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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)
committerasterisk-org-access-app[bot] <120671045+asterisk-org-access-app[bot]@users.noreply.github.com>
Tue, 6 Aug 2024 18:02:28 +0000 (18:02 +0000)
commit291c1e4c1f392926d593c76211dd7553d2f5b82e
tree3141ede5df6fe95e5c1fed1e884ffbeecd9607f0tree | snapshot
parentb7879224c120570cad8da12902a44ee85c5c67abcommit | diff
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 diff | blob | blame | history
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