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<DIV><SPAN class=090315414-21092005><FONT face=Arial
size=2>DV,</FONT></SPAN></DIV>
<DIV><SPAN class=090315414-21092005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=090315414-21092005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=090315414-21092005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=090315414-21092005><FONT face=Arial size=2>Kind regards,
</FONT></SPAN></DIV>
<DIV><SPAN class=090315414-21092005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV align=left><FONT face=Arial size=2><STRONG><EM>Fernando
Herrera</EM></STRONG></FONT></DIV>
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<DIV align=left><FONT face=Arial size=1><PRE><FONT size=3>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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