asterisk users - Aug 2003 - Grandstream firmware update DMTF Payload Type

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems
to be having problems.

The Grandstream and sip.conf were set to RFC2833 now with that setting I
get extra digits during "Mailbox" and "Password" phases.
222001 instead
of 2201 for example.

When both are changed to "SIP info" there is no problem.

But what is the new setting "DTMF Payload Type:" it appears to default
to 101 is that likely to be the culprit?
   
-- 
Dave Cotton <dcotton@linuxautrement.com>

For what its worth, "info" is the only way I could ever get the GS to
send
reliable DTMF even with 1.0.3.78.

I can confirm that 1.0.3.81 fixes the NTP issue I had with one phone.
However qualify=y still causes the phone to randomly lockup. (Still not
convinced this is a GS only issue since * is once again showing non-timed
out sip channels)

John
-------------------------------------------------------------------------
NetRom Internet Services			973-208-1339 voice
john@netrom.com					973-208-0942 fax
http://www.netrom.com
-------------------------------------------------------------------------


On Mon, 25 Aug 2003, Dave Cotton wrote:
> Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems
> to be having problems.
>
> The Grandstream and sip.conf were set to RFC2833 now with that setting I
> get extra digits during "Mailbox" and "Password"
phases. 222001 instead
> of 2201 for example.
>
> When both are changed to "SIP info" there is no problem.
>
> But what is the new setting "DTMF Payload Type:" it appears to
default
> to 101 is that likely to be the culprit?
>
> --
> Dave Cotton <dcotton@linuxautrement.com>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

John Vozza wrote:
> For what its worth, "info" is the only way I could ever get the
GS to send
> reliable DTMF even with 1.0.3.78.
> 
> I can confirm that 1.0.3.81 fixes the NTP issue I had with one phone.
> However qualify=y still causes the phone to randomly lockup. (Still not
> convinced this is a GS only issue since * is once again showing non-timed
> out sip channels)
Thanks for the info. I had to switch to info as well, using RTP seems 
iffy at best at the moment.

What NTP issue? My 1.0.3.78 phones are NATting out just fine.

The qualify=yes problem bit me as well. Thanks for the heads-up on that.

-- 
- Ian C. Blenke <icblenke@nks.net>
(This message bound by the following:
http://www.nks.net/email_disclaimer.html)

Out of 4 phones running 1.0.3.78, one would not do NTP. I have no idea
why. Same network. Same tftp update process/server, yet one did not work.

1.0.3.81 fixed that lone hold out.

John
-------------------------------------------------------------------------
NetRom Internet Services			973-208-1339 voice
john@netrom.com					973-208-0942 fax
http://www.netrom.com
-------------------------------------------------------------------------


On Mon, 25 Aug 2003, Ian Blenke wrote:
> John Vozza wrote:
> > For what its worth, "info" is the only way I could ever get
the GS to send
> > reliable DTMF even with 1.0.3.78.
> >
> > I can confirm that 1.0.3.81 fixes the NTP issue I had with one phone.
> > However qualify=y still causes the phone to randomly lockup. (Still
not
> > convinced this is a GS only issue since * is once again showing
non-timed
> > out sip channels)
>
> Thanks for the info. I had to switch to info as well, using RTP seems
> iffy at best at the moment.
>
> What NTP issue? My 1.0.3.78 phones are NATting out just fine.
>
> The qualify=yes problem bit me as well. Thanks for the heads-up on that.
>
> --
> - Ian C. Blenke <icblenke@nks.net>
> (This message bound by the following:
> http://www.nks.net/email_disclaimer.html)
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>