[Asterisk-Dev] New channel

[Marcin Kwiatkowski]

Hi,

We are developing a new channel for Asterisk based on VopLib, because we 
need to comunicate between SIP, H.323 and AudioCodes TP240 which only 
supports MGCP. Previously we tried to use MGCP protocol but it doesn't 
works for us.
Basicly - we'd copied a large part from chan_modem and implemented 
AudioCodes's voplib in it. Almost works - we can make a call from sip to 
PSTN, but voice is simplex. What do I mean. I can hear other side when 
I'm receiving call (PSTN) but SIP part doesn't hear me.
We do some debug - ie. read procedure should generate some message. but 
as it's registered - do nothing.

nativeformat, readformat and writeformat are AST_FORMAT_G7231 (we need 
pass-through mode - its only termination PSTN-VoIP), and registered 
functions are send_digit, call, hangup, answer, read, write, bridge. Has 
anyone any idea?

Thanks.


-- 
Marcin Kwiatkowski
Senior IT Specialist
Telebonus Sp. z o.o.
Legionow 30
43-300 Bielsko-Biala
pho/fax: +48 (33) 828 25 21
mob: +48 605 923 944