[Freeswitch-users] How to configure g729 pass though on Freeswitch

Tue Apr 6 01:51:09 PDT 2010

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 at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
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www.ipeva.fr  -   www.ipeva-studio.com

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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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>
>
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>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
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