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| 00:58.56 | k054 | hey everyone! |
| 00:59.36 | k054 | any idea if it is possible to redirect an incomming call based on the caller number to an specific extension? |
| 01:00.15 | WIMPy | Yes, but your terminology isn't Asterisk compatible. |
| 01:00.32 | k054 | oh, great! |
| 01:00.38 | WIMPy | You can use extension/callerid or use function CALLERID. |
| 01:01.19 | k054 | how could I phrase that? (maybe I'm using the wrong term to find out how to do it) |
| 01:01.51 | WIMPy | It's part if the extension syntax. |
| 01:02.15 | WIMPy | Should be explained in the sample extensions.conf. |
| 01:04.45 | k054 | WIMPy: not sure what to look for. |
| 01:05.48 | WIMPy | extension/callerid <= that's a literal / in the extension. |
| 01:06.18 | k054 | my final goal would be to have a list of CALLERID (that being an analog phone number) so if the CALLERID matches an entry of that list, it should be directed to certain extension. |
| 01:06.33 | k054 | is that possible? |
| 01:07.48 | WIMPy | "Extension" is what is being dialled. You send calls to a "device" or "peer". |
| 01:08.45 | Samot | k054: Yes, it's possible. |
| 01:09.26 | k054 | Samot: great! thanks! any link or something I could take a look to figure out how? |
| 01:10.32 | Samot | Well you should read up on anything about Asterisk dialplan since that's what you need to use to make this happen. |
| 01:10.54 | Samot | But this well documented across the Internet. |
| 01:13.03 | k054 | Samot: thanks. any specific term I should use? "Asterisk dialplan" looks like a too wide one. |
| 01:13.24 | Samot | "How to route incoming call based on callerid asterisk" |
| 01:13.30 | Samot | Pretty much your question. |
| 01:14.58 | k054 | Samot: I was having issue phrasing it since Englush is not my native language but that phrase did the trick. Thanks! |
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| 06:54.32 | jwpierce3 | my queue is ignoring my time value, It is currently only ringing 3 times. I had it set to 30, then tried 60, same 3 rings |
| 06:57.41 | scv | how are you dialing your queue members |
| 06:58.56 | jwpierce3 | incoming calls from provider. exten => 16014501263,1,NoOp(Incoming Call) |
| 06:59.02 | jwpierce3 | <PROTECTED> |
| 06:59.09 | jwpierce3 | <PROTECTED> |
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| 07:07.14 | [TK]D-Fender | Show us the call & the actual config to back it |
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| 07:10.23 | jwpierce3 | http://pastebin.com/HWJthjSv |
| 07:10.34 | jwpierce3 | extensions.conf |
| 07:10.38 | jwpierce3 | http://pastebin.com/Ntu1gWHY |
| 07:10.41 | jwpierce3 | queue.conf |
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| 07:15.55 | jwpierce3 | http://pastebin.com/DS4Ax44C |
| 07:16.00 | jwpierce3 | Master.csv |
| 07:16.12 | [TK]D-Fender | csv = irrelevant |
| 07:16.14 | [TK]D-Fender | show the call |
| 07:16.21 | [TK]D-Fender | CLI <------------------------ |
| 07:17.56 | jwpierce3 | http://pastebin.com/vmT00CJu |
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| 07:21.24 | [TK]D-Fender | Well you haven't defined an agent timeout in your queue config |
| 07:21.41 | [TK]D-Fender | So there is no defined interval to wait. So what do you thing * should wait before trying again? |
| 07:22.03 | [TK]D-Fender | Then take that numebr and add it to the time that has ALREADY passed. Should that EXCEED the timeout you specified? |
| 07:22.25 | [TK]D-Fender | You should actually make proper explicit setting in the queue config... |
| 07:22.32 | [TK]D-Fender | you left a lot out |
| 07:23.07 | jwpierce3 | Thank you, I'll research that. |
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| 14:39.14 | Lope | Previously I've been setting up my asterisk server such that trusted SIP peers are allowed to connect on 5060 and all other IP' |
| 14:39.18 | Lope | 's are dropped. |
| 14:39.25 | Lope | But what about all of the RTP ports? |
| 14:39.57 | Lope | is it an issue to let randoms send data to them? |
| 14:40.35 | Lope | I suppose there's always potential for an exploit |
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| 15:04.59 | Lope | whats the point of specifying localnet when a asterisk box is behind NAT? |
| 15:05.08 | Lope | so it knows what net the router is on etc? |
| 15:06.38 | file | so it knows when to put the external IP address in signaling |
| 15:08.19 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:08.53 | skyroveRR | Hey guys, I'm attempting to cross compile asterisk 13.1 certified for my raspberry pi using musl libc. So far I've been getting "configure: error: cannot run test program while cross compiling. See 'config.log' for more details". In the config.log, I just see that 'cannot run test program while cross compiling' error again. I tried using this patch: |
| 15:08.55 | skyroveRR | https://issues.asterisk.org/jira/secure/attachment/51783/asterisk-13-anonymous-semaphores.patch which can be found of this: https://issues.asterisk.org/jira/browse/ASTERISK-24663 however that patch didn't do anything.. any ideas? |
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| 15:09.20 | skyroveRR | I'm cross compiling in a x86 chroot btw. |
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| 15:18.09 | [TK]D-Fender | cert is a complete waste of time |
| 15:18.28 | [TK]D-Fender | You should not use it unless you're paying Digium for a support contract |
| 15:18.41 | skyroveRR | Hmm. |
| 15:33.01 | Lope | I've got my asterisk server behind nat with the port forwarded to it. If I test connecting to it with nc -z then I can connect successfully, and I can see the request and SYN response back etc in wireshark. |
| 15:33.27 | Lope | But when I try register my softphone, my softphone just sends a TON of register requests and NOTHING comes back... |
| 15:33.48 | Lope | oh, but I see the register requests are on UDP. nc -z is TCP. |
| 15:35.11 | Lope | yep, mistake in my FW |
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| 16:23.04 | Lope | after moving my asterisk install suddenly none of my sounds want to play |
| 16:23.20 | Lope | it says the file does not exist in any format, but it sure does :) |
| 16:24.00 | [TK]D-Fender | Not in a place that is readable to * using a format/codec module that loaded |
| 16:24.04 | [TK]D-Fender | It is NOT lying. |
| 16:24.17 | [TK]D-Fender | You have done something wrong. Time to look at the call & files properly |
| 16:24.32 | Lope | oh, ownership changed |
| 16:24.51 | [TK]D-Fender | Then there is also the qwuestion of the syntax of the app you're using |
| 16:26.46 | Lope | hmm |
| 16:30.55 | Lope | the only possible reason I can think of at the moment is the version change. |
| 16:31.08 | Lope | I had this setup working on asterisk 13.6. Now I've moved it to 13.1 |
| 16:31.22 | [TK]D-Fender | Versions don't magically change permissions |
| 16:31.27 | [TK]D-Fender | 13.1? |
| 16:31.34 | [TK]D-Fender | BACKWARDS? |
| 16:31.35 | Lope | what I'm doing with regard to playing sounds is I've got a directory symlink inside the /var/lib/asterisk/sounds dir |
| 16:31.35 | [TK]D-Fender | insane |
| 16:31.53 | Lope | then I set the language to the name of that symlink dir |
| 16:32.34 | Lope | then play something like "foo/bar" which will play bar, in the subdir of foo. foo being inside the language symlinked dir. |
| 16:33.00 | Lope | well, now I'm going to see if 13.1 is okay then I don't have to compile manually. |
| 16:33.00 | [TK]D-Fender | It will follow symlinks. |
| 16:33.11 | [TK]D-Fender | that is not an issue |
| 16:33.40 | Lope | okay. I've also temporarily given the asterisk user /bin/bash shell and su'd into asterisk and I can view the wav files etc in nano in the asterisk user. |
| 16:34.04 | Lope | File foo/bar does not exist in any format. |
| 16:34.15 | Lope | Maybe I don't have wav enabled or something... |
| 16:34.42 | [TK]D-Fender | Maybe you shouldn't be telling a story of how things aren't working and not show us |
| 16:35.11 | Lope | Well there would be a lot to show. |
| 16:35.12 | Lope | hehe |
| 16:35.25 | Lope | I'll scratch around for a bit, thanks. |
| 16:35.26 | [TK]D-Fender | nonsense |
| 16:35.32 | [TK]D-Fender | it's a dumb file |
| 16:35.41 | [TK]D-Fender | show us the call and the file |
| 16:36.03 | [TK]D-Fender | Exact app call & result and the file & folder |
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| 16:41.26 | [TK]D-Fender | heads out for a while.... |
| 16:58.03 | tompaw | I'm writing a documentation for our ARI-based dialer. The simplest scenario, i.e. setting up an outbound call, has 53 steps and spans across 3 separate contexts. |
| 16:58.23 | tompaw | How am I supposed to explain it to a new developer... |
| 17:03.16 | scv | tompaw why the heck is it so complicated? |
| 17:09.00 | Lope | I'm I've got a whole bunch of ast_db_put errors. How can I disable the asterisk DB? |
| 17:09.07 | Lope | or trying to interact with it? |
| 17:09.56 | scv | why are you trying to do that |
| 17:10.21 | scv | how about you fix the errors instead? |
| 17:12.58 | Lope | TKD: solved it. The old install had the sounds located at /var/lib/... the new one /usr/share/... |
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| 17:18.32 | Lope | scv: well I don't know what ast_db does. And I've never done anything with it before, so I don't know why I would need it? |
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| 17:25.55 | drmessano | ast_db is Asterisk's internal DB. You kinda need it |
| 17:29.17 | scv | yeah it's a vital component of asterisk. |
| 17:29.26 | scv | pastebin the errors and maybe we can help you? |
| 17:29.59 | Lope | WARNING[53]: db.c:332 ast_db_put: Couldn't execute statment: SQL logic error or missing database |
| 17:31.04 | Lope | When I start asterisk it says this also (which is a new error) tarting Asterisk PBX: Unable to install capabilities. |
| 17:31.14 | Lope | Starting Asterisk PBX: Unable to install capabilities. |
| 17:31.28 | Lope | but it seems to work aside from the ast_db warning. |
| 17:32.58 | Lope | Apparently that error is related to running asterisk inside a container. |
| 17:33.29 | Lope | I googled it and previously someone got the error running it in OpenVZ. I'm trying to run it in docker at the moment. |
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| 17:45.35 | Lope | wow wow wow!. running asterisk on my dedicated server now. halfway across the frikkin planet, and the audio quality is just beautiful. |
| 17:46.12 | Lope | so it turns out that shite quality I was having on digital ocean was due to too much congestion, whether it was CPU or network, whateve. |
| 17:46.21 | scv | it was probably cpu scheduling |
| 17:46.23 | scv | they oversell heavily |
| 17:46.28 | Lope | I bet. |
| 17:46.44 | Lope | I've never been as happy to hear myself :) |
| 17:46.45 | Lope | hahaha |
| 17:46.49 | scv | everybody always says "dont run asterisk in virtualization" but its fine as long as you ensure you aren't stealing cpu across guests |
| 17:47.19 | scv | 13:28 < Lope> WARNING[53]: db.c:332 ast_db_put: Couldn't execute statment: SQL logic error or missing database |
| 17:47.21 | Lope | hmm what do you mean by stealing CPU across guests? |
| 17:47.35 | scv | lope in 'top' or other monitoring tools you'll see a st% |
| 17:47.36 | Lope | you mean CPU overloaded, basically? |
| 17:47.36 | scv | steal % |
| 17:47.57 | scv | time the vm is paused to schedule other guests, introducing random latency to your guest |
| 17:48.08 | scv | if it's ever over 0% on a vm from a provider, they're overselling |
| 17:48.58 | Lope | Looking in HTOP, I don't see it |
| 17:49.10 | Lope | I've gone into setup where you choose the columns |
| 17:51.39 | Lope | any ideas about my ast_db_put issue? |
| 17:52.05 | scv | does it show anything else? |
| 17:52.12 | scv | where is your astdb currently stored |
| 17:52.32 | Lope | it's a standard 13.1 installation, I've only changed extensions.conf modules.conf, queues.conf musiconhold.conf, sip.conf, voicemail.conf |
| 17:53.04 | Lope | astdbdir => /var/lib/asterisk |
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| 17:53.35 | Lope | in that dir I see sqlite.db |
| 17:53.49 | Lope | lol however it's owned by root, and read only to other... |
| 17:53.57 | Lope | hehe same as astdb.sqlite3 lol |
| 17:54.01 | Lope | I'll chowm them asterisk. |
| 17:54.34 | tompaw | scv: cause it sets up a unique bridge, recording, crm data, callbacks, etc. |
| 17:54.57 | tompaw | It works perfectly fine, I'm just concerned about explaining it to people. |
| 17:54.59 | scv | tompaw: sounds like you need some detailed documentation then |
| 17:55.39 | tompaw | It allows our QAs to walk around with ipads and jump on agent's calls by specifying their cubicle number, etc. |
| 17:55.57 | tompaw | But damn, documenting this is going to be more difficult that writing it. |
| 17:56.07 | scv | some sort of interactive flowchart? :p |
| 17:56.12 | Lope | what the hell is this? hehe media_index_update: Failed to stat /usr/share/asterisk/sounds/snd/snd/snd/snd/snd/snd/snd/snd/snd/snd/snd/out/out/out/out/snd/snd/out/out/out/out/snd/out/snd/out/out/out: Too many levels of symbolic links |
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| 17:56.38 | scv | Lope: you have a recursive symlink |
| 17:56.42 | Lope | inside /usr/share/asterisk/sounds I only have one symlink that I've created |
| 17:56.51 | scv | you should use absolute symlinks, not relative ones |
| 17:56.51 | Lope | and it points to /data/snd/out |
| 17:57.02 | scv | you might have created the link in reverse accidentally |
| 17:57.14 | Lope | ohhh |
| 17:57.29 | Lope | looks like asterisk has gone and made symlinks inside my symlinked dir |
| 17:57.43 | Lope | one where snd points back to itself. |
| 17:57.51 | scv | that doesnt seem likely |
| 17:57.55 | Lope | or it was actually me trying to be clever with shell expansion. |
| 17:58.02 | Lope | (likely just me) haha |
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| 18:00.15 | Lope | looks like the ast_db error si gone buddy, thanks. |
| 18:01.34 | Lope | When I speak to myself using echo it sounds like crap. Lots of pops and crackles. But if I call and just listen to sounds it's 98% perfect. and when I call one softphone to another through my asterisk box the quality is also 98% perfect. |
| 18:04.02 | Lope | Okay nevermind about Echo. The problem was my softphone. I had "noise reduction" and "Automatic Gain" enabled. When I disabled both of those it became CRYSTAL clear. 100%. |
| 18:04.45 | Lope | That should also improve the quality of calls to my softphones. |
| 18:07.30 | scv | sounds like you should be pretty well off at this point |
| 18:07.44 | Lope | yeah, my bro |
| 18:07.53 | Lope | It's sleep time for me :) |
| 18:07.58 | scv | nn |
| 18:08.03 | Lope | night night, have a good one |
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