How to configure g729 pass though on Freeswitch
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN. What configuration changes do I need to allow freeswitch-codec-passthru-g729. -- Frank Church ======================= http://devblog.brahmancreations.com
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it. My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it.
My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think their call volume does not make it easy on them and their customers as well. I did some googling and came up with freeswitch-codec-passthru-g729. I have also read http://wiki.freeswitch.org/wiki/Proxy_Media and http://wiki.freeswitch.org/wiki/Bypass_Media. In my module.conf.xml there is also <load module="mod_g729"/>. Does that mean that my installlation is configured for pass thru if I make the right adjustments? I have looked at http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h... and http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h... which speak of modifying the dialplan. This is a basic freeswitch setup using the defaults. I just added the extensions to conf/directory/default and changed the provider in vars.xml and I want to be able to do the same in conf/dialplan/default.xml. In conf/dialplan/default.xml the extension is matched by the destination. Is there an option for not falling through to other extensions if they also match?
_______________________________________________ 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
-- Frank Church ======================= http://devblog.brahmancreations.com
Frank, mod_g729 needs to be loaded, and then G729 needs to be negotiated on both legs. I really recommend you enable G729 on the Linksys, enable SIP trace on FS console: sofia profile external siptrace on sofia profile internal siptrace on then make a (failing) call, capture the log on FS console and paste that to: http://pastebin:freeswitch@pastebin.freeswitch.org/ Then, send us back the link to your paste. You can also join us on #freeswitch (irc.freenode.net), for some live help. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it.
My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think their call volume does not make it easy on them and their customers as well.
I did some googling and came up with freeswitch-codec-passthru-g729. I have also read http://wiki.freeswitch.org/wiki/Proxy_Media and http://wiki.freeswitch.org/wiki/Bypass_Media. In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I make the right adjustments?
I have looked at http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h... and http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h... which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the extensions to conf/directory/default and changed the provider in vars.xml and I want to be able to do the same in conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the destination. Is there an option for not falling through to other extensions if they also match?
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
Frank, sorry, I completely forgot one important detail: in the default conf, G729 is not allowed on any SIP profiles, so you have to modify vars.xml. You will find the following lines: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/> For now, I recommend you replace them by: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/> Then in FS console, do: sofia profile external restart reloadxml sofia profile internal restart reloadxml It should then work far better. What we did there is to make G729 an accepted codec in inbound INVITEs and a proposed codec for outbound INVITEs. Look in external.xml or internal.xml, and look at the variables inbound-codec-prefs and outbound-codec-prefs. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 02/04/2010 à 14:30, David Ponzone a écrit :
Frank,
mod_g729 needs to be loaded, and then G729 needs to be negotiated on both legs. I really recommend you enable G729 on the Linksys, enable SIP trace on FS console: sofia profile external siptrace on sofia profile internal siptrace on
then make a (failing) call, capture the log on FS console and paste that to: http://pastebin:freeswitch@pastebin.freeswitch.org/
Then, send us back the link to your paste.
You can also join us on #freeswitch (irc.freenode.net), for some live help.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it.
My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think their call volume does not make it easy on them and their customers as well.
I did some googling and came up with freeswitch-codec-passthru- g729. I have also read http://wiki.freeswitch.org/wiki/Proxy_Media and http://wiki.freeswitch.org/wiki/Bypass_Media. In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I make the right adjustments?
I have looked at http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h... and http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h... which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the extensions to conf/directory/default and changed the provider in vars.xml and I want to be able to do the same in conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the destination. Is there an option for not falling through to other extensions if they also match?
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
On 2 April 2010 13:46, David Ponzone <david.ponzone@gmail.com> wrote:
Frank, sorry, I completely forgot one important detail: in the default conf, G729 is not allowed on any SIP profiles, so you have to modify vars.xml. You will find the following lines: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/> For now, I recommend you replace them by: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/> Then in FS console, do: sofia profile external restart reloadxml sofia profile internal restart reloadxml It should then work far better. What we did there is to make G729 an accepted codec in inbound INVITEs and a proposed codec for outbound INVITEs. Look in external.xml or internal.xml, and look at the variables inbound-codec-prefs and outbound-codec-prefs. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:30, David Ponzone a écrit :
Frank, mod_g729 needs to be loaded, and then G729 needs to be negotiated on both legs. I really recommend you enable G729 on the Linksys, enable SIP trace on FS console: sofia profile external siptrace on sofia profile internal siptrace on then make a (failing) call, capture the log on FS console and paste that to: http://pastebin:freeswitch@pastebin.freeswitch.org/
This the pastebin link http://pastebin.freeswitch.org/12616 There is also one below it with the successful call. I have set the default provider in vars.xml rather than under the sip_profiles/external. Is there a section in the default dialplan that handles the default context? Can the options below be used in the default for the dialplan and sip_profiles? In the dial plan <action application="set" data="proxy_media=true"/> In the sip profile <param name="inbound-proxy-media" value="true"/> The Linksys accounts are in the default contexts.
Then, send us back the link to your paste. You can also join us on #freeswitch (irc.freenode.net), for some live help. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think their call volume does not make it easy on them and their customers as well.
I did some googling and came up with freeswitch-codec-passthru-g729. I have also read http://wiki.freeswitch.org/wiki/Proxy_Media and http://wiki.freeswitch.org/wiki/Bypass_Media. In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I make the right adjustments?
I have looked at http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h... and http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h... which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the extensions to conf/directory/default and changed the provider in vars.xml and I want to be able to do the same in conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the destination. Is there an option for not falling through to other extensions if they also match?
_______________________________________________
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
-- Frank Church
======================= http://devblog.brahmancreations.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
_______________________________________________ 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
-- Frank Church ======================= http://devblog.brahmancreations.com
Frank, re-read my second mail. You have to enable G729 in FS prefs (vars.xml). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 02/04/2010 à 20:52, Frank Church a écrit :
On 2 April 2010 13:46, David Ponzone <david.ponzone@gmail.com> wrote:
Frank, sorry, I completely forgot one important detail: in the default conf, G729 is not allowed on any SIP profiles, so you have to modify vars.xml. You will find the following lines: <X-PRE-PROCESS cmd="set" data ="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/> For now, I recommend you replace them by: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/> Then in FS console, do: sofia profile external restart reloadxml sofia profile internal restart reloadxml It should then work far better. What we did there is to make G729 an accepted codec in inbound INVITEs and a proposed codec for outbound INVITEs. Look in external.xml or internal.xml, and look at the variables inbound-codec-prefs and outbound-codec-prefs. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:30, David Ponzone a écrit :
Frank, mod_g729 needs to be loaded, and then G729 needs to be negotiated on both legs. I really recommend you enable G729 on the Linksys, enable SIP trace on FS console: sofia profile external siptrace on sofia profile internal siptrace on then make a (failing) call, capture the log on FS console and paste that to: http://pastebin:freeswitch@pastebin.freeswitch.org/
This the pastebin link
http://pastebin.freeswitch.org/12616
There is also one below it with the successful call.
I have set the default provider in vars.xml rather than under the sip_profiles/external. Is there a section in the default dialplan that handles the default context?
Can the options below be used in the default for the dialplan and sip_profiles?
In the dial plan <action application="set" data="proxy_media=true"/>
In the sip profile <param name="inbound-proxy-media" value="true"/>
The Linksys accounts are in the default contexts.
Then, send us back the link to your paste. You can also join us on #freeswitch (irc.freenode.net), for some live help. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think their call volume does not make it easy on them and their customers as well.
I did some googling and came up with freeswitch-codec-passthru- g729. I have also read http://wiki.freeswitch.org/wiki/Proxy_Media and http://wiki.freeswitch.org/wiki/Bypass_Media. In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I make the right adjustments?
I have looked at http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h... and http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h... which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the extensions to conf/directory/default and changed the provider in vars.xml and I want to be able to do the same in conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the destination. Is there an option for not falling through to other extensions if they also match?
_______________________________________________
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
-- Frank Church
======================= http://devblog.brahmancreations.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
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
On 2 April 2010 20:36, David Ponzone <david.ponzone@gmail.com> wrote:
Frank, re-read my second mail.
http://pastebin.freeswitch.org/12617
You have to enable G729 in FS prefs (vars.xml). Was that obvious from the sip trace? I missed it the first time.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 20:52, Frank Church a écrit :
On 2 April 2010 13:46, David Ponzone <david.ponzone@gmail.com> wrote:
Frank,
sorry, I completely forgot one important detail:
in the default conf, G729 is not allowed on any SIP profiles, so you have to
modify vars.xml.
You will find the following lines:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
For now, I recommend you replace them by:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
Then in FS console, do:
sofia profile external restart reloadxml
sofia profile internal restart reloadxml
It should then work far better.
What we did there is to make G729 an accepted codec in inbound INVITEs and a
proposed codec for outbound INVITEs.
Look in external.xml or internal.xml, and look at the variables
inbound-codec-prefs and outbound-codec-prefs.
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:30, David Ponzone a écrit :
Frank,
mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
legs.
I really recommend you enable G729 on the Linksys, enable SIP trace on FS
console:
sofia profile external siptrace on
sofia profile internal siptrace on
then make a (failing) call, capture the log on FS console and paste that to:
http://pastebin:freeswitch@pastebin.freeswitch.org/
This the pastebin link
http://pastebin.freeswitch.org/12616
There is also one below it with the successful call.
I have set the default provider in vars.xml rather than under the sip_profiles/external. Is there a section in the default dialplan that handles the default context?
Can the options below be used in the default for the dialplan and sip_profiles?
In the dial plan <action application="set" data="proxy_media=true"/>
In the sip profile <param name="inbound-proxy-media" value="true"/>
The Linksys accounts are in the default contexts.
Then, send us back the link to your paste.
You can also join us on #freeswitch (irc.freenode.net), for some live help.
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think
their call volume does not make it easy on them and their customers as
well.
I did some googling and came up with freeswitch-codec-passthru-g729. I
have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
http://wiki.freeswitch.org/wiki/Bypass_Media.
In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I
make the right adjustments?
I have looked at
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h...
and
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h...
which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the
extensions to conf/directory/default and changed the provider in
vars.xml and I want to be able to do the same in
conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the
destination. Is there an option for not falling through to other
extensions if they also match?
_______________________________________________
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
--
Frank Church
=======================
http://devblog.brahmancreations.com
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- Frank Church
======================= http://devblog.brahmancreations.com
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-- Frank Church ======================= http://devblog.brahmancreations.com
Yes, in the lines: 2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001@192.168.1.133 [CS_NEW] [INCOMPATIBLE_DESTINATION] send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d From: Booth1 <sip:1001@192.168.4.156>;tag=bff390fd4255c0f9o0 To: <sip:02074379497@192.168.4.156>;tag=m006c20Fg5Spa Call-ID: 1c27844d-e299e5e9@192.168.4.154 That is the answer from FS to the phone, just after receiving the INVITE that contains only G729 and NSE (??) in the SDP. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 02/04/2010 à 22:53, Frank Church a écrit :
On 2 April 2010 20:36, David Ponzone <david.ponzone@gmail.com> wrote:
Frank, re-read my second mail.
http://pastebin.freeswitch.org/12617
You have to enable G729 in FS prefs (vars.xml). Was that obvious from the sip trace? I missed it the first time.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 20:52, Frank Church a écrit :
On 2 April 2010 13:46, David Ponzone <david.ponzone@gmail.com> wrote:
Frank,
sorry, I completely forgot one important detail:
in the default conf, G729 is not allowed on any SIP profiles, so you have to
modify vars.xml.
You will find the following lines:
<X-PRE-PROCESS cmd="set"
data ="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
For now, I recommend you replace them by:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
Then in FS console, do:
sofia profile external restart reloadxml
sofia profile internal restart reloadxml
It should then work far better.
What we did there is to make G729 an accepted codec in inbound INVITEs and a
proposed codec for outbound INVITEs.
Look in external.xml or internal.xml, and look at the variables
inbound-codec-prefs and outbound-codec-prefs.
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:30, David Ponzone a écrit :
Frank,
mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
legs.
I really recommend you enable G729 on the Linksys, enable SIP trace on FS
console:
sofia profile external siptrace on
sofia profile internal siptrace on
then make a (failing) call, capture the log on FS console and paste that to:
http://pastebin:freeswitch@pastebin.freeswitch.org/
This the pastebin link
http://pastebin.freeswitch.org/12616
There is also one below it with the successful call.
I have set the default provider in vars.xml rather than under the sip_profiles/external. Is there a section in the default dialplan that handles the default context?
Can the options below be used in the default for the dialplan and sip_profiles?
In the dial plan <action application="set" data="proxy_media=true"/>
In the sip profile <param name="inbound-proxy-media" value="true"/>
The Linksys accounts are in the default contexts.
Then, send us back the link to your paste.
You can also join us on #freeswitch (irc.freenode.net), for some live help.
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 02/04/2010 à 14:03, Frank Church a écrit :
On 2 April 2010 10:23, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
The carrier insists on G729, although they can accept G711. I think
their call volume does not make it easy on them and their customers as
well.
I did some googling and came up with freeswitch-codec-passthru- g729. I
have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
http://wiki.freeswitch.org/wiki/Bypass_Media.
In my module.conf.xml there is also <load module="mod_g729"/>.
Does that mean that my installlation is configured for pass thru if I
make the right adjustments?
I have looked at
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg12303.h...
and
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg20949.h...
which speak of modifying the dialplan.
This is a basic freeswitch setup using the defaults. I just added the
extensions to conf/directory/default and changed the provider in
vars.xml and I want to be able to do the same in
conf/dialplan/default.xml.
In conf/dialplan/default.xml the extension is matched by the
destination. Is there an option for not falling through to other
extensions if they also match?
_______________________________________________
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
--
Frank Church
=======================
http://devblog.brahmancreations.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
_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
-- Frank Church
======================= http://devblog.brahmancreations.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
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module? -Max On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it.
My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
_______________________________________________ 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
proxy media isn't needed. /b On Apr 4, 2010, at 1:12 PM, Max Clark wrote:
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really need it.
My recommendation would be to use a codec other than G.729 unless you have a compelling reason, for example a carrier that only supports G.729.
_______________________________________________ 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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How would one detect T38 and convert the session into proxy media? On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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
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As far as I know, you can't, because T38 is not advertised at first. T38 starts with regular codecs in the SDP, and then later, a T38 REINVITE is negotiated. I guess the easiest way to handle that is: -enable proxy media based on DID for inbound fax -enable proxy media based on CLID for outbound fax This requires to have specific extensions in your dialplan. For outbound, you can also have a dedicated SIP profile for T38 ATAs so you can enable proxy-media for the whole profile. For inbound, you may do the same if your ITSP/gateway can send you the fax DIDs on a specific trunk (so to a specific SIP profile). Be aware that T38 is a PITA, and that you really need to validate that your T38 device is compatible with the other endpoint, which is probably a gateway. If this gateway is yours, that's fine because you control it, and you can rely on it for your T38 service. If it's not yours, you could have issues if some day, your iTSP decices to change it. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 05/04/2010 à 00:00, Max Clark a écrit :
How would one detect T38 and convert the session into proxy media?
On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone@gmail.com
wrote: No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway I have checked it again a few times using the bypass_media, proxy_media settings. And with those settings the call ends as soon as ringing starts or as sonn as the call is answered. I will do another one just to confirm On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
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-- Frank Church ======================= http://devblog.brahmancreations.com
Frank, again: ------------------------------------------------------------------------ recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500: ------------------------------------------------------------------------ INVITE sip:02074379497@192.168.4.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3 From: Booth1 <sip:1001@192.168.4.156>;tag=3bb06cff17c3ecefo0 To: <sip:02074379497@192.168.4.156> Remote-Party-ID: Booth1 <sip: 1001@192.168.4.156>;screen=yes;party=calling Call-ID: 68993a3f-e38f80af@192.168.4.154 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1001",realm="192.168.4.156",nonce="9567 e05b-d541-4a5e-9e47-2152eb90a199",uri="sip: 02074379497@192.168.4.156",algorithm= MD5 ,response ="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d a1954" Contact: Booth1 <sip:1001@192.168.4.154:5060> Expires: 240 User-Agent: Linksys/SPA2102-3.2.8(d) Content-Length: 260 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 2612067 2612067 IN IP4 192.168.4.154 s=- c=IN IP4 192.168.4.154 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3 From: Booth1 <sip:1001@192.168.4.156>;tag=3bb06cff17c3ecefo0 To: <sip:02074379497@192.168.4.156> Call-ID: 68993a3f-e38f80af@192.168.4.154 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M Content-Length: 0 ------------------------------------------------------------------------ 2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001@192.168.4.156 [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8] 2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001@192.168.1.133 [CS_NEW] [INCOMPATIBLE_DESTINATION] send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d From: Booth1 <sip:1001@192.168.4.156>;tag=bff390fd4255c0f9o0 To: <sip:02074379497@192.168.4.156>;tag=m006c20Fg5Spa Call-ID: 1c27844d-e299e5e9@192.168.4.154 That is the answer from FS to the phone, just after receiving the INVITE that contains only G729 and NSE (??) in the SDP. If you're sure you enabled G729 in vars.xml, you should check you restarted your SIP profile. If you are not sure how to do it right, check my previous emails, or restart FS . David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 05/04/2010 à 11:49, Frank Church a écrit :
Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- Frank Church
======================= http://devblog.brahmancreations.com
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G729a is incorrect codec name, it must be changed to G729 in SPA settings. On Monday 05 April 2010, David Ponzone wrote:
Frank,
again:
------------------------------------------------------------------------ recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
------------------------------------------------------------------------ INVITE sip:02074379497@192.168.4.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3 From: Booth1 <sip:1001@192.168.4.156>;tag=3bb06cff17c3ecefo0 To: <sip:02074379497@192.168.4.156> Remote-Party-ID: Booth1 <sip: 1001@192.168.4.156>;screen=yes;party=calling Call-ID: 68993a3f-e38f80af@192.168.4.154 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1001",realm="192.168.4.156",nonce="9567 e05b-d541-4a5e-9e47-2152eb90a199",uri="sip: 02074379497@192.168.4.156",algorithm= MD5 ,response ="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d a1954" Contact: Booth1 <sip:1001@192.168.4.154:5060> Expires: 240 User-Agent: Linksys/SPA2102-3.2.8(d) Content-Length: 260 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp
v=0 o=- 2612067 2612067 IN IP4 192.168.4.154 s=- c=IN IP4 192.168.4.154 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
------------------------------------------------------------------------ send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3 From: Booth1 <sip:1001@192.168.4.156>;tag=3bb06cff17c3ecefo0 To: <sip:02074379497@192.168.4.156> Call-ID: 68993a3f-e38f80af@192.168.4.154 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M Content-Length: 0
------------------------------------------------------------------------ 2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001@192.168.4.156 [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8] 2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001@192.168.1.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]
send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d From: Booth1 <sip:1001@192.168.4.156>;tag=bff390fd4255c0f9o0 To: <sip:02074379497@192.168.4.156>;tag=m006c20Fg5Spa Call-ID: 1c27844d-e299e5e9@192.168.4.154
That is the answer from FS to the phone, just after receiving the INVITE that contains only G729 and NSE (??) in the SDP.
If you're sure you enabled G729 in vars.xml, you should check you restarted your SIP profile. If you are not sure how to do it right, check my previous emails, or restart FS .
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 11:49, Frank Church a écrit :
Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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======================= http://devblog.brahmancreations.com
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G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab). Ognjen On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com>wrote:
Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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
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-- Frank Church
======================= http://devblog.brahmancreations.com
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Ognjen, very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name. Am I wrong ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to allow G729 as inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP path, it relays the audio stream between endpoints, but can still detect DTMFs -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF detection is impossible. In this mode, FS is really a "dumb" transparent RTP-forwarder (this is required to get T38 working between the 2 endpoints) -bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass media" or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net> wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.
My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- Frank Church
======================= http://devblog.brahmancreations.com
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FS looks at codec name too. On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to
allow G729 as
inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
path, it relays the audio stream between endpoints, but can still
detect
DTMFs -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
detection is impossible. In this mode, FS is really a "dumb"
transparent
RTP-forwarder (this is required to get T38 working between the 2
endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et
établis à
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est
susceptible
d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass
media"
or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
really
need it.
My recommendation would be to use a codec other than G.729 unless
you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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-user s
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-- Frank Church
======================= http://devblog.brahmancreations.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
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Yup, thats why we even have a param called "NDLB-allow-bad-iananame" in sofia profiles. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
FS looks at codec name too.
On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to
allow G729 as
inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
path, it relays the audio stream between endpoints, but can still
detect
DTMFs -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
detection is impossible. In this mode, FS is really a "dumb"
transparent
RTP-forwarder (this is required to get T38 working between the 2
endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et
établis à
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est
susceptible
d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass
media"
or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
really
need it.
My recommendation would be to use a codec other than G.729 unless
you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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-user s
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-- Frank Church
======================= http://devblog.brahmancreations.com
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I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end. Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending? On 5 April 2010 11:59, Mathieu Rene <mrene_lists@avgs.ca> wrote:
Yup, thats why we even have a param called "NDLB-allow-bad-iananame" in sofia profiles.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
FS looks at codec name too.
On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to
allow G729 as
inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
path, it relays the audio stream between endpoints, but can still
detect
DTMFs -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
detection is impossible. In this mode, FS is really a "dumb"
transparent
RTP-forwarder (this is required to get T38 working between the 2
endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et
établis à
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est
susceptible
d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass
media"
or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
really
need it.
My recommendation would be to use a codec other than G.729 unless
you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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-user s
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-- Frank Church
======================= http://devblog.brahmancreations.com
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-- Frank Church ======================= http://devblog.brahmancreations.com
Visit www.freeswitch.org, Click G729 at the top and buy some licenses. Problem solved ;) It supports the project and solves a problem all at the same time. Thanks, Brian PS: In depth install instructions are coming shortly for G729. On Apr 5, 2010, at 12:00 PM, Frank Church wrote:
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
On 5 April 2010 18:09, Brian West <brian@freeswitch.org> wrote:
Visit www.freeswitch.org, Click G729 at the top and buy some licenses. Problem solved ;) It supports the project and solves a problem all at the same time.
I am experimenting with Freeswitch for embedded bundled product offered freely to a service provider's customers, which is already running well on an Asterisk VM. I don't see them forking up for licenses on every installation. The product and the adapters that are bundled with it is a loss leader, sending out an installer costs them even more. I am looking at Freeswitch because of its native Windows support, as it avoids the VM route which makes the installation more complicated for less savvy or impatient users. For the Freeswitch developers I would say that there is a large installed base of Linksys adapters,their clones and other VoIP adapters working well with Asterisk's passthru. If the Linksys G729 passthru is failing because of an additional letter 'a' where it is not expected, then it should be fixed or reverted. If the "NDLB-allow-bad-iananame" is what fixes it should be documented upfront. After 3 days playing with Freeswitch and checking the documentation I see Freeswitch as the way forward for a development oriented user, on both Linux and Windows, but minor end user issues like these ought to be avoided, especially if the existing alternatives work well. It is only a developer or an enthusiast who would put in the effort of switching, because they can see the advantage. PS. After using "NDLB-allow-bad-iananame" in both the internal and external profiles the problem still exists. Does it work on its own, or does it permit the use of 'G729a' in the codec parameters in vars.xml?
Thanks, Brian PS: In depth install instructions are coming shortly for G729.
On Apr 5, 2010, at 12:00 PM, Frank Church wrote:
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
_______________________________________________ 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
-- Frank Church ======================= http://devblog.brahmancreations.com
Frank, can you please try that with another phone or softphone ? You will then know if it's the Linksys which is faulty or your provider. Also, you can try to upgrade the ATA to the latest firmware. I recently bought a PAP2T, and the firmware on that was not the latest one (it was v4, but v5 is the current version). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 05/04/2010 à 19:00, Frank Church a écrit :
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
On 5 April 2010 11:59, Mathieu Rene <mrene_lists@avgs.ca> wrote:
Yup, thats why we even have a param called "NDLB-allow-bad-iananame" in sofia profiles.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
FS looks at codec name too.
On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to
allow G729 as
inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
path, it relays the audio stream between endpoints, but can still
detect
DTMFs -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
detection is impossible. In this mode, FS is really a "dumb"
transparent
RTP-forwarder (this is required to get T38 working between the 2
endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et
établis à
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est
susceptible
d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass
media"
or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
really
need it.
My recommendation would be to use a codec other than G.729 unless
you have a
compelling reason, for example a carrier that only supports G. 729.
_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch-user s
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-- Frank Church
======================= http://devblog.brahmancreations.com
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======================= http://devblog.brahmancreations.com
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Try to set ptime from 0.030 (default) to 0.020 (20ms). Ognjen On Mon, Apr 5, 2010 at 7:00 PM, Frank Church <vfclists@googlemail.com>wrote:
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
On 5 April 2010 11:59, Mathieu Rene <mrene_lists@avgs.ca> wrote:
Yup, thats why we even have a param called "NDLB-allow-bad-iananame" in sofia profiles.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
FS looks at codec name too.
On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote:
No, FreeSWITCH does NOT need to be in bypass media or proxy media. You just need the regular passthrough module: mod_g729 and to
allow G729 as
inbound and outbound codecs in vars.xml. To summarize: -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
path, it relays the audio stream between endpoints, but can still
detect
DTMFs -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
detection is impossible. In this mode, FS is really a "dumb"
transparent
RTP-forwarder (this is required to get T38 working between the 2
endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et
établis à
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
non autorisée est interdite. Tout message électronique est
susceptible
d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
Le 04/04/2010 à 20:12, Max Clark a écrit :
Clarification - for G729 does freeswitch need to be in "bypass
media"
or "proxy media"? My understanding was that G729 would work with "proxy media" enabled and without the new fangled module?
-Max
On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote:
Frank Church <vfclists@googlemail.com> wrote:
I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.
FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
really
need it.
My recommendation would be to use a codec other than G.729 unless
you have a
compelling reason, for example a carrier that only supports G.729.
_______________________________________________
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-user s
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-- Frank Church
======================= http://devblog.brahmancreations.com
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======================= http://devblog.brahmancreations.com
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I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page? It is the only thing that has the value 0.030 On 5 April 2010 19:29, Ognjen Seslija <oseslija@gmail.com> wrote:
Try to set ptime from 0.030 (default) to 0.020 (20ms).
Ognjen
On Mon, Apr 5, 2010 at 7:00 PM, Frank Church <vfclists@googlemail.com> wrote:
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
On 5 April 2010 11:59, Mathieu Rene <mrene_lists@avgs.ca> wrote:
Yup, thats why we even have a param called "NDLB-allow-bad-iananame" in sofia profiles.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
FS looks at codec name too.
On Monday 05 April 2010, David Ponzone wrote:
Ognjen,
very good point, but I used to think that for G729 (and all payload id smaller than 97), FS was relying on the payload id, and not the name.
Am I wrong ?
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 05/04/2010 à 12:34, Ognjen Seslija a écrit :
G729a is invalid as a codec name. FS used to allow it but not anymore afaik. You should change codec name to G729 (I assume you're using Linksys product; there is a setting to change under SIP tab).
Ognjen
On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists@googlemail.com> wrote: Have you reviewed http://pastebin.freeswitch.org/12617 ? It has the G729 set in the codecs section. In this one it seems the call does not get to the external gateway. Freeswitch stops the call before calling the external gateway
I have checked it again a few times using the bypass_media, proxy_media settings.
And with those settings the call ends as soon as ringing starts or as sonn as the call is answered.
I will do another one just to confirm
On 4 April 2010 19:35, David Ponzone <david.ponzone@gmail.com> wrote: > No, FreeSWITCH does NOT need to be in bypass media or proxy media. > You just need the regular passthrough module: mod_g729 and to
allow G729 as
> inbound and outbound codecs in vars.xml. > To summarize: > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
the RTP
> path, it relays the audio stream between endpoints, but can still
detect
> DTMFs > -proxy media enabled: FreeSWITCH relays the audio stream
transparently, DTMF
> detection is impossible. In this mode, FS is really a "dumb"
transparent
> RTP-forwarder (this is required to get T38 working between the 2
endpoints)
> -bypass media enabled: FreeSWITCH is not in the RTP path > David Ponzone Direction Technique > email: david.ponzone@ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > Ce message et toutes les pièces jointes sont confidentiels et
établis à
> l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
> non autorisée est interdite. Tout message électronique est
susceptible
> d'altération. IPeva décline toute responsabilité au titre de ce
message s'il
> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
de ce
> message, merci de le détruire immédiatement et d'avertir
l'expéditeur.
> Le 04/04/2010 à 20:12, Max Clark a écrit : > > Clarification - for G729 does freeswitch need to be in "bypass
media"
> or "proxy media"? My understanding was that G729 would work with > "proxy media" enabled and without the new fangled module? > > -Max > > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason@jasonjgw.net>
wrote: > Frank Church <vfclists@googlemail.com> wrote: > > I am just trialling Freeswitch with Linksys adapters, whose default > > codec I have set to G729 with 'Use Pref Codec Only:' set to no. > > When I change that setting to 'yes' the calls don't go through. I > am > > using the latest Windows SVN. > > FreeSWITCH only supports G.729 in pass-through mode (i.e., with
bypass media
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
you
> really > > need it. > > My recommendation would be to use a codec other than G.729 unless
you have a
> compelling reason, for example a carrier that only supports G.729. > > > _______________________________________________ > > 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-user > s > > 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-user > s 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-user > s http://www.freeswitch.org
-- Frank Church
======================= http://devblog.brahmancreations.com
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======================= http://devblog.brahmancreations.com
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change it to 0.020 /b On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
I have changed it and this time I get this error 2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error! When it is on 0.030 I get the garbled sound. I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that? On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
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-- Frank Church ======================= http://devblog.brahmancreations.com
There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable-transcoding. Since this is becoming a neverending saga, join us on IRC for help. On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com>wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.com
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Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case. I have documented my findings on my <http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura> devblog<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura> On 6 April 2010 08:10, Ognjen Seslija <oseslija@gmail.com> wrote:
There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable-transcoding.
Since this is becoming a neverending saga, join us on IRC for help.
I have got to start using IRC more often, I have never really gotten into it, but I see that it is necessary when time is important. I prefer mailing lists though as the keep a better record of what has transpired for all to see. Thanks to all. My next question will be coming up shortly, this time on IRC first.
On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com> wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.com
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-- Frank Church ======================= http://devblog.brahmancreations.com
Frank Church <vfclists@googlemail.com> wrote:
Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case.
sofia profile <profile-name> rescan reloadxml should suffice for a codec change.
Frank, Restarting FS should not be necessary. If you change a SIP profile xml, or some parameters affecting it (some variables in vars.xml for example), you have to restart the profile, I think I told you that before: sofia profile internal restart reloadxml David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/04/2010 à 09:42, Frank Church a écrit :
Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case.
I have documented my findings on my devblog
On 6 April 2010 08:10, Ognjen Seslija <oseslija@gmail.com> wrote:
There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable- transcoding.
Since this is becoming a neverending saga, join us on IRC for help.
I have got to start using IRC more often, I have never really gotten into it, but I see that it is necessary when time is important. I prefer mailing lists though as the keep a better record of what has transpired for all to see.
Thanks to all.
My next question will be coming up shortly, this time on IRC first.
On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com
wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729
decoder
error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
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-- Frank Church
======================= http://devblog.brahmancreations.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
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-- Frank Church
======================= http://devblog.brahmancreations.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
What does reloadxml do on its own? I thought it was equivalent it executed a restart as well. On 6 April 2010 09:06, David Ponzone <david.ponzone@gmail.com> wrote:
Frank,
Restarting FS should not be necessary.
If you change a SIP profile xml, or some parameters affecting it (some variables in vars.xml for example), you have to restart the profile, I think I told you that before: sofia profile internal restart reloadxml
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
*Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. **IPeva** décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.* * *
Le 06/04/2010 à 09:42, Frank Church a écrit :
Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case.
I have documented my findings on my <http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura> devblog<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura>
On 6 April 2010 08:10, Ognjen Seslija <oseslija@gmail.com> wrote:
There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable-transcoding.
Since this is becoming a neverending saga, join us on IRC for help.
I have got to start using IRC more often, I have never really gotten into it, but I see that it is necessary when time is important. I prefer mailing lists though as the keep a better record of what has transpired for all to see.
Thanks to all.
My next question will be coming up shortly, this time on IRC first.
On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com> wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729
decoder
error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
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-- Frank Church
======================= http://devblog.brahmancreations.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
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-- Frank Church ======================= http://devblog.brahmancreations.com
reloadxml reads again all the XML tree into memory (and you can see this tree in freeswitch.xml.fsxml). That is enough for the XML dialplan or for the user directory, because the XML tree is parsed from memory when an INVITE or REGISTER hits FS. But SIP profiles are not "dynamic". If you change some parameters, you have to restart the profile using the new XML tree. You may think of a SIP profile as a kind of sub-FS instance. restart will do a complete stop/start on the profile (stopping existing calls) rescan is softer and re-configure the profile without restarting it I suspect there are situations where rescan is not enough (perhaps when you change the IP/port the profile is bound to). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/04/2010 à 10:29, Frank Church a écrit :
What does reloadxml do on its own?
I thought it was equivalent it executed a restart as well.
On 6 April 2010 09:06, David Ponzone <david.ponzone@gmail.com> wrote: Frank,
Restarting FS should not be necessary.
If you change a SIP profile xml, or some parameters affecting it (some variables in vars.xml for example), you have to restart the profile, I think I told you that before: sofia profile internal restart reloadxml
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 06/04/2010 à 09:42, Frank Church a écrit :
Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case.
I have documented my findings on my devblog
On 6 April 2010 08:10, Ognjen Seslija <oseslija@gmail.com> wrote:
There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable- transcoding.
Since this is becoming a neverending saga, join us on IRC for help.
I have got to start using IRC more often, I have never really gotten into it, but I see that it is necessary when time is important. I prefer mailing lists though as the keep a better record of what has transpired for all to see.
Thanks to all.
My next question will be coming up shortly, this time on IRC first.
On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com
wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.
729 decoder
error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
_______________________________________________ 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
-- Frank Church
======================= http://devblog.brahmancreations.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
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-- Frank Church
======================= http://devblog.brahmancreations.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
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-- Frank Church
======================= http://devblog.brahmancreations.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
Hi Frank, On 04/06/2010 01:00 AM, Frank Church wrote:
I changed the 'G729a Codec Name' in the Linksys to G729 and the calls were completely garbled, even the ringing. It could be affecting something on the provider end.
Could it be that the provider has a different G729 codec that is not compatible with the actual G729a the Linksys is sending?
G.729 and G.729A are identical on the wire, which is why there is only a single SDP code for them. The only time you would ever see an incompatible G.729 is if you are working with an ancient Cisco. They had a form of G.729 at one time that packed all the bits in the opposite order. Its rare to see that these days, though. Its more likely something strange happens with the ptime. Steve
participants (9)
-
Brian West
-
David Ponzone
-
Frank Church
-
Jason White
-
Mathieu Rene
-
Max Clark
-
Ognjen Seslija
-
Sergey Okhapkin
-
Steve Underwood