On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote: > Hello, > > today I found in FS logfile lines like this: > > 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1 > channel > 20ms > > > It looks like L16 codec is used for incoming calls: > > 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:18/2799 [BREAK] > 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1 > channel > 20ms > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 > switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] > 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:[email protected]:5060;user=phone;transport=udp > entering state [proceeding] > 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() > Ring-Ready > sofia/internal/sip:[email protected]:5060;user=phone;transport=udp! > 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 > switch_core_session_write_frame() OpenZAP/1:18/2799 receive message > [TRANSCODING_NECESSARY] > 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, > State: 0) timed out > 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:[email protected]:5060;user=phone;transport=udp > entering state [ready] > 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() > Remote SDP: > v=0^M > o=2799 121183017 121183017 IN IP4 85.16.245.254^M > s=ATA186 Call^M > c=IN IP4 85.16.245.254^M > t=0 0^M > m=audio 16384 RTP/AVP 8 101^M > a=rtpmap:8 PCMA/8000/1^M > a=rtpmap:101 telephone-event/8000^M > a=fmtp:101 0-15^M > > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 > sofia_glue_tech_set_codec() Set Codec > sofia/internal/sip:[email protected]:5060;user=phone;transport=udp > PCMA/8000 20 ms 160 samples > > The audio codec compare function finds slightly different codecs for A > and B party. > > The dialplan for incoming calls via openzap is this. I set the codec > to > use in extensions "bridge" line: > > <extension name="fp_Local_Extension"> > <condition field="destination_number" > expression="(491[0-9]|492[0-8])$"> > <action application="ring_ready"/> > <action application="set" data="ringback=${de-ring}"/> > <action application="export" > data="nolocal:sip_secure_media=${user_data(${dialed_extensi...@$ > {domain_name} > var sip_secure_media)}"/> > <action application="bridge" > data="{absolute_codec_string=PCMA}user/$...@$${domain}"/> > </condition> > </extension> > > > In my vars.xml config I have these codecs configured: > > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMA"/> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMA"/> > > So where can I disable the L16 codec, or why is a transcoding > necessary?
Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
