in the
> audio fileIf you're feeling REALLY frisky, do a frequency
> analysis...I'll bet you'll see that the voice that is speaking at the
> time of the DTMF event on your various captures will have a frequency
> range in common...a very close range...maybe
SIP/5211 is a Grandstream device.
Did not add relaxdtmf=no, but sip show settings verifies it's already set to
no.
Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10
Travis
On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen
> wrote:
> Travis Langhals writes:
>
I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are
receiving random DTMF tones on their side, but that are not heard by the
outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have
always had this issue. I am only using SIP on the Asterisk server and all
e
I ended up getting this to work using Endstream for termination with the
same setup. I may try Gafachi again if I can upgrade to 1.6.2 in the near
future as they seems to be recommended for faxing.
On Tue, May 18, 2010 at 9:02 AM, David Backeberg wrote:
> On Tue, May 18, 2010 at 6:14 AM, Jonas K
I have been trying to get this working with an HT-502, Asterisk 1.4.31, and
Gafachi but no luck so far.
The VSP should send a re-invite for the T.38 media change on detection of
the fax tone.
I'm using canreinvite=yes on the trunk and canreinvite=no on the HT-502
extension. Also have t38pt_udptl
