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| 02:31.37 | snuff-work | file: you around ? |
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| 09:22.22 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk |
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| 18:03.35 | oej | Good evening |
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| 18:40.31 | mjordan | oej: evening |
| 18:48.26 | jsmith | Ugh... why do Cisco phones have to break every time I update Asterisk :-( |
| 18:48.45 | pabelanger | jsmith, eep |
| 18:48.50 | *** join/#asterisk-dev rgagnon (~rgagnon@d24-57-198-219.home.cgocable.net) |
| 18:48.53 | jsmith | Any known issues with Cisco 7960s and Asterisk 1.8.20.1? |
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| 18:49.11 | jsmith | feels like he's gone back in time to the 1.8.4 release |
| 18:49.28 | pabelanger | Ha |
| 18:49.30 | pabelanger | ya, good times |
| 18:50.10 | jsmith | Phone sends a REGISTER, Asterisk throws up a debug message about the AUTH begin correct but the nonce being stale, and sends a 401 Unauthorized back |
| 18:50.29 | jsmith | Rinse and repeat, except that I no longer see the debug message |
| 18:50.32 | Juggie | would you prefer the cisco sucks or asterisk sucks answer :) |
| 18:51.06 | pabelanger | jsmith, cisco specific? I've had that issue in the past with asterisk and polycom |
| 18:51.16 | pabelanger | infact, i thought leifmadsen found a reason for it too |
| 18:51.19 | jsmith | pabelanger: Well, in this case, yes... |
| 18:51.41 | jsmith | Looks very much like https://issues.asterisk.org/jira/browse/ASTERISK-17830 |
| 18:52.33 | jsmith | Almost identical, in fact |
| 18:52.43 | pabelanger | that would be bad |
| 18:53.48 | Juggie | looks about right |
| 18:55.05 | jsmith | Anybody feel like helping a friend debug? |
| 18:55.38 | Juggie | old bug |
| 18:55.49 | jsmith | Should have been fixed in 1.8.5 |
| 18:55.55 | jsmith | but I think it came back in the 1.8.11 timeframe |
| 18:55.59 | Juggie | theres a post saying in returns |
| 18:56.20 | Juggie | did the fix get removed for causing problems elsewhere? |
| 18:56.57 | jsmith | No that I can find |
| 18:57.29 | pabelanger | that would be 11months ago |
| 18:57.32 | pabelanger | 1.8.11 |
| 19:00.06 | Juggie | older bug is here, https://issues.asterisk.org/jira/browse/ASTERISK-17535 |
| 19:01.52 | pabelanger | jsmith, pb a SIP trace |
| 19:04.11 | pabelanger | jsmith, Also, you upgrading to fix a specific issue, or just testing latest release? |
| 19:06.32 | jsmith | pabelanger: Fixing other issues |
| 19:06.38 | jsmith | pabelanger: Will pastebin a trace shortly |
| 19:08.56 | jsmith | pabelanger: http://pastebin.com/Jmdk0WgD |
| 19:09.04 | Qwell | Oh. Snap. |
| 19:09.34 | jsmith | Snap? |
| 19:09.42 | file | jsmith, where is Asterisk sending the reponses? |
| 19:09.50 | Qwell | jsmith: I just broke trunk, is all. |
| 19:10.36 | jsmith | file: 10.10.100.244 (local network, no localnet or externip settings set, on a flat local network) |
| 19:10.45 | file | need port |
| 19:12.39 | Juggie | looks like a bug with the via maybe |
| 19:13.02 | file | if this is what I think it is... there is no bug |
| 19:13.20 | Juggie | cisco not adhearing to spec? :) |
| 19:13.30 | file | always! |
| 19:13.46 | jsmith | file: Register came from port 51060 on the phone, to Asterisk's port 5060 |
| 19:13.46 | jsmith | Response goes back to port 51060 |
| 19:13.46 | jsmith | (from Asterisk's port 5060, of course) |
| 19:13.55 | file | then that's why |
| 19:14.02 | file | set "nat=no" for that entry in sip.conf |
| 19:14.31 | Juggie | Via: SIP/2.0/UDP 10.10.100.244:5060;branch=z9hG4bK2be2d211 |
| 19:14.48 | Juggie | so, that means reply to 10.10.100.244:5060 obviously? |
| 19:15.00 | file | that is where it wants it. |
| 19:15.41 | jsmith | file: Setting "nat=no" seems to have fixed the problem |
| 19:16.05 | Juggie | ic... nat was overriding what was found in parse_via(..) |
| 19:16.19 | file | force_rport got switched on by default sometime and some Cisco stuff is incompatible with rport |
| 19:16.20 | Juggie | interesting... oh well... |
| 19:19.19 | mjordan | Occurred in 1.8.7.2, due to AST-2011-013 |
| 19:19.37 | mjordan | To lessen the frequency of unintended username disclosure, the default NAT setting was changed to always respond to the port from which we received the request—the most commonly used option. |
| 19:19.48 | mjordan | except, of course, if you're Cisco. |
| 19:26.56 | pabelanger | Oh, yes. AST-2011-013. Never did back port that :) Need to revisit it |
| 19:44.11 | wdoekes | auto-nat would've been best.. but that didn't exist back then |
| 19:44.21 | wdoekes | s/auto-nat/auto-rport/ |
| 19:44.37 | file | auto rport? |
| 19:44.51 | wdoekes | auto-force-rport |
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| 20:26.32 | jsmith | FWIW, http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz is pointing at 2.6.1+2.6.1, instead of 2.6.2+2.6.2 |
| 20:27.13 | rmeyerriecks | thanks jsmith |
| 20:27.54 | jsmith | It also appears that the 2.6.2+2.6.2 version didn't put a changelog, etc. on the downloads site |
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| 20:34.02 | rmeyerriecks | jsmith: that link downloads 2.6.2 for me |
| 20:34.14 | rmeyerriecks | jsmith: it also looks like it's correct in the svn repo |
| 20:34.21 | jsmith | Weird |
| 20:35.25 | rmeyerriecks | jsmith: do you have a script that grabs the changelog link or something. tbh I think the *-complete changelog is pretty useless |
| 20:35.48 | jsmith | rmeyerriecks: No, I just manually noticed the changelog thing |
| 20:36.23 | jsmith | rmeyerriecks: But I do have a script that downloaded the latest DAHDI tarball |
| 20:36.28 | rmeyerriecks | jsmith: the switch to git kind of made us re-visit some of our processes, if that causes any problems for you just lemme know |
| 20:36.35 | jsmith | rmeyerriecks: Will do |
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| 21:42.00 | jmls | evening all |
| 21:42.44 | jmls | at what point is it worth a semi-literate asterisk person like me getting involved with testing / playing with the new cool stuff in -12 ? |
| 21:44.48 | mjordan | jmls: what are you interested in? |
| 21:44.49 | mjordan | :-) |
| 21:46.01 | file | Join the Pimp My SIP crew! |
| 21:50.46 | mjordan | that is probably the furthest along and easiest to test |
| 21:50.53 | mjordan | the bridge work/API work is still a bit more in flux |
| 21:51.03 | mjordan | but the API work is going to need people looking at it at some point in time :-) |
| 21:55.26 | file | yeah, we have... uh... cake! and stuff! |
| 21:59.02 | file | that being said we could use CLI commands >_> |
| 22:03.10 | mjordan | I'd go so far as to say that we need a plan for interfacing with the SIP resources outside of Dial |
| 22:03.35 | mjordan | that is, what do we want to expose via AMI |
| 22:03.39 | mjordan | or through dialplan functions |
| 22:03.42 | mjordan | and who exposes it |
| 22:03.48 | mjordan | (CLI commands are a handy start though) |
| 22:05.10 | mjordan | jmls: http://svn.asterisk.org/svn/asterisk/team/group/pimp_my_sip |
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| 23:37.43 | jmls | API is going to be most interesting going forwards |
| 23:38.03 | jmls | but AOR (?) sounds like that it may solve a lot of existing problems |
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| 23:57.28 | lathama | Did something change in menuselect today? Looking at 382900 |