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chan_pjsip: Assign SIPDOMAIN after creating a channel

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session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.

This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.

ASTERISK-29240

Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755
This commit is contained in:
Ivan Poddubnyi
2020-12-29 19:16:00 +01:00
committed by George Joseph
parent 134d2e729d
commit f2aa6c7017
2 changed files with 16 additions and 7 deletions
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15
channels/chan_pjsip.c
View File

@@ -2980,6 +2980,18 @@ static void chan_pjsip_session_end(struct ast_sip_session *session)
SCOPE_EXIT_RTN(); SCOPE_EXIT_RTN();
} }
static void set_sipdomain_variable(struct ast_sip_session *session)
{
pjsip_sip_uri *sip_ruri = pjsip_uri_get_uri(session->request_uri);
size_t size = pj_strlen(&sip_ruri->host) + 1;
char *domain = ast_alloca(size);
ast_copy_pj_str(domain, &sip_ruri->host, size);
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
return;
}
/*! \brief Function called when a request is received on the session */ /*! \brief Function called when a request is received on the session */
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata) static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{ {
@@ -3031,6 +3043,9 @@ static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct p
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n", SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
ast_sip_session_get_name(session)); ast_sip_session_get_name(session));
} }
set_sipdomain_variable(session);
/* channel gets created on incoming request, but we wait to call start /* channel gets created on incoming request, but we wait to call start
so other supplements have a chance to run */ so other supplements have a chance to run */
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session)); SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));

8
res/res_pjsip_session.c
View File

@@ -3733,7 +3733,7 @@ static enum sip_get_destination_result get_destination(struct ast_sip_session *s
*/ */
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten); AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
pickup_cfg = ast_get_chan_features_pickup_config(session->channel); pickup_cfg = ast_get_chan_features_pickup_config(NULL); /* session->channel doesn't exist yet, using NULL */
if (!pickup_cfg) { if (!pickup_cfg) {
ast_log(LOG_ERROR, "%s: Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n", ast_log(LOG_ERROR, "%s: Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n",
ast_sip_session_get_name(session)); ast_sip_session_get_name(session));
@@ -3745,12 +3745,6 @@ static enum sip_get_destination_result get_destination(struct ast_sip_session *s
if (!strcmp(session->exten, pickupexten) || if (!strcmp(session->exten, pickupexten) ||
ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) { ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) {
size_t size = pj_strlen(&sip_ruri->host) + 1;
char *domain = ast_alloca(size);
ast_copy_pj_str(domain, &sip_ruri->host, size);
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
/* /*
* Save off the INVITE Request-URI in case it is * Save off the INVITE Request-URI in case it is
* needed: CHANNEL(pjsip,request_uri) * needed: CHANNEL(pjsip,request_uri)