Number portability

Hi all,

I’ve to configure my softswitch to support number portability with Asterisk.

This is how i am thinking.

SSW will receive the call, it will send the call to asterisks, ast will look in database and will send back the SIP 302 release cause (temporary moved) and the new modified number. SSW will re send the call according to new number.

In above scenario, between SSW and Ast* only signalling will be used, i.e. without any media.

at this point i am not sure how to send back 302 with new modified number, as far i searched, we can send back the number by changing sip header contact.

Any help would be really appreciated.

PS: Is there any better option to get this solution?

Thanks in advance

Hi guys,

No reply?

Ok can we break this question in few steps; for now can someone only explain me how to send 302 release back with a modified number?


Since you are acting just as a proxy. I’m going to suggest using OpenSER/OpenSIPs for this solution as they make more sense to use in this situation than asterisk.

Hi daveg,

I had this point in mind, but later the same asterisk box will be used for TDM based LNP

Thanks anyway.

The Transfer application, used before answer, will do this.

Caution: Transfer returns success immediately. In particular, if used after answer, it can completely fail without the dialplan realising.


Can we do the same for PRI connection?

I am connected with TDM switch using PRI line and want to do the same, that means, when call comes from TDM then i want to do a blind transfer without answering, with a “Number changed” release cause with new number.


Hi Guys,

Is it somehow possible?

Thanks for yourtime, i am in an urgent need :frowning:

Look in the dahdi (zaptel) channel driver source code to see if there is an implementation of the transfer method. If there is, it might be possible; if there isn’t, it is definitely not possible.

Hi thanks for reply.

could you please guide me where can i see source code?