]> git.ipfire.org Git - thirdparty/asterisk.git/commit
projects / thirdparty / asterisk.git / commit
? search:
summary | shortlog | log | commit | commitdiff | tree
(parent: f5083d4) | patch
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 Development Team <asteriskteam@digium.com>
Thu, 12 Sep 2024 18:44:38 +0000 (18:44 +0000)
commit36fc592a5bf1991c1db63f7294a28cc9e3a64988
treec8385414f7c75c1a030db9c50008840dcef8814btree | snapshot
parentf5083d4d483ed2d1e81b8600ebf0d426782096eacommit | 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
(cherry picked from commit 4aae8194b4389c4570368ea42152d2f1eff1e1b2)
res/res_pjsip_sdp_rtp.c diff | blob | blame | history
Mirror of https://github.com/asterisk/asterisk.git