[Asterisk-code-review] chan_pjsip.c: allow_180_ringing_with_sdp (asterisk[16])

[Joshua Colp]

Attention is currently required from: N A, Alexei Gradinari, Mark Petersen.
Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/17785 )

Change subject: chan_pjsip.c: allow_180_ringing_with_sdp
......................................................................


Patch Set 9: Code-Review-1

(4 comments)

Patchset:

PS9: 
This also needs an alembic migration added[1] for the ps_globals table.

[1] https://alembic.sqlalchemy.org/en/latest/tutorial.html#create-a-migration-script


File channels/chan_pjsip.c:

https://gerrit.asterisk.org/c/asterisk/+/17785/comment/c4a9b642_7e9f0c3a 
PS9, Line 1626: 				if(ast_sip_get_allow_180_ringing_with_sdp()) {
if (ast_sip


File configs/samples/pjsip.conf.sample:

https://gerrit.asterisk.org/c/asterisk/+/17785/comment/0d23e0e6_679a8a8d 
PS9, Line 1259: ;allow_180_ringing_with_sdp=yes	; preserve 180 Ringing after reciving an 183 with SDP
I think this is confusing as described, because you're talking about receiving when in reality this is about what we send to an endpoint.

"Allow Asterisk to send 180 Ringing with SDP to an endpoint after sending a 183 Session Progress with SDP to an endpoint. If disabled Asterisk will instead send only a 183 Session Progress with SDP to the endpoint."


File doc/CHANGES-staging/chan_pjsip_180_sdp.txt:

https://gerrit.asterisk.org/c/asterisk/+/17785/comment/a5050e08_44efd780 
PS9, Line 3: added new global config option "allow_180_ringing_with_sdp"
My comment applies here as well.



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Gerrit-Project: asterisk
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Gerrit-Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18
Gerrit-Change-Number: 17785
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