[Freeswitch-users] SBC In-band DTMF

jay binks jaybinks at gmail.com
Wed Jan 16 12:17:27 MSK 2013


on your internal sip profile ( one facing PBX's ) set  <param
name="dtmf-type" value="rfc2833"/>

then note the dialplan context that super dodgey sip carrier ( the one
doing inband only )
sends calls to, and use <action application="start_dtmf" /> on calls from
them.

Id advise against <action application="pre_answer"/> in that location
unless you REALLY want that.

that should do what you want.


On 16 January 2013 14:02, support at ecn.net.au <support at ecn.net.au> wrote:

>  Hi All
>
>
>
> We're quite new to Freeswitch and are in the process of migrating from
> OpenSer (as an SBC) to Freeswitch.
>
>
>
> Mostly all is working well, except an oddity on DTMF.
>
>
>
> Our scenario:
>
>
>
> Telco/SIP Provider A is passing us calls using DTMF inband.
>
>
>
> We have a freeswitch configured as a SBC using 2 sip profiles (telco and
> internal) to topology hide and manage
>
> distribution of calls to the PBX servers located behind the SBC.
>
>
>
> The freeswitch will be handling up to a few hundred calls so we're trying
> to keep it lightweight.
>
>
>
> Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling
> customer needs.
>
>
>
> An example inbound call profile looks like this:
>
>
>
> <extension name="Inbound 124356">
>
>         <condition field="destination_number" expression="^(123456)$">
>
>                 <action application="pre_answer"/>
>
>                 <action application="start_dtmf" />
>
>                 <action application="bridge" data="sofia<sofia/external/123456 at INTERNAL.PBX.IP:5060%22/>
> /external/123456 at INTERNAL.PBX.IP:5060"/<sofia/external/123456 at INTERNAL.PBX.IP:5060%22/>
> >
>
>         </condition>
>
> </extension>
>
>
>
> Initially when calling into the platform IVR type applications runinng on
> our PBX boxes would not
>
> work (you could hear the DTMF but the platform did not recognise the
> tones).
>
>
>
> We have had to add the appliation start_dtmf in order for Freeswitch to
> pass the DTMF to the Asterisk
>
> PBX behind the SBC.   Interestingly on our OpenSer platform we just
> proxied the media (rtpproxy) with
>
> inband DTMF from the Telco and our PBX boxes recognised the inband DTMF
> tones on the PBX platforms and
>
> IVR type applications just worked.
>
>
>
> However under freeswitch if we don't start_dtmf before the bridge the
> backend PBX boxes don't recognise
>
> the DTMF inband (even though the tones are audible ie you can hear them on
> a call recording on the
>
> PBX).
>
>
>
> Have we missed something here?  We would have thought with inband DTMF on
> non compressed codec (no
>
> transcoding) that the tones would just work with the media stream?
>
>
>
> We have confirmed both legs are PCMA and when using start_dtmf the first
> second of the call is clipped.
>
>
>
>
>
>
>  Kind Regards,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>


-- 
Sincerely

Jay
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