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| 01:56.17 | opticron | crud...just missed maxus2 |
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| 02:53.24 | maxhbp204 | Hi, i have installed asterisk 1.8-cert version, i have created 250 iax users, and i have done port forwarding 8080 to 4569 and rules are also ok, but i am not able register iax phone on server and it is also not showing any error, i have also put calltokenoptional=0.0.0.0/0.0.0.0 and requirecalltoken=no in general section as well, can anybody help me how can i enable registeration for iax? please |
| 02:54.02 | WIMPy | #asterisk |
| 02:54.28 | maxhbp204 | Hi, i have installed asterisk 1.8-cert version, i have created 250 iax users, and i have done port forwarding 8080 to 4569 and rules are also ok, but i am not able register iax phone on server and it is also not showing any error, i have also put calltokenoptional=0.0.0.0/0.0.0.0 and requirecalltoken=no in general section as well, can anybody help me how can i enable registeration for iax? please |
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| 12:08.05 | file | rawr |
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| 12:30.15 | file | definitely a Tuesday |
| 12:30.32 | leifmadsen | definitely |
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| 13:27.14 | acidfoo | moo |
| 13:29.30 | opticron | oom |
| 13:34.31 | russellb | yawn. |
| 13:34.56 | *** topic/#asterisk-dev by russellb -> Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk |
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| 14:11.05 | jrose_atDigium | A tear pie does not sound very appetizing. |
| 14:16.02 | jrose_atDigium | and here I thought I was in another channel. |
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| 16:08.38 | pabelanger | Does SIP message header order matter to asterisk? |
| 16:09.11 | pabelanger | not SIP messages, but their headers |
| 16:12.39 | wdoekes | no |
| 16:28.54 | jmls1 | congrats on the release of 11 - great work, and is much appreciated. |
| 16:29.03 | jmls1 | is a very happy -11 user |
| 16:31.16 | russellb | negative 11? |
| 16:31.21 | russellb | crazy |
| 16:31.24 | jmls1 | heh |
| 16:31.37 | jmls1 | nothing -ve about 11 ;) |
| 16:31.50 | russellb | yeah, some cool stuff. |
| 16:31.53 | russellb | wish i had a use for asterisk |
| 16:32.22 | russellb | instead i'll just sit around like an old man and bitch now and then |
| 16:33.18 | Qwell | jmls1 uses the heck out of Asterisk abs(11). |
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| 16:34.28 | jmls1 | 129799 calls processed :) |
| 16:36.11 | jmls1 | my kids saw a genuine rotary dial phone the other day, and asked "how does that work?" |
| 16:36.43 | jmls1 | "what do you mean you have to remember the number ? It has no phonebook ???" |
| 16:37.07 | jmls1 | the youngest girl _actually_ shivered :P |
| 16:40.20 | mjordan | russellb: bitch & moan away. And make sure you shout get off my lawn every now and again |
| 16:40.44 | mjordan | jmls1: thanks for the positiveness :-) |
| 16:41.01 | leifmadsen | jmls1: lolz |
| 16:41.16 | leifmadsen | I ran into a crashing issue with Asterisk 11 this morning |
| 16:41.25 | leifmadsen | ConfBridge() apparently really wants a timing source :D |
| 16:41.29 | leifmadsen | filed an issue |
| 16:41.36 | mjordan | I think there is one for that |
| 16:41.37 | leifmadsen | but it works great once you recompile dahdi after a kernel upgrade! |
| 16:41.48 | leifmadsen | mjordan: oh there might be? I didn't search because I'm a douche :D |
| 16:42.06 | leifmadsen | when it gets triaged, go head and link it and mark as dupe |
| 16:42.16 | leifmadsen | I wouldn't be surprised if I was the one who filed the original as well... |
| 16:42.54 | mjordan | ASTERISK-19448 |
| 16:45.22 | leifmadsen | mjordan: nicely done sir |
| 16:46.26 | mjordan | is a little scared he remember that issue |
| 16:48.59 | jmls1 | leifmadsen: what crashing ? We're using confbridge all the time ... now I'm worried :) |
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| 16:56.17 | leifmadsen | jmls1: no, it only crashes if you start asterisk without a timing source |
| 16:56.27 | leifmadsen | if it doesn't crash on initiation, you're good :) |
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| 17:00.11 | jmls1 | well, going to make bloody sure that I don't restart 8-) |
| 17:00.56 | mjordan | jmls1: still shouldn't be a problem - as long as you have any timing module loaded at all, you should be fine |
| 17:15.20 | jmls1 | don't beleive this. Just got a message from HQ. "System is running very slowly" |
| 17:15.22 | jmls1 | :( |
| 17:15.26 | jmls1 | what is WARNING[29006][C-00018692]: channel.c:1304 __ast_queue_frame: Exceptionally long voice queue length queuing |
| 17:15.34 | jmls1 | loads of these warnings on the console |
| 17:29.51 | acidfoo | who I need to ask for a team branch ? |
| 17:30.40 | Qwell | acidfoo: me! but send an email to mjordan. |
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| 17:39.19 | mjordan | jmls1: that usually occurs when frames start to not get serviced on a particular channel fast enough. Its usually more of an indication of a problem, rather than the actual problem itself |
| 17:39.34 | acidfoo | Qwell, ok ;) |
| 17:49.33 | jmls1 | jmls1: thanks, found that my jabber process had stopped running |
| 17:49.57 | jmls1 | I suspect that asterisk was trying to send *lots* of jabber messages to a server that wasn't there |
| 17:50.12 | jmls1 | is removing jabber from the mix. Stomp all the way now. |
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| 19:02.37 | pabelanger | is markm not around today? |
| 19:02.50 | pabelanger | Had a question about https://issues.asterisk.org/jira/browse/ASTERISK-17883 and CANCEL headers |
| 19:03.18 | pabelanger | specifically, if the CANCEL headers needs to match the original INVITE message |
| 19:05.18 | putnopvut | when you say markm, do you mean me or kobaz? |
| 19:05.30 | pabelanger | putnopvut: no, you |
| 19:05.36 | pabelanger | forgot your handle |
| 19:05.38 | pabelanger | :) |
| 19:05.53 | putnopvut | looks at the issue. |
| 19:06.21 | putnopvut | Ah yes, I already commented on it. |
| 19:07.09 | putnopvut | I remember that one. There are specific items in the CANCEL that need to match the original INVITE: to, from, branch, call-id, and I think cseq. |
| 19:11.57 | pabelanger | putnopvut: Ya, I am testing a prototype phone with asterisk and having the same issue with 481 back from asterisk |
| 19:12.03 | pabelanger | I suspect it is a header issue on the phone side |
| 19:12.17 | pabelanger | since using a polycom 560 works fine |
| 19:18.59 | pabelanger | putnopvut: but the order of those items in the CANCEL message does not matter does it? |
| 19:19.10 | pabelanger | or should it be the same order as the INVITE |
| 19:19.38 | file | order shouldn't matter |
| 19:21.07 | pabelanger | The only difference between it and the polycom seems to be the lack of: Contact, Allow, User-Agent and Authorization |
| 19:21.25 | Qwell | That's quite a lot to be lacking... |
| 19:22.56 | pabelanger | working: http://pastebin.com/mtpxG1Hg |
| 19:23.26 | pabelanger | 481 call/transaction does not exist: http://pastebin.com/32ZtByVX |
| 19:26.27 | pabelanger | So, the missing Authorization might be the key... and makes the most sense. |
| 19:30.39 | pabelanger | Oooh, interesting |
| 19:30.47 | pabelanger | <PROTECTED> |
| 19:31.14 | pabelanger | I mean: /* RFC 3261 section 9. "CANCEL has no effect on a request to which a UAS has already given a final response |
| 19:31.23 | pabelanger | is Ringing() considered a final response? |
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| 19:41.30 | pabelanger | putnopvut: So setting pedantic = false in sip.conf causes the phone to cancel properly |
| 19:42.36 | putnopvut | I haven't looked at the pastebin yet. Let me have a look. |
| 19:43.09 | putnopvut | Yeah, I think the problem is on Asterisk's end. |
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| 19:44.14 | pabelanger | putnopvut: Ya, there is a blob of data in handle_incoming() that check for pedantic, then does some additional checks with headers |
| 19:45.13 | pabelanger | I think I see where the issue is, but my SIP foo is weak. Will have to read up on the RFC and see if I can figure out what we are doing |
| 19:45.54 | putnopvut | Yeah, at first I thought it might be that Asterisk was expecting a to-tag in the CANCEL (which it shouldn't), but the Polycom doesn't send a to-tag and you said that worked. |
| 19:47.23 | pabelanger | I think this is the problem code: |
| 19:47.25 | pabelanger | <PROTECTED> |
| 19:47.36 | pabelanger | then we transmit 481 |
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| 20:32.54 | wdoekes | pabelanger: you are running a rather old version there (like the reporter) |
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| 20:40.30 | pabelanger | wdoekes: Yup, we run 1.8.7.1 in house. And backport fixes when needed. But you are right, I should test with branches/1.8 and see if it is already fixed |
| 20:48.46 | mjordan | pabelanger: there was a lot of SIP CANCEL bug fixes that went in more recent revisions - so there's a chance it is fixed. That being said, when we looked at ASTERISK-17883, it looked like it was still a problem - so there's a chance that your problem is exactly the same as theirs and its still broke |
| 20:55.08 | pabelanger | mjordan: okay |
| 20:55.26 | wdoekes | mjordan: can you reproduce that? reporter was using 1.8.4 |
| 20:56.18 | mjordan | wdoekes: haven't really tried. I was trolling through issues trying to close out the ones that we thought we fixed, and is it turns out, I couldn't verify (by inspection) that we had gotten that one. The CANCEL bug fixes were... varied? And there were a lot of them |
| 20:56.22 | mjordan | lots of niche cases |
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| 21:45.39 | pabelanger | putnopvut: mjordan: Soo, if I don't register my phone to asterisk (allowguest=yes) then dial, get ringing and cancel. Asterisk will terminate the call properly, if I setup registration of the phone, I get 481 Call/Transaction not found |
| 21:45.52 | pabelanger | and this is with branches/1.8 |
| 21:49.51 | putnopvut | that's...weird. |
| 21:52.23 | pabelanger | putnopvut: line 25662 of chan_sip.c in branches/1.8 is sending the 481 |
| 21:52.33 | putnopvut | pabelanger: 1.8 branch? |
| 21:52.49 | pabelanger | yup |
| 21:52.50 | pabelanger | if (!p->pendinginvite && (req->method == SIP_CANCEL)) { |
| 21:53.12 | pabelanger | so, if I understand asterisk doesn't think it is a pending invite |
| 21:54.42 | putnopvut | Yeah, for whatever reason p->pendinginvite has been cleared even though the INVITE transaction has not completed. |
| 21:55.39 | pabelanger | putnopvut: cool, so I am on the right track |
| 21:55.42 | pabelanger | now to figure out why |
| 21:56.43 | mjordan | is scared of p->pendinginvite getting cleared |
| 21:57.11 | putnopvut | It's possible that rather than matching the incoming CANCEL to the existing sip_pvt, it's creating a new one. The new one wouldn't have pendinginvite set. |
| 21:57.28 | putnopvut | This would be consistent with your finding that setting pedantic=no fixed the problem. |
| 22:01.43 | pabelanger | putnopvut: would we use find_call() to find it? |
| 22:02.19 | putnopvut | pabelanger: indeed, we would. |
| 22:02.23 | pabelanger | woot |
| 22:02.27 | pabelanger | look at me |
| 22:02.30 | pabelanger | hacking the chan_sip |
| 22:05.00 | pabelanger | so how can I get the logging from an unref to show up? For example: dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search"); |
| 22:05.13 | pabelanger | Is that text a debug message some place? |
| 22:05.47 | mjordan | pabelanger: it goes in the ref debug log |
| 22:05.58 | putnopvut | pabelanger: yeah, what you have to do is #define REF_DEBUG at the top of chan_sip.c. It will then log that info to a file. |
| 22:06.06 | putnopvut | sees if there is a wiki page about that. |
| 22:06.11 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging |
| 22:06.13 | mjordan | there is! |
| 22:06.16 | putnopvut | Yes! |
| 22:06.18 | mjordan | documentation ftw |
| 22:08.52 | pabelanger | hack the planet |
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| 22:17.53 | pabelanger | okay, I'll have to continue this in the morning. ast_debug() here I come |
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