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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