[asterisk-bugs] [JIRA] (ASTERISK-25790) Unable to set X-P-Asserted-Identity with PJSIP_HEADER - the header doesn't make it into the SIP packet.

Mon Feb 15 19:09:33 CST 2016

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25790?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229512#comment-229512 ] 

Oliver Nauliv commented on ASTERISK-25790:
------------------------------------------

Hello Josh,

Yes, we changed the dialplan in accordance to Richard's recommendation. See the last test we did below:

{code}
[dial-out-nextiva]
exten => _X.,1,NoOp()
same => n,Set(CALLERID(num)=7143861234)
same => n,Dial(PJSIP/${EXTEN}@nextiva1,,B(addh^1^1))
same => n,NoOp(Status: ${DIALSTATUS})
same => n,Hangup()

[addh]
exten => 1,1,NoOp()
same => n,Verbose(0, In caller pre-dial handler!)
same => n,Set(PJSIP_HEADER(add,X-Nextiva-SayHello)=Hello)
same => n,Return()
{code}

When placing a call, we see that all the dialplan is being executed without error, and that the handler is being executed:

{code}
-nextiva:3] Set("PJSIP/100-00000025", "CALLERID(num)=7143861234") in new stack
-- Executing [9497951234 at dial-out-nextiva:4] Dial("PJSIP/100-00000025", "PJSIP/9497951234 at nextiva1,,B(addh^1^1)") in new stack
-- PJSIP/100-00000025 Internal Gosub(addh,1,1) start
-- Executing [1 at addh:1] NoOp("PJSIP/100-00000025", "") in new stack
-- Executing [1 at addh:2] Verbose("PJSIP/100-00000025", "0, In caller pre-dial handler!") in new stack
In caller pre-dial handler!
-- Executing [1 at addh:3] Set("PJSIP/100-00000025", "PJSIP_HEADER(add,X-Nextiva-SayHello)=Hello") in new stack
-- Executing [1 at addh:4] Return("PJSIP/100-00000025", "") in new stack
== Spawn extension (dial-out-nextiva, 9497951234, 4) exited non-zero on 'PJSIP/100-00000025'
-- PJSIP/100-00000025 Internal Gosub(addh,1,1) complete GOSUB_RETVAL=
-- Called PJSIP/9497951234 at nextiva1
{code}

However when we look at the SIP packet, there is still no header in it.
We tried all sort of combinations with the header, make it as simple as X-Test=hello; it's always executed without error, but never been added to the headers of the SIP transaction.

{code}
<--- Transmitting SIP request (901 bytes) to UDP:208.73.123.45:5060 --->
INVITE sip:9497951234 at bt.voipdnsservers.com:5060 SIP/2.0
Via: SIP/2.0/UDP 107.207.12.34:5060;rport;branch=z9hG4bKPjRXQkKMQGIZtMGkCMdxtYGPZ-JPsPVj0n
From: "Desk" ;tag=17Huzx1HCGrq-cMOeuldTY
To: sip:9497951234 at bt.voipdnsservers.com
Contact: 
Call-ID: RMn2GlpyVFPyxPZ5VCn.ZJrjE9vm
CSeq: 16525 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 237

v=0
o=- 453472723 453472723 IN IP4 107.207.12.34
s=Asterisk
c=IN IP4 107.207.12.34
t=0 0
m=audio 15224 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
{code}

> Unable to set X-P-Asserted-Identity with PJSIP_HEADER - the header doesn't make it into the SIP packet.
> -------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25790
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25790
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.1.0
>            Reporter: Oliver Nauliv
>            Assignee: Oliver Nauliv
>
> We are unable to send the X-P-Asserted-Identity, because while the dialplan runs PJSIP_HEADER without error, the option doesn't make it into the SIP packet.
> h2. SIP TRACE FOR OUTBOUND CALL
> {noformat}
> <--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
> INVITE sip:9492345566 at 208.73.140.70:5060 SIP/2.0
> Via: SIP/2.0/UDP 107.207.90.90:5060;rport;branch=z9hG4bKPja-5qMYTknEobutnWKzOTPpSrCg3swCz
> From: "Manager Office"
> <sip:7143861212 at 107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
> To: <sip:9492345566 at 208.73.140.70>
> Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
> Call-ID: QmF69h0tpNPtZhE6IrwYqtoQ67kcJ5pn
> CSeq: 9147 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
> PRACK, REFER, REGISTER, MESSAGE
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 237
> v=0
> o=- 455864548 455864548 IN IP4 107.207.90.90
> s=Asterisk
> c=IN IP4 107.207.90.90
> t=0 0
> m=audio 19628 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> {noformat}
> h2. DIALPLAN FOR OUTBOUND CALL
> {noformat}
> [dial-out]
> exten => _X.,1,NoOp()
> same => n,Set(PJSIP_HEADER(add,X-P-Asserted-Identity)=sip:7147890101)
> same => n,Set(CALLERID(num)=7143861212)
> same => n,Dial(PJSIP/${EXTEN}@nextiva1,45,r)
> same => n,Hangup()
> {noformat}
> h2. PJSIP CONFIGURATION FOR SIP TRUNK
> {noformat}
> [nextiva1]
> type=endpoint
> transport=transport-udp
> context=from-nextiva
> disallow=all
> allow=ulaw
> aors=nextiva1-aor
> [nextiva1-auth]
> type=auth
> auth_type=userpass
> username=7143861212
> password=NotMyRealPassword
> [nextiva1-reg]
> type=registration
> outbound_auth=nextiva1-auth
> server_uri=sip:bt.voipdnsservers.com
> client_uri=sip:7143861212 at bt.voipdnsservers.com
> [nextiva1]
> type=identify
> endpoint=nextiva1
> match=208.73.140.70
> [nextiva1-aor]
> type=aor
> contact=sip:bt.voipdnsservers.com:5060
> {noformat}

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