Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

Don't recreate stuff on authentication (so it works on reinvites)

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Mark Spencer 20 years ago
parent 7348cba9c7
commit bcd06b8582
1 changed files with 18 additions and 15 deletions
Show Stats Download Patch File Download Diff File

33
channels/chan_sip.c
Unescape View File

@ -1978,7 +1978,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
if ( res != -1 ) { if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres; p->callingpres = ast->cid.cid_pres;
p->jointcapability = p->capability; p->jointcapability = p->capability;
transmit_invite(p, SIP_INVITE, 1, 1); transmit_invite(p, SIP_INVITE, 1, 2);
if (p->maxtime) { if (p->maxtime) {
/* Initialize auto-congest time */ /* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p); p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
@ -4422,7 +4422,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
onlydigits = 0; onlydigits = 0;
break; break;
} }
} }
/* If we have only digits, add ;user=phone to the uri */ /* If we have only digits, add ;user=phone to the uri */
if (onlydigits) if (onlydigits)
@ -4491,16 +4491,16 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
} }
/* If custom URI options have been provided, append them */ /* If custom URI options have been provided, append them */
if (p->options && p->options->uri_options) if (p->options && p->options->uri_options)
ast_build_string(&invite, &invite_max, ";%s", 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 there is a VXML URL append it to the SIP URL */
if (p->options && p->options->vxml_url) { if (p->options && p->options->vxml_url) {
snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url); snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
} else { } else {
snprintf(to, sizeof(to), "<%s>", p->uri); snprintf(to, sizeof(to), "<%s>", p->uri);
} }
memset(req, 0, sizeof(struct sip_request)); memset(req, 0, sizeof(struct sip_request));
init_req(req, sipmethod, p->uri); init_req(req, sipmethod, p->uri);
@ -4529,7 +4529,10 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
/* Bump branch even on initial requests */ /* Bump branch even on initial requests */
p->branch ^= rand(); p->branch ^= rand();
build_via(p, p->via, sizeof(p->via)); 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 } else
reqprep(&req, p, sipmethod, 0, 1); reqprep(&req, p, sipmethod, 0, 1);
@ -4792,7 +4795,7 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
char *t = tmp; char *t = tmp;
size_t maxbytes = sizeof(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, "Event", "message-summary");
add_header(&req, "Content-Type", default_notifymime); add_header(&req, "Content-Type", default_notifymime);
@ -8281,7 +8284,7 @@ static int sip_notify(int fd, int argc, char *argv[])
continue; continue;
} }
initreqprep(&req, p, SIP_NOTIFY); initreqprep(&req, p, SIP_NOTIFY);
for (var = varlist; var; var = var->next) for (var = varlist; var; var = var->next)
add_header(&req, var->name, var->value); add_header(&req, var->name, var->value);
@ -10654,9 +10657,9 @@ static int sip_poke_peer(struct sip_peer *peer)
ast_set_flag(p, SIP_OUTGOING); ast_set_flag(p, SIP_OUTGOING);
#ifdef VOCAL_DATA_HACK #ifdef VOCAL_DATA_HACK
ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username)); 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 #else
transmit_invite(p, SIP_OPTIONS, 0, 1); transmit_invite(p, SIP_OPTIONS, 0, 2);
#endif #endif
gettimeofday(&peer->ps, NULL); gettimeofday(&peer->ps, NULL);
peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer); peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);

Write Preview
Loading…
Cancel
Save