[Freeswitch-users] No ring tone while recording incoming call. Please help.

msc msc at freeswitch.org
Thu Sep 24 09:08:28 PDT 2009


On Wed, Sep 23, 2009 at 7:34 AM, Svetik VOIP <svetikvoip at gmail.com> wrote:

> Brian,
>
> Thank yo very much for your reply.
>
> I have tried to add transfer_ringback action, but it did not solve my
> problem.
> Destination phone is ringing, but the person who is calling does not hear
> ringing tone in hte handset.
>
> Is there anything in the logfile that can help you to identify the problem?
>
What kind of system is the calling party connected to? It looks like a 180
is sent out by FS:

2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
sip:main at 192.168.0.121:5060 entering state [proceeding][180]

At that point the server at the originating side *should* generate pretend
ringing for the calling phone. If that is not happening then you need to see
what's going on at the originating side. Is it a SIP provider?

-MC


>
> Closest I can see is:
> 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec
> Activation Success L16 at 8000hz 1 channel 20ms
> 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play
> Ringback Tone [%(2000,4000,440.0,480.0)]
> 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/
> 4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY]
> 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
> sip:main at 192.168.0.121:5060 entering state [proceeding][180]
> 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/
> sip:main at 192.168.0.121:5060!
> 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/
> 4163641113 at 67.205.74.164 entering state [terminated][487]
> 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/
> 4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL]
>
> Thank you,
>
> Igor
>
> >set ringback before record_session and also set transfer_ringback
> >because record_session causes an pre-answer.
> >
> >/b
> >
> >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
> >
> >> Hi,
> >>
> >> I have trouble recording incoming calls with FreeSwitch.
> >>
> >> I have followed the instruction from Misc. Dialplan Tools record
> >> session
> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session)
> >> It works well for outgoing calls, but I have the problem with
> >> incoming calls.
> >>
> >> The person who is calling does not hear ring tone, he hears just the
> >> silence until
> >> I pick up the phone. Everything else is working, we can talk,
> >> conversation is recorded.
> >>
> >> Here is a copy of my dialplan for incoming calls
> >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml
> >>
> >> <include>
> >>     <extension name="voipms">   <!-- your provider or any name you'd
> >> like to call it -->
> >>         <condition field="destination_number"
> >> expression="XXXXXXXXXX">  <!-- your DID for this gateway-->
> >>             <action application="set" data="RECORD_TITLE=Recording $
> >> {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:
> >> %M)}"/>
> >>             <action application="set" data="RECORD_COPYRIGHT=(c)
> >> 2009"/>
> >>             <action application="set"
> >> data="RECORD_SOFTWARE=FreeSwitch"/>
> >>             <action application="set"
> >> data="RECORD_ARTIST=FreeSwitch"/>
> >>             <action application="set"
> >> data="RECORD_COMMENT=FreeSwitch"/>
> >>             <action application="set" data="RECORD_DATE=${strftime
> >> (%Y-%m-%d %H:%M)}"/>
> >>             <action application="set" data="RECORD_STEREO=true"/>
> >>             <action application="set" data="RECORD_ANSWER_REQ=true"/>
> >>             <action application="set" data="ringback=${us-ring}"/>
> >>             <action application="record_session" data="$${base_dir}/
> >> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$
> >> {destination_number}_${caller_id_number}.wav"/>
> >>             <action application="bridge" data="user/user1@$
> >> {domain_name}"/>
> >>     </condition>
> >> </include>
> >
> >
>
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