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
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 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 mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
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=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 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 mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
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? > > bkw > > On Wed, 30 Jul 2003, 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. > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > 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 mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
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. > > > On Wed, 30 Jul 2003, Brian West wrote: > > > 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 mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)
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 > > 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 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 mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
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 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 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 mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users >
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 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. > > > > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > > > 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 mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >