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

Svetik VOIP svetikvoip at gmail.com
Wed Sep 23 07:34:21 PDT 2009


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 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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