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| 02:46.25 | bhans | what's the best operator panel that is opensource for asterisknow? |
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| 02:50.33 | Penguin | I don't even know what that means. |
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| 02:56.28 | bhans | Penguin: like a Dial Manager |
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| 08:52.40 | [sr] | howdy |
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| 09:33.09 | frederikBE | hello all |
| 09:33.50 | frederikBE | periodically I'm seeing Peer 'XXXXX' is now UNREACHABLE! Last qualify: 1008 |
| 09:34.04 | frederikBE | it's happening with all trunks (from different providers) |
| 09:34.13 | frederikBE | and all extensions (on different locations) |
| 09:34.25 | frederikBE | so I assume it's a server issue |
| 09:34.38 | Rasputin3711 | version? |
| 09:34.52 | frederikBE | Asterisk 11.20.0 |
| 09:37.31 | frederikBE | when doing different network checks (mtr, ping,...) the network seems to be OK on the server |
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| 09:42.40 | frederikBE | not sure, but the endusers also complain they have a call delay |
| 09:42.49 | frederikBE | when transfer, but also with direct dialing |
| 09:47.45 | frederikBE | I noticed in an online thread: port 5060 become overfilled and the system wait for asterisk to process these packets. In the same time you can see that phones goes down in the logs |
| 09:47.50 | frederikBE | is it possible? |
| 09:49.23 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 09:50.44 | frederikBE | anyone has an idea? |
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| 10:05.39 | frederikBE | ? |
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| 11:12.50 | UncleKiwi | hey people |
| 11:13.35 | UncleKiwi | is there a nice way to make phone numbers clickable in windows so that they can initiate a call to asterisk |
| 11:14.30 | UncleKiwi | im thinking of having a softphone on my windows machine and when i make a call it dials my deskphone and also the clicked extention and then gets out of the call |
| 11:15.05 | UncleKiwi | or is there a better way to get this functionality |
| 11:15.31 | UncleKiwi | im attempting to not have to dial number on my deskphone |
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| 12:07.00 | s8y | hello, I need some help with mixmode recording in fop2. We would like to record entire conversation once record button is pressed (not just part after recording button is pressed. Can anyone help? |
| 12:08.12 | s8y | I think I need to use MixMonitor and change channel variable but don't know where to start with it. |
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| 12:44.07 | catphish | what's the difference between the username in the "request line" of a SIP request, and the username in the "to" line? which does asterisk use? |
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| 12:45.56 | leon_ | hi :) |
| 12:46.04 | wdoekes | catphish: asterisk doesn't use the To for anything, at least chan_sip doesn't |
| 12:46.28 | catphish | so it uses the username from the request line? |
| 12:46.41 | wdoekes | yes, the user-part is used to match an exten |
| 12:46.43 | leon_ | I can't access asterisk download page. is there some problem ? |
| 12:46.56 | wdoekes | (.. from the R-URI) |
| 12:47.14 | wdoekes | leon_: http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 12:47.19 | wdoekes | works for me |
| 12:47.23 | catphish | wdoekes: thanks |
| 12:47.45 | catphish | http://www.asterisk.org/downloads WFM |
| 12:48.00 | leon_ | i tried to access from www.asterisk.org/downloads |
| 12:48.52 | catphish | leon_: gather some debugging info, see if you can ping it, etc |
| 12:49.06 | catphish | not that i can help, i have nothing to do with the project |
| 12:50.27 | leon_ | catphish: ok. |
| 12:51.00 | leon_ | thanks |
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| 13:19.24 | gregs | file: I tried the Silk codec for Asterisk 13.9.1 today. I'm getting errors when trying to load the module (see next). If I restart Asterisk then the codec_silk module seems to be either crashing Asterisk or at the very least triggering a full config reload. If I delete codec_silk.so from usr/lib64/asterisk/modules then everything goes back to normal. |
| 13:19.24 | gregs | Are there any tricks to getting this module working in Asterisk 13? I have a codecs.conf file from Asterisk 12 with silk codec setup info in it. I've tried renaming it to see if that caused the problem, but no luck. |
| 13:19.34 | gregs | https://www.irccloud.com/pastebin/BOvPrRFa/ |
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| 13:21.52 | file | it required core Asterisk changes to support, it will be in the 13.11 release |
| 13:22.34 | gregs | Ah. OK. That makes sense then. Guess 13.11 is a while off? |
| 13:22.46 | file | 13.10 is in rc3 and hasn't been released yet |
| 13:22.55 | file | so, a few weeks minimum |
| 13:23.00 | gregs | Thanks |
| 13:23.09 | file | the changes themselves are in the 13 branch |
| 13:23.42 | file | and https://gerrit.asterisk.org/#/c/3136/ is the change itself |
| 13:24.04 | gregs | I'll have a look. Thankyou. Any chance you could put that into a README in the Asterisk 13 Silk Codec folders? Might stop others falling into the same trap? |
| 13:24.42 | file | if you grab putnopvut when he appears he took care of silk |
| 13:25.45 | gregs | OK |
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| 14:09.24 | twoion[shell] | vim ub |
| 14:09.56 | twoion[shell] | (Sorry, please disregard ^.) |
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| 14:45.20 | *** join/#asterisk iheartlinux (~jwpierce3@173-16-50-64.client.mchsi.com) |
| 14:45.58 | iheartlinux | Is there a way to hide voicemailmain in voicemail and make it accessable via # or *? |
| 14:48.10 | WIMPy | * or 0 |
| 14:48.29 | WIMPy | See the a and o extensions. |
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| 15:01.24 | iheartlinux | WIMPy: I don't need to exit to operator, so I dont need o. But how would I put (a) in my dial plan so I can exit to voicemailmain? |
| 15:01.48 | [TK]D-Fender | you .. put it in |
| 15:02.01 | [TK]D-Fender | make the extension |
| 15:02.37 | iheartlinux | exten => 108,1,Answer(108) |
| 15:02.52 | iheartlinux | same => n,VoiceMail(108) |
| 15:03.46 | [TK]D-Fender | That first one looks pretty bad..... |
| 15:03.57 | [TK]D-Fender | And so does your assumed VM context |
| 15:04.28 | iheartlinux | I'm not claiming to be an asterisk expert. Someday perhaps |
| 15:05.54 | [TK]D-Fender | 108 MS wait after answer doesn't seem like a specific thing to do... |
| 15:06.13 | [TK]D-Fender | just saying you seem to be slapping that number around kinda loose is all... |
| 15:06.35 | [TK]D-Fender | anyway for your question it's just like I said.. Make the extension |
| 15:07.14 | iheartlinux | does not exten => 108,1,Answer(108) create the extension? |
| 15:07.38 | [TK]D-Fender | WIMPy> See the a and o extensions. |
| 15:07.46 | [TK]D-Fender | THE A EXTENSION |
| 15:08.16 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 15:13.16 | iheartlinux | [TK]D-Fender: ok, got that to work |
| 15:13.19 | iheartlinux | ty |
| 15:13.32 | [TK]D-Fender | \o/ |
| 15:13.36 | iheartlinux | is there a way to pass current extension into () |
| 15:13.45 | iheartlinux | using @default atm |
| 15:14.34 | [TK]D-Fender | there is another standard asterisk dialplan variable for that you should already know... |
| 15:14.43 | [TK]D-Fender | The most common one of all |
| 15:15.07 | iheartlinux | ${EXTEN} |
| 15:15.14 | [TK]D-Fender | in fact it's almost pointless to know patterns at all if you don't know this one. |
| 15:15.16 | [TK]D-Fender | Yes |
| 15:22.41 | iheartlinux | Getting @default after putting in ${EXTEN} into voicemailmain. I suppose it's grabbing extenion a |
| 15:22.56 | iheartlinux | instead of 108 |
| 15:23.12 | WIMPy | Indeed. |
| 15:23.27 | WIMPy | You need to save the extension to another variable whule you're there. |
| 15:23.36 | iheartlinux | ah |
| 15:23.54 | iheartlinux | tnx |
| 15:27.25 | iheartlinux | WIMPy: awesome, love yall |
| 15:27.44 | iheartlinux | [TK]D-Fender: |
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| 15:39.51 | volga629 | Hello Everyone, version 13.9 segfaulting |
| 15:39.52 | volga629 | kernel: asterisk[12741]: segfault at 7f8960241000 ip 00007f89981887fb sp 00007f89399ec9d8 error 4 in libc-2.12.so[7f89980ff000+18a000] |
| 15:42.49 | file | instructions for getting a backtrace are at https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace which can show where it is crashing |
| 15:45.05 | volga629 | yes I am collecting right now |
| 15:45.13 | volga629 | I will open new issue |
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| 16:12.11 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-26222 |
| 16:14.00 | scv | volga629: you need to install the debugging symbols for your asterisk |
| 16:14.15 | scv | the majority of the backtrace is useless |
| 16:14.29 | scv | it would typically be the -debuginfo or -dbg packages |
| 16:14.41 | volga629 | you mean this asterisk13-debuginfo.x86_64 |
| 16:14.43 | scv | yes |
| 16:15.57 | scv | you would also benefit from installing the debuginfo for other related modules there like openssl-debuginfo and iksemel-debuginfo |
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| 16:21.31 | volga629 | I updated ticket |
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| 17:32.08 | *** join/#asterisk Janos (~cramos@190.171.120.15) |
| 17:33.44 | Janos | hey there, does anyone knows of an asterisk function or app that would allow me to do extension pattern matching programatically ? let's say I have a string value returned from the database and I would like to check of it matches 80XX using the asterisk extension matching code |
| 17:35.20 | WIMPy | The only thing is DIALPLAN_EXISTS. |
| 17:35.33 | Janos | WIMPy, thanks a lot, will check it out |
| 17:36.27 | Janos | another question, do you know if there is any kind of non exact lookup on the asterisk database ? |
| 17:42.30 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 17:52.29 | [TK]D-Fender | whatt is a non-exact lookup? |
| 17:52.42 | [TK]D-Fender | And unless you're running realtime... there is no database either |
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| 17:57.29 | Janos | [TK]D-Fender, sorry, I mean the asterisk internal database, the one you can access with DB_EXISTS etc, and by non exact lookup I mean to obtain all values that match a pattern instead of the exact key |
| 17:57.52 | Janos | I pretty sure it's not possible, just wanted to confirm it |
| 17:59.03 | [TK]D-Fender | it doesn't obtain values. |
| 17:59.11 | [TK]D-Fender | it returns true or false |
| 17:59.36 | [TK]D-Fender | And it'll confirm a match against all patterns in the path |
| 18:04.17 | Janos | indeed it returns true or false and if true it places the value in the variable DB_RESULT, but that was not my question, the question was if I could use a pattern as lookup key instead of the exact value |
| 18:07.44 | volga629 | I am trying press "Send Back" in jira and get Workflow Action Invalid |
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| 18:15.31 | [TK]D-Fender | Janos, No because your pattern can be multiple values and * will not try to loop them for all possiblities |
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| 19:05.07 | thiagoc | Hi, is there a way to escape the pipe character in the SHELL function? |
| 19:15.40 | Janos | [TK]D-Fender, thanks a log, I wanted to confirm that, cheers |
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| 22:38.33 | sanmarcos | does anybody have experience with PJSIP decoding of H264 over RTP and why it shows green frames initially? |
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| 23:00.47 | celibate | ahh it's a wonderful day |
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| 23:27.19 | infinity1 | is there something in the newer asterisk that would cause 1 way audio? i'm setting up a new ppbx and the remote phones will not connect. |
| 23:27.56 | infinity1 | well, they connect, but you can't hear anything. the phone says its not reiving packets, only sending. very strange. |
| 23:28.31 | robmal | It's always NAT. |
| 23:28.45 | infinity1 | thats what i always thought ... |
| 23:29.13 | infinity1 | but after spending the last few hours on this i'm starting to wonder. |
| 23:29.51 | robmal | You get used to it after some time. But it's always NAT. Or some router with SIP ALG. |
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| 23:31.36 | infinity1 | well, if its sonicwall, i can't seem to figure out the right combo. i have it working on an old tz200 at another site. this one is a tz400 and also different asterisk versions |
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| 23:33.13 | robmal | Did you set extenip in sip.conf? |
| 23:34.18 | infinity1 | <PROTECTED> |
| 23:34.49 | robmal | sip set debug peer whatever and check if the ip addresses match |
| 23:35.20 | infinity1 | hmmm . i'll try that. |
| 23:35.59 | infinity1 | phone shows registered. |
| 23:36.00 | infinity1 | 444/444 166.171.250.160 D Yes Yes A 63662 OK (58 ms) |
| 23:36.48 | robmal | Since it can make a call thats not a surprise. Debug your siptrunk. |
| 23:36.52 | infinity1 | what should i look for in the debug ? |
| 23:37.12 | infinity1 | its not a trunk issue. local phones work. WAN phones do not. |
| 23:38.35 | infinity1 | i don't see anything obvious in the debug but i'm not really sure what to look for . heh |
| 23:38.51 | robmal | Outbound calls only? |
| 23:39.36 | infinity1 | hmmm ..not sure. i only have one remote phone ...and i'm remote. |
| 23:40.12 | infinity1 | all internal phones are fine though. |
| 23:40.14 | robmal | Ok, set localnet in sip [general], should help. |
| 23:40.51 | infinity1 | hmmm ..shit |
| 23:40.58 | robmal | If it wasn't almost 2am my time i'd be more helpful but 90% of my brain is already asleep ;-) |
| 23:41.00 | infinity1 | i checked that earlier and forgot the phones are on a different vlan. |
| 23:41.04 | infinity1 | lets see. |
| 23:42.59 | infinity1 | weird. i'm getting different results. |
| 23:43.12 | infinity1 | i guess thats good :) |
| 23:44.15 | infinity1 | now my bria client is stuck saying "calling..." |
| 23:44.39 | infinity1 | i probably broke the sonicwall trying to fix it. |
| 23:44.45 | infinity1 | thanks. i'll be back soon :) |
| 23:45.00 | robmal | Good luck :-) |
| 23:46.10 | infinity1 | thanks. i'll need it! |
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| 23:58.48 | Uranio | hi there, I'm trying to setup asterisk for sending message between 2 sip extensions |
| 23:58.50 | Uranio | but... |
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| 23:59.10 | Uranio | it fails with Code problem's error |
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