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

[Mark Michelson]

Mark Michelson has posted comments on this change. ( https://gerrit.asterisk.org/4651 )

Change subject: res_pjsip_session: Access SIPDOMAIN via Dialplan.
......................................................................


Patch Set 1: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/#/c/4651/1/res/res_pjsip_session.c
File res/res_pjsip_session.c:

PS1, Line 2003: 		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);
              : 		}
I think malloc() is unnecessary here, given the short lifetime of the buffer. Instead, you can use ast_alloca() instead to get a stack allocation. A SIP hostname should not be large enough to cause a stack overflow.


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Gerrit-MessageType: comment
Gerrit-Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243
Gerrit-PatchSet: 1
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>
Gerrit-HasComments: Yes