Hello,
I have installed oh323 channel driver. Outgoing calls to H.323 world do not
include RFC2833 in the outgoing TerminalCapabilitiesSet despite that
userInputMode=RFC2833 has already been set.
Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel?
Kind regards,
Fernando Herrera
_____
De: Fernando Herrera [mailto:fherrera@iplan.com.ar]
Enviado el: Mi?rcoles, 21 de Septiembre de 2005 12:51
Para: 'asterisk-users@lists.digium.com'
Asunto: [Asterisk-Users] Help with asterisk-oh323 driver
DV,
Have you solved this? I am facing the same problem. I am running Asterisk
1.0.9 and outgoing TCS does not show the
receiveRTPAudioTelephonyEventCapability.
Kind regards,
Fernando Herrera
_____
Hi all,
Sorry if this has been answered previously, but I have not had any
luck trying to find it.
I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2,
kernel 2.6.8-1.521) to connect to a gateway that can only support
H323. I have installed the asterisk-oh323 channel driver (version
0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's
instructions) and PWLIB 1.6.6. This is all working fine for very basic
call setup and tear down, from any of my SCCP, SIP, H323 or POTS
(X100P card) phones.
NB: The gateway only handles signalling, so all media will flow
between the endpoints and the gateway will handle signalling to the
receiving gateway, as such (excuse the dodgy diagram :) ):
----------------->[Gateway]<-----------
| |
(H323) (H323 or MGCP/ISUP)
| |
V V
[Asterisk]-----------(RTP)----------[Terminating gateway]
|
(Signalling + RTP)
|
(Zaptel/SIP/H323/SCCP phones)
There are some requirements for me to connect to this switch:
1. I must support H245 tunneling and faststart (working fine)
2. I must dynamically negotiate the codecs (i.e. send multiple codecs
as part of the faststart and the softswitch will decide which of the
codecs to use based on the terminating gateway's capabilities). The
codec picked will be passed back in the return faststart from the
gateway.
3. It must support RFC2833 for OOB DTMF.
The problems I am facing are that my faststart in my setup messages
only ever has one codec, regardless of what I have set in the [codecs]
section of oh323.conf, and even if I specify userInputMode=RFC2833 in
oh323.conf my TCS does not include the capability
receiveRTPAudioTelephonyEventCapability hence RFC2833 is never
neogitated. I'm sure this is just a minor tweak of the source code,
but not being an expert in C I am having problems figuring out what
needs to be done and where.
Any help on this matter would be appreciated.
Cheers
DV
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Which version of the driver do you use? Fernando Herrera wrote:> Hello, > > I have installed oh323 channel driver. Outgoing calls to H.323 world do > not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that > userInputMode=RFC2833 has already been set. > > Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? > > Kind regards, > > */Fernando Herrera/* > > > ------------------------------------------------------------------------ > *De:* Fernando Herrera [mailto:fherrera@iplan.com.ar] > *Enviado el:* Mi?rcoles, 21 de Septiembre de 2005 12:51 > *Para:* 'asterisk-users@lists.digium.com' > *Asunto:* [Asterisk-Users] Help with asterisk-oh323 driver > > > > DV, > > Have you solved this? I am facing the same problem. I am running > Asterisk 1.0.9 and outgoing TCS does not show the > receiveRTPAudioTelephonyEventCapability. > > Kind regards, > > */Fernando Herrera/* > > ------------------------------------------------------------------------ > > Hi all, > > Sorry if this has been answered previously, but I have not had any > luck trying to find it. > > I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2, > kernel 2.6.8-1.521) to connect to a gateway that can only support > H323. I have installed the asterisk-oh323 channel driver (version > 0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's > instructions) and PWLIB 1.6.6. This is all working fine for very basic > call setup and tear down, from any of my SCCP, SIP, H323 or POTS > (X100P card) phones. > > NB: The gateway only handles signalling, so all media will flow > between the endpoints and the gateway will handle signalling to the > receiving gateway, as such (excuse the dodgy diagram :) ): > > ----------------->[Gateway]<----------- > | | > (H323) (H323 or MGCP/ISUP) > | | > V V > [Asterisk]-----------(RTP)----------[Terminating gateway] > | > (Signalling + RTP) > | > (Zaptel/SIP/H323/SCCP phones) > > > There are some requirements for me to connect to this switch: > > 1. I must support H245 tunneling and faststart (working fine) > 2. I must dynamically negotiate the codecs (i.e. send multiple codecs > as part of the faststart and the softswitch will decide which of the > codecs to use based on the terminating gateway's capabilities). The > codec picked will be passed back in the return faststart from the > gateway. > 3. It must support RFC2833 for OOB DTMF. > > The problems I am facing are that my faststart in my setup messages > only ever has one codec, regardless of what I have set in the [codecs] > section of oh323.conf, and even if I specify userInputMode=RFC2833 in > oh323.conf my TCS does not include the capability > receiveRTPAudioTelephonyEventCapability hence RFC2833 is never > neogitated. I'm sure this is just a minor tweak of the source code, > but not being an expert in C I am having problems figuring out what > needs to be done and where. > > Any help on this matter would be appreciated. > > Cheers > DV > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users