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
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