passing dtmf rpt-event packet directly for 2-way call.

Add a new frame type AST_FRAME_DTMF_BYPASS that used by chan-voicemngr which will forward the dtmf rtp-event packet directly for 2-way call when dtmf_mode is rfc4733.

for MR: asterisk-chan-voicemngr!70 (merged)

assigned to @wenpeng.song

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• b23a21f5 - update
• afdce05b - update

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mentioned in merge request asterisk-chan-voicemngr!70 (merged)

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marked this merge request as ready

requested review from @Yalu

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• 4abdd325 - correction

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resolved all threads

See the comment in channel.h for this redefinition.

changed this line in version 4 of the diff

Why not just include res_pjsip.h as other samples but redefining it here? Duplication of code is not a good practice and shall be avoided as possible as we can.

~/git/voip/asterisk$ grep -rsn "res_pjsip.h"|grep "#include"
channels/chan_pjsip.c:67:#include "asterisk/res_pjsip.h"
channels/pjsip/dialplan_functions.c:604:#include "asterisk/res_pjsip.h"
channels/pjsip/cli_commands.c:36:#include "asterisk/res_pjsip.h"
funcs/func_pjsip_aor.c:44:#include "asterisk/res_pjsip.h"
funcs/func_pjsip_endpoint.c:45:#include "asterisk/res_pjsip.h"
funcs/func_pjsip_contact.c:44:#include "asterisk/res_pjsip.h"
...

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• 4d71f4a2 - update header

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merged