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SIP call forward wo/302 problem

Hi.

I am useing sip redirecting with

no supplementary-service sip moved-temporarily

so.
I get a call via sip-ua to my router(6811231)
I have a dial-peer, wich sends this call to the phone
dial-peer voice 6631 voip
destination-pattern 6756
session protocol sipv2
session target ipv4:10.0.0.2
codec g711alaw
now the phone is forwarded to some number(lets say 53341662) wich gets routed to the sip provider via sip-ua.
the phone responds to my invite with 302 moved-temporarily.
because the router has
no supplementary-service sip moved-temporarily
it does not send 302 to sip provider, instead it makes a new call-leg with diversion header attached.
now here is the problem:
because of redirecting..the router sends the wrong calling number, it sends the original calling number (6811231) not 6631 as it should
now, because the calling number is 6811231, i get calling user not registred from my sip provider.
So... can somebody tell me how can i copy the original called number to the calling number field in new INVITE's.
i tried
voice translation-rule 5
rule 1 /.+/ /6631/
voice translation-profile test
translate calling 5
this work, but when I have multiple numbers..then i have a problem, because all the outgoing calls calling number will be changed to 6631.
Keep in mind that i am not useing telephony-service nor, voice registrar.
1 REPLY

Re: SIP call forward wo/302 problem

In my opinion translation rules are the right way.

You say "this work, but when I have multiple numbers..then i have a problem, because all the outgoing calls calling number will be changed to 6631.".

How many numbers are configured in your PBX? Have these numbers a common root part?

You can create up to 16 rules under a voice translation rule.

Can you post your PBX configuration and provide more examples?

Thanks.

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