[Asterisk-code-review] res pjsip session: Access SIPDOMAIN via Dialplan. (asterisk[master])

Alexander Traud asteriskteam at digium.com
Thu Dec 22 09:28:05 CST 2016


Alexander Traud has uploaded a new change for review. ( https://gerrit.asterisk.org/4651 )

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, 9 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/51/4651/1

diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index be07e94..bd8f9e8 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -2000,6 +2000,15 @@
 
 	if (!strcmp(session->exten, pickupexten) ||
 		ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) {
+		size_t size = pj_strlen(&sip_ruri->host) + 1;
+		char *domain = ast_malloc(size);
+
+		if (domain) {
+			ast_copy_pj_str(domain, &sip_ruri->host, size);
+			pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
+			ast_free(domain);
+		}
+
 		return SIP_GET_DEST_EXTEN_FOUND;
 	}
 	/* XXX In reality, we'll likely have further options so that partial matches

-- 
To view, visit https://gerrit.asterisk.org/4651
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>



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