[Asterisk-Users] Help with asterisk-oh323 driver

[Fernando Herrera]

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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