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

[Private Name (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24015?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221816#comment-221816 ] 

Private Name commented on ASTERISK-24015:
-----------------------------------------

As noted by Rusty, the correct behavior is how the old SIP channels worked

    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.


> 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: SVN, 12.3.2, 12.5.0
>         Environment: Linux Fedora 20
>            Reporter: Private Name
>         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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