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chan_pjsip: Register for "BEFORE_MEDIA" responses 

chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses.  If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".

* Removed chan_pjsip_incoming_response from the original session
  supplement (which was handling only "AFTER MEDIA") and added it to a
  new session supplement which accepts both "BEFORE_MEDIA" and
  "AFTER_MEDIA".

* Also cleaned up some cleanup code in load module.

ASTERISK-27902

Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a
This commit is contained in:
George Joseph 2018-06-07 07:46:03 -06:00
parent 7b5fc5d20f
commit 1725eaf8fb
1 changed files with 15 additions and 5 deletions
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20
channels/chan_pjsip.c
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@ -144,11 +144,18 @@ static struct ast_sip_session_supplement chan_pjsip_supplement = {
.session_begin = chan_pjsip_session_begin,
.session_end = chan_pjsip_session_end,
.incoming_request = chan_pjsip_incoming_request,
.incoming_response = chan_pjsip_incoming_response,
/* It is important that this supplement runs after media has been negotiated */
.response_priority = AST_SIP_SESSION_AFTER_MEDIA,
};
/*! \brief SIP session supplement structure just for responses */
static struct ast_sip_session_supplement chan_pjsip_supplement_response = {
.method = "INVITE",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
.incoming_response = chan_pjsip_incoming_response,
.response_priority = AST_SIP_SESSION_BEFORE_MEDIA | AST_SIP_SESSION_AFTER_MEDIA,
};
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
@ -3109,6 +3116,7 @@ static int load_module(void)
}
ast_sip_session_register_supplement(&chan_pjsip_supplement);
ast_sip_session_register_supplement(&chan_pjsip_supplement_response);
if (!(pjsip_uids_onhold = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK,
AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, 37, uid_hold_hash_fn,
@ -3123,10 +3131,6 @@ static int load_module(void)
if (pjsip_channel_cli_register()) {
ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
goto end;
}
@ -3142,6 +3146,11 @@ static int load_module(void)
end:
ao2_cleanup(pjsip_uids_onhold);
pjsip_uids_onhold = NULL;
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
@ -3160,6 +3169,7 @@ static int unload_module(void)
pjsip_channel_cli_unregister();
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);