diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-05-02 20:31:39 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-05-02 20:31:39 +0000 |
| commit | f6dbbaaa8ecb3a4462048edfd5e8475493bdb2d4 (patch) | |
| tree | 41edb97dfa9f29a55bbf5bfbfca794fe6c0dc9a9 | |
| parent | 70b436e19748dbdb9cb6a0180852d4dbcffa3398 (diff) | |
- fix typo in rtp.c, devicestate.h
- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24342 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | configs/sip.conf.sample | 17 | ||||
| -rw-r--r-- | include/asterisk/devicestate.h | 2 | ||||
| -rw-r--r-- | rtp.c | 2 |
3 files changed, 14 insertions, 7 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 26966cd08..84200a53a 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -29,7 +29,6 @@ context=default ; Default context for incoming calls ; this can also be set to 'osp' ; if asterisk was compiled with OSP support.) allowoverlap=no ; Disable overlap dialing support. (Default is yes) -;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 @@ -114,10 +113,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;compactheaders = yes ; send compact sip headers. ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration -;subscribecontext = default ; Set a specific context for SUBSCRIBE requests - ; Useful to limit subscriptions to local extensions - ; Settable per peer/user also -;notifyringing = yes ; Notify subscriptions on RINGING state ; ;videosupport=yes ; Turn on support for SIP video ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) @@ -126,6 +121,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;callevents=no ; generate manager events when sip ua ; performs events (e.g. hold) +;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) --------- +; You can subscribe to the status of extensions with a "hint" priority +; (See extensions.conf.sample for examples) +; chan_sip support two major formats for notifications: dialog-info and SIMPLE +; Note: Subscriptions does not work if you have a realtime dialplan and use the +; realtime switch. +; +;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) +;subscribecontext = default ; Set a specific context for SUBSCRIBE requests + ; Useful to limit subscriptions to local extensions + ; Settable per peer/user also +;notifyringing = yes ; Notify subscriptions on RINGING state ; ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with diff --git a/include/asterisk/devicestate.h b/include/asterisk/devicestate.h index e41b9ee27..1892ab51b 100644 --- a/include/asterisk/devicestate.h +++ b/include/asterisk/devicestate.h @@ -79,7 +79,7 @@ int ast_device_state_changed(const char *fmt, ...) /*! \brief Tells Asterisk the State for Device is changed - * \param device devicename like a dialstrin + * \param device devicename like a dialstring * Asterisk polls the new extensionstates and calls the registered * callbacks for the changed extensions * Returns 0 on success, -1 on failure @@ -23,7 +23,7 @@ * * \author Mark Spencer <markster@digium.com> * - * \note RTP is deffined in RFC 3550. + * \note RTP is defined in RFC 3550. */ #include <stdio.h> |