[Freeswitch-dev] Problems with intercept application and attended transfer

[Anthony Minessale]

First of all attended transfer is already supported within the SIP itself by
the sip phone.
The intercept app is not for attended transfer.
If you insist on doing emulated attended transfer, there is an example in
the default config using the bind_meta_app function.



On Fri, Sep 5, 2008 at 10:09 AM, Francisco de Ezcurra <
francisco at deezcurra.com.ar> wrote:

> Hi
>
> I'm trying to make an attended transfer from the A leg of a bridge that was
> created using the intercept application. This works if the transfer is made
> from the B leg. I've run the tests on a Polycom 501.
>
> 1) Dialplan extensions:
>
>        <extension name="bleg">
>        <condition field="destination_number" expression="1005">
>          <action application="set_global" data="leg_b=${uuid}" />
>          <action application="set"
>
> data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/>
>          <action application="gentones" data="${us-ring}|-1" />
>        </condition>
>    </extension>
>
>    <extension name="aleg">
>        <condition field="destination_number" expression="1006">
>          <action application="set"
>
> data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/>
>          <action application="intercept" data="${leg_b}"/>
>        </condition>
>    </extension>
>
>        <extension name="2003">
>        <condition field="destination_number" expression="2003">
>          <action application="set"
>
> data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/>
>          <action application="bridge" data="sofia/default/user3.fezcurra"/>
>        </condition>
>    </extension>
>
>
> 2) Steps in which the attended transfer DOES NOT work.
>
>        1) With a linksys phone dial the extension 1005
>        2) With the Polycom 501 dial the extension 1006 which intercept the
> linksys
> channel.
>        3) With the Polycom 501 start an attended transfer to the extension
> 2003
>        4) The linksys starts playing the hold music.
>        5) Answer the extension 2003
>        6) Confirm the transfer and hang up the Polycom 501.
>
>        Instead of completed the transfer, FS plays the ring again on the
> linksys and
> the extension
>        2003 stays connected without any bridge.
>
>
> 3) Steps in which the attended transfer works.
>
>        1) With the Polycom 501 phone dial the extension 1005
>        2) With the Linksys dial the extension 1006 which intercepts the
> Polycom
> channel.
>        3) With the Polycom 501 start an attended transfer to the extension
> 2003
>        4) The linksys starts playing the hold music.
>        5) Answer the extension 2003
>        6) Confirm the transfer and hang up the Polycom 501.
>        7) The linksys and the extension 2003 are connected properly.
>
>
> Has anybody else experienced this problem ?
>
> Thanks
> Panchi
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20080908/35f6a06c/attachment.html