This commit introduces COLP/CONP and Redirecting party information into Asterisk.

The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 109 insertions, 85 deletions

diff --git a/configs/misdn.conf.sample b/configs/misdn.conf.sample
index ca7f45497..e813953b3 100644
--- a/configs/misdn.conf.sample
+++ b/configs/misdn.conf.sample
@@ -7,13 +7,13 @@
 ; for debugging and general setup, things that are not bound to port groups
 ;
 
-[general] 
+[general]
 ;
 ; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
 ;
 misdn_init=/etc/misdn-init.conf
 
-; set debugging flag: 
+; set debugging flag:
 ;   0 - No Debug
 ;   1 - mISDN Messages and * - Messages, and * - State changes
 ;   2 - Messages + Message specific Informations (e.g. bearer capability)
@@ -26,8 +26,8 @@ debug=0
 
 
 
-; set debugging file and flags for mISDNuser (NT-Stack) 
-; 
+; set debugging file and flags for mISDNuser (NT-Stack)
+;
 ; flags can be or'ed with the following values:
 ;
 ; DBGM_NET        0x00000001
@@ -57,7 +57,7 @@ ntdebugflags=0
 ntdebugfile=/var/log/misdn-nt.log
 
 
-; some pbx systems do cut the L1 for some milliseconds, to avoid 
+; some pbx systems do cut the L1 for some milliseconds, to avoid
 ; dropping running calls, we can set this flag to yes and tell
 ; mISDNuser not to drop the calls on L2_RELEASE
 ntkeepcalls=no
@@ -76,26 +76,13 @@ ntkeepcalls=no
 bridging=no
 
 
-;
-; watches the L1s of every port. If one l1 is down it tries to 
-; get it up. The timeout is given in seconds. with 0 as value it
-; does not watch the l1 at all
-; 
-; default value: 0
-;
-; this option is only read at loading time of chan_misdn, 
-; which means you need to unload and load chan_misdn to change the 
-; value, an asterisk restart should do the trick
-; 
-l1watcher_timeout=0
-
 ; stops dialtone after getting first digit on nt Port
 ;
 ; default value: yes
 ;
 stop_tone_after_first_digit=yes
 
-; whether to append overlapdialed Digits to Extension or not 
+; whether to append overlapdialed Digits to Extension or not
 ;
 ; default value: yes
 ;
@@ -122,19 +109,6 @@ crypt_prefix=**
 ;
 crypt_keys=test,muh
 
-; users sections:
-; 
-; name your sections as you which but not "general" ! 
-; the sections are Groups, you can dial out in extensions.conf
-; with Dial(mISDN/g:extern/101) where extern is a section name, 
-; chan_misdn tries every port in this section to find a 
-; new free channel
-; 
-
-; The default section is not a group section, it just contains config elements
-; which are inherited by group sections.
-;
-
 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
 ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                               ; SIP channel. Defaults to "no". An enabled jitterbuffer will
@@ -161,6 +135,17 @@ crypt_keys=test,muh
 ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
 ;-----------------------------------------------------------------------------------
 
+; users sections:
+;
+; name your sections as you wish but not "general" or "default" !
+; the sections are Groups, you can dial out in extensions.conf
+; with Dial(mISDN/g:extern/101) where extern is a section name,
+; chan_misdn tries every port in this section to find a
+; new free channel
+;
+; The default section is not a group section, it just contains config elements
+; which are inherited by group sections.
+;
 [default]
 
 ; define your default context here
@@ -182,7 +167,7 @@ musicclass=default
 
 ;
 ; Either if we should produce DTMF Tones ourselves
-; 
+;
 senddtmf=yes
 
 ;
@@ -205,14 +190,26 @@ far_alerting=no
 ;
 allowed_bearers=all
 
-; Prefixes for national and international, those are put before the 
-; oad if an according dialplan is set by the other end. 
-;
-; default values: nationalprefix      : 0
-;                 internationalprefix : 00
-;
-nationalprefix=0
+; Prefixes for national and international Type-Of-Number.  These are
+; inserted before any number (caller, dialed, connected, redirecting,
+; redirection) received from the ISDN link if that number has the
+; correspondng Type-Of-Number.
+; See the dialplan options.
+;
+; default values:
+;    unknownprefix=
+;    internationalprefix=00
+;    nationalprefix=0
+;    netspecificprefix=
+;    subscriberprefix=
+;    abbreviatedprefix=
+;
+;unknownprefix=
 internationalprefix=00
+nationalprefix=0
+;netspecificprefix=
+;subscriberprefix=
+;abbreviatedprefix=
 
 ; set rx/tx gains between -8 and 8 to change the RX/TX Gain
 ;
@@ -222,7 +219,7 @@ internationalprefix=00
 rxgain=0
 txgain=0
 
-; some telcos especially in NL seem to need this set to yes, also in 
+; some telcos especially in NL seem to need this set to yes, also in
 ; switzerland this seems to be important
 ;
 ; default value: no
@@ -232,7 +229,20 @@ te_choose_channel=no
 
 
 ;
-; This option defines, if chan_misdn should check the L1 on  a PMP 
+; Monitors L1 of the port.  If L1 is down it tries
+; to bring it up.  The polling timeout is given in seconds.
+; Setting the value to 0 disables monitoring L1 of the port.
+;
+; default value: 0
+;
+; This option is only read at chan_misdn loading time.
+; You need to unload and load chan_misdn to change the
+; value.  An asterisk restart will also do the trick.
+;
+l1watcher_timeout=0
+
+;
+; This option defines, if chan_misdn should check the L1 on  a PMP
 ; before making a group call on it. The L1 may go down for PMP Ports
 ; so we might need this.
 ; But be aware! a broken or plugged off cable might be used for a group call
@@ -245,19 +255,19 @@ pmp_l1_check=no
 
 
 ;
-; in PMP this option defines which cause should be sent out to 
+; in PMP this option defines which cause should be sent out to
 ; the 3. caller. chan_misdn does not support callwaiting on TE
-; PMP side. This allows to modify the RELEASE_COMPLETE cause 
+; PMP side. This allows to modify the RELEASE_COMPLETE cause
 ; at least.
 ;
 reject_cause=16
 
 
 ;
-; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING), 
-; this requests additional Infos, so we can waitfordigits 
+; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
+; this requests additional Infos, so we can waitfordigits
 ; without much issues. This works only for PTP Ports
-; 
+;
 ; default value: no
 ;
 need_more_infos=no
@@ -280,29 +290,30 @@ nttimeout=no
 method=standard
 
 
-; specify if chan_misdn should collect digits before going into the 
+; specify if chan_misdn should collect digits before going into the
 ; dialplan, you can choose yes=4 Seconds, no, or specify the amount
 ; of seconds you need;
-; 
+;
 overlapdial=yes
 
 ;
-; dialplan means Type Of Number in ISDN Terms (for outgoing calls)
+; dialplan means Type Of Number in ISDN Terms
+; There are different types of the dialplan:
 ;
-; there are different types of the dialplan:
+; dialplan -> for outgoing call's dialed number
+; localdialplan -> for outgoing call's callerid
+;       (if -1 is set use the value from the asterisk channel)
+; cpndialplan -> for incoming call's connected party number sent to caller
+;       (if -1 is set use the value from the asterisk channel)
 ;
-; dialplan -> outgoing Number
-; localdialplan -> callerid
-; cpndialplan -> connected party number
-;
-; dialplan options: 
+; dialplan options:
 ;
 ; 0 - unknown
 ; 1 - International
 ; 2 - National
+; 3 - Network-Specific
 ; 4 - Subscriber
-;
-; This setting is used for outgoing calls
+; 5 - Abbreviated
 ;
 ; default value: 0
 ;
@@ -313,7 +324,7 @@ cpndialplan=0
 
 
 ;
-; turn this to no if you don't mind correct handling of Progress Indicators  
+; turn this to no if you don't mind correct handling of Progress Indicators
 ;
 early_bconnect=yes
 
@@ -321,16 +332,16 @@ early_bconnect=yes
 ;
 ; turn this on if you like to send Tone Indications to a Incoming
 ; isdn channel on a TE Port. Rarely used, only if the Telco allows
-; you to send indications by yourself, normally the Telco sends the 
+; you to send indications by yourself, normally the Telco sends the
 ; indications to the remote party.
-; 
+;
 ; default: no
 ;
 incoming_early_audio=no
 
 ; uncomment the following to get into s extension at extension conf
 ; there you can use DigitTimeout if you can't or don't want to use
-; isdn overlap dial. 
+; isdn overlap dial.
 ; note: This will jump into the s exten for every exten!
 ;
 ; default value: no
@@ -338,7 +349,7 @@ incoming_early_audio=no
 ;always_immediate=no
 
 ;
-; set this to yes if you want to generate your own dialtone 
+; set this to yes if you want to generate your own dialtone
 ; with always_immediate=yes, else chan_misdn generates the dialtone
 ;
 ; default value: no
@@ -346,9 +357,9 @@ incoming_early_audio=no
 nodialtone=no
 
 
-; uncomment the following if you want callers which called exactly the 
+; uncomment the following if you want callers which called exactly the
 ; base number (so no extension is set) jump to the s extension.
-; if the user dials something more it jumps to the correct extension 
+; if the user dials something more it jumps to the correct extension
 ; instead
 ;
 ; default value: no
@@ -369,6 +380,8 @@ nodialtone=no
 ;callgroup=1
 ;pickupgroup=1
 
+; Set the outgoing caller id to the value.
+;callerid="name" <number>
 
 ;
 ; these are the exact isdn screening and presentation indicators
@@ -376,11 +389,31 @@ nodialtone=no
 ; from asterisks CALLERPRES function.
 ; s=0, p=0 -> callerid presented
 ; s=1, p=1 -> callerid restricted (the remote end does not see it!)
-; 
+;
 ; default values s=-1, p=-1
 presentation=-1
 screen=-1
 
+; Put a display ie in the CONNECT message containing the following
+; information if it is available (nt port only):
+;
+; 0 - Do not put the connected line information in the display ie.
+; 1 - Put the available connected line name in the display ie.
+; 2 - Put the available connected line number in the display ie.
+; 3 - Put the available connected line name and number in the display ie.
+;
+display_connected=0
+
+; Put a display ie in the SETUP message containing the following
+; information if it is available (nt port only):
+;
+; 0 - Do not put the caller information in the display ie.
+; 1 - Put the available caller name in the display ie.
+; 2 - Put the available caller number in the display ie.
+; 3 - Put the available caller name and number in the display ie.
+;
+display_setup=0
+
 ; This enables echo cancellation with the given number of taps.
 ; Be aware: Move this setting only to outgoing portgroups!
 ; A value of zero turns echo cancellation off.
@@ -391,18 +424,9 @@ screen=-1
 ;
 ;echocancel=no
 
-; Set this to no to disable echotraining. You can enter a number > 10
-; the value is a multiple of 0.125 ms. 
-;
-; default value: no 
-; yes = 2000
-; no = 0
-;
-echotraining=no
-
 ;
 ; chan_misdns jitterbuffer, default 4000
-; 
+;
 jitterbuffer=4000
 
 ;
@@ -412,7 +436,7 @@ jitterbuffer_upper_threshold=0
 
 
 ;
-; change this to yes, if you want to bridge a mISDN data channel to 
+; change this to yes, if you want to bridge a mISDN data channel to
 ; another channel type or to an application.
 ;
 hdlc=no
@@ -420,8 +444,8 @@ hdlc=no
 
 ;
 ; defines the maximum amount of incoming calls per port for
-; this group. Calls which exceed the maximum will be marked with 
-; the channel variable MAX_OVERFLOW. It will contain the amount of 
+; this group. Calls which exceed the maximum will be marked with
+; the channel variable MAX_OVERFLOW. It will contain the amount of
 ; overflowed calls
 ;
 max_incoming=-1
@@ -433,7 +457,7 @@ max_incoming=-1
 max_outgoing=-1
 
 [intern]
-; define your ports, e.g. 1,2 (depends on mISDN-driver loading order) 
+; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
 ports=1,2
 ; context where to go to when incoming Call on one of the above ports
 context=Intern
@@ -445,21 +469,21 @@ context=Intern
 ; configs. For backwards compatibility you can still set ptp here.
 ;
 ports=3
-	
+
 [first_extern]
 ; again port defs
 ports=4
 ; again a context for incoming calls
 context=Extern1
-; msns for te ports, listen on those numbers on the above ports, and 
+; msns for te ports, listen on those numbers on the above ports, and
 ; indicate the incoming calls to asterisk
-; here you can give a comma separated list or simply an '*' for 
-; any msn. 
+; here you can give a comma separated list or simply an '*' for
+; any msn.
 msns=*
 
 ; here an example with given msns
 [second_extern]
 ports=5
 context=Extern2
-callerid=15
+callerid="Asterisk" <1234>
 msns=102,144,101,104