[Freeswitch-users] calling multiple destinations
Francis
sms at icefire.qza.net.au
Mon May 26 02:49:01 MSD 2014
Try adding this:
<action application="set" data="ignore_early_media=true"/>
Looking at the log snippet you posted earlier, I suspect this *might* be
the cause.
On 2014-05-26 12:57, Vincent Xia wrote:
> the entire dialplan extension is something like:
> <extension name="simul_call" continue="false">
> <condition field="destination_number" expression="^(b1d{3}b)$">
> <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features" />
> <action application="set" data="continue_on_fail=true" />
> <action application="set" data="transfer_ringback=$${hold_music}" />
> <action inline="true" application="lua" data="get_callee.lua" />
> </condition>
> <condition field="${is_call_allowed}" expression="allowed" continue="false">
> <action application="bridge" data="${callee1},${callee2}" />
> <action application="answer" />
> <action application="sleep" data="1000" />
> </condition>
> </extension>
> (the variables callee1 and callee2 are assigned within get_callee.lua)
>
> it works fine with multiple linphone endpoints, but not with some others as sipdroid,
> maybe in these scenarios the FS is acting differently against 180 and 183 response from the endpoints,
> any ideas?
>
> 2014-05-24 19:56 GMT+08:00 Francis <sms at icefire.qza.net.au>:
>
>> Hi Vincent, the action looks correct, what else is in this extension?
>> Also, (this is a trap that one of my users created) are there any active
>> forward or divert settings on the endpoints themselves?
>>
>> - Francis
>>
>> On 23/05/2014 8:34 PM, Vincent Xia wrote:
>>> i was trying to ring multiple endpoints (A calls B1 and B2
>>> simultaneously) by doing
>>> <action application="bridge" data="user/${callee1},user/${callee2}"/>
>>> i was expecting one callee hung up once the other answers the call,
>>> but what i was experiencing is one hung up once the other rings, the
>>> FS console log says:
>>>
>>> [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1 at 1.2.3.4!
>>> [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2
>>> [CS_CONSUME_MEDIA] [LOSE_RACE]
>>>
>>> so how can i make the calls happen as expected?
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com [1]
>>
>>
>> [2]
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org [3]
>> http://wiki.freeswitch.org [4]
>> http://www.cluecon.com [5]
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com [1]
>
>
> [2]
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org [3]
> http://wiki.freeswitch.org [4]
> http://www.cluecon.com [5]
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
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[2]
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[5] http://www.cluecon.com
[6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[7] http://lists.freeswitch.org/mailman/options/freeswitch-users
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