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

Sun Feb 8 17:47:34 CST 2015

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24015?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224808#comment-224808 ] 

Private Name edited comment on ASTERISK-24015 at 2/8/15 5:47 PM:
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The second set of traces and done without any g729 codec. My app does not need any codec, all it does is 
SIP/2.0 302 Moved Temporarily
But issue is not mine, it is an issue of Asterisk. I am just the messenger.
Besides this, the patent applies to companies that sale equipment of software to third parties. I am a one-person company. I use virtual machines and containers to separate customers, so to pay Digium G729 fees, it would take several times my gross income. Digium has not come up with a reasonable licensing methodology for thousands of mini-companies like mine, which simply could not afford the fees. And we are, I believe, 80% of the Asterisk industry. I also don't think that you are a judge of who and how people use g729 to survive and pay their bills. Do you have legal training? 
In any case, I invite you to look into what is killing my very simple app. As I mentioned, if I use regular SIP, same exact code, the FIFO count is about 72, versus hundreds of thousands.

Now that I think about it, I do own a G729 license fro Digium, a few, from a few years ago. If you have somebody contact me, I can give them my email and name and they will find me in the database. I also used to have a license, paid,for the supported version of Asterisk, years ago. I spend already thousands of $$ in Digium. As recently as last week, I wanted to spend even more, but Digium declined the request of one of my customers, who needed to outpulse the CALLERID(ANI2) field. It was a 10 line patch. We ended up getting the patch in Ukraine. Digium could have gotten that money, but, sadly, declined. I am sure it would have taken you 10 mins to write the 10 lines. Digium is leaving money on the table just because they decline to create ad-hoc patches that business need, and of course, if you need it, you need it now, not in one year.
Sorry about my ramblings.

was (Author: falves11):
The second set of traces and done without any g729 codec. My app does not need any codec, all it does is 
SIP/2.0 302 Moved Temporarily
But issue is not mine, it is an issue of Asterisk. I am just the messenger.
Besides this, the patent applies to companies that sale equipment of software to third parties. I am a one-person company. I use virtual machines and containers to separate customers, so to pay Digium G729 fees, it would take several times my gross income. Digium has not come up with a reasonable licensing methodology for thousands of mini-companies like mine, which simply could not afford the fees. And we are, I believe, 80% of the Asterisk industry. I also don't think that you are a judge of who and how people use g729 to survive and pay their bills. Do you have legal training? 
In any case, I invite you to look into what is killing my very simple app. As I mentioned, if I use regular SIP, same exact code, the FIFO count is about 72, versus hundreds of thousands.

> 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: backtrace.txt, full_answered.txt, full_no_answer.txt, myDebugLog, pjsip_trace.txt, valgrind.core.txt, valgrind.txt, valgrind.txt
>
>
> 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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