Fix a crash caused by freeing a dialog directly instead of using dialog_unref.

(closes issue #16097)
Reported by: steinwej
Patches:
      no_RTP.diff uploaded by steinwej (license 841)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Joshua Colp 17 years ago
parent c24d9bdb6d
commit 8d5fab9e52

@ -6218,7 +6218,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
return NULL;
if (ast_string_field_init(p, 512)) {
ast_free(p);
dialog_unref(p);
return NULL;
}
@ -6278,27 +6278,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
p->t38_maxdatagram = global_t38_maxdatagram;
}
if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
if (p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
|| (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno));
if (p->rtp) {
ast_rtp_destroy(p->rtp);
}
if (p->vrtp) {
ast_rtp_destroy(p->vrtp);
}
if (p->udptl) {
ast_udptl_destroy(p->udptl);
}
ast_mutex_destroy(&p->pvt_lock);
if (p->chanvars) {
ast_variables_destroy(p->chanvars);
p->chanvars = NULL;
}
ast_string_field_free_memory(p);
ast_free(p);
dialog_unref(p);
return NULL;
}
ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP");

Loading…
Cancel
Save