[Freeswitch-users] forcing codec on leg b

Sameer Khan sameer2k3t at gmail.com
Tue Dec 21 11:58:49 MSK 2010


in asterisk i m doing it like

i call perl file via agi and in that perl file i check in  the database what
should be the forwarding address for the incoming number
lets say call arrives at my box for 12121212 then a perl file is called
which checks in the database for the destination address and lets say the
destination address is xyz at 1.1.1.1

this all is achieved by xml curl

now in asterisk i add these lines in sip.conf
[1.1.1.1]
host=1.1.1.1
type=peer
disallow=all
allow=alaw

so when asterisk find that ip in sip.conf it force the codec on leg b to
alaw.


this i want to achieve in fs


On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> sorry, I dont understand what you want to achieve
>
> 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 21/12/2010 à 09:32, Sameer Khan a écrit :
>
> thanks for your reply david,
>
> but for that i will need to call absolute_Codec_string value from datbase
> which i dont want to do now.
>
> as i have one data base and running two asterisk on them with this fs
>
> cant i define in sip users or profile ,
>
> i saw an option there in brian.xml for cidr but all examples are username
> dependent
>
> i dont want user authentication as i am just forwarding the calls to my
> user
>
> i just want to check if ip is 1.1.1.1 it uses pcma for all
> numbers/extension going to the ip like this XXX at 1.1.1.1
>
>
> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> in the dialplan, use a condition to match network_addr and then set
>> absolute_codec_string to PCMA.
>>
>>  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 21/12/2010 à 09:18, Sameer Khan a écrit :
>>
>> Hello guyz,
>> Need your help
>>
>> I want my fs to behave like this
>>
>> when call arrives for a number 1234656789 at fs i call a php file in which
>> i check in the database that what should be the forwarding address of this
>> number. for example the forwarding address is 123456789 at 1.1.1.1 then i
>> return that forwarding address in bridge application in the end
>> i want to define some setting in sip profiles, user directory or wherever
>> so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>> like it is defined in asterisk as
>> [1.1.1.1]
>> host=1.1.1.1
>> type=peer
>> disallow=all
>> allow=alaw
>>
>>
>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>> defult
>> and this i want to define for around 40 ip addresses
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