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asterisk RTP透传相关参数整理

xiaot99
 xiaot99
发布于 2014/02/07 11:54
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1 directmedia/canreinvite

The "canreinvite" option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: "yes", "no", "nonat", "update".Please consult sip.conf.sample for detailed information.

canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does.See also the closely related setting directrtpsetup.

canreinvite = yes "allow RTP media direct"

canreinvite = no "deny re-invites"

canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"

canreinvite = update "use UPDATE instead of INVITE"

canreinvite = update,nonat "use UPDATE when local, deny when NAT"

2 directmediapermit/directmediadeny

Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other.

3 directrtpsetup=yes

directrtpsetup=yes is similar to directmedia=, except the audio is redirected in the initial INVITEs rather than reinviting the media a few RTP packets in. Note: canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does

4 media_address

configuration option which can be used to explicitly specify the IP address to use in the SDP for media (audio, video, and text) streams.

5 NOTICE

If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.

If the clients use different codecs, Asterisk will not issue a re-invite.

If the Dial() command contains ''t'', ''T", "h", "H", "w", "W" or "L" (with multiple arguments)

Asterisk will not issue a re-invite.

6 DIAL()

T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf). Does not affect transfers initiated through other methods.

    If you have set the variable GOTO_ON_BLINDXFR then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_BLINDXFR=woohoo^s^1); works with both t and T

t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf) Does not affect transfers initiated through other methods.

    If you have set the variable GOTO_ON_BLINDXFR then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_BLINDXFR=woohoo^s^1); works with both t and T

H: Allow the caller to hang up by dialing * ( * is defined in features.conf -> featuremap -> disconnect )

h: Allow the callee to hang up by dialing * ( * is defined in features.conf -> featuremap -> disconnect )

W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)

w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)

L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. Numbers must be integers- beware of AGI scripts that may return long integers in scientific notation (esp PHP 5.2.5&6) The following special variables are optional for limit calls: (pasted from app_dial.c)

    LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.

    LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.

    LIMIT_TIMEOUT_FILE - File to play when time is up.

    LIMIT_CONNECT_FILE - File to play when call begins.

    LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce ("You have [XX minutes] YY seconds").


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asterisk RTP透传相关参数整理

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