[asterisk-users] passing '302 moved temporarily' back to the SIP provider

[Abhishek]

Satish Barot <satish4asterisk <at> gmail.com> writes:

> 
> On 5/9/13, Carlos Alvarez <carlos <at> televolve.com> wrote:
> > On Tue, May 7, 2013 at 10:05 PM, Satish Barot
> > <satish4asterisk <at> gmail.com>wrote:
> >
> >>
> >>
> >> promiscredir= yes in sip.conf should help you achieve your 
requirement.
> >>
> >
> > I haven't been able to get that to work in a similar situation, 
except we
> > are the provider.  It results in the new invite being from the CLID 
of the
> > original caller, and fails.
> >
> >
> > --
> > Carlos Alvarez
> > TelEvolve
> > 602-889-3003
> >
> Completely misunderstood the OP!
> Revised solution:
> Set promiscredir= no in sip.conf. I assume you land your dids in
> [incoming-trunk] and here is the basic dialplan tested on 11 but
> should work on 1.8.
> 
> [incoming-trunk]
> ;-- Handle Incoming DIDs. Mine start with 89 and are of 4 digits --;
> exten => 
_89XX,1,Noop(RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALL
ERID(dnid)})
> same => n,Set(__ORIGCHANNEL=${CHANNEL})
> same => n,Dial(SIP/${EXTEN},30)
> 
> ;-- Dialplan to handle 302 Moved temporarily --;
> exten => 
_X..,1,Noop(ORIGCHANNEL=${ORIGCHANNEL}::RDNIS=${CALLERID(rdnis)}::ANI=${
CALLERID(ani)}::DNID=${CALLERID(dnid)}::CHANNELTYPE=${CHANNEL(channeltyp
e)})
> same => n,ExecIf($["${CALLERID(rdnis)}"!=""]?
ChannelRedirect(${ORIGCHANNEL},back2provider,${EXTEN},1)
> same => n,Hangup()
> 
> [back2provider]
> ;--Send 302 back to provider --;
> exten => _X.,1,Transfer(${EXTEN})
> same => n,NoOp(TRANSFERSTATUS=${TRANSFERSTATUS})
> same => n,Hangup()
> 
> --Satish Barot
> Ahmedabad, India
> 
> --
> _____________________________________________________________________


Hi Satish,

We want to configure following setup:
  “A” initiated call to SIP1.
  SIP1 redirected CALL to SIP2(first redirection.
  SIP 2 return 302, and request a redirect to SIP3(with SIP3 IP in 
return packet).
  SIP1 receive a redirect from SIP2 with SIP3 IP.
  SIP1 makes a call to SIP3.
  SIP3 finally helps in landing a call to “B”

All SIP are asterisk servers.

Please help in configuring asterisk to send 302 request back to the 
server SIP1. 
We are not able to get anywhere.

Regards,
Abhishek