chan_dahdi: Add inband_on_setup_ack compatibility option.

The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.

Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP.  It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed.  This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.

NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.

The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.

This commit is a merge of the two patches indicated below.

ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
      pri-4.diff (license #6302) patch uploaded by Pavel Troller
      jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett

Review: https://reviewboard.asterisk.org/r/3633/
........

Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/UPGRADE.txt b/UPGRADE.txt
index 87256ef25a18acaec66a2fdfafa9391d6e24c784..42f9837d19ccd8d1e0f03080d3ea044af0c0ba2c 100644 (file)
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -35,16 +35,26 @@ From 12.3.2 to 12.4.0:
    - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
    - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
 
- - Added a compatibility option for ari, chan_sip, and chan_pjsip
-   'websocket_write_timeout'. When a websocket connection exists where Asterisk
-   writes a substantial amount of data to the connected client, and the connected
-   client is slow to process the received data, the socket may be disconnected.
-   In such cases, it may be necessary to adjust this value. Default is 100 ms.
+ARI:
+ - Added a compatibility option 'websocket_write_timeout'.  When a websocket
+   connection exists where Asterisk writes a substantial amount of data to
+   the connected client, and the connected client is slow to process the
+   received data, the socket may be disconnected.  In such cases, it may be
+   necessary to adjust this value.
+   Default is 100 ms.
 
- - Added support for persistent HTTP connections.  To enable persistent
-   HTTP connections configure the keep alive time between HTTP requests.  The
-   keep alive time between HTTP requests is configured in http.conf with the
-   session_keep_alive parameter.
+chan_dahdi:
+ - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
+   deal with switches that don't send an inband progress indication in the
+   SETUP ACKNOWLEDGE message.
+
+chan_pjsip:
+ - Added a compatibility option 'websocket_write_timeout'.  When a websocket
+   connection exists where Asterisk writes a substantial amount of data to
+   the connected client, and the connected client is slow to process the
+   received data, the socket may be disconnected.  In such cases, it may be
+   necessary to adjust this value.
+   Default is 100 ms.
 
  - Added a 'force_avp' option to chan_pjsip which will force the usage of
    'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
@@ -55,6 +65,14 @@ From 12.3.2 to 12.4.0:
    cause the SDP answer to use the media transport as received in the SDP
    offer.
 
+chan_sip:
+ - Added a compatibility option 'websocket_write_timeout'.  When a websocket
+   connection exists where Asterisk writes a substantial amount of data to
+   the connected client, and the connected client is slow to process the
+   received data, the socket may be disconnected.  In such cases, it may be
+   necessary to adjust this value.
+   Default is 100 ms.
+
  - Added a 'force_avp' option for chan_sip. When enabled this option will
    cause the media transport in the offer or answer SDP to be 'RTP/AVP',
    'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
@@ -72,6 +90,12 @@ From 12.3.2 to 12.4.0:
    hash to be specified for the DTLS fingerprint placed in SDP. Supported
    values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
 
+HTTP:
+ - Added support for persistent HTTP connections.  To enable persistent
+   HTTP connections configure the keep alive time between HTTP requests.  The
+   keep alive time between HTTP requests is configured in http.conf with the
+   session_keep_alive parameter.
+
 From 12.3.0 to 12.3.1:
 
  - MixMonitor AMI actions now require users to have authorization classes.

diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index ab6df48381453629dd1c3f1fe9e5bb284a5761ed..62164104d255a1fce3317478b94724c8b6ac1edf 100644 (file)
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -847,6 +847,7 @@ static struct dahdi_chan_conf dahdi_chan_conf_default(void)
                        .localdialplan = PRI_NATIONAL_ISDN + 1,
                        .nodetype = PRI_CPE,
                        .qsigchannelmapping = DAHDI_CHAN_MAPPING_PHYSICAL,
+                       .inband_on_setup_ack = 1,
 
 #if defined(HAVE_PRI_CCSS)
                        .cc_ptmp_recall_mode = 1,/* specificRecall */
@@ -12148,6 +12149,7 @@ static struct dahdi_pvt *mkintf(int channel, const struct dahdi_chan_conf *conf,
                                                        pris[span].pri.layer1_ignored = 0;
                                                }
                                                pris[span].pri.append_msn_to_user_tag = conf->pri.pri.append_msn_to_user_tag;
+                                               pris[span].pri.inband_on_setup_ack = conf->pri.pri.inband_on_setup_ack;
                                                pris[span].pri.inband_on_proceeding = conf->pri.pri.inband_on_proceeding;
                                                ast_copy_string(pris[span].pri.initial_user_tag, conf->chan.cid_tag, sizeof(pris[span].pri.initial_user_tag));
                                                ast_copy_string(pris[span].pri.msn_list, conf->pri.pri.msn_list, sizeof(pris[span].pri.msn_list));
@@ -17636,6 +17638,8 @@ static int process_dahdi(struct dahdi_chan_conf *confp, const char *cat, struct
 #endif /* defined(HAVE_PRI_MWI) */
                        } else if (!strcasecmp(v->name, "append_msn_to_cid_tag")) {
                                confp->pri.pri.append_msn_to_user_tag = ast_true(v->value);
+                       } else if (!strcasecmp(v->name, "inband_on_setup_ack")) {
+                               confp->pri.pri.inband_on_setup_ack = ast_true(v->value);
                        } else if (!strcasecmp(v->name, "inband_on_proceeding")) {
                                confp->pri.pri.inband_on_proceeding = ast_true(v->value);
 #if defined(HAVE_PRI_DISPLAY_TEXT)

diff --git a/channels/sig_pri.c b/channels/sig_pri.c
index 57c518c886b7fc010e453630ead5db0e354775e0..63cf776da20b24114062f4ff5ab825570ca25535 100644 (file)
--- a/channels/sig_pri.c
+++ b/channels/sig_pri.c
@@ -1629,6 +1629,9 @@ static int pri_fixup_principle(struct sig_pri_span *pri, int principle, q931_cal
 #if defined(HAVE_PRI_CALL_WAITING)
                new_chan->is_call_waiting = old_chan->is_call_waiting;
 #endif /* defined(HAVE_PRI_CALL_WAITING) */
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+               new_chan->no_dialed_digits = old_chan->no_dialed_digits;
+#endif /* defined(HAVE_PRI_SETUP_ACK_INBAND) */
 
 #if defined(HAVE_PRI_AOC_EVENTS)
                old_chan->aoc_s_request_invoke_id_valid = 0;
@@ -1644,6 +1647,9 @@ static int pri_fixup_principle(struct sig_pri_span *pri, int principle, q931_cal
 #if defined(HAVE_PRI_CALL_WAITING)
                old_chan->is_call_waiting = 0;
 #endif /* defined(HAVE_PRI_CALL_WAITING) */
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+               old_chan->no_dialed_digits = 0;
+#endif /* defined(HAVE_PRI_SETUP_ACK_INBAND) */
 
                /* More stuff to transfer to the new channel. */
                new_chan->call_level = old_chan->call_level;
@@ -7489,8 +7495,19 @@ static void *pri_dchannel(void *vpri)
                                         * We explicitly DO NOT want to check PRI_PROG_CALL_NOT_E2E_ISDN
                                         * because it will mess up ISDN to SIP interoperability for
                                         * the ALERTING message.
+                                        *
+                                        * Q.931 Section 5.1.3 says that in scenarios with overlap
+                                        * dialing where no called digits are received and the tone
+                                        * option requires dialtone, the switch MAY send an inband
+                                        * progress indication ie to indicate dialtone presence in
+                                        * the SETUP ACKNOWLEDGE.  Therefore, if we did not send any
+                                        * digits with the SETUP then we must assume that dialtone
+                                        * is present and open the voice path.  Fortunately when
+                                        * interoperating with SIP, we should be sending digits.
                                         */
-                                       && (e->setup_ack.progressmask & PRI_PROG_INBAND_AVAILABLE)
+                                       && ((e->setup_ack.progressmask & PRI_PROG_INBAND_AVAILABLE)
+                                               || pri->inband_on_setup_ack
+                                               || pri->pvts[chanpos]->no_dialed_digits)
 #endif /* defined(HAVE_PRI_SETUP_ACK_INBAND) */
                                        ) {
                                        /*
@@ -8120,7 +8137,12 @@ int sig_pri_call(struct sig_pri_chan *p, struct ast_channel *ast, const char *rd
        if (!keypad || !ast_strlen_zero(c + p->stripmsd + dp_strip))
 #endif /* defined(HAVE_PRI_SETUP_KEYPAD) */
        {
-               pri_sr_set_called(sr, c + p->stripmsd + dp_strip, pridialplan, s ? 1 : 0);
+               char *called = c + p->stripmsd + dp_strip;
+
+               pri_sr_set_called(sr, called, pridialplan, s ? 1 : 0);
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+               p->no_dialed_digits = !called[0];
+#endif /* defined(HAVE_PRI_SETUP_ACK_INBAND) */
        }
 
 #if defined(HAVE_PRI_SUBADDR)

diff --git a/channels/sig_pri.h b/channels/sig_pri.h
index ad6c4c5483e7a40af589efc29d2d079eafe18e77..2517730f20747d7573a93254236c88766b70ec6c 100644 (file)
--- a/channels/sig_pri.h
+++ b/channels/sig_pri.h
@@ -345,6 +345,10 @@ struct sig_pri_chan {
        /*! \brief TRUE if this is a call waiting call */
        unsigned int is_call_waiting:1;
 #endif /* defined(HAVE_PRI_CALL_WAITING) */
+#if defined(HAVE_PRI_SETUP_ACK_INBAND)
+       /*! TRUE if outgoing SETUP had no called digits */
+       unsigned int no_dialed_digits:1;
+#endif /* defined(HAVE_PRI_SETUP_ACK_INBAND) */
 
        struct ast_channel *owner;
 
@@ -483,6 +487,8 @@ struct sig_pri_span {
         * appended to the initial_user_tag[].
         */
        unsigned int append_msn_to_user_tag:1;
+       /*! TRUE if a SETUP ACK message needs to open the audio path. */
+       unsigned int inband_on_setup_ack:1;
        /*! TRUE if a PROCEEDING message needs to unsquelch the received audio. */
        unsigned int inband_on_proceeding:1;
 #if defined(HAVE_PRI_MCID)

diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample
index 12490fc926c6158e599906115dda2d48b2db2a57..bba9e456e68f8593527a451612c8e88c0fba634e 100644 (file)
--- a/configs/chan_dahdi.conf.sample
+++ b/configs/chan_dahdi.conf.sample
@@ -196,6 +196,23 @@ context=public
 ;
 ;resetinterval = 3600
 ;
+; Assume inband audio may be present when a SETUP ACK message is received.
+; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
+; dialtone is sent from the network side, progress indicator 8 "Inband info
+; now available" MAY be sent to the CPE if no digits were received with
+; the SETUP.  It is thus implied that the ie is mandatory if digits came
+; with the SETUP and dialtone is needed.
+; This option should be enabled, when the network sends dialtone and you
+; want to hear it, but the network doesn't send the progress indicator when
+; needed.
+;
+; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
+; dialing is also enabled because Q.SIG does not send the progress indicator
+; with the SETUP ACK.
+; Default yes in current release branches for backward compatibility.
+;
+;inband_on_setup_ack=yes
+;
 ; Assume inband audio may be present when a PROCEEDING message is received.
 ; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
 ; attached to the B channel at this time without explicitly sending the