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| 11:43.38 | MAbbas | Hi, Asterisk is missing AMI events under load. How to fix this issue? |
| 11:46.19 | MAbbas | I have verified this under asterisk ver 1.4.20 and 1.4.29 |
| 11:50.10 | snuff-home | check the tracker.. they might have something listed already |
| 11:59.55 | [psy] | hmm thats bad |
| 12:00.25 | snuff-home | another way i guess.. see if reducing the 'console' logging |
| 12:00.29 | snuff-home | helps |
| 12:00.37 | [psy] | and turn of debugging |
| 12:02.05 | snuff-home | ppl have known that logging onto the console can cause asterisk stalls or ami misses |
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| 12:43.21 | snuff-home | ello eliel |
| 12:44.14 | eliel | morning snuff-home! |
| 12:44.39 | snuff-home | not exactly here ;) 22:43 |
| 12:45.35 | snuff-home | you must be getting close to signoff on 'data get' right? |
| 12:57.13 | eliel | hope so! |
| 12:57.15 | eliel | :) |
| 12:58.23 | snuff-home | if i'm lucky after easter break i get HRtime CDR |
| 12:58.26 | snuff-home | in |
| 12:59.33 | snuff-home | decides to get some zzz's.. back to work tomorrow.. 4day weekend was nice |
| 12:59.39 | eliel | great! |
| 12:59.46 | eliel | good luck with that! :) |
| 13:00.00 | snuff-home | btw unixodbc is EVIL |
| 13:00.19 | snuff-home | it can't handle a 'space' in front of definitions |
| 13:00.30 | snuff-home | aka ' [blah]' |
| 13:00.38 | snuff-home | will not work at all |
| 13:01.06 | snuff-home | not user error friendly |
| 13:01.56 | snuff-home | u'd think unixODBC code would strip leading spaces.. |
| 13:02.13 | snuff-home | now goes to bed after bitchin about odbc.. |
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| 13:10.44 | pabelanger | morning |
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| 14:25.17 | pabelanger | Anybody tried using Eclipse for development? |
| 14:26.00 | jsmith | For Asterisk development? |
| 14:26.02 | jsmith | I haven't... |
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| 14:26.12 | jsmith | I've used it for some limited Android development though |
| 14:29.04 | pabelanger | I usually use 'vi' most of the time, but I was curious what other people did. Since eclipse is an IDE, figured somebody maybe using it |
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| 15:18.07 | *** join/#asterisk-dev cellofellow (~josh@137.190.202.223) |
| 15:18.40 | cellofellow | What kind of experience are you guys looking for in a GSoC student? |
| 15:19.34 | cellofellow | Your listing says C, Lua, and Python. I know Python really well, and am at least schooled in C. Lua I hear is similar to JavaScript, so I could learn that. |
| 15:21.18 | Corydon76-lap | cellofellow: that would depend upon which project you chose |
| 15:21.29 | cellofellow | ok, let's look at the list |
| 15:22.16 | Corydon76-lap | I don't think we have any projects that are not wholly C on the list, but we're open to additional ideas |
| 15:22.39 | cellofellow | I wonder how much of this depends on having PBX hardware. |
| 15:22.48 | cellofellow | I currently have none. |
| 15:23.35 | Corydon76-lap | I don't think any of it requires use of PBX hardware |
| 15:23.47 | cellofellow | ok |
| 15:24.11 | Corydon76-lap | You may need to have a SIP softphone to test certain items, though |
| 15:24.21 | cellofellow | that's no biggie |
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| 15:24.57 | Corydon76-lap | And in fact, we would prefer if the testing did not require a softphone (other than the sipp command line test tool) |
| 15:25.21 | cellofellow | oh, ok |
| 15:27.41 | cellofellow | honestly not familiar enough with asterisk (basically know what it is) to offer any new cool ideas. |
| 15:29.03 | Corydon76-lap | Then this may not be the project for you |
| 15:31.15 | cellofellow | ok, I'll look elsewhere. Thanks for your time. |
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| 16:20.46 | Corydon76-lap | Heh, b3nt_pin |
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| 19:09.15 | Kobaz | i think i found a bug |
| 19:09.58 | seanbright | heh |
| 19:11.02 | Kobaz | bet you haven't heard that one in a while :P |
| 19:12.08 | Kobaz | http://pastebin.com/wwJrXKrJ |
| 19:12.19 | Kobaz | and i think it has to do with local channels |
| 19:12.52 | Kobaz | so... you do the originate... it calls the first phone... first phone picks up... calls the destination... and the destination gets a callerid of "s <s>" |
| 19:15.35 | seanbright | that originate syntax looks weird |
| 19:16.00 | seanbright | is that manager? |
| 19:16.01 | putnopvut | your face looks weird |
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| 19:16.27 | putnopvut | yeah, that's manager as being set in a python script I believe. |
| 19:16.31 | Kobaz | it's perl |
| 19:16.41 | Kobaz | it's a hash of the stuff going to the manager |
| 19:16.43 | putnopvut | F#*(K |
| 19:16.48 | Kobaz | i can convert it to plain |
| 19:16.50 | seanbright | ahh |
| 19:16.52 | putnopvut | lol |
| 19:17.14 | Kobaz | i'll clean the whole thing up and make a bug report |
| 19:17.27 | Kobaz | but i don't see anything obviously wrong |
| 19:17.41 | Kobaz | callerid is being set in dialplan just fine according to the console |
| 19:18.17 | Kobaz | i think i've had this problem before... i forgot what the workaround was |
| 19:18.37 | seanbright | cool, cool. |
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| 19:23.55 | Kobaz | so that looks like a bug... right? |
| 19:24.13 | Kobaz | there's no stupid stuff in there? |
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| 20:13.01 | pabelanger | If we, asterisk, get a SIP URI that is not encoded properly, do we have any obligations to make it proper? |
| 20:13.13 | pabelanger | M17127 |
| 20:13.15 | MuffinMan | [new] [Asterisk] Channels/chan_sip/General 0017127: Wrong encoding of SIP URI reported by sdaniels https://issues.asterisk.org/view.php?id=17127 |
| 20:14.24 | pabelanger | or am I just reading the report wrong. |
| 20:14.25 | Marquis42 | I would say if Asterisk receives a URI that isn't encoded properly, we should reject it. |
| 20:15.09 | pabelanger | Ya, I'm not sure what this guy is reporting. From the sounds of it, he is getting an inbound call to asterisk, and the URI is being set improperly. |
| 20:15.25 | Marquis42 | But then I just read the actually issue you linked, and I'm not sure there is enough information to understand the problem. |
| 20:15.34 | putnopvut | It sounds like the URI is not written that way coming into Asterisk. Rather, he is using that URI in an outbound INVITE from Asterisk and a phone is rejecting it. |
| 20:16.04 | putnopvut | But I agree with Marquis42 in that the dude should provide more info. |
| 20:16.18 | pabelanger | Agree |
| 20:24.49 | Corydon76-lap | Marquis42: I have one difference from what you said previously. Even if the URI is not correct, if we can figure out what it should have been, we should go ahead and process it. "Be liberal in what you accept and conservative in what you send." |
| 20:26.14 | Marquis42 | Corydon76-lap: Yeah, that's probably not a bad idea. I tend to get pedantic sometimes. :) |
| 20:37.13 | Corydon76-lap | Do you know who Jon Postel was? |
| 20:37.58 | Qwell | Corydon76-dig: RFC2468 |
| 20:38.24 | Marquis42 | Corydon76-lap: Oh yes, most assuredly. I know exactly what you're talking about. |
| 20:39.18 | Corydon76-lap | Okay, good |
| 20:39.22 | Corydon76-lap | Qwell: heh |
| 20:41.17 | Qwell | Corydon76-dig: and more specifically, RFC793, Section 2.10 |
| 20:44.33 | Corydon76-lap | Qwell: as long as someone understood the quote, I'm good |
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| 20:47.35 | hardwire | so minexpirey in sip.conf simply sets an unused variable in chan_sip.c? |
| 20:48.10 | hardwire | oh nm.. I found the code that uses it :) |
| 20:48.39 | eliel | nice quote |
| 20:49.03 | hardwire | it's not what I expected, I think thats what my problem was. |
| 21:05.25 | Kobaz | https://issues.asterisk.org/view.php?id=17138 |
| 21:05.34 | Kobaz | there's my little buggy |
| 21:13.48 | hardwire | bugaboo |
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| 21:52.16 | Netgeeks | anyone here know if there is some code out there that will report either instant or averaged sound levels on a channel on a frequent basis? Basically I'm looking for something that will provide a sound level for multiple channels being fed into a conference room. This would allow moderators to determine if it is safe to un-mute someone without flooding the conference with noise? |
| 22:18.19 | Corydon76-lap | Don't know if there is, but it wouldn't be difficult to write |
| 22:18.45 | Corydon76-lap | Just need to ensure that you lock correctly when you're modifying each user's value |
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| 23:03.13 | pabelanger | M17127 |
| 23:03.15 | MuffinMan | [new] [Asterisk] Channels/chan_sip/General 0017127: Wrong encoding of SIP URI reported by sdaniels https://issues.asterisk.org/view.php?id=17127 |
| 23:03.50 | pabelanger | I'm going to try and create a patch but wanted some input on where the best place to fix this is. |
| 23:04.24 | pabelanger | should we check for URI encoding when we _build_ SIP headers? |
| 23:04.34 | pabelanger | if pedantic is enabled? |
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