using chan_h323 and g.711u
bkw
On Wed, 30 Jul 2003, Patrick wrote:
>
> Which codec are you using? and which H.323 channel driver? chan_h323 or
> chan_oh323 ?
>
>
> On 30 Jul 2003, Eric Wieling wrote:
>
> > That only works if you are using the G711 (ulaw/alaw) codecs. Other
> > codecs distort
Which codec are you using? and which H.323 channel driver? chan_h323 or
chan_oh323 ?
On 30 Jul 2003, Eric Wieling wrote:
> That only works if you are using the G711 (ulaw/alaw) codecs. Other
> codecs distort inband DTMF.
>
> On Wed, 2003-07-30 at 15:26, Patrick wrote:
> > I have the same s
thats all we use right now
On Wed, 30 Jul 2003, Eric Wieling wrote:
> That only works if you are using the G711 (ulaw/alaw) codecs. Other
> codecs distort inband DTMF.
>
> On Wed, 2003-07-30 at 15:26, Patrick wrote:
> > I have the same setup, and in the sip.conf file I set the dtmfmode=inband
>
That only works if you are using the G711 (ulaw/alaw) codecs. Other
codecs distort inband DTMF.
On Wed, 2003-07-30 at 15:26, Patrick wrote:
> I have the same setup, and in the sip.conf file I set the dtmfmode=inband
> for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
>
Try setting dtmf-relay h245-alphanumeric in the MC3810 dial-peer.
On Wed, 30 Jul 2003, Brian West wrote:
> I have done that but I think its the Asterisk => MC3810 via h323 thats
> causing that. Does anyone have an example on how i can dump sip to and
> from the MC3810 to my asterisk server?
>
I have done that but I think its the Asterisk => MC3810 via h323 thats
causing that. Does anyone have an example on how i can dump sip to and
from the MC3810 to my asterisk server?
bkw
On Wed, 30 Jul 2003, Patrick wrote:
>
> I have the same setup, and in the sip.conf file I set the dtmfmode=inb
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
> I have this setup:
>
> Sip Phones -> Asterisk -> h323 gateway -> ptsn
>
> Sip phones are setup for ou
I have this setup:
Sip Phones -> Asterisk -> h323 gateway -> ptsn
Sip phones are setup for out of band dtmf
but the h323 gateway is inband. Is their a way to pass the digits from
the sip phones to the ptsn via the h323 gateway?
bkw
___
Asterisk-Users
