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| 03:04.02 | *** topic/#asterisk-bugs is Asterisk Bugs Discussion -=- http://bugs.digium.com/ -=- http://asterisk.org/developers/bug-guidelines - If you are not discussing a report already posted to the bugtracker, the discussion probably belongs in #asterisk. -=- Overall issue count: 277 -=- Monitor new reports in #asterisk-commits -=- Test bugs marked Ready For Testing -=- Mantis happy now. |
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| 14:54.10 | file | dances in the door |
| 14:54.49 | caio1982 | do that robot thing :) |
| 15:31.44 | putnopvut | file: vvvvv |
| 15:31.47 | putnopvut | M12265 |
| 15:31.49 | MuffinMan | [new] [Asterisk] Applications/app_dial 0012265: SIP caller hanging up before answer does not stop Dial reported by kodomo http://bugs.digium.com/view.php?id=12265 |
| 15:32.06 | putnopvut | We discussed this a bit on Thursday or Friday, and I wanted to talk a bit more about it if you don't mind. |
| 15:33.01 | file | I'm logged onto someone's machine right now looking at a Mixmonitor issue but I can pencil you in afterwards |
| 15:33.16 | putnopvut | file: sure thing, just let me know when. |
| 15:33.46 | file | woah... |
| 15:33.49 | file | frowns |
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| 15:49.15 | file | this write factory is being kaboomed somehow |
| 15:49.19 | file | further frowns |
| 16:00.16 | file | putnopvut: okay, I needz me a break from this - let's talk |
| 16:00.29 | putnopvut | file: cool |
| 16:00.32 | putnopvut | So, a recap: |
| 16:00.47 | putnopvut | the person has a bug where if they call a phone and then hang up, the ringing phone continues ringing. |
| 16:01.04 | putnopvut | I looked at his sip debug vs a successful one of mine, and here's the difference. |
| 16:01.29 | putnopvut | With his, as soon as he sends the 200 OK to the calling phone (in response to the CANCEL), the calling phone responds with an ACK. |
| 16:01.54 | putnopvut | On reception of the ACK, Asterisk then cancels retransmission of the CANCEL to the called phone, meaning the far end never knew the call was canceled. |
| 16:02.20 | putnopvut | In my successful case, Asterisk sends the 200 OK to the calling phone, but then immediately sends a CANCEL to the far end too. |
| 16:02.34 | putnopvut | It's not until after this is sent that we receive the ACK from the calling phone. |
| 16:02.56 | putnopvut | Thus it appears there is a possible race condition. |
| 16:03.26 | putnopvut | So what I want to know is why we immediately respond to a CANCEL with a 200 OK instead of sending it on to the destination first? |
| 16:05.04 | file | this has IAX2 involved, I didn't notice that before |
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| 16:06.05 | file | I don't know what you mean by "sending it on" or "On reception of the ACK, Asterisk then cancels retransmission of the CANCEL to the called phone, meaning the far end never knew the call was canceled." |
| 16:06.07 | putnopvut | interesting, I hadn't either. |
| 16:06.34 | putnopvut | I'll pastebin the relevant part from the sip debug. |
| 16:07.21 | putnopvut | http://pastebin.ca/955372 |
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| 16:08.14 | putnopvut | Please let me know if I'm reading this incorrectly, but it appears that reception of the ACK causes us to cease retransmitting the CANCEL to the far end. |
| 16:09.08 | putnopvut | But I see now that there's a hangup called in handle_request_cancel which should take care of transmitting the CANCEL to the far end. |
| 16:09.22 | file | a hangup queued |
| 16:09.36 | putnopvut | Hmmm... |
| 16:12.07 | file | the question is, why aren't things acting on that hangup |
| 16:12.35 | putnopvut | Perhaps the invitestate is incorrect? |
| 16:15.46 | file | that shouldn't matter |
| 16:16.04 | file | ast_queue_hangup adds a frame to the channel's frame list, and writes data out the alert pipe |
| 16:16.20 | file | app_dial reads the channel but since it's a hangup frame nothing gets returned from ast_read, which is treated as though the channel hung up |
| 16:16.20 | putnopvut | Why not? It appears that the invitestate contributes towards whether a CANCEL is transmitted in sip_hangup. |
| 16:18.32 | putnopvut | I'm taking an early lunch. I'll talk later. |
| 16:18.44 | file | yeah, we need to talk verbally |
| 16:22.42 | file | putnopvut: give me a ring @6066 when you return |
| 16:42.42 | putnopvut | back |
| 16:55.25 | file | strikes a pose |
| 16:56.25 | putnopvut | strikes file |
| 16:56.34 | file | well now that is interesting... |
| 16:58.25 | putnopvut | May I ask what's going through your head, Mr. file? |
| 16:58.49 | file | putnopvut: I removed the ast_queue_hangup so that sip_scheddestroy logic executes... |
| 16:58.59 | file | quite... interesting |
| 17:00.16 | putnopvut | interesting in what way, my good man? |
| 17:00.29 | file | the channel eventually died, but it hung around for a bit... |
| 17:00.50 | putnopvut | That is interesting. |
| 17:01.08 | file | quite... odd |
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| 19:31.24 | putnopvut | M12279 |
| 19:31.25 | MuffinMan | [new] [Asterisk] Channels/chan_sip/General 0012279: [patch] Remove Event: header in Asterisk generated REGISTER requests reported by rjain http://bugs.digium.com/view.php?id=12279 |
| 19:31.33 | putnopvut | Seems fine to me any objections to committing? |
| 20:06.04 | jsmith-busy | putnopvut: Looks straightforward to me |
| 20:06.39 | putnopvut | Me too. I'm doing an svn blame right now to see if there was some specific reason for having that header there. |
| 20:08.30 | putnopvut | It is an incredibly old markster commit...I'm guessing it's safe to let go. |
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