[Asterisk-code-review] res pjsip: New endpoint option "refer blind progress" (asterisk[13])
Alexei Gradinari
asteriskteam at digium.com
Mon May 8 16:07:13 CDT 2017
Alexei Gradinari has uploaded a new change for review. ( https://gerrit.asterisk.org/5602 )
Change subject: res_pjsip: New endpoint option "refer_blind_progress"
......................................................................
res_pjsip: New endpoint option "refer_blind_progress"
This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
ASTERISK-26333 #close
Change-Id: Id606fbff2e02e967c02138457badc399144720f2
---
M CHANGES
A contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_refer.c
6 files changed, 68 insertions(+), 2 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/02/5602/1
diff --git a/CHANGES b/CHANGES
index d93903d..800b8b4 100644
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,15 @@
--- Functionality changes from Asterisk 13.15.0 to Asterisk 13.16.0 ----------
------------------------------------------------------------------------------
+res_pjsip
+------------------
+ * A new endpoint option "refer_blind_progress" was added to turn off notifying
+ the progress details on Blind Transfer. If this option is not set then
+ the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
+ On default is enabled.
+ Some SIP phones like Mitel/Aastra or Snom keep the line busy until
+ receive "200 OK".
+
res_rtp_asterisk
------------------
* Added the stun_blacklist option to rtp.conf. Some multihomed servers have
diff --git a/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py
new file mode 100644
index 0000000..9b0f6d4
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py
@@ -0,0 +1,30 @@
+"""add ps_endpoints.refer_blind_progress
+
+Revision ID: 86bb1efa278d
+Revises: 1d0e332c32af
+Create Date: 2017-05-08 16:52:36.615161
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '86bb1efa278d'
+down_revision = '1d0e332c32af'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+ ############################# Enums ##############################
+
+ # yesno_values have already been created, so use postgres enum object
+ # type to get around "already created" issue - works okay with mysql
+ yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+ op.add_column('ps_endpoints', sa.Column('refer_blind_progress', yesno_values))
+
+def downgrade():
+ op.drop_column('ps_endpoints', 'refer_blind_progress')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 59122b9..bc3b8d4 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -761,6 +761,8 @@
unsigned int rtcp_mux;
/*! Do we allow overlap dialling? */
unsigned int allow_overlap;
+ /*! Whether to notifies all the progress details on blind transfer */
+ unsigned int refer_blind_progress;
};
/*! URI parameter for symmetric transport */
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index d47815f..e3a1244 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -957,6 +957,14 @@
since they mandate this option's use.
</para></description>
</configOption>
+ <configOption name="refer_blind_progress" default="yes">
+ <synopsis>Whether to notifies all the progress details on blind transfer</synopsis>
+ <description><para>
+ Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK"
+ after REFER has been accepted. If set to <literal>no</literal> then asterisk
+ will not send the progress details, but immediately will send "200 OK".
+ </para></description>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 511ea41..77e31ab 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1940,6 +1940,7 @@
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "asymmetric_rtp_codec", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, asymmetric_rtp_codec));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtcp_mux", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, rtcp_mux));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_overlap", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allow_overlap));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "refer_blind_progress", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, refer_blind_progress));
if (ast_sip_initialize_sorcery_transport()) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
diff --git a/res/res_pjsip_refer.c b/res/res_pjsip_refer.c
index db50612..111c4b2 100644
--- a/res/res_pjsip_refer.c
+++ b/res/res_pjsip_refer.c
@@ -61,6 +61,8 @@
char *transferee;
/*! \brief Non-zero if the 100 notify has been sent */
int sent_100;
+ /*! \brief Whether to notifies all the progress details on blind transfer */
+ unsigned int refer_blind_progress;
};
/*! \brief REFER Progress notification structure */
@@ -372,6 +374,8 @@
ast_debug(3, "Created progress monitor '%p' for transfer occurring from channel '%s' and endpoint '%s'\n",
progress, ast_channel_name(session->channel), ast_sorcery_object_get_id(session->endpoint));
+ (*progress)->refer_blind_progress = session->endpoint->refer_blind_progress;
+
(*progress)->framehook = -1;
/* To prevent a potential deadlock we need the dialog so we can lock/unlock */
@@ -563,6 +567,8 @@
pjsip_replaces_hdr *replaces;
/*! \brief Optional Refer-To header */
pjsip_sip_uri *refer_to;
+ /*! \brief Attended transfer flag */
+ unsigned int attended:1;
};
/*! \brief Blind transfer callback function */
@@ -577,8 +583,16 @@
pbx_builtin_setvar_helper(chan, "SIPTRANSFER", "yes");
- /* If progress monitoring is being done attach a frame hook so we can monitor it */
- if (refer->progress) {
+ if (refer->progress && !refer->attended && !refer->progress->refer_blind_progress) {
+ /* If blind transfer and endpoint doesn't want to receive all the progress details */
+ struct refer_progress_notification *notification = refer_progress_notification_alloc(refer->progress, 200,
+ PJSIP_EVSUB_STATE_TERMINATED);
+
+ if (notification) {
+ refer_progress_notify(notification);
+ }
+ } else if (refer->progress) {
+ /* If attended transfer and progress monitoring is being done attach a frame hook so we can monitor it */
struct ast_framehook_interface hook = {
.version = AST_FRAMEHOOK_INTERFACE_VERSION,
.event_cb = refer_progress_framehook,
@@ -785,6 +799,7 @@
refer.rdata = rdata;
refer.replaces = replaces;
refer.refer_to = target_uri;
+ refer.attended = 1;
if (ast_sip_session_defer_termination(session)) {
ast_log(LOG_ERROR, "Received REFER for remote session on channel '%s' from endpoint '%s' but could not defer termination, rejecting\n",
@@ -839,6 +854,7 @@
refer.progress = progress;
refer.rdata = rdata;
refer.refer_to = target;
+ refer.attended = 0;
if (ast_sip_session_defer_termination(session)) {
ast_log(LOG_ERROR, "Channel '%s' from endpoint '%s' attempted blind transfer but could not defer termination, rejecting\n",
--
To view, visit https://gerrit.asterisk.org/5602
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: Id606fbff2e02e967c02138457badc399144720f2
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexei Gradinari <alex2grad at gmail.com>
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