[Freeswitch-users] FXO and analogue phones

[Michael Collins]

Merul,

My apologies, I had meant to follow up much earlier. Would you mind going
ahead and capturing full debug log output of a call in as well as a call
out? Please use pastebin.freeswitch.org to post the logs and then reply here
with the link to the pb post.

Thanks,
MC

On Sun, Aug 23, 2009 at 10:27 AM, Merul Patel <merul at mac.com> wrote:

> I have a Freeswitch setup working on an Alix embedded platform in
> conjunction with a USB FXO device from Sangoma. My goal is to be able to
> either answer incoming calls on a softphone or on a POTS handset elsewhere
> in the building, and to also be able to make outgoing calls from either. For
> clarity, the analogue line has two physical extensions, one connected to the
> POTS and the other to the FXO.
>
> I can make and receive calls fine, but have problems when the call is
> answered on the POTS handset.
>
> Here is the dialplan I initially used  in
> /opt/freeswitch/conf/dialplans/public/01_incoming.xml:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="sleep" data="23000"/>
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="transfer" data="1001 XML default"/>
>    </condition>
>  </extension>
> </include>
>
> It's pretty basic, and if the softphone is not registered or does not
> answer then the call goes to voicemail. However the call will always go to
> voicemail, and the voicemail application will begin to execute after the
> call has been answered on the POTS handset.
>
> I've been trying to make the dialplan more useful, by having it ring the
> softphone immediately, and only transfer the call to the voicemail
> application if the line is still ringing. I'm in the UK, hence my choice of
> frequencies in the tone_detect application:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="set" data="call_timeout=23"/>
>      <action application="set" data="continue_on_fail=true"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="bridge" data="sofia/internal/1001%$${domain}"/>
>      <action application="sleep" data="23000"/>
>      <action application="tone_detect" data="ring 400,450 r +5000 set
> RING=true"/>
>      <action application="transfer" data="public_answer_and_email"/>
>    </condition>
>  </extension>
>
> <extension name="public_answer_and_email">
>    <condition field="RING" expression="true">
>       <action application="answer"/>
>       <action application="voicemail" data="default $${domain} 1001"/>
>    </condition>
>  </extension>
> </include>
>
> Unfortunately, this is not working, and the logs are not yielding anything
> the is helpful to me. Is my use of the tone_detect application and the basic
> dialplan correct?
>
> Merul
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