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 PCMU@30i@8000h
2008-06-03 12:52:18 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec() Set Codec sofia/internal/1006@10.47.40.56:5090 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 516722008-06-03 12:57:03 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/internal/1004@10.47.40.56:5090 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/1004@10.47.40.56:5090 PCMU/8000 20 ms 160 samplesRegards,MiroslavOn Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian@freeswitch.org> wrote:
What codec are you using on the inbound side?
/bHi 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
Freeswitch-users@lists.freeswitch.org
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
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org