8 thoughts on - Retaining P-Asserted Info

• Am 16.02.2014 03:30, schrieb Nick Cameo:

  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}”
  \;privacy=off\;screen=no)

• Hello Markus,

  Thank you so much for your response. Our switch is already generating the needed P-Asserted header:

  P-Asserted-Identity: “John Doe”
  ; 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.

• 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

• 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” (which is asterisk peer extension and not)
  P-Asserted-Identity: “John Doe”
  ; 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

• 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

• 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—

• 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”
  ; user=phone; nat=yes.

  Forgive the noob,

  Nick.

• How about: SipAddHeader(${SIP_HEADER(P-Asserted-Identity)})

  Might have some issues with the ; character being see as start of comment.

  —–Original Message—