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
What codec are you using on the inbound side? /b On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip@gmail.com> 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
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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 51672 2008-06-03 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 samples Regards, Miroslav On Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian@freeswitch.org> wrote:
What codec are you using on the inbound side?
/b
On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip@gmail.com> 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
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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 <nuvovoip@gmail.com> 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 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 51672 2008-06-03 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 samples
Regards,
Miroslav On Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian@freeswitch.org> wrote:
What codec are you using on the inbound side?
/b
On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip@gmail.com> 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
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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 <nuvovoip@gmail.com> 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 <nuvovoip@gmail.com> 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 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 51672 2008-06-03 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 samples
Regards,
Miroslav On Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian@freeswitch.org> wrote:
What codec are you using on the inbound side?
/b
On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip@gmail.com> 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 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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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 < anthony.minessale@gmail.com> 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 <nuvovoip@gmail.com> 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 <nuvovoip@gmail.com> 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 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 51672 2008-06-03 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 samples
Regards,
Miroslav On Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian@freeswitch.org> wrote:
What codec are you using on the inbound side?
/b
On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip@gmail.com> 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 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
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_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ 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
participants (3)
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Anthony Minessale
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Brian West
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Miroslav Mostic