mod_sofia will now examine a variable in the channel to
see what the channel's originator was using for a codec and
try to put that to the top of the list in the sdp.
if this new sofia profile param is set:
<param name="disable-transcoding" value="true"/>
All outbound calls will use *only* the codec that thier originator
is using to ensure no transcoding.
(of course that could lead to a failed call where there is no way to do this, so use sparingly)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4073 d0543943-73ff-0310-b7d9-9358b9ac24b2
1) The xml_curl now has a more enterprise config where it can have more than 1
url configured so you could have failover. (*note the syntax change*)
2) dialplan modules now take an extra arguement making it possible to pass runtime params to
them. This is now used in mod_dialplan_xml to allow an alternate file path to be specified.
dialplans were already stackable meaning you can configure a sofia profile, for example,
to use enum followed by the default XML dialplan.
e.g. <param name="dialplan" value="enum,XML"/>
From now on, you can also specify :param after each dialplan name to allow param
to be passed to the module. mod_dialplan_xml uses this param as a way to override
where it looks for the dialplan making it possible to stack mutiple calls to the XML dialplan.
e.g. <param name="dialplan" value="XML:/some/xml/file.xml,XML"/>
With this you can search the local file file.xml first and if there is still no match
the hunt will move on to the standard XML using the onboard XML registry and or the external
gateways.
*NOTE* this alternate path does not use the external bindings but it does parse the #includes etc.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4066 d0543943-73ff-0310-b7d9-9358b9ac24b2
This updates mod_portaudio to use the new v19 api and also contains
major behavioural changes. This initial check-in should be tested to find
any obscure use cases that lead to crashes etc...
All of the old api interface commands are now depricated and any attempt to
use them should cause a polite warning asking you to try the new single "pa" command.
New Features:
*) Mulitiple calls with hold/call switching.
*) Inbound calls can play a ring file on specified device. (global and per call)
*) Optional hold music for backgrounded calls. (global and per call)
Example dialplan usage:
<extension name="2000">
<condition field="destination_number" expression="^2000$">
<!--if the next 3 lines are omitted the defaults will be used from portaudio.conf-->
<action application="set" data="pa_ring_file=/sounds/myring.wav"/>
<action application="set" data="pa_hold_file=/sounds/myhold.wav"/>
<action application="set" data="export_vars=pa_ring_file,pa_hold_file"/>
<action application="bridge" data="portaudio"/>
</condition>
</extension>
Example API interface usage:
call extension 1000
> pa call 1000
call extension 1001 putting the other call on hold
> pa call 1001
swap the calls between hold and active
> pa switch
view the current calls
> pa list
forground the call with id 1
> pa switch 1
background all calls
> pa switch none
send a dtmf string (1234) to the current call
> pa dtmf 1234
answer the oldest unanswered inbound call
> pa answer
answer the call with id 1
> pa answer 1
hangup the active call
> pa hangup
hangup the call with id 1
> pa hangup 1
get device info
> pa dump
print usage summary
> pa help
USAGE:
--------------------------------------------------------------------------------
pa help
pa dump
pa call <dest> [<dialplan> <cid_name> <cid_num> <rate>]
pa answer [<call_id>]
pa hangup [<call_id>]
pa list
pa switch [<call_id>|none]
pa_dtmf <digit string>
--------------------------------------------------------------------------------
The source of the portaudio v19 library will also be checked in for the
sake of the build system.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3981 d0543943-73ff-0310-b7d9-9358b9ac24b2
