Cosmetic update for outbound REFERs

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 627770634ad20562691a084945cf0b43d13ef26a..68c8b94fd170821640dbf1b4391f73fc247ca6f7 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5705,7 +5705,8 @@ static int transmit_refer(struct sip_pvt *p, const char *dest)
        char *of, *c;
        char referto[256];
 
-       if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) 
+       /* Are we transfering an inbound or outbound call? */
+       if (ast_test_flag(&p->flags[0], SIP_OUTGOING))
                of = get_header(&p->initreq, "To");
        else
                of = get_header(&p->initreq, "From");
@@ -5729,16 +5730,26 @@ static int transmit_refer(struct sip_pvt *p, const char *dest)
                snprintf(referto, sizeof(referto), "<sip:%s>", dest);
        }
 
+       add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+
        /* save in case we get 407 challenge */
        ast_string_field_set(p, refer_to, referto);
        ast_string_field_set(p, referred_by, p->our_contact);
 
        reqprep(&req, p, SIP_REFER, 0, 1);
        add_header(&req, "Refer-To", referto);
+       add_header(&req, "Allow", ALLOWED_METHODS);
+       add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
        if (!ast_strlen_zero(p->our_contact))
                add_header(&req, "Referred-By", p->our_contact);
        add_blank_header(&req);
        return send_request(p, &req, 1, p->ocseq);
+
+       /*! \todo In theory, we should hang around and wait for a reply, before
+       returning to the dial plan here. Don't know really how that would
+       affect the transfer() app or the pbx, but, well, to make this
+       useful we should have a STATUS code on transfer().
+       */
 }
 
 /*! \brief Send SIP INFO dtmf message, see Cisco documentation on cisco.com */