Re: [asterisk-users] Retaining P-Asserted Info
HI Have you tried: sendrpid = pai ; Use the P-Asserted-Identity header ; to send the identity of the remote party in the sip.conf? Regards Ish On 16 February 2014 20:29, Nick Cameo sym...@gmail.com wrote: Hello Markus, Thank you so much for your response. Our switch is already generating the needed P-Asserted header: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. I really did not want to have to rebuild it using `SIPAddHeader` however, if I have no choice, can someone please provide an extension rule that will include the exiting inbound leg line above in the outbound leg. Kind Regards, Nick. -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Retaining P-Asserted Info
Hello Ishfaq, I just tried it and it did create a P-Asserted header however it contains the extension of the asterisk peer not what was passed by our switch. From the previous example: P-Asserted-Identity: 222 sip:222@192.168.2.10 (which is asterisk peer extension and not) P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. (which is being passed by the call leg) Is there a flag that retains the rpid from the call leg? Kind Regards, Nick -- _ -- 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] Retaining P-Asserted Info
On Mon, Feb 17, 2014 at 4:29 AM, Nick Cameo sym...@gmail.com wrote: Hello Ishfaq, I just tried it and it did create a P-Asserted header however it contains the extension of the asterisk peer not what was passed by our switch. From the previous example: P-Asserted-Identity: 222 sip:222@192.168.2.10 (which is asterisk peer extension and not) P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. (which is being passed by the call leg) Is there a flag that retains the rpid from the call leg? No. Asterisk is a back to back user agent, not a proxy. Overriding the settings of a peer with the peer that it is bridged with is typically contrary to Asterisk's nature. If you want to copy information from one SIP channel to another, you should do as Markus suggested. Matt -- _ -- 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] Retaining P-Asserted Info
Asterisk is a B2BUA -- think of it as two calls, one inbound call from your switch to Asterisk and one for outbound call from Asterisk to the destination. Using SIPAddHeader or similar is the proper way to copy headers from the inbound call to the outbound call in Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo Sent: Sunday, February 16, 2014 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retaining P-Asserted Info Hello Markus, Thank you so much for your response. Our switch is already generating the needed P-Asserted header: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. I really did not want to have to rebuild it using `SIPAddHeader` however, if I have no choice, can someone please provide an extension rule that will include the exiting inbound leg line above in the outbound leg. Kind Regards, Nick. -- _ -- 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 -- _ -- 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] Retaining P-Asserted Info
Hey Guys, I really appreciate this and I apologize for asking however, we do not have any way to test in advance outside of our live environment. Can someone kindly provide a working extension rule that will retain the following P-Asserted info that is existent from the inbound-leg to the outbound-leg using `SIPAddHeader`: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. Forgive the noob, Nick. -- _ -- 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] Retaining P-Asserted Info
How about: SipAddHeader(${SIP_HEADER(P-Asserted-Identity)})
Might have some issues with the ; character being see as start of comment.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo
Sent: Monday, February 17, 2014 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Retaining P-Asserted Info
Hey Guys, I really appreciate this and I apologize for asking however, we do
not have any way to test in advance outside of our live environment. Can
someone kindly provide a working extension rule that will retain the following
P-Asserted info that is existent from the inbound-leg to the outbound-leg using
`SIPAddHeader`:
P-Asserted-Identity: John Doe
sip:14167493...@toronto.location.com; user=phone; nat=yes.
Forgive the noob,
Nick.
--
_
-- 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
--
_
-- 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] Retaining P-Asserted Info
Am 16.02.2014 03:30, schrieb Nick Cameo:
Tried setting `sendrpid = yes` and still same problem. We really don't want to
have to `SipAddHeader` as it is already being formed by our switch.
From http://www.voip-info.org/wiki/view/Asterisk+SIP+trustrpid :
-snip-
P-Asserted-Identity
Asterisk does nothing when it receives a P-Asserted-Identity header. It
ignores it totally no matter what settings you use for trustrpid or
sendrpid. It does not copy it from an inbound call leg to an outbound
call leg for a bridged SIP-to-SIP call.
-snip-
I use SIPAddHeader to achieve what you described.
For example, I have a DID provider who only sends P-Asserted-Identity,
and at the same time a customer who needs that data as Remote-Party-ID,
so I'm doing this to send it to him:
exten = _X.,n,Set(RPID=${SIP_HEADER(P-Asserted-Identity)})
exten = _X.,n,Set(RPID=${CUT(RPID,\+,2-)})
exten = _X.,n,Set(RPID=${CUT(RPID,@,-1)})
exten = _X.,n,SIPAddHeader(Remote-Party-ID: ${RPID}
sip:${RPID}@my-ip-address\;privacy=off\;screen=no)
--
_
-- 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] Retaining P-Asserted Info
Hello Markus, Thank you so much for your response. Our switch is already generating the needed P-Asserted header: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. I really did not want to have to rebuild it using `SIPAddHeader` however, if I have no choice, can someone please provide an extension rule that will include the exiting inbound leg line above in the outbound leg. Kind Regards, Nick. -- _ -- 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
