Asterisk SIP -> H.323
приветствую всех.
помогите разобраться. пытаюсь научить Asterisk звонить с клиента у которого используется SIP клиенту у которого используется H.323
Звонок доходит. дозванивается. снимаю трубку. коннект обрывается.
OpenPhone Version 1.9.2 by NetCall55 on Windows 2000 (v5.1.2600-i586)
Codec G.729A encoder created
RTP_UDP Failed to extract mediaControl transport for T-0
Codec G.729A encoder destroyed
Codec G.729A decoder created
RTP_UDP Failed to extract mediaControl transport for R-101
Codec G.729A decoder destroyed
Codec G.729A encoder created
RTP_UDP Failed to extract mediaControl transport for T-0
Codec G.729A encoder destroyed
Codec G.729A decoder created
RTP_UDP Failed to extract mediaControl transport for R-102
Codec G.729A decoder destroyed
RTP_UDP Failed to extract mediaControl transport for T-0
RTP_UDP Failed to extract mediaControl transport for R-104
RTP_UDP Failed to extract mediaControl transport for T-0
RTP_UDP Failed to extract mediaControl transport for R-105
Codec G.729A decoder created
RTP_UDP Failed to extract mediaControl transport for R-106
Codec G.729A decoder destroyed
H323 Connection ip$82.208.40.88:53043/16288 terminated.
в extensions.conf написано вот так
exten => 123456,1,Dial(H323/${EXTEN}@213.220.215.51,120,hHr)
Вот что мне сыпется в логи!
Jan 26 23:30:15 WARNING[96081]: channel.c:2328 set_format: Unable to find a codec translation path from unknown to gsm
Jan 26 23:30:15 WARNING[96081]: channel.c:2328 set_format: Unable to find a codec translation path from unknown to gsm
Jan 26 23:30:15 WARNING[96081]: channel.c:2690 ast_channel_make_compatible: No path to translate from SIP/200-e1b6(2) to H323/213.220.215.51-1(0)
Jan 26 23:30:15 WARNING[96081]: channel.c:3509 ast_channel_bridge: Can't make SIP/200-e1b6 and H323/213.220.215.51-1 compatible
Jan 26 23:30:15 WARNING[96081]: res_features.c:1371 ast_bridge_call: Bridge failed on channels SIP/200-e1b6 and H323/213.220.215.51-1
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