[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=224562#comment-224562 ] 

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

Version 13  does not compile pjsip
   [CC] res_pjsip/pjsip_distributor.c -> res_pjsip/pjsip_distributor.o
res_pjsip/pjsip_distributor.c: In function ‘find_dialog’:
res_pjsip/pjsip_distributor.c:145:21: error: ‘pjsip_transaction’ has no member named ‘mutex’
make[1]: *** [res_pjsip/pjsip_distributor.o] Error 1
make: *** [res] Error 2
don't know what version exactly is it, for it is a new computer but I did this today Sunday 19th 2015 at 8 PM
git clone https://github.com/asterisk/pjproject pjproject
and 
svn co http://svn.digium.com/svn/asterisk/branches/13 asterisk
I did compile pjproject like this
./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr
make dep
make
make install
ldconfig
Note: I also tried downloading 
svn co http://svn.pjsip.org/repos/pjproject/trunk/ pjproject-trunk
but it fails to compile with a larger number of errors.





> 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
>            Assignee: Matt Jordan
>         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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