Recording problem speed and noise
Hi All! Im facing a strange issue.. My recordings (in mp3, wav or gsm) are too fast.. now recordings are completely inaudible, only with noise and voice metallized... But sometimes it not happen, just speed issue... im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2) today i tryed to change to 1.3.0, but faced another problem, noise on calls. i checked load on my machine, and load was 0.2, almost 100% cpu FREE.. i have a khomp board with R2 signaling.. Can you help me to debug what is happening? Im quite new to FS!!! Hugs
On 24.09.2012 22:43, Tiago N. Sampaio wrote:
Hi All!
Im facing a strange issue.. My recordings (in mp3, wav or gsm) are too fast.. now recordings are completely inaudible, only with noise and voice metallized... But sometimes it not happen, just speed issue...
im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
today i tryed to change to 1.3.0, but faced another problem, noise on calls. i checked load on my machine, and load was 0.2, almost 100% cpu FREE.. i have a khomp board with R2 signaling..
Can you help me to debug what is happening? Im quite new to FS!!!
Hugs
Please update to the latest developing version using git checkout make clean This happens when it is recorded a transcoded bridge, ie, one leg have one codec and another leg - another codec. Also checkout VoIP phone's settings. If you see "signaling standard" is set to "Chinese" change it to some European standard. -- Mimiko desu.
Hi Vbvbrj! I updated to 1.2.3 but the problems still occurs... Im using Xlite and Zoiper as softphone, both with recording issue. I try change codecs, all codecs problem still occurs... Em 25/09/2012 03:17, Vbvbrj escreveu:
On 24.09.2012 22:43, Tiago N. Sampaio wrote:
Hi All!
Im facing a strange issue.. My recordings (in mp3, wav or gsm) are too fast.. now recordings are completely inaudible, only with noise and voice metallized... But sometimes it not happen, just speed issue...
im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
today i tryed to change to 1.3.0, but faced another problem, noise on calls. i checked load on my machine, and load was 0.2, almost 100% cpu FREE.. i have a khomp board with R2 signaling..
Can you help me to debug what is happening? Im quite new to FS!!!
Hugs Please update to the latest developing version using git checkout make clean
This happens when it is recorded a transcoded bridge, ie, one leg have one codec and another leg - another codec.
Also checkout VoIP phone's settings. If you see "signaling standard" is set to "Chinese" change it to some European standard.
On 25.09.2012 20:31, Tiago N. Sampaio wrote:
I updated to 1.2.3 but the problems still occurs... Im using Xlite and Zoiper as softphone, both with recording issue. I try change codecs, all codecs problem still occurs...
The latest version, I mean v.1.3.0. Set both softphones to same default codec PCMU for use. And see the logs when bot legs negotiate codecs. -- Mimiko desu.
I tested with 1.3.0, and the problem still occours... Testing to better debug the issue, i can reach this: PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs.. I know this is a transcoding problem, but why when i change codec to PCMA or PCMU nothing changes.. I tryed all codecs, without success... My pstn is an MFCr2 E1 circuit over a Khomp board! Im very lost and confused, this problem is killing me :/ 2012/9/25 Vbvbrj <vbvbrj@gmail.com>
On 25.09.2012 20:31, Tiago N. Sampaio wrote:
I updated to 1.2.3 but the problems still occurs... Im using Xlite and Zoiper as softphone, both with recording issue. I try change codecs, all codecs problem still occurs...
The latest version, I mean v.1.3.0. Set both softphones to same default codec PCMU for use. And see the logs when bot legs negotiate codecs.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- Tiago N. Sampaio BSD Certified Associate
I had this issue before, I tried so many things finally this version helped me. Everything was fine before and some combination of HW/SW did the same to me. FreeSWITCH Version 1.0.head (git-9f8d37d 2012-01-28 18-32-35 -0600) May be this is totally irrelevant to your issue, thought to share my experience. Thanks, Bala On Wed, Sep 26, 2012 at 7:25 AM, Tiago Sampaio <tnsampaio@bsd.com.br> wrote:
I tested with 1.3.0, and the problem still occours...
Testing to better debug the issue, i can reach this:
PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs..
I know this is a transcoding problem, but why when i change codec to PCMA or PCMU nothing changes.. I tryed all codecs, without success... My pstn is an MFCr2 E1 circuit over a Khomp board! Im very lost and confused, this problem is killing me :/
2012/9/25 Vbvbrj <vbvbrj@gmail.com>
On 25.09.2012 20:31, Tiago N. Sampaio wrote:
I updated to 1.2.3 but the problems still occurs... Im using Xlite and Zoiper as softphone, both with recording issue. I try change codecs, all codecs problem still occurs...
The latest version, I mean v.1.3.0. Set both softphones to same default codec PCMU for use. And see the logs when bot legs negotiate codecs.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Tiago N. Sampaio BSD Certified Associate
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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On 26.09.2012 04:55, Tiago Sampaio wrote:
I tested with 1.3.0, and the problem still occours...
Testing to better debug the issue, i can reach this:
PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA. -- Mimiko desu.
What method are you using to record? On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com> wrote:
On 26.09.2012 04:55, Tiago Sampaio wrote:
I tested with 1.3.0, and the problem still occours...
Testing to better debug the issue, i can reach this:
PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- 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
This is an example... <extension name="numero_local"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) SAMGAR"/> <action application="set" data="RECORD_SOFTWARE=SGIPBX v1.0"/> <action application="set" data="RECORD_ARTIST=SAMGAR"/> <action application="set" data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/> <action application="set" data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/> <action application="bridge" data="Khomp/b0/$1" /> </condition> </extension> in any format issue occurs... Em 26/09/2012 14:45, Anthony Minessale escreveu:
What method are you using to record?
On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com <mailto:vbvbrj@gmail.com>> wrote:
On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio wrote: > I tested with 1.3.0, and the problem still occours... > > Testing to better debug the issue, i can reach this: > > PSTN -> FS -> user/1000 (record ok!) > user/1000 -> FS -> user/1001 (record ok!) > user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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Must be a khomp specific issue. Maybe the frames coming from it are reporting the wrong size or something... On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br>wrote:
This is an example...
<extension name="numero_local"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) SAMGAR"/> <action application="set" data="RECORD_SOFTWARE=SGIPBX v1.0"/> <action application="set" data="RECORD_ARTIST=SAMGAR"/> <action application="set" data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/> <action application="set" data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/> <action application="bridge" data="Khomp/b0/$1" /> </condition> </extension>
in any format issue occurs...
Em 26/09/2012 14:45, Anthony Minessale escreveu:
What method are you using to record?
On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com> wrote:
On 26.09.2012 04 <26.09.2012%2004>:55, Tiago Sampaio wrote:
I tested with 1.3.0, and the problem still occours...
Testing to better debug the issue, i can reach this:
PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
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Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
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-- 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
The problem is to debug it... I didnt saw anything in logs or debug..... Can you tell me a way to debug it? Em 26/09/2012 15:30, Anthony Minessale escreveu:
Must be a khomp specific issue. Maybe the frames coming from it are reporting the wrong size or something...
On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br <mailto:tnsampaio@bsd.com.br>> wrote:
This is an example...
<extension name="numero_local"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) SAMGAR"/> <action application="set" data="RECORD_SOFTWARE=SGIPBX v1.0"/> <action application="set" data="RECORD_ARTIST=SAMGAR"/> <action application="set" data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/> <action application="set" data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/> <action application="bridge" data="Khomp/b0/$1" /> </condition> </extension>
in any format issue occurs...
Em 26/09/2012 14:45, Anthony Minessale escreveu:
What method are you using to record?
On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com <mailto:vbvbrj@gmail.com>> wrote:
On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio wrote: > I tested with 1.3.0, and the problem still occours... > > Testing to better debug the issue, i can reach this: > > PSTN -> FS -> user/1000 (record ok!) > user/1000 -> FS -> user/1001 (record ok!) > user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- 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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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maybe edit the code and print out values like the frame datalen or samples in the read_frame function in the mod and see if you can see anything odd. On Wed, Sep 26, 2012 at 1:49 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br>wrote:
The problem is to debug it... I didnt saw anything in logs or debug..... Can you tell me a way to debug it?
Em 26/09/2012 15:30, Anthony Minessale escreveu:
Must be a khomp specific issue. Maybe the frames coming from it are reporting the wrong size or something...
On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br>wrote:
This is an example...
<extension name="numero_local"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) SAMGAR"/> <action application="set" data="RECORD_SOFTWARE=SGIPBX v1.0"/> <action application="set" data="RECORD_ARTIST=SAMGAR"/> <action application="set" data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/> <action application="set" data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/> <action application="bridge" data="Khomp/b0/$1" /> </condition> </extension>
in any format issue occurs...
Em 26/09/2012 14:45, Anthony Minessale escreveu:
What method are you using to record?
On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com> wrote:
On 26.09.2012 04 <26.09.2012%2004>:55, Tiago Sampaio wrote:
I tested with 1.3.0, and the problem still occours...
Testing to better debug the issue, i can reach this:
PSTN -> FS -> user/1000 (record ok!) user/1000 -> FS -> user/1001 (record ok!) user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Now if i use speex@8000 it records normally now... I will let it as is for now! On next weekend i will try this to get more information!! Thx for help! Em 26/09/2012 16:52, Anthony Minessale escreveu:
maybe edit the code and print out values like the frame datalen or samples in the read_frame function in the mod and see if you can see anything odd.
On Wed, Sep 26, 2012 at 1:49 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br <mailto:tnsampaio@bsd.com.br>> wrote:
The problem is to debug it... I didnt saw anything in logs or debug..... Can you tell me a way to debug it?
Em 26/09/2012 15:30, Anthony Minessale escreveu:
Must be a khomp specific issue. Maybe the frames coming from it are reporting the wrong size or something...
On Wed, Sep 26, 2012 at 1:01 PM, Tiago N. Sampaio <tnsampaio@bsd.com.br <mailto:tnsampaio@bsd.com.br>> wrote:
This is an example...
<extension name="numero_local"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) SAMGAR"/> <action application="set" data="RECORD_SOFTWARE=SGIPBX v1.0"/> <action application="set" data="RECORD_ARTIST=SAMGAR"/> <action application="set" data="RECORD_COMMENT=GRAVACAO DE CONVERSACAO"/> <action application="set" data="RECORD_DATE=${strftime(%d/%m/%Y %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="/dados/ligacoes/saida/${strftime(%Y/%m/%d/%H-%M)}_${caller_id_number}_${destination_number}.wav"/> <action application="bridge" data="Khomp/b0/$1" /> </condition> </extension>
in any format issue occurs...
Em 26/09/2012 14:45, Anthony Minessale escreveu:
What method are you using to record?
On Wed, Sep 26, 2012 at 2:05 AM, Vbvbrj <vbvbrj@gmail.com <mailto:vbvbrj@gmail.com>> wrote:
On 26.09.2012 04 <tel:26.09.2012%2004>:55, Tiago Sampaio wrote: > I tested with 1.3.0, and the problem still occours... > > Testing to better debug the issue, i can reach this: > > PSTN -> FS -> user/1000 (record ok!) > user/1000 -> FS -> user/1001 (record ok!) > user/1000 -> FS -> PSTN problem occurs..
So problem occur when an VoIP client calls the PSTN client, ie an outbound call from FS to PSTN. Checkout in vars.xml the option outbound_codec_prefs and it must contain the same codecs, especialy PCMU or PCMA.
-- Mimiko desu.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- 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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- 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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
I forgot to mention, when a call come from PSTN it records fine, in mp3 or wav. Im using an R2 KHOMP board. Its driving me crazy hehe!!! Thx for help! Em 25/09/2012 14:31, Tiago N. Sampaio escreveu:
Hi Vbvbrj!
I updated to 1.2.3 but the problems still occurs... Im using Xlite and Zoiper as softphone, both with recording issue. I try change codecs, all codecs problem still occurs...
Em 25/09/2012 03:17, Vbvbrj escreveu:
On 24.09.2012 22:43, Tiago N. Sampaio wrote:
Hi All!
Im facing a strange issue.. My recordings (in mp3, wav or gsm) are too fast.. now recordings are completely inaudible, only with noise and voice metallized... But sometimes it not happen, just speed issue...
im running this version: FreeSWITCH Version 1.2.0-rc2 (1.2.0-rc2)
today i tryed to change to 1.3.0, but faced another problem, noise on calls. i checked load on my machine, and load was 0.2, almost 100% cpu FREE.. i have a khomp board with R2 signaling..
Can you help me to debug what is happening? Im quite new to FS!!!
Hugs Please update to the latest developing version using git checkout make clean
This happens when it is recorded a transcoded bridge, ie, one leg have one codec and another leg - another codec.
Also checkout VoIP phone's settings. If you see "signaling standard" is set to "Chinese" change it to some European standard.
Also checkout VoIP phone's settings. If you see "signaling standard" is set to "Chinese" change it to some European standard.
Ah, that's why the Chinese talk so fast... :-)
-- Brian D. Foster Endigo Computer LLC Email: bdfoster@endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.
On 27.09.2012 05:34, Brian Foster wrote:
Also checkout VoIP phone's settings. If you see "signaling standard" is set to "Chinese" change it to some European standard.
Ah, that's why the Chinese talk so fast... :-)
Actually, it indeed resolved the problem for VoIP phones made in China. -- Mimiko desu.
participants (6)
-
Anthony Minessale
-
Balamurugan Mahendran
-
Brian Foster
-
Tiago N. Sampaio
-
Tiago Sampaio
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Vbvbrj