Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
Satish Barot gmail.com> writes:
>
> On 5/9/13, Carlos Alvarez televolve.com> wrote:
> > On Tue, May 7, 2013 at 10:05 PM, Satish Barot
> > 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On Tue, May 7, 2013 at 10:05 PM, Satish Barot satish4aster...@gmail.comwrote: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
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=${CALLERID(dnid)})
same = n,Set(__ORIGCHANNEL=${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(channeltype)})
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On 5/9/13, Satish Barot satish4aster...@gmail.com wrote:
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
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=${CALLERID(dnid)})
same = n,Set(__ORIGCHANNEL=${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(channeltype)})
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
[incoming-trunk] is also a context of my SIP extensions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] passing '302 moved temporarily' back to the SIP provider
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone A has a CFW all calls to a phone number in public network (Mobile Phone)
incoming call to Phone A, Phone A sends back a 302 response. This results in:
-- Got SIP response 302 Moved Temporarily back from 192.168.2.188:5072
-- Now forwarding SIP/public-00e7 to
'Local/00662943825@internal_extensions' (thanks to SIP/25-00e8)
[May 7 12:22:15] NOTICE[31241]: app_dial.c:859 do_forward: Not accepting call
completion offers from call-forward recipient
Local/00662943825@internal_extensions-f967;1
and a Dial via Local channel to the SIP provider is made.
However, two channels are consumed and bandwith is wasted.
Second approach:
I tried to catch the redirect in the call forward context which is defined
before the Dial statement:
and do the redirect with app_transfer.
e.g. snippet from macro-stdexten:
exten = s,n,Set(_FORWARD_CONTEXT=from_sip_forward)
exten = s,n,Dial(${ARG2},${ARG4},tTfwW)
[from_sip_forward]
include = internal_devices
exten = _X.,1,Verbose(1,${CALLERID(num)} tries call forward to ${EXTEN} for
device ${CALLERID(rdnis)})
exten = _X.,n,Transfer(${EXT_TRUNK}/${EXTEN})
exten = _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS})
However, this does not work,
Is there a way to send the 302 response to the VoIP provider ?
Thanks.
Hans
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On 5/7/13, Johann Steinwendtner steinwendt...@gmx.net wrote:
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone A has a CFW all calls to a phone number in public network (Mobile
Phone)
incoming call to Phone A, Phone A sends back a 302 response. This results
in:
-- Got SIP response 302 Moved Temporarily back from 192.168.2.188:5072
-- Now forwarding SIP/public-00e7 to
'Local/00662943825@internal_extensions' (thanks to SIP/25-00e8)
[May 7 12:22:15] NOTICE[31241]: app_dial.c:859 do_forward: Not accepting
call completion offers from call-forward recipient
Local/00662943825@internal_extensions-f967;1
and a Dial via Local channel to the SIP provider is made.
However, two channels are consumed and bandwith is wasted.
Second approach:
I tried to catch the redirect in the call forward context which is defined
before the Dial statement:
and do the redirect with app_transfer.
e.g. snippet from macro-stdexten:
exten = s,n,Set(_FORWARD_CONTEXT=from_sip_forward)
exten = s,n,Dial(${ARG2},${ARG4},tTfwW)
[from_sip_forward]
include = internal_devices
exten = _X.,1,Verbose(1,${CALLERID(num)} tries call forward to ${EXTEN} for
device ${CALLERID(rdnis)})
exten = _X.,n,Transfer(${EXT_TRUNK}/${EXTEN})
exten = _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS})
However, this does not work,
Is there a way to send the 302 response to the VoIP provider ?
Thanks.
Hans
promiscredir= yes in sip.conf should help you achieve your requirement.
--Satish Barot
Ahmedabad, India.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
