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| 00:32.49 | latv | in asterisk 11 chan_sip doesn't see custom codec formats after restart. seems that they are created later. if unload chan_sip and then reload, there is no problem |
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| 00:55.47 | snuff-work | Maxus2: tab indented is fine for 'initial' parts.. aka dont use 'tabs' where u rely on it looking the same.. (after any other character) |
| 00:56.13 | Maxus2 | ahh cool thanks. |
| 00:56.28 | snuff-work | so.. tab int i=0; is ok.. but tab int i = tab 0; is bad |
| 00:57.15 | Maxus2 | cool understood :) |
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| 05:31.34 | jmls | morning |
| 05:32.02 | jmls | is there any documented process for getting asterisk to compile on an android-powered box ? |
| 05:32.06 | jmls | http://www.in4dealz.com/deals/Dual-Core-Mini-PC-UG802-Rockchip-3066-Android-4-0-4-1-2GHz-Cortex-A9-Quad-Core-GPU-1GB-RAM-4G-ROM-IPTV-TV-Box-600195/ |
| 05:32.21 | jmls | would seem quite powerful enough to run a soho pbx :) |
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| 07:42.39 | schmidts | good morning |
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| 08:43.59 | latv | hi! once again about custom codecs in codecs.conf, it seems build_peer() in chan_sip hits before they are loaded, so it doesn't see any custom codec, namely, ast_getformatbyname() fails |
| 08:44.45 | latv | later reload of chan_sip helps |
| 08:45.42 | latv | is it a bug, or i need "preload" some module in conf? |
| 09:30.52 | Juggie | id guess toying w/ preload might fix it but that doesnt mean its not a bug |
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| 10:59.36 | latv | preload => res_format_attr_*.so |
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| 13:13.32 | mcolombo | hi all! |
| 13:15.47 | mcolombo | anybody can explain me when asterisk sends the SIP 181 response? I must enable this message, but i don't know how to do it |
| 13:15.51 | mcolombo | thanks guys |
| 13:23.09 | schmidts | mcolombo i am not sure but imho asterisk cant do it. 181 means that the call will be forwarded to another destination and asterisk isnt a proxy, its a B2BUA |
| 13:23.57 | schmidts | ok i was wrong there is some situation to do this |
| 13:26.20 | putnopvut | Yeah, if you're using Asterisk 1.8 it should just send that. |
| 13:28.40 | schmidts | if a diversion header is present in the invite then it should work IMHO |
| 13:28.51 | putnopvut | Actually, it may only be that Asterisk will send a 181 if the 302 that Asterisk receives has a Diversion header. |
| 13:34.54 | schmidts | putnopvut i didnt made it through the whole handle_request_invite but i am not sure about the 302 response cause the part i mean is in the request function not the response |
| 13:37.20 | putnopvut | Gotcha, I was looking at update_redirecting() in chan_sip.c |
| 13:37.48 | putnopvut | That gets called when a redirecting update is sent to a channel. This typically happens when the call is forwarded. |
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| 13:46.55 | jmls | *bump* ;) |
| 13:47.08 | jmls | has anyone compiled asterisk for android ? |
| 13:47.20 | mcolombo | ok putnopvut, thanks |
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| 16:03.40 | _Corey_ | Hey, can someone take a look at a bit of app_dial.c and let me know what this code segment is trying to do? http://pastebin.com/N2m67XEy |
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| 16:04.17 | _Corey_ | I'm debugging something weird and am not a C guy, so I'm just trying to (first) get my arms around what this code section is all about |
| 16:07.31 | leifmadsen | _Corey_: how are you using the Local channel? |
| 16:07.35 | leifmadsen | suspects you need /n on the end |
| 16:07.56 | _Corey_ | hmm, interesting thought |
| 16:08.10 | leifmadsen | basically the Local channel is being optimized away |
| 16:08.17 | leifmadsen | at least that's my guess |
| 16:08.32 | _Corey_ | well, what's happening is it doesn't attempt to dial |
| 16:08.46 | _Corey_ | doesn't the /n just apply when the other end picks up? |
| 16:11.20 | bweschke | it doesn't look like this peer is available |
| 16:11.21 | bweschke | SIP/login_agent_to_OpenGate/2817||M(pogpsendsiprtpinfo) |
| 16:11.41 | bweschke | so the Local channel can't ever set up ? |
| 16:14.06 | _Corey_ | so, the SIP peer "login_agent_to_OpenGate" is unmonitored and goes to an OpenSER server |
| 16:14.26 | _Corey_ | Here's the bigger picture: |
| 16:14.59 | bweschke | I think you're going to need more than just that code sequence … probably relevant extensions.conf, sip.conf and a 'sip show peers' and a relevant sip dialog related to this call attempt |
| 16:15.10 | _Corey_ | only one or two Dials fail during a given window... and only to specific targets |
| 16:15.14 | _Corey_ | this SIP peer can be dialed with other extensions while this failure happens |
| 16:15.39 | bweschke | which makes me think that the sip debug dialog is even more important |
| 16:16.19 | _Corey_ | yeah, when I noticed there was only one spot that produced that particular log line, I wanted to understand what the test was in app_dial.c that caused it |
| 16:17.41 | _Corey_ | I'm not familiar with it overall, and at a quick glance it seems to be within this "/* only the input channel is available */" section |
| 16:17.53 | bweschke | the code spot is telling you that if there's only one channel to try to bridge … the bridge itself cannot work and, as a result, the dial fails. |
| 16:18.15 | bweschke | right |
| 16:18.18 | _Corey_ | so the target channel of the dial that's not available is what causes that, makes sense |
| 16:18.22 | bweschke | the input channel in your case if the Local channel |
| 16:18.59 | _Corey_ | so, I'm probably looking for a stuck SIP dialog or something in 'sip show channels' |
| 16:21.03 | bweschke | somewhere in the dialog of the attempted call setup, I'm fairly sure you're going to find why the other side isn't getting nailed up |
| 16:21.36 | _Corey_ | yeah, SIP traces are just telling me asterisk isn't sending an INVITE |
| 16:23.08 | _Corey_ | Dial just bombs with 'No one is available to answer at this time (1:0/0/0)' before doing anything external to Asterisk |
| 16:23.12 | bweschke | do you have some higher level debugging getting logged in asterisk? try that as a first step if a trace is showing that you're never sending an INVITE out at all. The "called login_agent…." line implies that it thinks it did |
| 16:23.35 | _Corey_ | hmm, interesting |
| 16:23.50 | _Corey_ | yeah, i'm going to turn up the logging and dig into this more |
| 16:24.13 | _Corey_ | thanks for the explanation... it was really helpful |
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| 17:10.12 | italorossi | putnopvut: ping |
| 17:11.32 | putnopvut | italorossi: pong |
| 17:12.17 | italorossi | putnopvut: you can tell if it was implemented? or started? http://lists.digium.com/pipermail/asterisk-dev/2009-February/036734.html |
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| 17:15.27 | putnopvut | italorossi: I don't think so. |
| 17:15.41 | jkroon | https://code.asterisk.org/code/changelog/asterisk?cs=373342 which seemingly is in response to https://issues.asterisk.org/jira/browse/ASTERISK-20337 looks wrong, can someone confirm? |
| 17:17.16 | jkroon | ast_db_get will return -1 on failure, which is fine, so the inversion of the test is correct, however, this also removes an ast_mutex_unlock and return, and then thereafter a sscanf() is run on (the possibly uninitialized) tmp string ? |
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| 17:20.44 | italorossi | putnopvut: ok, thanks man. |
| 17:23.08 | mjordan | jrose_atDigium: ^^^^ ASTERISK-20337 |
| 17:24.04 | jrose_atDigium | looks |
| 17:29.23 | jrose_atDigium | jkroon: I think you are right about the sscanf running uninitialized... that's definitely bad. I'll need to check a little deeper before I say for sure. |
| 17:30.44 | jrose_atDigium | jkroon: Nevermind. tmp can't be uninitialized. |
| 17:31.05 | jrose_atDigium | It's declared as an 80 character array already set to "" |
| 17:31.54 | jrose_atDigium | I don't believe it will be mucked up by ast_db_get if that function returns an error either. |
| 17:36.31 | jkroon | jrose_atDigium, no, it's not. |
| 17:36.37 | jkroon | correct. |
| 17:36.55 | jkroon | is there any way to just avoid that code path in affected versions? |
| 17:37.15 | jkroon | we've got a user on gentoo that's using IAX/2 but apparently not iaxprov that's complaining about this. |
| 17:37.40 | jkroon | i can't see an easy way from looking at iax-provision.c |
| 17:40.12 | jrose_atDigium | jkroon: What is 'this' that it's complaining about exactly? |
| 17:40.27 | jkroon | https://bugs.gentoo.org/show_bug.cgi?id=437740 |
| 17:40.39 | jkroon | basically he's seeing the error message in his logs. |
| 17:41.29 | jkroon | and whilst it's almost trivial to back-port the patch (we try to avoid release candidates) I'm wondering whether there isn't just a simpler way to avoid the error message. |
| 17:42.21 | jrose_atDigium | I don't really know unfortunately. I don't really touch IAX2 on any kind of regular basis. |
| 17:42.37 | jkroon | thanks, i'll dig a bit. |
| 17:42.49 | jrose_atDigium | good luck |
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| 18:39.42 | file | silence. |
| 18:41.06 | bweschke | hi file! |
| 18:41.56 | file | woah bweschke |
| 18:42.27 | bweschke | in the virtual flesh, yep! |
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| 19:13.55 | mjordan | file: now that I've caught up a bit, yes, 374850 |
| 19:14.17 | mjordan | file: as in, is this something that we think absolutely must be fixed in 11.0.0 |
| 19:14.52 | file | 374850 and 374833 |
| 19:14.54 | file | yes |
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| 19:24.12 | mjordan | kk |
| 19:36.44 | leifmadsen | mjordan: ping |
| 19:36.59 | leifmadsen | mjordan: now that RC1 is out -- it's time that the commercial modules be made available :) |
| 19:37.29 | Qwell | leifmadsen: we know |
| 19:37.30 | leifmadsen | since there should be no underlying ABI/API changes at this point |
| 19:37.32 | jrose_atDigium | It's been a topic of discussion. |
| 19:37.42 | jrose_atDigium | I think the work has already been assigned |
| 19:37.55 | leifmadsen | ok, just wanted to pass it along since they usually are supposed to be released at the same time as RC1 |
| 19:38.06 | Qwell | ish |
| 19:38.15 | leifmadsen | that was the general consensus last time we did releases |
| 19:38.38 | leifmadsen | although that might have only been a discussion between russell and myself, and terry was the one building the modules |
| 19:39.08 | Qwell | rc1 triggered the work to start |
| 19:39.14 | Qwell | couldn't really start it beforehand |
| 19:39.34 | leifmadsen | can't read minds |
| 19:39.39 | Qwell | Well why not? |
| 19:39.52 | mjordan | leifmadsen: you should work on your mind reading skills :-) |
| 19:40.00 | leifmadsen | ok |
| 19:40.15 | mjordan | we're plugging our way through them - a number of them are in testing right now |
| 19:40.23 | Qwell | zero is a number |
| 19:40.38 | mjordan | 0 <= x < n |
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