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rodrigoes
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hi

 

i h ave a problem please help me, how i can transfer a call that cames from sip trunk the trunk change the real numeber to an extention of the gateway and the call dont terminate because the pbx change the real numnber to a extencion are there any patterns for trnuks o what i have to configure i dont need to change the real number the did to cid the user will call a number and that munber must arive to other gateway , well in the meadle is the pbx and that is changing the real number. please help me.

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i h ave a problem please help me, how i can transfer a call that cames from sip trunk the trunk change the real numeber to an extention of the gateway and the call dont terminate because the pbx change the real numnber to a extencion are there any patterns for trnuks o what i have to configure i dont need to change the real number the did to cid the user will call a number and that munber must arive to other gateway , well in the meadle is the pbx and that is changing the real number. please help me.

 

Ehhhh... I have difficulties understanding what the problem is. Can you make an example?

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Ehhhh... I have difficulties understanding what the problem is. Can you make an example?

 

hi

 

sorry now the calls arive in the pbxnsip but the calls dont go to outbound for example , my sip provider is sending me calls to my pbxnsip and my pbxnsip must redirect the calls to audiocodes , but now the calls dont doit that for example

 

sip:59170533006@2xxxxxxxxx:5060 SIP/2.0 <this is the real number but the pbx transform the call to this To: "300" <sip:300@localhost> so tha call never terminate how can i fax that i need to arive to pbxnsip whit the real number and redirec that mumber to the audiocodes to terminate well de pbxnsip must do the CDR

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hi

 

sorry now the calls arive in the pbxnsip but the calls dont go to outbound for example , my sip provider is sending me calls to my pbxnsip and my pbxnsip must redirect the calls to audiocodes , but now the calls dont doit that for example

 

sip:59170533006@2xxxxxxxxx:5060 SIP/2.0 <this is the real number but the pbx transform the call to this To: "300" <sip:300@localhost> so tha call never terminate how can i fax that i need to arive to pbxnsip whit the real number and redirec that mumber to the audiocodes to terminate well de pbxnsip must do the CDR

 

Oh you mean having calls coming from a trunk and going out to a trunk. That is actually possible if you set "Accept Redirect" and "Assume that call comes from user".

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