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https://github.com/asterisk/asterisk.git
synced 2026-07-01 20:43:15 -07:00
Don't recreate stuff on authentication (so it works on reinvites)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
+18
-15
@@ -1978,7 +1978,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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if ( res != -1 ) {
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p->callingpres = ast->cid.cid_pres;
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p->jointcapability = p->capability;
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transmit_invite(p, SIP_INVITE, 1, 1);
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transmit_invite(p, SIP_INVITE, 1, 2);
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if (p->maxtime) {
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/* Initialize auto-congest time */
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p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
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@@ -4422,7 +4422,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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onlydigits = 0;
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break;
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}
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}
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}
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/* If we have only digits, add ;user=phone to the uri */
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if (onlydigits)
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@@ -4491,16 +4491,16 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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}
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/* If custom URI options have been provided, append them */
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if (p->options && p->options->uri_options)
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ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options);
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if (p->options && p->options->uri_options)
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ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options);
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ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
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ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
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/* If there is a VXML URL append it to the SIP URL */
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if (p->options && p->options->vxml_url) {
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else {
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snprintf(to, sizeof(to), "<%s>", p->uri);
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/* If there is a VXML URL append it to the SIP URL */
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if (p->options && p->options->vxml_url) {
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else {
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snprintf(to, sizeof(to), "<%s>", p->uri);
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}
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memset(req, 0, sizeof(struct sip_request));
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init_req(req, sipmethod, p->uri);
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@@ -4529,7 +4529,10 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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/* Bump branch even on initial requests */
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p->branch ^= rand();
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build_via(p, p->via, sizeof(p->via));
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initreqprep(&req, p, sipmethod);
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if (init > 1)
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initreqprep(&req, p, sipmethod);
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else
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reqprep(&req, p, sipmethod, 0, 1);
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} else
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reqprep(&req, p, sipmethod, 0, 1);
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@@ -4792,7 +4795,7 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
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char *t = tmp;
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size_t maxbytes = sizeof(tmp);
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initreqprep(&req, p, SIP_NOTIFY);
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initreqprep(&req, p, SIP_NOTIFY);
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add_header(&req, "Event", "message-summary");
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add_header(&req, "Content-Type", default_notifymime);
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@@ -8281,7 +8284,7 @@ static int sip_notify(int fd, int argc, char *argv[])
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continue;
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}
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initreqprep(&req, p, SIP_NOTIFY);
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initreqprep(&req, p, SIP_NOTIFY);
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for (var = varlist; var; var = var->next)
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add_header(&req, var->name, var->value);
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@@ -10654,9 +10657,9 @@ static int sip_poke_peer(struct sip_peer *peer)
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ast_set_flag(p, SIP_OUTGOING);
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#ifdef VOCAL_DATA_HACK
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ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
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transmit_invite(p, SIP_INVITE, 0, 1);
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transmit_invite(p, SIP_INVITE, 0, 2);
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#else
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transmit_invite(p, SIP_OPTIONS, 0, 1);
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transmit_invite(p, SIP_OPTIONS, 0, 2);
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#endif
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gettimeofday(&peer->ps, NULL);
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peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);
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