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@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
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@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@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 _______________________________________________ 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
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@1.1.1.1 On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@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 _______________________________________________ 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
sorry, I dont understand what you want to achieve 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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@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 _______________________________________________ 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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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@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@ipeva.fr>wrote:
sorry, I dont understand what you want to achieve
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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@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 _______________________________________________ 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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Hi, You can use this, absolute_codec_string=PCMA http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_paramete... Regds Sam On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t@gmail.com> wrote:
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@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@ipeva.fr>wrote:
sorry, I dont understand what you want to achieve
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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@1.1.1.1then 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 _______________________________________________ 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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i know that but i want such things from sip profiles, i dont want to pass this value from dialplan as my dialplan is binded in xml curl can i set this in sip settings? different codecs for different ips On Tue, Dec 21, 2010 at 2:54 PM, Sam <u2nsam@gmail.com> wrote:
Hi,
You can use this,
absolute_codec_string=PCMA
http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_paramete...
Regds Sam
On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t@gmail.com> wrote:
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@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@ipeva.fr>wrote:
sorry, I dont understand what you want to achieve
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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@1.1.1.1then 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 _______________________________________________ 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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No, you cannot do this in SIP profiles. The whole point of xml_curl is to allow you to make these kinds of decisions outside of FreeSWITCH using the dialplan binding. When using xml_curl you are not *forced* into using a dynamic dialplan. You can always fall back to the static XML. You would still need to create a dialplan extension that matches based on the criteria that are important to you. If I understand correctly you want to route the call based upon the source IP address. If that's the case then you could have a few extensions in the public context like this: <extension name="sample route, PCMA"> <condition field="network_address" expression="^10\.20\.30\.40$"/> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="{absolute_codec_string=PCMA} sofia/gateway/your_gateway/$1"/> </condition> </extension> <extension name="sample route, GSM"> <condition field="network_address" expression="^11\.22\.33\.44$"/> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="{absolute_codec_string=GSM} sofia/gateway/your_other_gateway/$1"/> </condition> </extension> To keep the dialplan simple I would use one extension per IP address. Hope this helps, -MC On Tue, Dec 21, 2010 at 2:54 AM, Sameer Khan <sameer2k3t@gmail.com> wrote:
i know that but i want such things from sip profiles,
i dont want to pass this value from dialplan as my dialplan is binded in xml curl
can i set this in sip settings? different codecs for different ips
On Tue, Dec 21, 2010 at 2:54 PM, Sam <u2nsam@gmail.com> wrote:
Hi,
You can use this,
absolute_codec_string=PCMA
http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_paramete...
Regds Sam
On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t@gmail.com>wrote:
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@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@ipeva.fr>wrote:
sorry, I dont understand what you want to achieve
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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@1.1.1.1then 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 _______________________________________________ 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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You can try looking at mod_easyroute. It should do what you want. On Tue, Dec 21, 2010 at 2:58 AM, Sameer Khan <sameer2k3t@gmail.com> wrote:
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@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@ipeva.fr>wrote:
sorry, I dont understand what you want to achieve
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 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@1.1.1.1
On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone@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@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@1.1.1.1then 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 _______________________________________________ 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
_______________________________________________ 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
_______________________________________________ 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
-- -Rupa
participants (5)
-
David Ponzone
-
Michael Collins
-
Rupa Schomaker
-
Sam
-
Sameer Khan