Thanks a lot Anthony !!! It helped when I loaded mod_native_file before mod_voipcodecs.
Many thanks again, Miroslav On Fri, Jun 6, 2008 at 3:53 PM, Anthony Minessale < [EMAIL PROTECTED]> wrote: > try reversing the order that you load mod_voipcodecs and mod_sndfile in > your modules.conf.xml > iirc there is some symbol collision between the 2 that we need to resolve > somehow. > > > On Fri, Jun 6, 2008 at 1:29 PM, Miroslav Mostic <[EMAIL PROTECTED]> > wrote: > >> It is not inbound codec issue. When I enable only GSM codec in my x-lite I >> can see that GSM is negotiated inbound codec. Even in that case recording in >> gsm is not working. However recording in uncompressed PCM wav format is >> working properly. >> I have the same behaviour with rc3 and some snapshot release after that, >> but for the sake of this test I have default Freeswitch 1.0 configuration on >> the separate box. >> >> Could it be that compressed gsm encoded recordings are not supported >> anymore? >> >> Regards, >> >> Miroslav >> >> On Tue, Jun 3, 2008 at 1:49 PM, Miroslav Mostic <[EMAIL PROTECTED]> >> wrote: >> >>> >>> G711 u-Law. I was calling from aastra 9133i and X-lite v3.0 and I have >>> default codec configuration of Freeswitch 1.0. After installation of >>> Freeswitch 1.0 the only thing that are changed from default config are sip >>> port 5090 on internal sip profile and additional extensions 9991 and 9992 in >>> default.xml dialplan. In both scenarios I have the same result. >>> >>> Console output when I dial from aastra 9133i : >>> >>> 2008-06-03 12:52:18 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state() >>> Remote SDP: >>> v=0 >>> o=MxSIP 0 1966242209 IN IP4 172.22.2.102 >>> s=SIP Call >>> c=IN IP4 172.22.2.102 >>> t=0 0 >>> m=audio 3000 RTP/AVP 0 8 18 101 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=ptime:30 >>> a=silenceSupp:on - - - - >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[G722:9:8000] >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[PCMA:8:8000] >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[GSM:3:8000] >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2137 sofia_glue_negotiate_sdp() >>> Substituting codec [EMAIL PROTECTED]@8000h >>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec() >>> Set Codec sofia/internal/[EMAIL PROTECTED]:5090 PCMU/8000 30 ms 240 >>> samples >>> Console output when I dial from X-Lite Version 3.0: >>> >>> 2008-06-03 12:57:03 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state() >>> Remote SDP: >>> v=0 >>> o=- 4 2 IN IP4 172.22.2.141 >>> s=CounterPath X-Lite 3.0 >>> c=IN IP4 172.22.2.141 >>> t=0 0 >>> m=audio 51672 RTP/AVP 0 8 3 101 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=alt:1 1 : uY3hci3U 9llPveYf 172.22.2.141 51672 >>> 2008-06-03 12:57:03 [DEBUG] switch_core_state_machine.c:365 >>> switch_core_session_run() sofia/internal/[EMAIL PROTECTED]:5090 Running >>> State Change CS_NEW >>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[G722:9:8000] >>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp() >>> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] >>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec() >>> Set Codec sofia/internal/[EMAIL PROTECTED]:5090 PCMU/8000 20 ms 160 >>> samples >>> >>> Regards, >>> >>> Miroslav >>> On Mon, Jun 2, 2008 at 8:52 PM, Brian West <[EMAIL PROTECTED]> >>> wrote: >>> >>>> What codec are you using on the inbound side? >>>> >>>> /b >>>> >>>> >>>> >>>> >>>> On 6/2/08 4:27 PM, "Miroslav Mostic" <[EMAIL PROTECTED]> wrote: >>>> >>>> Hi everyone! >>>> >>>> I am probably missing something very basic, but I have problems with >>>> simple file record and playback in gsm file format. >>>> >>>> I have just installed 1.0 release system and I changed only default >>>> dialplan with following two extensions: >>>> >>>> <extension name="gsm_record"> >>>> <condition field="destination_number" expression="^9991$"> >>>> <action application="answer"/> >>>> <action application="record" data="/tmp/rec.gsm"/> >>>> </condition> >>>> </extension> >>>> <extension name="gsm_playback"> >>>> <condition field="destination_number" expression="^9992$"> >>>> <action application="answer"/> >>>> <action application="playback" data="/tmp/rec.gsm"/> >>>> </condition> >>>> </extension> >>>> >>>> When I place call to 9991extension I see INFO message about opening >>>> /tmp/rec.gsm, no error messages and when I hangup I can see that file >>>> /tmp/rec.gsm exists. However when I try to play this file using 9992 >>>> extension the only thing I can hear is noise. >>>> >>>> When I just change name to /tmp/rec.wav in both extensions everything is >>>> working fine. >>>> >>>> What am I missing? >>>> >>>> Many regards, >>>> >>>> Miroslav >>>> >>>> ------------------------------ >>>> _______________________________________________ >>>> 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 >>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>> >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > pstn:213-799-1400 > _______________________________________________ > 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 > >
_______________________________________________ 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
