[Asterisk-code-review] res pjsip sdp rtp: Set hangup cause for RTP timeouts (asterisk[13])

[Mark Michelson]

Mark Michelson has posted comments on this change. ( https://gerrit.asterisk.org/5316 )

Change subject: res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts
......................................................................


Patch Set 1: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/#/c/5316/1/res/res_pjsip_sdp_rtp.c
File res/res_pjsip_sdp_rtp.c:

PS1, Line 166: 	ast_channel_lock(chan);
             : 	ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
             : 	ast_channel_unlock(chan);
             : 
             : 	ast_softhangup(chan, AST_SOFTHANGUP_DEV);
Since you're adding explicit channel locking here, move the ast_softhangup() up to inside the scope of the channel lock, and change it to ast_softhangup_nolock().


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Gerrit-MessageType: comment
Gerrit-Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-HasComments: Yes