| 00:05.44 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 00:10.37 | tuxd00d | * Now with more fur in the can! * |
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| 00:11.11 | carrar | mmmm more fur in every bite! |
| 00:13.43 | tuxd00d | thinks ânom nom nomâ |
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| 01:36.52 | WIMPy | Micc: What kind of buggy are you talking about? |
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| 03:03.19 | UncleKiwi | May 15 14:56:02] NOTICE[7194][C-000000a7]: pbx.c:4948 pbx_extension_helper: No such label 'cfim' in extension '0' in context 'home' |
| 03:03.20 | UncleKiwi | [May 15 14:56:02] WARNING[7194][C-000000a7]: pbx.c:12329 pbx_parseable_goto: Priority 'cfim' must be a number > 0, or valid label |
| 03:03.27 | UncleKiwi | hey guys what does this mean |
| 03:03.50 | UncleKiwi | im using a label in exten 0 |
| 03:04.03 | UncleKiwi | but its not doing what i say |
| 03:04.07 | UncleKiwi | :-/ |
| 03:04.12 | WIMPy | Yes, one that doesn't seem to exist. |
| 03:04.28 | UncleKiwi | yeah but i can show you |
| 03:04.31 | UncleKiwi | its odd |
| 03:04.43 | UncleKiwi | !pb |
| 03:06.24 | WIMPy | Do you see it with 'dialplan show home'? |
| 03:07.03 | superscrat | yeah, it's looking for cfim label in the home context |
| 03:07.21 | UncleKiwi | http://pastebin.com/MXppMe1c |
| 03:08.05 | superscrat | question, UncleKiwi |
| 03:08.16 | superscrat | why do you preface it with the pattern match symbol, when it is just ext 0 |
| 03:08.28 | UncleKiwi | i duno |
| 03:08.29 | superscrat | you are not using any pattern, just a straight digit literal |
| 03:08.42 | UncleKiwi | do you think this could cause the issue |
| 03:08.51 | themrrobert | My asterisk is crashing multiple times per day with core dumps, what can be done about this? I look in the logs (without debug) and there is absolutely no hint as to why it crashed. Asterisk 11.17.1 |
| 03:08.53 | WIMPy | No |
| 03:09.04 | themrrobert | segmentation fault |
| 03:09.17 | WIMPy | Show us 'dialplan show home' |
| 03:09.20 | superscrat | UncleKiwi, no, that is not the issue |
| 03:09.23 | superscrat | I was just curious |
| 03:09.33 | UncleKiwi | WIMPy: Hey, yep is shows with that command you asked me to do |
| 03:10.14 | WIMPy | themrrobert: As the bot is in the pub again, look in the wiki for collectdebug. |
| 03:10.17 | UncleKiwi | Include => 'did' [pbx_config] |
| 03:10.17 | UncleKiwi | <PROTECTED> |
| 03:10.17 | UncleKiwi | <PROTECTED> |
| 03:10.17 | UncleKiwi | <PROTECTED> |
| 03:10.50 | WIMPy | An who BTW came up with "cfim"? |
| 03:11.15 | UncleKiwi | maybe me |
| 03:11.23 | UncleKiwi | im not sure its for call forwarding |
| 03:11.35 | UncleKiwi | and it works in other parts of the dailplan |
| 03:11.42 | superscrat | the label name should be arbitrary, so long as it does not conflict with any reserved words |
| 03:11.48 | superscrat | and since it is in lower case, it should be fine |
| 03:12.19 | WIMPy | Reserved words? Do any exist for lables? |
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| 03:13.09 | superscrat | WIMPy, not that I am aware. Just variables. |
| 03:13.23 | superscrat | Be kind, please :) I am half asleep |
| 03:13.39 | UncleKiwi | it would it be that i am using that lable in did |
| 03:13.45 | superscrat | UncleKiwi, is this the line that is choking: exten => _0,n,GotoIf(${DB_EXISTS(CFIM/0)}?cfim) |
| 03:13.59 | UncleKiwi | yep |
| 03:14.30 | UncleKiwi | i want ti to jump tot he line with label cfim |
| 03:15.07 | superscrat | do me a favor, lets take out that _, just for fun. It may be a red herring, but it certainly is not needed for your particular use-case |
| 03:15.25 | superscrat | you do not have any pattern matching symbols, just the 0 |
| 03:17.17 | UncleKiwi | testing now with out the _ |
| 03:17.23 | UncleKiwi | ... |
| 03:17.32 | [TK]D-Fender | <UncleKiwi> http://pastebin.com/MXppMe1c <- broken |
| 03:17.44 | [TK]D-Fender | UncleKiwi, Line #5 |
| 03:17.58 | [TK]D-Fender | UncleKiwi, You put the HIINT right in the MIIDDLE of that pile of exten lines |
| 03:18.07 | superscrat | ah, yes |
| 03:18.17 | [TK]D-Fender | UncleKiwi, And that BREAK the continuity of the "n"'s |
| 03:18.25 | UncleKiwi | hahah i like the way you say pile of lines |
| 03:18.35 | superscrat | good catch, [TK]D-Fender |
| 03:20.44 | UncleKiwi | thanks TKD-Fender but where can I put the HINT |
| 03:21.00 | UncleKiwi | or should I put the hint |
| 03:21.05 | WIMPy | Before or after the rest. |
| 03:21.16 | UncleKiwi | ok thanks |
| 03:21.21 | UncleKiwi | i'll test now... |
| 03:21.38 | WIMPy | Just not before using n. |
| 03:22.10 | UncleKiwi | nice thanks WIMPy |
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| 03:23.36 | superscrat | also, UncleKiwi, one thing and this is just a matter of convention |
| 03:24.20 | UncleKiwi | yeah my convention is always getting better... I bought the book |
| 03:24.25 | superscrat | exten => <ext_name>,1,<op> |
| 03:24.26 | superscrat | <PROTECTED> |
| 03:24.42 | UncleKiwi | yeah I know about this |
| 03:24.56 | UncleKiwi | this was a system that I was being a bit lazy |
| 03:25.07 | UncleKiwi | and i copied some config from another |
| 03:25.19 | UncleKiwi | i need to tidy up my workmanship |
| 03:25.22 | [TK]D-Fender | Order matters |
| 03:25.24 | WIMPy | wonder how writing more counts as lazy |
| 03:25.54 | [TK]D-Fender | n=next. Hint itself is a priority. You shouldn't take a pile of "n" steps and insert something in the middle that isn't actually consecutive |
| 03:25.55 | superscrat | you don't have to indent the lines with "same", I just do that to make it more readable. cool =). it's not a scold against your style, which is perfectly valid as-is, just trying to pass along some tricks in case you weren't aware. |
| 03:26.56 | [TK]D-Fender | superscrat, Actually, that doesn't make it more readable per-se. That jst makes it easier to copy/paste chunks around without have to change the pattern for "nesxt6" step instructions |
| 03:26.57 | superscrat | [TK]D-Fender is correct. Asterisk will automatically determine what the 'n' should be, and if you inject your own something in there, it's going to mess it all up. |
| 03:27.27 | superscrat | ehh, I suppose I should clarify: it makes it more readable for me =) |
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| 03:33.07 | superscrat | Anyway, UncleKiwi, now that you are all squared away, I am out =) Night, y'all. I am officially turning into a pumpkin. |
| 03:34.40 | UncleKiwi | thanks superscat - its all soved |
| 03:35.38 | UncleKiwi | im just talking with customer |
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| 03:45.23 | UncleKiwi | i had to completely remove that hint |
| 03:45.34 | UncleKiwi | to make it work ... i dont need it anyway |
| 03:45.52 | UncleKiwi | little bit confusing |
| 03:46.12 | WIMPy | No, you just need to make sure it's not followed by a line using n. |
| 03:47.07 | [TK]D-Fender | You don't break a pile of stuff you intend to be consecutive with it, that's all |
| 03:47.13 | UncleKiwi | ah, so put 2 int he next line for example |
| 03:47.17 | UncleKiwi | ? |
| 03:47.28 | [TK]D-Fender | next, next, next, COMPLETELY DIFFERENT, next, next. |
| 03:47.48 | WIMPy | You /could/ do that. But just put it before the first line with 1 or after the last. |
| 03:47.48 | [TK]D-Fender | Put the hint BEFORE step 1 |
| 03:47.53 | UncleKiwi | 1,hint,2,n,n,n, ? |
| 03:47.57 | [TK]D-Fender | NO |
| 03:48.22 | UncleKiwi | 1,hint,n,n,n,n,n ? |
| 03:48.30 | [TK]D-Fender | NO |
| 03:48.42 | [TK]D-Fender | <[TK]D-Fender> Put the hint BEFORE step 1 <---------------- |
| 03:48.52 | UncleKiwi | ah ok |
| 03:48.53 | WIMPy | <UncleKiwi> 1,hint,2,n,n,n, ? <- Well, it would work, but it's ugly and confusing. |
| 03:49.04 | UncleKiwi | cool thanks |
| 03:49.12 | UncleKiwi | customer is happy |
| 03:49.20 | UncleKiwi | they always want more and more |
| 03:49.45 | [TK]D-Fender | And just dumb. Don't shove stuff in the MIDDLE of a pile of things are supposed to be consecutive. That's just straight-up TOURETTES |
| 03:50.22 | UncleKiwi | sure thing |
| 03:51.01 | UncleKiwi | i guess i need to read the definitive guide more than I realise |
| 03:51.35 | [TK]D-Fender | n = next priority. hint IS a priority. You're breaking the ordering |
| 03:51.55 | UncleKiwi | understood |
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| 04:06.01 | superscrat | [TK]D-Fender, if you are there, I misunderstood what you were saying earlier with respect to my readability comment. I thought you were referencing my spaces, so that is why I answered oddly... You are 100% correct about the 'same' construct. My gosh, lateness is not working for me tonight. Y'all be good. |
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| 04:30.53 | phix | [TK]D-Fender: I have updated the version of dahdi I was using. It picks up my card and channels (2 FXS and 1 FXO) however it tells me channel 4 doesn't exist any more. Why would that be? does it count from 0 now instead of 1? |
| 04:31.17 | phix | (channel 4 is FXO which dmesg lists) |
| 04:31.41 | [TK]D-Fender | just show all of it |
| 04:31.49 | [TK]D-Fender | I don't remember the backstory. |
| 04:31.56 | [TK]D-Fender | And need to see what the current state is anyway |
| 04:35.39 | phix | was that directed to me? |
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| 04:42.29 | [TK]D-Fender | yes |
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| 04:49.16 | phix | [TK]D-Fender: http://pastebin.com/Nv9WmrNY |
| 04:51.05 | [TK]D-Fender | and configs.... |
| 04:51.48 | phix | http://pastebin.com/kRJ4KBqT |
| 04:52.19 | phix | Do you need the asterisk config too? |
| 04:52.37 | phix | If I do a dahdi_genconf it doesn't pick up my device |
| 04:52.44 | phix | dahdi_scan fails too |
| 04:57.25 | [TK]D-Fender | dahdi_cfg -vvvv |
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| 06:53.38 | nitr0n | Hi everyone |
| 06:54.01 | nitr0n | I have an issue with odbc for several days |
| 06:54.15 | nitr0n | https://gist.github.com/anonymous/230485ddb07f58b01c0b |
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| 07:05.13 | nitr0n | does anyone meet the same issue ? |
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| 08:38.26 | kalbanker19 | Anyone here got experience with integrating Active Directory with Asterisk? |
| 08:40.17 | sekil | shouldn't be a problem witg asterisk having ldap connector |
| 08:40.23 | sekil | s/witg/with/ |
| 08:41.00 | kalbanker19 | Oh never heard of it. I'll check it out thanks! |
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| 08:48.17 | sekil | kalbanker19: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver |
| 08:57.27 | kalbanker19 | I've read that one. Have some trouble understanding it with my limited knowledge of pretty much everything |
| 08:58.28 | sekil | kalbanker19: that's easy...just rtfm |
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| 10:23.20 | dan_j | I'm thinking about capacity and looking at the current CPU usage. Currrently asterisk is running at about 30% CPU with around 60 channels. Are there any ways to make asterisk more efficient? Or has anyone seen any papers on the subject? |
| 10:35.46 | eirirs | g711? |
| 10:47.08 | EmleyMoor | On my Asterisk system, where is the default ring cadence for the analog phones set? (the one that gets used if there's no directive such as "r3") |
| 10:48.03 | EmleyMoor | Or, probably more likely, which one does it use by default? |
| 10:48.07 | Tim_Toady | EmleyMoor: if you are using dahdi then this is configurable in /etc/dahdi/system.conf |
| 10:48.52 | EmleyMoor | Tim_Toady: Tell me more |
| 10:49.50 | Tim_Toady | and I think you you can also set it per port in /etc/asterisk/chan_dadhi.conf |
| 10:50.34 | EmleyMoor | Tim_Toady: Not "per port", "per setup" - I am looking to find what is used as the default, as I'm sure the cadence changed on my recent upgrade. |
| 10:53.25 | Tim_Toady | EmleyMoor: for the VoIP channels the tone settings are in indications.conf, im not sure if dadhdi and analog lines use the same config |
| 10:53.52 | Tim_Toady | iof not i guess they must be hardcoded somewhere in the dahdi source |
| 10:54.37 | EmleyMoor | I do know what you mean about chan_dahdi.conf, but there's no reference to where it picks up the cadence to use if no "r" directive. |
| 10:55.03 | WIMPy | the country setting |
| 10:55.24 | EmleyMoor | Which points to... |
| 10:56.11 | WIMPy | Either indications.conf unless it has something of its own. |
| 10:56.13 | EmleyMoor | Hmmm... according to zonedata.c it should be using 400,200,400,2000 - but I don't think it is |
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| 10:57.22 | EmleyMoor | indications.conf specifies same |
| 10:58.38 | EmleyMoor | Hmmm... so dahdi.conf now overrides if cadences are specified there... that's why I'm getting 1000,2000 |
| 10:59.30 | EmleyMoor | chan_dahdi.conf, that is |
| 11:00.43 | EmleyMoor | Are cadences in chan_dahdi.conf using 0 or 1 as base? |
| 11:01.26 | EmleyMoor | Hmmm... 1 |
| 11:01.35 | EmleyMoor | Need to rework to take that into account |
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| 11:08.12 | EmleyMoor | It used to use the default in older versions - now it defaults to "r1" |
| 11:09.04 | EmleyMoor | Hmmm... still "no" |
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| 11:15.18 | EmleyMoor | Still ringing the phone in the wrong cadence... despite r1 now being the desired pattern - let me see if specifying it helps... |
| 11:20.27 | EmleyMoor | Right - r1 has to be specified - so it's really not clear where the "default" cadence comes from, still |
| 11:28.24 | EmleyMoor | Very mysterious... can achieve desired behaviour now, but where the default cadence is coming from is still a mystery |
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| 11:35.20 | EmleyMoor | Is there a way I can "lump" two DAHDI channels together, so that I can call them as one, with one cadence specifier? |
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| 13:12.56 | EmleyMoor | And still the Stoke Ambulance Chasers keep calling! |
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| 13:27.10 | EmleyMoor | staff tech: suspect my router is becoming a bit flaky - anything I can do about it? |
| 13:29.09 | EmleyMoor | Er, wrong window |
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| 13:48.16 | qakhan | can we restrict incoming calls from same callerid, we will accept 1 call per 10 sec from same callerid |
| 13:48.31 | [TK]D-Fender | You can do whatever you want. |
| 13:48.34 | [TK]D-Fender | It's your dialplan. |
| 13:48.38 | [TK]D-Fender | We've been through this already |
| 13:48.42 | WIMPy | It's your dialplan. |
| 13:49.12 | [TK]D-Fender | Get coding. |
| 13:50.10 | qakhan | i need lil hint on it |
| 13:50.25 | [TK]D-Fender | What is there to need a hint on? |
| 13:50.41 | [TK]D-Fender | Call comes in, shove the callerid and time in a DB entry. |
| 13:51.09 | [TK]D-Fender | before doing that check if the LAST time was more than 10 seconds ago. |
| 13:53.12 | qakhan | DB mean CDR table? |
| 13:53.42 | [TK]D-Fender | No. |
| 13:54.11 | [TK]D-Fender | CDR logs when a call is OVER |
| 13:54.29 | [TK]D-Fender | You could have 100 calls coming in and NOT being over |
| 13:55.06 | [TK]D-Fender | So use something else |
| 13:55.58 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 13:56.13 | qakhan | do i need to create a separat table (incoming calls) which record incoming call and time? and then query that incoming call table in incoming calls |
| 13:56.29 | *** join/#asterisk drazoro (~drazoro@105.210.62.87) |
| 13:56.44 | WIMPy | yes |
| 13:57.12 | qakhan | ok thanks |
| 14:01.43 | [TK]D-Fender | [09:50][TK]D-FenderCall comes in, shove the callerid and time in a DB entry. |
| 14:02.05 | [TK]D-Fender | You should be over-writing any previous one |
| 14:03.07 | qakhan | Thanks [TK]D-Fender and WIMPy |
| 14:03.32 | WIMPy | And a garbage collector might be a good idea as well. |
| 14:05.14 | [TK]D-Fender | very easy to do. |
| 14:08.16 | qakhan | WIMPy garbage collector in DB ? |
| 14:08.30 | WIMPy | yes |
| 14:08.45 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-bkbuflfhspuktkpe) |
| 14:09.52 | qakhan | ok |
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| 15:01.18 | sublime | Im having issues with one of our digium phones disconnecting after 30 seconds when an outside user calls the dedicated line. We have a PRI through TWC |
| 15:01.31 | sublime | Any ideas as to where to start troubleshooting? |
| 15:02.43 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:04.08 | [TK]D-Fender | Look at the call. |
| 15:04.31 | [TK]D-Fender | SIP debug & PRI debug and see what side initiates the disconnect and why. |
| 15:04.35 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 15:05.25 | sublime | I started by running a packet capture. Trying to sift through it. Can i turn on sip debug per line? |
| 15:05.44 | WIMPy | per IP |
| 15:05.51 | sublime | Awesome! |
| 15:15.58 | sublime | do sip debugs just to do the messages log file? |
| 15:17.02 | [TK]D-Fender | sublime: You need what, about 30 seconds of tracking for this? just make a test call with a big scrollback buffer |
| 15:17.30 | *** join/#asterisk superscrat (asanders@nat/digium/x-btnmchykekdxkomb) |
| 15:18.39 | sublime | I cant always replicate the issue its only with a few of our customers. when i call from my cell phone it works fine. |
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| 15:28.21 | sublime | [TK]D-Fender, does this make sense? Span 1: Channel 0/2 got hangup request, cause 16 |
| 15:30.02 | sublime | Cant replicate it again. Sounds like first call to direct line drops. Second and on work fine. I will try and get some more data. thanks for the help |
| 15:30.03 | [TK]D-Fender | mattsl: that isn't PRI debug and I don't see what else happened there |
| 15:30.07 | [TK]D-Fender | sublime: rather |
| 15:31.19 | sublime | whats was a console message using asterisk -vvvvvr |
| 15:32.47 | [TK]D-Fender | The is not PRI debug. |
| 15:32.52 | [TK]D-Fender | We need to see the actual call. |
| 15:33.08 | *** join/#asterisk Draecos (~Draecos@58-7-103-204.dyn.iinet.net.au) |
| 15:33.28 | sublime | I know that was what i got before i went on here. but now i cant replicate the issue again. |
| 15:34.16 | [TK]D-Fender | Come back with it when you have and we'll take a look |
| 15:34.21 | sublime | Will do. |
| 15:34.22 | [TK]D-Fender | Until then we have nothing to go on |
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| 15:48.05 | *** join/#asterisk wchance (~chatzilla@c-75-74-197-203.hsd1.fl.comcast.net) |
| 15:49.20 | wchance | we have the following process taking 90%+ of cpu /usr/bin/perl /var/www/html/admin/modules/fw_fop_op_server.pl |
| 15:49.29 | wchance | anyone know the function of this? |
| 15:52.21 | [TK]D-Fender | FOP <- |
| 15:52.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:52.30 | [TK]D-Fender | Flash Operator Panel. |
| 15:52.42 | wchance | thanks |
| 15:52.55 | wchance | we should be able to disable that? |
| 15:52.59 | wchance | if we do not use it? |
| 15:53.32 | WIMPy | If you don't want it, you probably shouldn't have enabled or even not installed it. |
| 15:53.55 | [TK]D-Fender | Find whatever script that is starting it... and stop it |
| 15:54.44 | WIMPy | Captain obvious says: Things that aren't installed are hard to start. |
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| 16:01.55 | sublime | [TK]D-Fender, logs: http://pastebin.com/MarK3Kxx and http://pastebin.com/2aGz6mRF |
| 16:02.42 | sublime | problem extension is 213 |
| 16:04.50 | [TK]D-Fender | I don't see an actual dial, answer, OR disconnect |
| 16:05.25 | WIMPy | Which call is the interesting one? |
| 16:05.34 | sublime | that was just pri and console log. working on sip log. |
| 16:06.01 | [TK]D-Fender | Also don't jsut try to redirect CLI to a file.. you are getting ANSI colour crap that makes this an unreadable mess |
| 16:06.20 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 16:06.40 | sublime | noted |
| 16:07.13 | WIMPy | And there is no timing information |
| 16:07.32 | sublime | i was wondering the same thing |
| 16:07.44 | WIMPy | So we don't know if the call was disconnected immediately, after 30s or after many minutes, |
| 16:07.58 | sublime | first try it was 13 seconds second try it was 19 seconds |
| 16:09.59 | WIMPy | Oh, and doinf search/replace on the numbers doesn't work. The original ones are still in plain sight. |
| 16:10.28 | *** join/#asterisk Sudravirodhin (~Sudraviro@162.218.22.124) |
| 16:11.20 | Sudravirodhin | Do remote attended transfers leave call legs open? |
| 16:11.24 | sublime | where do the sip debug logs go? |
| 16:12.18 | WIMPy | Where you configured them to go. |
| 16:12.25 | WIMPy | logger.conf |
| 16:12.26 | [TK]D-Fender | Sudravirodhin: There is no such thing as "remote". An attended transfer is an attended transfer. The channel doing it ENDS at the end. It is gone. |
| 16:14.39 | *** join/#asterisk svm_invictvs (~patricktw@unaffiliated/svminvictvs/x-938456) |
| 16:14.57 | Sudravirodhin | [TK]D-Fender: So an inbound trunk channel transferred to an outbound trunk channel would no longer run througn Asterisk? |
| 16:15.29 | [TK]D-Fender | Sudravirodhin: You are mixing topics. |
| 16:15.35 | WIMPy | Depends on the abilities of that trunk. |
| 16:15.52 | [TK]D-Fender | Sudravirodhin: If you expect the 2 ends to end up in DIRECT comminucatio... that is another thing entirely |
| 16:16.07 | [TK]D-Fender | Sudravirodhin: Both call legs are still on ASTERISK |
| 16:16.20 | [TK]D-Fender | Sudravirodhin: Unless you have a reinvite/ 2BCT option somewhere |
| 16:16.55 | [TK]D-Fender | Sudravirodhin: and "trunk channel" clearly depends on the kind of trunk and the provider |
| 16:17.07 | [TK]D-Fender | Sudravirodhin: As to whether it's possible to free it up or not. |
| 16:17.16 | [TK]D-Fender | Sudravirodhin: As well as the dial options passed, etc |
| 16:17.28 | Sudravirodhin | [TK]D-Fender: I apologize for mixing topics, my concerns are more in the realm of "am I using outbound and inbound minutes for an inbound call transferred to a phone number." |
| 16:17.50 | [TK]D-Fender | Sudravirodhin: See above |
| 16:18.27 | WIMPy | That would most probably be a yes, even if you get rid of the call. |
| 16:18.55 | Sudravirodhin | WIMPy: but if I were to blind transfer rather than attend it, the call would leave Asterisk. |
| 16:18.56 | [TK]D-Fender | Sudravirodhin: AFAIK there is no provider where you will NOT get billed for this regardless. The only difference would be possibly freeing up actual comm channels |
| 16:18.56 | WIMPy | Billing is yet another topic. |
| 16:19.12 | [TK]D-Fender | [12:18]SudravirodhinWIMPy: but if I were to blind transfer rather than attend it, the call would leave Asterisk. <- No. |
| 16:19.47 | WIMPy | It could work the same, no matter if it's a bling or an attended transfer. |
| 16:19.49 | [TK]D-Fender | You don't seem to understand Asterisk. You a device transfers... it just redirects to DIALPLAN. |
| 16:19.58 | [TK]D-Fender | When your* |
| 16:20.13 | WIMPy | If you used diversion services, that might make a difference. |
| 16:21.26 | Sudravirodhin | [TK]D-Fender WIMPy: Thanks for your thoughts. |
| 16:21.38 | sublime | [TK]D-Fender, sip log: http://pastebin.com/mB6WhkJU |
| 16:21.59 | sublime | Call started at 11:10 lasted 12 seconds |
| 16:23.43 | WIMPy | Wow. That looks seriousely borked. |
| 16:24.31 | sublime | Its a pretty bare FreePBX install |
| 16:25.38 | WIMPy | Well, not the best starting point, either. |
| 16:25.55 | WIMPy | But your log is somewhat garbled. |
| 16:25.58 | sublime | Yeah I know. I'm not proud of it. |
| 16:26.04 | [TK]D-Fender | And there is no PRI debug in there. |
| 16:26.25 | sublime | Boss kinda pushed to get it done as soon as possible. i sent the pri log eariler |
| 16:26.43 | WIMPy | We need all in one file. |
| 16:27.07 | WIMPy | And it should be readable text without binary garbage. |
| 16:27.27 | [TK]D-Fender | We can't be expected to try and stich this crap together... |
| 16:27.29 | [TK]D-Fender | http://pbxbook.com/other/isdncode.html#57 |
| 16:27.29 | sublime | pri logs to a seperate file? |
| 16:27.36 | WIMPy | And verbose of at least 3 so we can se what the dialplan is doing. |
| 16:27.36 | [TK]D-Fender | For the one PRI return code we see... |
| 16:27.42 | [TK]D-Fender | NOT SEPARATE |
| 16:27.54 | [TK]D-Fender | Don't make us patch random crap together. |
| 16:28.05 | WIMPy | It *can* log to a seperate file. |
| 16:28.46 | WIMPy | Why do I believe that the cause 57 is used wrongly? |
| 16:28.48 | sublime | i'm trying to make it as easy as i can. Im saying that pri log is set to log to a separate file. I will try and change it. |
| 16:29.07 | [TK]D-Fender | * CLI |
| 16:29.12 | [TK]D-Fender | Flogs = trash |
| 16:29.16 | [TK]D-Fender | Logs too |
| 16:29.43 | WIMPy | Send both debug and verbose to the same file via the logger. That's the only way to get usable output. |
| 16:30.29 | sublime | Alright I will make it happen |
| 16:30.31 | WIMPy | And that was already seen in the very first pri debug. |
| 16:30.48 | WIMPy | But without timig or dialplan we have no idea if the two calls are related. |
| 16:31.41 | WIMPy | But if they are, I'd take a look at the dialplan, as the first call is answered AFTER the second one failed. |
| 16:32.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:33.53 | sublime | So to set the pri debug file can just have it output to asteris/full? |
| 16:33.58 | sublime | asterisk/full* |
| 16:34.05 | WIMPy | So they don't like what you're trying to call there. They say they don't allow speech, but somehow I doubt that is true. |
| 16:34.31 | WIMPy | Don't set a file. Just use logger.conf. |
| 16:35.07 | WIMPy | But you can also just anwer if those two calls are related and that's the issue. |
| 16:35.53 | sublime | 9202654307 is the cell phone direct line is 9206326663 other calls are unrelated. |
| 16:36.45 | WIMPy | Looks different to me, as we have the caller ID the same. |
| 16:37.12 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 16:38.46 | WIMPy | First you get a call from 9202654307 to 9206326663 and then you place a call with caller ID 9202654307 (i.e. the first aller) to 9202410965, which is rejected. |
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| 16:39.10 | WIMPy | *After it's rejected, the 1st call is answered and hung up. |
| 16:39.15 | sublime | It was only 1 call. |
| 16:39.26 | WIMPy | No. There are two calls. |
| 16:39.48 | WIMPy | Loos like you told that extension to call out again. |
| 16:40.20 | sublime | looks like the 241 number is 213's cell phone |
| 16:40.31 | WIMPy | And the 2nd call using the Caller ID of the 1st call surely looks like they are related. |
| 16:40.32 | sublime | what the fuck are people doing. no one is supposed to be touching this damn thin |
| 16:41.06 | sublime | he said he was trying to get it to forward numbers to his cell phone |
| 16:41.38 | WIMPy | Looks like he succeeded locally, jut not with your telco. |
| 16:42.33 | sublime | thats weird that its only on his direct line though. if someone is transfered to his extension it works fine |
| 16:42.38 | WIMPy | Anyway, I'd file the message from your provider as an error. And so far that's the only error I see. |
| 16:43.12 | WIMPy | No idea if FreePBX would make a difference between internal and external calls. |
| 16:43.31 | WIMPy | Asterisk itself doesn't have that concept. |
| 16:44.35 | sublime | i wouldnt think so? that number is in his follow me list. But if the extension is answered it shouldn't try to transfer to his cellphone? |
| 16:45.50 | WIMPy | Well, if there's more going on, we need the full debug. Although FreePBX dialplan isn't realyy readable :-( |
| 16:46.12 | WIMPy | So it looks like you have multiple issues. |
| 16:47.50 | sublime | yeah i can imagine there are a few issues. I would like to switch over to raw asterisk but im more of a linux novice so im sure it would be a PIA |
| 16:49.12 | sublime | also FreePBX logger.conf just references other conf files. None of wich have really anything in them |
| 16:50.00 | WIMPy | There's a reason we don't support FreePBX. It's just too hard. |
| 16:52.02 | sublime | Know any good resources to become an asterisk guru? |
| 16:52.17 | WIMPy | ~primer |
| 16:52.17 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 16:52.22 | WIMPy | ~book |
| 16:52.22 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:54.00 | sublime | sweet! thanks |
| 16:54.48 | sublime | i assume that you can do pretty much everything freebpx does in regular asterisk |
| 16:55.26 | WIMPy | That's what FreePBX does. It uses Asterisk. |
| 16:55.46 | WIMPy | And you can do even more yourself. |
| 16:56.11 | sublime | Well right. I just know they messed with it alot. Wasnt sure if that included adding features. like forwarding calls to a cellphone after an extension didnt answer |
| 16:56.18 | sublime | Alright good to hear. |
| 16:56.41 | WIMPy | Well, you have to write that in your dialplan/ |
| 16:56.55 | WIMPy | Asterisk won't do anything without you telling it to do so. |
| 16:57.16 | sublime | ahh ok. so dial plan is all the logic behind call routing? |
| 16:57.26 | WIMPy | So that can become quite a lot of configuration. |
| 16:57.31 | WIMPy | Yes |
| 16:58.36 | sublime | is there a light weight FreePBX type system? |
| 16:58.46 | WIMPy | Just the part that lets users manage call forwarding makes 300 lines for me. |
| 16:59.17 | WIMPy | There are other GUIs, but I'm not sure any of them are still maintained. |
| 16:59.27 | sublime | oh ok. Do you use digium phones? |
| 16:59.45 | sublime | those have a forward feature on them as well but i could never get that to work properly |
| 16:59.46 | WIMPy | Sometimes |
| 16:59.59 | sublime | i assume that works differently than a dial plan? |
| 17:00.51 | WIMPy | In telephony terms, forwarding is a switch/server based feature. What phones do is deflection. |
| 17:01.05 | WIMPy | And for obvious reasons a phone can't divert calls wehn it's not reachable. |
| 17:01.39 | sublime | ahh. ok makes sense. |
| 17:01.42 | *** join/#asterisk nix8n82 (~AndChat27@67-130-74-235.dia.static.qwest.net) |
| 17:01.54 | sublime | looks like i have a weekend full of reading and a ton of beer |
| 17:02.05 | WIMPy | SIP doesn't have a concept of setting or inquiring server based features. |
| 17:02.31 | WIMPy | One of the many things missing. |
| 17:02.42 | carrar | might jsut skip the sip and drink the beer |
| 17:02.42 | sublime | Thats what i thought it would make security a pain. Thats why i was confused with that setting. |
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| 18:03.37 | WIMPy | Nice. Just used an RPi for a conference :-) |
| 18:06.46 | themrrobert | I followed the info here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information but there was absolutely nothing in the log indicating why it crashed |
| 18:07.27 | WIMPy | Did you look at the core dump with gdb? |
| 18:07.28 | themrrobert | so i've rebuilt asterisk with DONT_OPTIMIZE and BETTER_BACKTRACES but i can't reinstall until people are off the phone (production system, impossible to predict what's going to cause it) |
| 18:07.58 | themrrobert | is it going to be useful without those flags? |
| 18:08.15 | WIMPy | No matter what, you won't find any onformation on crashes in any log. |
| 18:08.27 | themrrobert | lol then i'm pissed someone told me to follow those directions |
| 18:08.34 | WIMPy | Definitely less. But it's woth looking anyway. |
| 18:08.38 | themrrobert | i mentioned it was a segfault |
| 18:08.44 | WIMPy | Yes. I did. |
| 18:08.58 | WIMPy | But you need to look at the core dump. |
| 18:09.15 | themrrobert | is all good, i just need it to not crash this is gonna cost me my job lol |
| 18:09.25 | WIMPy | Logs may tell you what happened before, but they never contain information on the crash itself. |
| 18:09.26 | themrrobert | they're jerks anyway but i need the money xD |
| 18:09.50 | themrrobert | well i dont have the debuginfo installed, hopefulyl there's enough space left for it |
| 18:10.22 | WIMPy | Well, yes, they can become big files. |
| 18:10.55 | themrrobert | how do i examine the dump ? I know the command but my experience with gdb is limited. can i run it alongside a running asterisk? |
| 18:11.01 | themrrobert | and how do i jump to the relevant data |
| 18:11.37 | WIMPy | You can, but what you need is to run it with the core file after it has crashed. |
| 18:11.56 | WIMPy | That will tell you how/where it crashed. |
| 18:13.03 | WIMPy | Using gdb on a running Asterisk is what you need if it locked up without crashing. |
| 18:14.31 | themrrobert | hopefully it works with out libtiff debuginfo as it conflicts on my system |
| 18:15.24 | themrrobert | terminated in _int_malloc, from what i've read on the asterisk wiki, sounds like memory corruption, and i know malloc allocates memory |
| 18:17.46 | themrrobert | also seems backtrace.py isn' |
| 18:17.58 | themrrobert | isnt included in gdb anymore, so it fails to import it -.- |
| 18:19.48 | themrrobert | ugh amazon linux sucks XD missing debuginfo packages for several components. oh well i can see it crashed in int_malloc |
| 18:20.58 | WIMPy | libtiff? Where does that come in to play? |
| 18:22.03 | WIMPy | Hmm. if it crashes in malloc that's a little odd. Smells like something more serious than Asterisk. Or you just ran out of RAM. |
| 18:22.04 | themrrobert | idk gdb told me to install debuginfo for it |
| 18:22.19 | themrrobert | Program terminated with signal 11, Segmentation fault. #0 0x00007fedb38a1906 in _int_malloc () from /lib64/libc.so.6 |
| 18:22.38 | themrrobert | server is definitely not out of memory, has 8GB, and 80% is taken by cached files |
| 18:22.55 | themrrobert | and theres a fat swap file |
| 18:23.37 | WIMPy | Broken RAM? |
| 18:27.02 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:27.30 | themrrobert | ya that has been a theory i've had for a few days now |
| 18:27.41 | themrrobert | its on an aws server =/ |
| 18:27.56 | WIMPy | Or some broken lib. |
| 18:28.15 | WIMPy | The best RAM test I know it to compile gcc. |
| 18:28.46 | themrrobert | is to, you mean, right? |
| 18:29.03 | WIMPy | yes |
| 18:29.06 | themrrobert | how do we know if it works or not |
| 18:29.22 | themrrobert | will it fail if ram is bad, or just silently finish, then breaking stuff when you try to compile |
| 18:29.39 | WIMPy | I had random aborts when running make when I had a faulty RAM. |
| 18:30.35 | themrrobert | 0: _int_malloc /lib64/libc.so.6, 1: malloc /lib64/libc.so.6, 2: _ast_malloc ./include/asterisk/utils.h |
| 18:35.15 | themrrobert | i'll test the memory when the agents have gone home for the day |
| 18:39.07 | themrrobert | its showing like 160 threads in the backtrace, is that high? |
| 18:39.40 | themrrobert | from what i'm reading, this type of seg fault can be caused from heap corruption arising from a buffer overflow ANYWHERE else in the code, making it very hard to trace |
| 18:42.44 | themrrobert | i wonder if an unlimited stack size would solve this issue, maybe it's running out of stack space? |
| 18:43.19 | themrrobert | i'll give it a shot and let you know monday if that was it, also i should have better backtraces by then if not anywy |
| 18:44.06 | WIMPy | Yes, something would have to overwrite what the memory management needs. |
| 18:44.44 | WIMPy | Off course that could happen anywhere, but you usually hit places around what is being used. |
| 18:45.06 | WIMPy | So things are most likely to break themselves. |
| 18:45.23 | WIMPy | But, as said, wit a little "luck" anything can happen. |
| 18:45.34 | themrrobert | ugh oh, it crashed in another spot this time: __strlen_sse42 |
| 18:45.59 | themrrobert | also in libc.so.6 |
| 18:49.08 | WIMPy | You could try to reinstall your libs in case something could be corrupted. |
| 18:49.25 | themrrobert | that's exactly what i was thinking when i saw another crash in the same library |
| 18:49.55 | themrrobert | i should probably also recompile asterisk afterwards correct? |
| 18:50.13 | *** join/#asterisk hyperized (~Hyper@unaffiliated/hyperized) |
| 18:50.30 | themrrobert | probably not necessary, doubt the API has changed, the entry points are probably still the same, the next version up is a minor upgrade |
| 18:54.25 | themrrobert | u ever had any trouble installing a newly compiled asterisk while an old one was running? it's all in memory so shouldn't be an issue unless it tries to reload something |
| 18:56.42 | WIMPy | No, you just can't load modules any more. But that souldn't eb an issue. |
| 18:56.47 | WIMPy | be |
| 19:04.08 | themrrobert | u a dcap wimpy? |
| 19:04.41 | themrrobert | I like that you write with good grammer, even in IRC. |
| 19:04.45 | themrrobert | grammar* |
| 19:08.30 | WIMPy | No. I just took the dcaa I was offered. |
| 19:09.41 | *** join/#asterisk Jacoby6000 (~Jacoby600@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:10.12 | *** join/#asterisk drsprite (drsprite@madness/drsprite) |
| 19:10.27 | Jacoby6000 | does anybody know what format is being used for origtime in the voicemail text files? Parsing them as an epoch instant, it gives me a 1970's date: 1970-01-17 06:58:34 |
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| 19:11.19 | drsprite | Hey there, I'm trying to get an Avaya 4621SW working with Asterisk. So far, so good. It registers and I can call outbound from it. When calling into the Avaya 4621SW, I get error SIP/2.0 415 Unsupported Media Type. According to Avaya, this means The server refuses to accept the request because it is in |
| 19:11.19 | drsprite | an unsupported format |
| 19:11.23 | drsprite | Any insight? |
| 19:11.53 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:13.19 | [TK]D-Fender | drsprite: The phone doesn't like what is being offered to it. Go look at the SIP debug to see what each side offered |
| 19:13.35 | [TK]D-Fender | drsprite: And look at the call that did work to prove what peer it matched against, etc |
| 19:14.01 | drsprite | I have set sip set debug peer 6003, but I'm not sure I know what exactly to look for. I'm still relatively new at this |
| 19:14.31 | drsprite | let me compare a working call, vs, non-working and see if anything stands out |
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| 19:15.04 | themrrobert | WIMPy - bad news, the stack limit was already set to 8GB, so doubt that's going to help :( |
| 19:15.59 | WIMPy | That should be more than neccessary. |
| 19:17.18 | themrrobert | yea it never gets close to that |
| 19:17.27 | themrrobert | so it wasn't running out of stack space =/ which would have been an easy fix |
| 19:17.59 | themrrobert | i tried reinstalling (upgrading) glibc, and recompiling asterisk against it just in case, and have installed it so we'll see if t hat helps |
| 19:20.57 | *** part/#asterisk wakko (~wakko@78.233.162.32) |
| 19:21.06 | drsprite | [TK]D-Fender, i have a copy of the working outbound call and failed inbound call to/from the Avaya, however I have no idea what I'm looking for. Any tips? Should I pastebin it? |
| 19:21.07 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 19:21.19 | [TK]D-Fender | yes |
| 19:22.02 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 19:23.13 | drsprite | [TK]D-Fender, thanks. Working call is up top. and the failed call is after the "------" line break. http://pastebin.com/TGRDc1B8 |
| 19:23.45 | *** join/#asterisk areski (~areski@80.174.128.14.dyn.user.ono.com) |
| 19:29.47 | [TK]D-Fender | drsprite: You are offering encryption there and haven't set that up properly |
| 19:31.02 | drsprite | interesting. Truth be told, I'm using a pbx in a flash image. Wonder where encryption is defined |
| 19:31.50 | drsprite | [TK]D-Fender, thanks for checking into it. which line was the clue? just so i can learn for myself |
| 19:32.06 | [TK]D-Fender | That looks more like Asterisk GUI.... |
| 19:32.20 | [TK]D-Fender | PIAF is supposed to be using FreePBX underneath last I heard |
| 19:32.34 | drsprite | You're right, I'm actually using IncrediblePBX (a spin off from the guys who make PIAF" |
| 19:32.39 | drsprite | and it uses Asterisk GUI |
| 19:32.51 | [TK]D-Fender | AsteriskGUI is ***DEAD*** |
| 19:32.57 | [TK]D-Fender | And should never be used |
| 19:33.02 | [TK]D-Fender | has been for years |
| 19:33.24 | drsprite | I've used the gui maybe once. Everything else I've been doing is through the configs |
| 19:33.50 | drsprite | Interesting though, this IncrediblePBX image is maybe a month old? heh |
| 19:34.05 | [TK]D-Fender | The GUI depends on users.conf and that makes a real mess of trying to debug and properly configure stuff. |
| 19:34.23 | [TK]D-Fender | Either go full DIY or full FreePBX the proper way |
| 19:36.10 | drsprite | Yeah that's the plan. Right now I'm messing around in a lab. |
| 19:36.17 | drsprite | Is it "easy" to remove the encryption? |
| 19:51.09 | [TK]D-Fender | Yes. Fix your peers |
| 19:51.19 | [TK]D-Fender | give or take the use of users.conf <- |
| 19:51.23 | [TK]D-Fender | ~users.conf |
| 19:51.23 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
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| 19:53.34 | drsprite | hehe, ok so go to FreePBX |
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| 20:38.10 | themrrobert | well it was crashing fairly consistently every 5 - 30ish minutes, and it's been about 100 minutes since the last crash... |
| 20:38.18 | themrrobert | hopefully i didn't jinx it |
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| 20:46.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 20:48.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 21:28.46 | themrrobert | WIMPy almost 3 hours now :) too bad i did too much, it's possible though extremely unlikely that removing the optimizations fixed it, more likely though that upgrading glibc fixed it. We'll see as time goes on if it's still an issue. |
| 21:30.51 | themrrobert | Any ideas how would i go about troubleshooting this? Asterisk 13: WebRTC -> Works fine, but after placing a few calls, it stops working. IIRC i was still able to register via WSS, and then I would send an INVITE from SIPML5, and Asterisk would never see it. It didn't show up in SIP DEBUG, issue affected all users at the same time, after restarting asterisk, it worked again, but the problem |
| 21:30.51 | themrrobert | continued. Asterisk didn't crash or freeze (well didn't seem to be frozen) and there was no information immediately useful in the logs |
| 21:31.21 | themrrobert | the problem would resume after a few more calls i mean |
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| 22:26.27 | wchance | I have customer with many processes of /usr/sbin/asterisk -f -U asterisk -G asterisk |
| 22:26.29 | wchance | is that one for each call that is up? |
| 22:35.25 | Katty | STRAWBERRY JAM |
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