[Asterisk-code-review] res pjsip session: Access SIPDOMAIN via Dialplan. (asterisk[master])
Alexander Traud
asteriskteam at digium.com
Wed Jan 4 07:12:30 CST 2017
Hello Mark Michelson, Anonymous Coward #1000019,
I'd like you to reexamine a change. Please visit
https://gerrit.asterisk.org/4651
to look at the new patch set (#2).
Change subject: res_pjsip_session: Access SIPDOMAIN via Dialplan.
......................................................................
res_pjsip_session: Access SIPDOMAIN via Dialplan.
This feature was available in the SIP channel driver chan_sip. For example,
Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not
local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect
and dial remote SIP-URIs. This change here sets the SIP destination domain of
an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well.
ASTERISK-26670 #close
Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243
---
M res/res_pjsip_session.c
1 file changed, 6 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/51/4651/2
--
To view, visit https://gerrit.asterisk.org/4651
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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