[Freeswitch-users] Originate call into conference - no early media

Mon Apr 23 23:29:11 MSD 2012

Excellent description and cool new chan var.

Alex, this is definitely something you should put on the wiki. If you need
any help please let me know. Don't forget that we have a template that you
can use to add a channel variable:

http://wiki.freeswitch.org/wiki/Variable_skeleton

The rest of the documentation can go in
http://wiki.freeswitch.org/wiki/Mod_loopback. For bonus points you can use
the wiki {{:foo}} syntax to include the channel variable docs in the
mod_loopback page w/o actually duplicating your work. See the main channel
variables page for examples on how to do that.

-MC

On Mon, Apr 23, 2012 at 12:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You should definitely help on the wiki.  It's always worth it.
>
> I did not mean that the book would teach you about L16 on loopback but
> codecs and setting up calls.
> Since you said you cannot understand why there is no early media in
> your conference I thought you might benefit but studying the nuances
> of call origination.
>
> As for loopback, if its important enough to you to get that ring
> sound, its really the only choice you have.
> I check and it's easy enough to add a patch to let you specify the
> initial codec so try out git HEAD and use the new variable
> loopback_initial_codec you can specify it in the originate string like
>
> originate {loopback_initial_codec=L16 at 16000h}loopback/1234 4321
>
> Transcoding fees may apply.
>
>
>
>
> On Mon, Apr 23, 2012 at 12:29 PM, Alex Crow <acrow at integrafin.co.uk>
> wrote:
> > On 23/04/12 16:09, Anthony Minessale wrote:
> >> If you originate loopback channels in that manner it chooses raw PCM
> >> (L16) because it has no idea what other codec to use.
> >> You want to avoid loopback unless its absolutely necessary since it
> >> adds extra resource consumption dynamically allocating the audio for
> >> each frame.
> >>
> >> I suggest you study the books we have (not a plug, just that's where
> >> its best documented) To get the full descriptions of how the dialplan
> >> and codec stuff works.  It's very complicated and hard to describe in
> >> a simple reply.
> >
> >
> > Thanks Anthony,
> >
> > I do have the FreeSwitch book and just got the cookbook. However I'd not
> > come across any docs stating that loopback defaults to L16/8000 so far,
> > so perhaps I should reread them.
> >
> > Is there any reason why loopback can't use L16/16000 or higher so HD
> > codecs can be of advantage? Just curious as my new dialplan seems to
> > have solved the problem. For anyone else that is interested I have
> > pasted it below (it is slightly adapted from the example):
> >
> > <extension name="confringback">
> > <condition field="destination_number" expression="^confringback$">
> > <action application="set" data="ringback=$${uk-ring}"/>
> > <action application="bridge"
> > data="{ignore_early_media=true}loopback/confringback_media"/>
> > </condition>
> > </extension>
> >
> > <extension name="confringback_media">
> > <condition field="destination_number" expression="^confringback_media$">
> > <action application="pre_answer"/>
> > <action application="sleep" data="200000"/>
> > <action application="hangup"/>
> > </condition>
> > </extension>
> >
> > <extension name="Add new OB call to conference">
> > <condition field="destination_number" expression="^add_a_call$">
> > <!-- ask caller for a number + #, collect into ${target_num} variable -->
> > <action application="play_and_get_digits" data="3 13 4 5000 #
> >
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> > <!-- add this call to the conference -->
> > <action application="execute_extension"
> > data="ADD_CALL_TO_CONF__${target_num}"/>
> > </condition>
> > </extension>
> >
> > <extension name="Remove last OB call added to conference">
> > <condition field="destination_number" expression="^remove_a_call$">
> > <!-- remove a call from the conference -->
> > <action application="play_and_get_digits" data="3 13 1 5000 #
> >
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> > <action application="set" data="res=${uuid_kill
> > ${hash(select/domain-${domain_name}/last_user_${target_num})}}"/>
> > <action application="set" data="res=${uuid_kill
> > ${hash(select/domain-${domain_name}/last_user_ring_${target_num})}}"/>
> > </condition>
> > </extension>
> >
> > <extension name="add that call">
> > <!-- if we have a four-digit number then attempt to dial it as a user
> > ... -->
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/confringback
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(6\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.43/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> >
> expression="^ADD_CALL_TO_CONF__(49\d\d|5[3-4]\d\d|74\d\d|8\d{3}|0|9\d{3,13}|0\d{10,12})$"
> > break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/*9181
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> > ${pb_uuid}'}sofia/gateway/10.10.0.2/$1&conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(3\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.15/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/confringback
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> > ${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
> > </condition>
> > <!-- ... otherwise inform moderator that the operation was not exactly
> > successful -->
> > <condition field="destination_number" expression="^ADD_CALL_TO_CONF__$">
> > <action application="playback" data="ivr/ivr-please_try_again.wav"/>
> > </condition>
> > </extension>
> >
> >
> > So for 3 destinations (1st local extensions at 1xxx or 11xxxx, 2nd
> > 350-399 and 3rd 600-649) we don't need to fake it, for the other 2
> > gateways we do.
> >
> > Seems to work pretty well for now. Would it be worth getting Wiki write
> > access to add that to the article about joining members by calling out
> > from a conference?
> >
> > Cheers
> >
> > Alex
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
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>
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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