[Freeswitch-users] SBC In-band DTMF

Wed Jan 16 16:15:54 MSK 2013

Following up on this.

We've tested a few more configurations and it seems that when we exec start_dtmf it is the first perhaps half a second of any sound file that is played on a back end PBX seems to be clipped.   If we download the same sound file and play it it is fine, if we use Freeswitch 1.0.6 (intstead of the stable version) it is also fine.

an example sounds file is www.ecn.net.au/example.gsm - this file played off an Astiersk PBX befind the Freeswitch SBC plays, however the first 1/2 second or is not heard by the caller (calling through the telco, routed off the SBC and answered in the backend PBX).  (if you listen the first part that says "please press 1 to" is not heard, but the rest of the file is.

Is there some Freeswitch configuration that may cause this?

Regards
Mark

________________________________________
From: support at ecn.net.au
Sent: Wednesday, 16 January 2013 10:55 PM
To: FreeSWITCH Users Help
Subject: RE: [Freeswitch-users] SBC In-band DTMF

Hi

This is what we now have, however there is an interesting side effect (major issues actally) we're getting when using start_dtmf on the aleg from the "dodgy" telco.  the oddity is that when start_dtmf is executed prior to bridge some audio content from the back end PBX's (behind the Freeswitch SBC) does not seem to transmit back through to the caller (on the aleg of the freeswitch SBC (for example play back of a canned audio file from an IVR on an Asterisk PBX behind the SBC).

In testing Freeswitch on version 1.2 (1.2.5 and 1.2.3) and 1.3 testing all versions have this issue, however on an old 1.0.6 legacy version that we compiled today -  does not produce the same problem (it works perfectly); could it be a bug in the 1.2/3 freeswitch do you think?  Or is it something we've missed?

To reconfirm:

context A (telco context) we have a sip profile (dtmf_mode=none),
context B (to.pbx context) we have a sip profile (dtmf_mode=rfc2833)

On the xml dial plan for context A we start_dtmf before we bridge the call from the telco to the backend pbx.

When executing start_dtmf the system correctly transmits the dtmf digits to the backend pbx's however the by product (which we earlier thought was clipping the start of the call) is that some audio content (such as some audio file playback) doesn't get heard by the caller (dialed via the telco through the SBC to the pbx).

Any help appreciated!  Is there a change in how start_dtmf effects calls from the 1.2 and onward versions?

Regards
Mark

________________________________________
From: jay binks [jaybinks at gmail.com]
Sent: Wednesday, 16 January 2013 7:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SBC In-band DTMF

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<mailto:support at ecn.net.au> <support at ecn.net.au<mailto: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<mailto:sofia/external/123456 at INTERNAL.PBX.IP:5060%22/>/external/123456 at INTERNAL.PBX.IP:5060"/<mailto: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<mailto: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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
Sincerely

Jay