]> git.ipfire.org Git - thirdparty/asterisk.git/commit
res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
authorTinet-mucw <mucw@ti-net.com.cn>
Fri, 2 Aug 2024 08:49:58 +0000 (16:49 +0800)
committerTinet-mucw <mucw@ti-net.com.cn>
Tue, 6 Aug 2024 18:02:24 +0000 (18:02 +0000)
commit7f8391e9ff28895be619a3a20b87aefc9c409d8c
tree9e570a943e01ece788024e63e51e88d4a6015cb3
parent1ca9661c6d343861d1954e1c855b4e5caa2529c0
res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto

When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.

Resolves: asterisk#826
res/res_pjsip_sdp_rtp.c