<br><br><div class="gmail_quote">On Wed, Sep 23, 2009 at 7:34 AM, Svetik VOIP <span dir="ltr"><<a href="mailto:svetikvoip@gmail.com">svetikvoip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Brian,<br><br>Thank yo very much for your reply.<br><br>I have tried to add transfer_ringback action, but it did not solve my problem.<br>Destination phone is ringing, but the person who is calling does not hear ringing tone in hte handset.<br>
<br>Is there anything in the logfile that can help you to identify the problem?<br></blockquote><div>What kind of system is the calling party connected to? It looks like a 180 is sent out by FS:<br><br>2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/<a href="http://sip:main@192.168.0.121:5060/" target="_blank">sip:main@192.168.0.121:5060</a> entering state [proceeding][180]<br>
<br>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?<br>
<br>-MC<br> <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Closest I can see is:<br>2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec Activation Success L16@8000hz 1 channel 20ms<br>
2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play Ringback Tone [%(2000,4000,440.0,480.0)]<br>2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/<a href="mailto:4163641113@67.205.74.164" target="_blank">4163641113@67.205.74.164</a> receive message [TRANSCODING_NECESSARY]<br>
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/<a href="http://sip:main@192.168.0.121:5060" target="_blank">sip:main@192.168.0.121:5060</a> entering state [proceeding][180]<br>2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/<a href="http://sip:main@192.168.0.121:5060" target="_blank">sip:main@192.168.0.121:5060</a>!<br>
2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/<a href="mailto:4163641113@67.205.74.164" target="_blank">4163641113@67.205.74.164</a> entering state [terminated][487]<br>2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/<a href="mailto:4163641113@67.205.74.164" target="_blank">4163641113@67.205.74.164</a> [CS_EXECUTE] [ORIGINATOR_CANCEL]<br>
<br>Thank you,<br><br>Igor<br><br>>set ringback before record_session and also set transfer_ringback <br>>because record_session causes an pre-answer.<br>><br>>/b<br>><br>>On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:<br>
><br>>> Hi,<br>>><br>>> I have trouble recording incoming calls with FreeSwitch.<br>>><br>>> I have followed the instruction from Misc. Dialplan Tools record <br>>> session<br>>> (<a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session" target="_blank">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session</a>)<br>
>> It works well for outgoing calls, but I have the problem with <br>>> incoming calls.<br>>><br>>> The person who is calling does not hear ring tone, he hears just the <br>>> silence until<br>
>> I pick up the phone. Everything else is working, we can talk, <br>>> conversation is recorded.<br>>><br>>> Here is a copy of my dialplan for incoming calls<br>>> /usr/local/freeswitch/conf/dialplan/public/voipms.xml<br>
>><br>>> <include><br>>> <extension name="voipms"> <!-- your provider or any name you'd <br>>> like to call it --><br>>> <condition field="destination_number" <br>
>> expression="XXXXXXXXXX"> <!-- your DID for this gateway--><br>>> <action application="set" data="RECORD_TITLE=Recording $ <br>>> {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H: <br>
>> %M)}"/><br>>> <action application="set" data="RECORD_COPYRIGHT=(c) <br>>> 2009"/><br>>> <action application="set" <br>>> data="RECORD_SOFTWARE=FreeSwitch"/><br>
>> <action application="set" <br>>> data="RECORD_ARTIST=FreeSwitch"/><br>>> <action application="set" <br>>> data="RECORD_COMMENT=FreeSwitch"/><br>
>> <action application="set" data="RECORD_DATE=${strftime <br>>> (%Y-%m-%d %H:%M)}"/><br>>> <action application="set" data="RECORD_STEREO=true"/><br>
>> <action application="set" data="RECORD_ANSWER_REQ=true"/><br>>> <action application="set" data="ringback=${us-ring}"/><br>>> <action application="record_session" data="$${base_dir}/ <br>
>> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ <br>>> {destination_number}_${caller_id_number}.wav"/><br>>> <action application="bridge" data="user/user1@$ <br>
>> {domain_name}"/><br>>> </condition><br>>> </include><br>><br>><br>
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<br></blockquote></div><br>