Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv22814/channels
Modified Files:
chan_sip.c
Log Message:
Don't recreate stuff on authentication (so it works on reinvites)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.836
retrieving revision 1.837
diff -u -d -r1.836 -r1.837
--- chan_sip.c 2 Sep 2005 19:24:32 -0000 1.836
+++ chan_sip.c 6 Sep 2005 20:03:31 -0000 1.837
@@ -1978,7 +1978,7 @@
if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres;
p->jointcapability = p->capability;
- transmit_invite(p, SIP_INVITE, 1, 1);
+ transmit_invite(p, SIP_INVITE, 1, 2);
if (p->maxtime) {
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime * 4,
auto_congest, p);
@@ -4422,7 +4422,7 @@
onlydigits = 0;
break;
}
- }
+ }
/* If we have only digits, add ;user=phone to the uri */
if (onlydigits)
@@ -4491,16 +4491,16 @@
}
/* If custom URI options have been provided, append them */
- if (p->options && p->options->uri_options)
- ast_build_string(&invite, &invite_max, ";%s",
p->options->uri_options);
+ if (p->options && p->options->uri_options)
+ ast_build_string(&invite, &invite_max, ";%s",
p->options->uri_options);
- ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
+ ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
- /* If there is a VXML URL append it to the SIP URL */
- if (p->options && p->options->vxml_url) {
- snprintf(to, sizeof(to), "<%s>;%s", p->uri,
p->options->vxml_url);
- } else {
- snprintf(to, sizeof(to), "<%s>", p->uri);
+ /* If there is a VXML URL append it to the SIP URL */
+ if (p->options && p->options->vxml_url) {
+ snprintf(to, sizeof(to), "<%s>;%s", p->uri,
p->options->vxml_url);
+ } else {
+ snprintf(to, sizeof(to), "<%s>", p->uri);
}
memset(req, 0, sizeof(struct sip_request));
init_req(req, sipmethod, p->uri);
@@ -4529,7 +4529,10 @@
/* Bump branch even on initial requests */
p->branch ^= rand();
build_via(p, p->via, sizeof(p->via));
- initreqprep(&req, p, sipmethod);
+ if (init > 1)
+ initreqprep(&req, p, sipmethod);
+ else
+ reqprep(&req, p, sipmethod, 0, 1);
} else
reqprep(&req, p, sipmethod, 0, 1);
@@ -4792,7 +4795,7 @@
char *t = tmp;
size_t maxbytes = sizeof(tmp);
- initreqprep(&req, p, SIP_NOTIFY);
+ initreqprep(&req, p, SIP_NOTIFY);
add_header(&req, "Event", "message-summary");
add_header(&req, "Content-Type", default_notifymime);
@@ -8281,7 +8284,7 @@
continue;
}
- initreqprep(&req, p, SIP_NOTIFY);
+ initreqprep(&req, p, SIP_NOTIFY);
for (var = varlist; var; var = var->next)
add_header(&req, var->name, var->value);
@@ -10654,9 +10657,9 @@
ast_set_flag(p, SIP_OUTGOING);
#ifdef VOCAL_DATA_HACK
ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__",
sizeof(p->username));
- transmit_invite(p, SIP_INVITE, 0, 1);
+ transmit_invite(p, SIP_INVITE, 0, 2);
#else
- transmit_invite(p, SIP_OPTIONS, 0, 1);
+ transmit_invite(p, SIP_OPTIONS, 0, 2);
#endif
gettimeofday(&peer->ps, NULL);
peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2,
sip_poke_noanswer, peer);
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