| 00:07.04 | Katty | props file back up. |
| 00:07.24 | Katty | checks for signs of life |
| 00:07.44 | file | wobbles |
| 00:08.01 | Katty | Hrm. |
| 00:08.16 | Katty | restores a previous file version |
| 00:08.25 | file | crashes |
| 00:08.32 | Katty | Oops! |
| 00:08.43 | Katty | reverts to previous snapshot |
| 00:20.17 | *** part/#asterisk kharwell (kharwell@nat/digium/x-mucwrazngzbafmhj) |
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| 01:25.58 | b6 | is "from-trunk" a special official magic context? |
| 01:29.11 | [TK]D-Fender | in Asterisk, no. In FreePBX, yes |
| 01:32.04 | b6 | i see. i use asterisk, but followed some instructions that said to append a [from-trunk] to sip.conf. does it make sense that my from-trunk context would still work somehow in asterisk? |
| 01:32.58 | [TK]D-Fender | that is a device definition.... not a trunk |
| 01:33.08 | [TK]D-Fender | that is a device definition.... not a CONTEXT |
| 01:33.13 | [TK]D-Fender | (correction) |
| 01:33.25 | b6 | i see, sorry, still catching up on terminology. |
| 01:33.28 | [TK]D-Fender | and "some instructions" is dangerously vague |
| 01:36.03 | b6 | sorry again, i mispoke. i added my dialplan under [from-trunk] in extensions.conf. it's effective, but i wasn't sure what if anything was special about the name from-trunk. |
| 01:38.36 | [TK]D-Fender | No, the name itself is not special. There are only 3 places in extensions.conf that are: [default], [globals], and [general] |
| 01:46.26 | b6 | i see, thanks. |
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| 03:02.11 | sbrath | Well I'm back to my timing issue from this morning, Asterisk 11 on a HP DC5800 workstation, when I do a voicemail recording, and play it back it sounds like I'm fast forwarding it at 4x speed.... I checked timing test, and I get "1019 milliseconds and 51 ticks" .. is that good? |
| 03:05.43 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
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| 03:23.44 | sbrath | What's interesting is that if I call in to the system over VoIP and leave the message, it's normal. If I call in from an onsite phone, it's garbled? |
| 03:23.52 | sbrath | ethernet card maybe? |
| 03:28.14 | sbrath | everyone must be sleeping :) |
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| 04:27.40 | ChannelZ | using what timer? |
| 04:41.06 | sbrath | right now dahdi |
| 04:41.20 | sbrath | But I had all 3 enabled, and it was the same thing. |
| 04:59.52 | *** join/#asterisk timahvo1 (~rogue@105.164.222.72) |
| 05:04.07 | WIMPy | So what's the difference between those phones? |
| 05:05.53 | sbrath | Well the system only has 1 yealink T28 on it, and a 24 port wildcard. |
| 05:06.50 | WIMPy | So it's either dahdi or sip? |
| 05:06.52 | sbrath | If I call in from the yealink ( that I know is good as I just used it elsewhere and it was fine ) and dial *4000 to go direct to voicemail, leave a message. It's super fast. |
| 05:07.09 | WIMPy | Leaves us with 4 options for record/playback. |
| 05:07.22 | sbrath | If I call in over the internet via voip-provider then the recording is normal. |
| 05:07.46 | sbrath | It seems like it can't figure out what codec to use with the phone and it's going from "unknown" to ulaw. and then something about slin. |
| 05:07.56 | sbrath | Maybe it's doing a wierd transcoding? |
| 05:08.40 | WIMPy | Probably. |
| 05:09.09 | WIMPy | Ok, so you have 3 options? A SIP phone, a SIP ITSP and dahdi? |
| 05:10.00 | WIMPy | And what combinations of recording and playbak works/doesn't work? |
| 05:10.48 | sbrath | If I call in on the SIP phone, and leave a message... Then check my voice mail on the phone, it's garbled. |
| 05:11.07 | sbrath | If I call in via voip provider, and leave a message. And then check the phone's voice mail, it's fine. |
| 05:11.44 | sbrath | I can try to call in via the DAHDI. let me try that. |
| 05:12.24 | WIMPy | And all pre-recorded propt play fine all the time? |
| 05:13.18 | sbrath | pre records are fine. |
| 05:13.53 | sbrath | Ok, called in via dahdi and I get no audio, so something is wrong with that... This is a PRI connected to a Toshiba CTX-100 in dumb mode. |
| 05:15.31 | WIMPy | So it's probably recording that fails. |
| 05:16.52 | WIMPy | Oh, BTW: If you repeatedly do 'timing test' do you frequently get odd numbers? |
| 05:18.09 | sbrath | if you mean Odd as in not-even numbers, I did when I was forcing it to use only the dahdi timer. But now it says it's using the timerfd and it 1000 ms and 50 ticks |
| 05:18.50 | WIMPy | That's not a good thing. |
| 05:19.14 | WIMPy | If you have dahdi, that should be your timing source. |
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| 05:20.46 | sbrath | should I force it to dahdi? |
| 05:20.55 | sbrath | I'm the clock master for dahdi |
| 05:21.41 | WIMPy | I'm not sure at all that's realated to your issue, but it's not good, either. |
| 05:21.41 | WIMPy | Maybe you need to check for shared IRQ. |
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| 05:22.21 | WIMPy | Still the cards timer should be better than any software version. |
| 05:22.30 | sbrath | r |
| 05:23.02 | WIMPy | And if it isn't, that would be an issue. |
| 05:25.45 | sbrath | I switched to dahdi, and now my timer test is like 1019 ms with 51 ticks |
| 05:26.12 | sbrath | how do I check the IRQ |
| 05:26.28 | WIMPy | You should run the dahdi tests from the system shell. |
| 05:26.40 | WIMPy | cat /proc/interrupts for a start. |
| 05:27.31 | WIMPy | Troule is you can have shared IRQ without seeing them if you don't have a driver loaded for the other device. |
| 05:27.53 | sbrath | dahdi_test -vvv almost 100% all the time. |
| 05:28.48 | WIMPy | It should be very close to 100. |
| 05:29.08 | sbrath | all APIC on the /proc/interrupts |
| 05:29.59 | sbrath | eth is on 27 and wcte11xp is on 20 ?? |
| 05:30.19 | WIMPy | And they are alone? |
| 05:30.34 | sbrath | bunch of USB devices on each of them |
| 05:30.45 | sbrath | well eth0 is alone |
| 05:30.56 | sbrath | and wildcard is sharing with usbb 2,3,6 |
| 05:31.00 | WIMPy | That is not good. |
| 05:31.19 | sbrath | ok, what do I do? |
| 05:31.38 | WIMPy | So how close to 100% did the test get? |
| 05:31.46 | sbrath | almost 100% everytime |
| 05:32.02 | WIMPy | I still think the main issue must be some transcoding/resampling thing. |
| 05:32.02 | sbrath | 99.996 or better |
| 05:32.21 | WIMPy | That should be ok, I think. |
| 05:32.32 | sbrath | this is on a HP DC5800 workstation, if that makes a difference. |
| 05:32.38 | WIMPy | NFI |
| 05:33.23 | sbrath | in a previous build I had a problem that hdparm -t /dev/sda returned poor numbers and there was a busmaster issue, on another box like this one. |
| 05:33.24 | WIMPy | Did you build it yourself? |
| 05:33.27 | sbrath | but this one seems ok. |
| 05:34.00 | sbrath | As in did I load Asterisk myself on the box.. Yes. |
| 05:34.06 | sbrath | via --- PIAF |
| 05:34.26 | WIMPy | That would be a no. |
| 05:34.44 | sbrath | I could rebuild asterisk. |
| 05:34.55 | WIMPy | This alls seems rather strange to me. |
| 05:35.14 | sbrath | me too, as I've built MANY boxes and non have done this. |
| 05:35.21 | WIMPy | But maybe you shoudl try to get dahdi working first, so you have one more option for comparison. |
| 05:35.47 | sbrath | winder why I wouldn't have audio on it one way? |
| 05:36.26 | WIMPy | Yes. One way audio is easy with SIP, but really hard with dahdi. |
| 05:36.43 | WIMPy | (really hard = not normally possible) |
| 05:38.18 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:40.23 | sbrath | I think I'm going to grab a working asterisk 1.8 box tomorrow, and replace the asterisk 11 box here and see if the known working install works... |
| 05:40.36 | sbrath | Then I can eliminate the card / server / install / et al. |
| 05:41.31 | WIMPy | It always helps to eliminate as many unknowns as possible,. |
| 05:50.56 | sbrath | for grinns... I setup the voicemail to email, and played the resulting wav, and it's super fast. |
| 05:51.03 | sbrath | I can forward it to you if you'd like to hear :) |
| 05:52.28 | WIMPy | Not that I take anythign for granted. But I think that playing faster than it should is a description I can trust :-) |
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| 06:13.19 | WIMPy | Damn. What I wrote there is a clear indication that I shold go to sleep. |
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| 06:33.54 | ChannelZ | rblsmtpd: 24.155.162.97 pid 30373: FROM:<fxC4480@amoricanexpress.com> |
| 06:34.02 | ChannelZ | Amorican Express.. sounds legit |
| 06:43.41 | [TK]D-Fender | <WIMPy> Yes. One way audio is easy with SIP, but really hard with dahdi. <- had it happen once. Early Sangoma Otasic HWEC buffer got caught and swallowed up everything :) |
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| 15:21.32 | phpdave11 | i want to try out asterisk on a linux virtual machine. what's the best distro for asterisk? |
| 15:21.50 | wanda__ | hullo o/ |
| 15:22.39 | kleszcz | phpdave11: debian and centos |
| 15:22.46 | kleszcz | 4me ofc. |
| 15:23.05 | phpdave11 | ubuntu server a good choice? |
| 15:23.31 | phpdave11 | based on debian |
| 15:24.02 | [TK]D-Fender | phpdave11: Whichever you can properly manage |
| 15:24.33 | [TK]D-Fender | phpdave11: * doesn't care what you run it on as long as you satisfy its dependencies |
| 15:24.59 | phpdave11 | ok, thanks! |
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| 15:31.20 | SpaceInvaders | when evaluating SIP service providers (within the US) do I need to be concerned with geographic location of POPs (TTL, latency)? |
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| 15:39.39 | stkoch | hi WIMPy thanks for the local channel trick, it works now fine. configuration: http://pastebin.com/dpCChLg8 |
| 15:40.45 | stkoch | The warning at too much packet loss Exceptionally long voice queue length queuing ... is now not for TAPI, it's for Local |
| 15:41.14 | stkoch | I can hear the jitter, but the call is no more terminated |
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| 15:53.33 | solmsted | Hello, anyone using UniMRCP and speech recognition? I can set MRCP params globally in a config file. Is there any way to set them during a call? Ideally I'd like to be able to call Asterisk's SpeechBackground application with custom MRCP values. |
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| 15:59.48 | noko | Hello guys. I need such scheme: |
| 15:59.50 | noko | exten => s,1,Queue(myqueue,tT,,,20) |
| 15:59.50 | noko | exten => s,2,Answer() |
| 16:00.31 | [TK]D-Fender | noko: I don't see a point in the Answer following your Queue there... |
| 16:00.50 | [TK]D-Fender | noko: It will already answer the channel. |
| 16:01.02 | noko | I want ring in queue without answer and if queue is timeout the answer |
| 16:01.30 | noko | no! My queue don't answer unit somebody answer in queue |
| 16:01.34 | [TK]D-Fender | noko: "core show application queue" <- |
| 16:01.38 | noko | * until |
| 16:02.31 | noko | <PROTECTED> |
| 16:02.39 | WIMPy | stkoch: Well, at least you've got a workaround then. |
| 16:03.20 | noko | [TK]D-Fender, i need to answer only when queue in not answered and exiting by timeout |
| 16:03.58 | [TK]D-Fender | noko: "core show application queue" <- |
| 16:04.04 | noko | but now I get "Spawn extension (menu, s, 1) exited non-zero on..." |
| 16:04.28 | noko | [TK]D-Fender, give me a hint, please |
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| 16:04.55 | [TK]D-Fender | noko: Hint : read the instructions. They tell you what to do to have it not answer the channel |
| 16:06.42 | noko | n: No retries on the timeout; will exit this application and go to the next step. |
| 16:06.45 | noko | you mean this? |
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| 16:08.05 | noko | [TK]D-Fender, still the same. |
| 16:08.22 | [TK]D-Fender | No, that clearly says nothing about answering the call or not. |
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| 16:10.09 | noko | [TK]D-Fender, are you sure that this option exist in asterisk 1.8.13? |
| 16:10.26 | noko | maybe option "C: Mark all calls as "answered elsewhere" when cancelled."? |
| 16:10.38 | [TK]D-Fender | noko: pastebin the instructions..... |
| 16:12.39 | noko | [TK]D-Fender, http://dumpz.org/1315594/text/ |
| 16:13.23 | [TK]D-Fender | noko: And a call attempt from CLI.... |
| 16:15.53 | noko | [TK]D-Fender, http://dumpz.org/1315595/text/ |
| 16:16.59 | noko | queue working as i expect, it ringing without answering |
| 16:17.46 | noko | but it hangup when timeout and not go to the next step |
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| 17:05.49 | [TK]D-Fender | noko: pastebin your actual dialplan |
| 17:11.25 | *** part/#asterisk stkoch (58d12049@gateway/web/freenode/ip.88.209.32.73) |
| 17:33.49 | *** join/#asterisk Xaviertoor (~jerson.ju@177.99.205.154) |
| 17:34.16 | Xaviertoor | Someone already made ââSiemens HiPath 4000 integration with Asterisk ? |
| 17:43.56 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 17:44.14 | phpdave11 | it's possible to place an automated test call from my asterisk box to a SIP URI, isn't it? |
| 17:45.05 | WIMPy | Xaviertoor: "Integration"? dould mean anything. |
| 17:46.15 | *** join/#asterisk JeffC_NN (32cad19e@gateway/web/freenode/ip.50.202.209.158) |
| 17:46.54 | JeffC_NN | Anyone know why ##XXX wouldn't match ##122 in the dialplan? |
| 17:47.16 | JeffC_NN | if I specify ##122, it works, but if I use the pattern X, it doesn't |
| 17:47.36 | mjordan | what is your actual extension in the dialplan? |
| 17:48.59 | JeffC_NN | ;call subscriptions/pickupsexten = _##XXX,hint,SIP/${EXTEN:2}) exten = _##XXX,1,PickUp(${EXTEN:2}@default) |
| 17:49.05 | JeffC_NN | arg |
| 17:49.13 | phpdave11 | i'm trying to place an automatic test call by creating a .call file but i'm getting this error message: Call from '' (127.0.0.1:5060) to extension 'sip:17771234567@callcentric.com' rejected because extension not found in context 'public'. |
| 17:49.33 | JeffC_NN | http://pastebin.com/PL8wjeuS |
| 17:49.52 | noko | [TK]D-Fender, http://pastebin.com/raw.php?i=7M23qcj1 |
| 17:49.52 | JeffC_NN | the underscores are a new test I'm trying |
| 17:50.34 | [TK]D-Fender | jeffShow us the actual call attempt |
| 17:50.39 | [TK]D-Fender | JeffC_NN: Show us the actual call attempt |
| 17:50.47 | file | the underscores are required or else it won't be considered a pattern match |
| 17:51.40 | JeffC_NN | It's a subscription and a call, 2 pieces |
| 17:52.06 | [TK]D-Fender | JeffC_NN: Show us |
| 17:53.33 | [TK]D-Fender | noko: Look at the SIP debug to prove which side dropped it. |
| 17:53.39 | *** join/#asterisk JBITX (~jbl@200.150.121.118) |
| 17:55.57 | noko | [TK]D-Fender, maybe you can say what queue option you mean earlier? |
| 17:56.07 | JeffC_NN | Hmm. seems to be a problem with how my softphone handles # in speed dials |
| 17:56.56 | doop | JeffC_NN lots of equipment interprets # as "i'm finished dialing so put my call through right now" -- maybe you should use * |
| 17:57.55 | JeffC_NN | aha! I should pay closer attention to the warnings when I reload the dialplan. I had a syntax error. :) |
| 17:58.42 | JBITX | hi guys... 2 or 3 times a day my asterisk stop accept new calls but remains responsive for some commands |
| 17:59.00 | JBITX | sip show peers ok... but core show channels no |
| 17:59.07 | JBITX | any ideia? |
| 17:59.21 | mjordan | JBITX: what version of Asterisk? |
| 18:00.07 | JBITX | i have same problems in 1.8.15 cert2, 1.8.31 and now with 11.15 |
| 18:00.12 | JeffC_NN | How do I have strings/arguments that contain semicolons in dialplan? ex: exten => _X.,1,SIPAddHeader(Call-Info: sip: ;answer-after=0) |
| 18:00.38 | mjordan | JBITX: most likely, you have a deadlock of some sort. |
| 18:01.12 | mjordan | Use the instructions here to gather information: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 18:01.18 | [TK]D-Fender | noko: Right now I can't prove that it's not working fine. We need to see the call |
| 18:02.08 | Xaviertoor | WIMPy, Interconnection using Digium E1 ISDN |
| 18:03.13 | WIMPy | Xaviertoor: Off course you can connect them. |
| 18:06.45 | JBITX | mjordan, when i compile asterisk with debug flags, minutes later my sip peers goes to unreachable |
| 18:07.39 | *** join/#asterisk danjenkins_ (~anonymous@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:12.58 | JBITX | the problem only happens with a high volume of calls, with transcoding (GSM>G729 or GSM>ISDN). Which is a high volume to you? |
| 18:13.47 | JBITX | mjordan, some deadlock on different versions? something on dialplan? Is the same on all installs |
| 18:14.04 | JBITX | same deadlock... |
| 18:15.08 | *** join/#asterisk generalhan_ (~tester@about/windows/staff/generalhan) |
| 18:19.45 | mjordan | JBITX: there's no way I can speculate on what might be causing the issue without any evidence. |
| 18:24.40 | noko | [TK]D-Fender, here is the part where call hangup'd http://dumpz.org/1315686/text/ |
| 18:28.26 | noko | "c - Continue in the dialplan if the callee hangs up." |
| 18:28.37 | noko | wow, i don't see this option in my asterisk |
| 18:34.04 | *** join/#asterisk killfill (~killfill@190.107.178.50) |
| 18:34.09 | killfill | Hi |
| 18:34.29 | killfill | anyone knows if its possible to connect this phone to asterisk? http://www2.panasonic.com/webapp/wcs/stores/servlet/BTSModelDetail?storeId=11201&catalogId=13051&catGroupId=141510&itemId=134566&modelNo=KX-NT343 |
| 18:34.35 | killfill | it doesnt seem to be SIP, right? |
| 18:44.26 | Xaviertoor | noko, use option q of queue_app |
| 18:45.50 | Xaviertoor | noko, sorry, c - Continue in the dialplan if the callee hangs up |
| 18:46.14 | noko | Xaviertoor, i'm already try option "c" |
| 18:46.24 | noko | with the same result |
| 18:48.03 | *** join/#asterisk adoodle (~adeeln@fw1.ridgeway.scc-zip.net) |
| 18:50.24 | adoodle | i'm using the ODBC connector to connect to an Oracle DB and am trying to execute a package procedure which takes in and out variables....is it possible to execute that kind of a procedure from the asterisk dialplan? i've tried using an anonymous block; but i can't get that to work...it fails after the begin statement |
| 18:51.46 | adoodle | the skeleton definition for the procedure looks like: procedure getBackendNum(idnum in varchar2, ilang in varchar2, itermcode in number, retcode out number, benum out number); |
| 18:57.52 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 19:10.26 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-45-135.dynamic.qsc.de) |
| 19:11.30 | phpdave11 | can anyone tell me why my outgoing test sip call fails? https://gist.github.com/anonymous/914aaea3ef98284138b4 |
| 19:15.34 | adoodle | phpdave11: Failed to authenticate on INVITE to 'asterisk' .... |
| 19:15.47 | adoodle | phpdave11: so your auth fields are off.. |
| 19:16.21 | adoodle | phpdave11: http://www.callcentric.com/support/device/asterisk/1_4 |
| 19:16.22 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:17.20 | phpdave11 | i see... so in order to place a sip call from my asterisk server to a callcentric account, i must set up auth fields? |
| 19:18.57 | *** join/#asterisk funtriaco (~funtriaco@mail.brickellmotors.com) |
| 19:21.29 | funtriaco | can i execute an shell script or perl script from an agi script inside agi-bin? |
| 19:22.35 | WIMPy | Thqt |
| 19:22.39 | WIMPy | oops |
| 19:22.44 | WIMPy | That' swhat AGI is there for. |
| 19:23.57 | funtriaco | currently doing a shell_exec, but doesnt seem to be doing anything. |
| 19:24.59 | funtriaco | so the mainscript.php gets executed from the extensions.conf and theres a line in the mainscript.php that needs to execute a perl script. |
| 19:25.49 | WIMPy | Ah. That way. Still a clear yes. You can do whatever you want, as long as you have permissions. |
| 19:26.19 | *** join/#asterisk CeBe (~CeBe@dhcp-213-167.vpn.tu-berlin.de) |
| 19:26.30 | funtriaco | hmm |
| 19:29.42 | adoodle | funtriaco: are you calling your perl script via full path, or relative path? |
| 19:39.09 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:40.59 | funtriaco | adoodle: full path.. |
| 19:41.08 | funtriaco | found.. there was actually an error on the script.. sorry |
| 19:41.13 | adoodle | ah ok |
| 19:42.18 | *** join/#asterisk MrSparks (~twanny@c103-220.i07-26.onvol.net) |
| 19:45.29 | *** part/#asterisk phpdave11 (~phpdave11@ec2-23-22-82-220.compute-1.amazonaws.com) |
| 19:48.40 | solmsted | Hello, anyone using UniMRCP and speech recognition? I can set MRCP params globally in a config file. Is there any way to set them during a call? Ideally I'd like to be able to call Asterisk's SpeechBackground application with custom MRCP values. |
| 19:50.23 | *** join/#asterisk phpdave11 (~phpdave11@ec2-23-22-82-220.compute-1.amazonaws.com) |
| 19:51.24 | *** part/#asterisk phpdave11 (~phpdave11@ec2-23-22-82-220.compute-1.amazonaws.com) |
| 20:04.15 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 20:05.49 | *** join/#asterisk funnymanva (~funnymanv@pool-72-73-20-234.clppva.fios.verizon.net) |
| 20:10.58 | *** join/#asterisk areski (~areski@80.174.128.98.dyn.user.ono.com) |
| 20:12.02 | *** join/#asterisk cyford (allen@76.122.73.37) |
| 20:12.19 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:13.48 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:16.40 | *** part/#asterisk mjordan (mjordan@nat/digium/x-lwmjghsxmnsncpyc) |
| 20:19.55 | eppigy | hello |
| 20:19.57 | eppigy | i am dave |
| 20:21.31 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:340f:15e6:559d:e94a) |
| 20:22.47 | [TK]D-Fender | yes you is |
| 20:22.51 | *** part/#asterisk killfill (~killfill@190.107.178.50) |
| 20:24.52 | eppigy | thank you |
| 20:25.04 | eppigy | how's life [TK]D-Fender? |
| 20:25.39 | [TK]D-Fender | Still breathing. SSDD |
| 20:26.05 | *** join/#asterisk webmonkey (~clint@2607:fcc8:b7c5:9600:dce1:f772:ee3a:aadb) |
| 20:27.15 | eppigy | excellent |
| 20:29.38 | averythomas | any way to get a free VOIP number to work with asterisk? |
| 20:30.23 | webmonkey | I am having an issue using the MixMonitor command. The call recording works just fine, but I am trying to use the "command" option to execute a script when the recording has completed. The script works from the command line as the Asterisk user, and there do not seem to be any file permissions issues. Any advice on where I can find error output? |
| 20:30.52 | webmonkey | I see in the console that it is trying to call the command: == MixMonitor close filestream (mixed) |
| 20:30.52 | webmonkey | <PROTECTED> |
| 20:30.52 | webmonkey | <PROTECTED> |
| 20:34.06 | webmonkey | I cannot find any errors even when enabling debug logging. |
| 20:45.10 | [TK]D-Fender | * doesn't pipe any output back to the console. |
| 20:45.21 | mbowie | With the demise of the Asterisk GUI project, is there something comparable worth looking at for adding a general UI to Asterisk 11? (Adding users, extensions and such) |
| 20:45.24 | [TK]D-Fender | Any logging has to be done by your script itseld |
| 20:45.46 | [TK]D-Fender | First to think about is PATHS. Don't assume the folder it is executed from when referencing other files |
| 20:46.22 | [TK]D-Fender | mbowie: Pretty much everything out there is a "full service" GUI, for which FreePBX is the most mature, featured, and free |
| 20:48.08 | mbowie | [TK]D-Fender: Thanks, I'll give it another look. My only encounter with it previously has been as it's own distro, rather than layering it over a clean * install. |
| 20:49.57 | [TK]D-Fender | mbowie: it has always been a "bolt-on" that is separately downloadable and installable over you current server |
| 20:50.10 | [TK]D-Fender | mbowie: Them providing a distro is a new OPTION |
| 20:50.19 | [TK]D-Fender | mbowie: They have a tarball w/ instructions |
| 20:50.41 | mbowie | Roger that... will get into it and get educated. Gracias! |
| 20:51.11 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 20:54.43 | mbowie | Ouch... it has a dependency list that makes an addict look good. |
| 20:58.22 | *** join/#asterisk ipengineer_ (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:59.17 | [TK]D-Fender | no worse than * itself... |
| 20:59.50 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 21:00.35 | mbowie | We can install * totally without depends, or with speex. I think that's pretty light by comparison. |
| 21:03.23 | WIMPy | And Asterisk certainly doesn't require PHP. |
| 21:03.25 | WIMPy | (yet) |
| 21:04.13 | mbowie | Yeah, this is basically an AMP stack with a few extras. |
| 21:05.04 | mbowie | Don't get me wrong, but coming from using what was basically a bolt-on GUI, this is significantly more surface area etc. |
| 21:05.16 | WIMPy | And no mysql or apache, either. |
| 21:06.12 | mbowie | (That's the "M" and the "A" in AMP. ;-) ) |
| 21:10.59 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 21:13.57 | webmonkey | okay thank you [TK]D-Fender, I will check that out. |
| 21:18.38 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 21:20.38 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 21:27.04 | webmonkey | This tells me my filepath is accurate: == Executing [/var/spool/asterisk/monitor/test.sh] |
| 21:28.09 | [TK]D-Fender | webmonkey: I'm talking about once it's RUNNING |
| 21:28.32 | [TK]D-Fender | webmonkey: If your script is attempting to touch files... that those paths had better bee absolute in your script |
| 21:28.51 | webmonkey | ah! Thank you for clarifying. |
| 21:29.27 | [TK]D-Fender | [15:45][TK]D-FenderFirst to think about is PATHS. Don't assume the folder it is executed from when referencing other files |
| 21:30.34 | webmonkey | Specified absolute filepaths in the script; still doesn't work. |
| 21:33.55 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 21:40.21 | drmessano | webmonkey, that funky monkey |
| 21:41.07 | drmessano | webmonkey, junkie, that funky monkey |
| 21:41.14 | webmonkey | ha. The funky webmonkey is currently ticked at what is probably a stupid simple issue. |
| 21:42.05 | drmessano | If its permissions, chown, if you can't execute, sudo. For all the rest, reboot |
| 21:50.23 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 21:51.04 | webmonkey | Pretty much ruled all that out; it's like Asterisk isn't actually launching the script for some reason... |
| 21:51.24 | webmonkey | Trying this on two different systems and am having the same problem on each. |
| 21:51.38 | webmonkey | All I want is a hint in a log somewhere... |
| 21:54.17 | [TK]D-Fender | Might not happen |
| 21:54.29 | [TK]D-Fender | verify permissions on the script, the files it's supposed to work on, etc |
| 21:58.18 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 22:02.32 | webmonkey | everything in question is owned and writable by asterisk. |
| 22:02.46 | adoodle | webmonkey: but is it executable by asterisk? |
| 22:04.19 | webmonkey | effing balls *facepalm* |
| 22:04.57 | [TK]D-Fender | senses a One D Ten Tee error :) |
| 22:05.29 | webmonkey | yup.... |
| 22:05.52 | webmonkey | I told ya it was probably stupid simple ha |
| 22:06.08 | webmonkey | Still, you would think Asterisk would through an error when it can't run it. |
| 22:06.29 | webmonkey | Thanks adoodle and [TK]D-Fender ! |
| 22:06.56 | [TK]D-Fender | You're welcome |
| 22:10.25 | webmonkey | I think I was testing a different script when I ran it as the asterisk user with success previously... oh well, doesn't matter now. |
| 22:11.17 | [TK]D-Fender | Mistake discoverd and fixed. That's what matters. And it didn't even take all that long |
| 22:29.24 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 22:31.10 | webmonkey | Yay, the real script works now too :-) |
| 22:31.23 | [TK]D-Fender | \o/ |
| 22:50.40 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 23:07.20 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 23:21.28 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 23:23.48 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:46.10 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 23:48.38 | *** join/#asterisk siptipping (~moonman@modemcable069.209-22-96.mc.videotron.ca) |
| 23:48.42 | siptipping | hello |
| 23:49.08 | siptipping | Is anyone here familiar with Elastix 3.0 MT and can provide Inbound/Outbound route support? |
| 23:50.12 | *** part/#asterisk kharwell (kharwell@nat/digium/x-czukuaednqkfwarm) |
| 23:53.39 | WIMPy | ~elastix |
| 23:53.39 | infobot | i heard elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 23:54.29 | siptipping | <PROTECTED> |
| 23:54.49 | siptipping | thanks for being a smartass though much appreciated.... |
| 23:54.57 | siptipping | nobody is active in that channe |
| 23:54.58 | siptipping | l |
| 23:55.39 | WIMPy | Well, Asterisk doesn't knwo such things as inbound/outbound routes. Is that of more help to you? |
| 23:57.59 | WIMPy | Maybe you could try #freepbx. Not sure how they see elastix. |