[asterisk-bugs] [JIRA] (ASTERISK-24015) app_transfer fails with PJSIP channels

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24015?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24015:
------------------------------------

    Attachment: full_no_answer.txt
                full_answered.txt

I was able to reproduce and further define the issue.

Attaching full_answered.txt showing debug and trace for a PJSIP call that is answered before calling app_transfer. Then, full_no_answer.txt showing an unanswered call hitting app_transfer.

Using the same phone, and similar configuration in chan_sip and res_pjsip:

When using SIP channels with app_transfer:

 * Call is *NOT* answered, hits Transfer, Asterisk responds with "302 Moved Temporarily"
 * Call is *answered*, his Transfer, Asterisk responds with REFER and the phone responds appropriately.

When using PJSIP channels with app_transfer:

 * Call is *NOT* answered, hits Transfer, Asterisk responds with "603 Decline"
 * Call is *answered*, his Transfer, Asterisk responds with REFER and the phone *fails to process it*.

I noted that in the PJSIP REFER it does not include a Referred-by header and the Refer-to header does not include the domain/IP part of the address.



> app_transfer fails with PJSIP channels
> --------------------------------------
>
>                 Key: ASTERISK-24015
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24015
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_transfer
>    Affects Versions: 12.3.2
>         Environment: Linux Fedora 20
>            Reporter: Private Name
>            Assignee: Rusty Newton
>         Attachments: full_answered.txt, full_no_answer.txt, myDebugLog
>
>
> When using PJSIP, the Transfer application does not behave like the when using the old SIP channel, i.e., generate 301 Redirect messages
> Here is the trace
> {noformat}
> [Jul  9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer'
>     -- Executing [17274428141 at redirect:30] Transfer("PJSIP/Client.1.1.1.1-00000002", "PJSIP/17274428141;rn=+18134029999;npdi at 1.1.1.1") in new stack
> [Jul  9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869 pbx_extension_helper: Launching 'Verbose'
>     -- Executing [17274428141 at redirect:31] Verbose("PJSIP/Client.1.1.1.1-00000002", "2,Transferred: 17274428141;rn=+18134029999;npdi at 1.1.1.1") in new stack
>   == Transferred: 17274428141;rn=+18134029999;npdi at 1.1.1.1
>     -- Auto fallthrough, channel 'PJSIP/Client.1.1.1.1-00000002' status is 'UNKNOWN'
> [Jul  9 21:39:29] DEBUG[47716][C-00000002]: channel.c:2597 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'PJSIP/Client.1.1.1.1-00000002'
> [Jul  9 21:39:29] DEBUG[47716][C-00000002]: channel.c:2753 ast_hangup: Hanging up channel 'PJSIP/Client.1.1.1.1-00000002'
> [Jul  9 21:39:29] DEBUG[47716][C-00000002]: chan_pjsip.c:1578 hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP)
> <--- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 --->
> SIP/2.0 603 Decline
> v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z-
> i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY
> f: "9544447408" <sip:9544447408 at 8.26.191.189>;tag=82c82c1d
> t: <sip:17274428141 at 8.26.191.189>;tag=09f3a67a-f457-46d1-8d16-243478ac3859
> CSeq: 1 INVITE
> Reason: Q.850;cause=0
> l:  0
> {noformat}
> Note: it makes no difference if the endpoint has "allow_transfer" in false or true, yes or now. The behavior is identical.
> This issue is a blocker for me, since I process several million redirects per day. Hence the importance. I already converted everything else and it works perfectly,



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