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? regards Helmut _______________________________________________ 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
