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[svetikvoip at gmail.com]

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>

for outcoming calls I have a similar code added to the
/usr/local/freeswitch/conf/dialplan/default/user1.xml and it works well.


I have tried to move the line
<action application="set" data="ringback=${us-ring}"/>
between the lines
<action application="record_session
and
<action application="bridge"
but it did not solve my problem.

Any ideas what am I doing wrong and how to fix it?

Igor

[brian at freeswitch.org]

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:

[svetikvoip at gmail.com]

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?

Closest I can see is:
2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec Activation Success L16@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@67.205.74.164 (4163641113@67.205.74.164) receive message [TRANSCODING_NECESSARY]
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/sip:main@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@192.168.0.121:5060!
2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/4163641113@67.205.74.164 (4163641113@67.205.74.164) entering state [terminated][487]
2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/4163641113@67.205.74.164 (4163641113@67.205.74.164) [CS_EXECUTE] [ORIGINATOR_CANCEL]

Thank you,

Igor

[msc at freeswitch.org]

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

[svetikvoip at gmail.com]

Hi guys,

I have solve my problem by adding
<action application="ring_ready" />
before
<action application="set" data="ringback=${us-ring}"/>

I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and noticed this line. Tried it and it works like a charm.

Thanks everybody, especially Brian and MC.

Igor

[brian at freeswitch.org]

Doesn't make sense because ring_ready sends only a 180.... set ringback then pre_answer would make use do 183.

/b

On Sep 25, 2009, at 1:27 PM, Svetik VOIP wrote: