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

Mon Feb 9 09:23:35 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 2/9/15 9:22 AM:
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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, for the supported version of Asterisk, years ago. I spent 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.
By the way we could take the debate to the community and let them take sides. I happen to be followed by 5 times more people than @Digium. 
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.

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, for the supported version of Asterisk, years ago. I spent 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.

> 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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