[Freeswitch-users] Using presence or state to avoid intercom barge-in

Yehavi Bourvine yehavi.bourvine at gmail.com
Sun Oct 9 12:22:03 MSD 2016


Hello Phil,

  Here I use the "limit" application to avoid such situation. This means
that you have to add calls to "limit" whenever you establish a call. Once
this is implemened, you can use this call to check whether the "parent
extension" is busy before doing the "bridge" call.
If this is acceptable approach for you, then I can send you some dialplan
fragments that I am using here for exactly this situation.

           Regards, __Yehavi:

2016-09-01 22:20 GMT+03:00 Phil Quesinberry <philq at qsystemsengineering.com>:

> Hi,
>
> We’re running into an issue with Cisco SPA50x phones where an intercom
> call to an extension either places the currently active call on hold or
> hangs up on it and switches to the intercom call.  We’d like to not have
> this happen if the destination extension is in use.
>
> The only way I know of to do this would involve reconfiguring all of the
> phones to share presence and then check it in the dialplan before
> bridging.  Is there a convenient “no-barge” variable or similar that can be
> set so FS won’t send the invite if the extension is in use or a way to
> check the extension’s state from the dialplan?  I’m guessing that’s a NO
> but I had to ask!
>
> Alternatively, is there a way to get the Cisco phones not to interrupt the
> call in progress when a call comes in?  Instead of setting
> sip_auto_answer=true, I’ve also tried setting sip_h_Call-Info=<sip:ip.
> address.of.server>;answer-after=0 which appears to stop this behavior on
> the Aastra phones but not the Cisco phones, although I’m not entirely sure
> about this as I now can’t seem to reproduce the undesired behavior on the
> Aastra phones...
>
> Dialplan entry for the intercom as it stands is as follows:
>
> <extension name="extension-intercom-fix" continue="true">
>
>    <condition field="destination_number" expression="^\*8(\d{2,7})$" >
>
>        <action application="set" data="dialed_extension=$1" />
>
>        <action application="bridge" data="user/${dialed_extension}@${domain_name}"
> />
>
>        <action application="export" data="sip_auto_answer=true" />
>
>    </condition>
>
> </extension>
>
> Any thoughts or advice would be appreciated.  Thanks!
>
> *Phil Quesinberry*
>
> Q Systems Engineering, Inc.
>
> Embedded Systems, Telecom, IT
>
> (410) 969-8002
>
> *http://www.qsystemsengineering.com* <http://www.qsystemsengineering.com/>
>
>
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