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

[Svetik VOIP]

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,
>>
>> Thank you 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_sessi
>> >> on) 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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>>
>
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