Hi When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/> And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either. Global Codecs definition. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/> Is this known issue? -- Piotr Sobolewski sobolewski@gmail.com
It is a known-issue that you never know what you're going to get with Asterisk.. yes. On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
Hi
When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK
I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/>
And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either.
Global Codecs definition.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/>
Is this known issue?
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It is a known-issue that you never know what you're going to get with Asterisk.. yes.
Is there a way to solve this issue without taking away asterisk?
On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
Hi
When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK
I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/>
And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either.
Global Codecs definition.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/>
Is this known issue?
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ 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
-- Piotr Sobolewski sobolewski@gmail.com
possibly but I am not quite sure what the issue is caused by. you say the audio is slower in the recording? Does it sound normal during the actual conversation? What format are you recording to? On Tue, Oct 12, 2010 at 9:58 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It is a known-issue that you never know what you're going to get with Asterisk.. yes.
Is there a way to solve this issue without taking away asterisk?
On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
Hi
When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK
I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/>
And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either.
Global Codecs definition.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/>
Is this known issue?
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ 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
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
possibly but I am not quite sure what the issue is caused by. you say the audio is slower in the recording?
yes, it's slower to such extent that it's unreadable
Does it sound normal during the actual conversation?
yes, it's normal
What format are you recording to?
wav -- Piotr Sobolewski sobolewski@gmail.com
maybe you can produce a sample recording On Tue, Oct 12, 2010 at 10:45 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
possibly but I am not quite sure what the issue is caused by. you say the audio is slower in the recording?
yes, it's slower to such extent that it's unreadable
Does it sound normal during the actual conversation?
yes, it's normal
What format are you recording to?
wav
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
possibly but I am not quite sure what the issue is caused by. you say the audio is slower in the recording?
yes, it's slower to such extent that it's unreadable
Does it sound normal during the actual conversation?
yes, it's normal
What format are you recording to?
wav -- Piotr Sobolewski sobolewski@gmail.com
possibly but I am not quite sure what the issue is caused by. you say the audio is slower in the recording? Does it sound normal during the actual conversation? What format are you recording to? On Tue, Oct 12, 2010 at 9:58 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It is a known-issue that you never know what you're going to get with Asterisk.. yes.
Is there a way to solve this issue without taking away asterisk?
On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
Hi
When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK
I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/>
And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either.
Global Codecs definition.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/>
Is this known issue?
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ 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
-- Piotr Sobolewski sobolewski@gmail.com
_______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (2)
-
Anthony Minessale
-
Piotr Sobolewski