| 00:19.16 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.16 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:41.17 | *** join/#asterisk phunter (46a7da6e@gateway/web/freenode/ip.70.167.218.110) |
| 00:41.49 | phunter | Is this the right place to ask for help with AMI stuff? |
| 00:42.36 | phunter | I am trying to call a batch of numbers via NodeJS using Asterisk AMI, and it works, but the second stream of call audio doesn't start from the beginning. |
| 00:47.41 | phunter | Heading home, be back in an hour. |
| 01:46.46 | *** join/#asterisk hdon_ (~hdon@71-38-80-194.lsv2.qwest.net) |
| 02:22.59 | *** join/#asterisk jeffspeff (~Jeff@209.141.208.197) |
| 02:32.45 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-icxzmyhlueykdinm) |
| 03:05.31 | hdon_ | hi :) |
| 03:05.48 | hdon_ | so there's this "Remote UNIX Connection" message in my console a lot |
| 03:05.56 | hdon_ | now with freepbx i assumed it was some part of freepbx connecting all the time |
| 03:06.07 | hdon_ | but i'm running pure asterisk at home and now i'm seeing this message here, too |
| 03:14.28 | hdon_ | restarting my rasterisk seemed to make it go away |
| 03:21.14 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 03:22.19 | drmessano | hdon_: Do you have AMI open to the outside? |
| 04:02.29 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 04:03.57 | FarhaadN | in asterisk 14 , in asterisk -r , don't show any verbose or debug |
| 04:04.10 | FarhaadN | even with core set verbose 10 |
| 04:04.26 | FarhaadN | but in full log i have the log |
| 04:06.55 | [TK]D-Fender | Show us |
| 04:09.00 | FarhaadN | [TK]D-Fender: what? |
| 04:09.15 | [TK]D-Fender | SHOW US <- |
| 04:10.17 | FarhaadN | when i nothing too see , how can i send to u |
| 04:10.25 | [TK]D-Fender | You connectg at CLI |
| 04:10.33 | [TK]D-Fender | * gives OUTPUT |
| 04:10.37 | [TK]D-Fender | it TELLS you you connected. |
| 04:10.47 | [TK]D-Fender | Show us where you connect to CLI |
| 04:12.37 | Samot | And your logger.conf |
| 04:12.47 | Samot | Let's make sure you have it configured to show stuff in the console. |
| 04:13.05 | [TK]D-Fender | skip tha |
| 04:13.10 | FarhaadN | i conectet to cli |
| 04:13.13 | [TK]D-Fender | I should be seeing what you are doing NOW |
| 04:13.18 | Samot | Yes.. |
| 04:13.20 | Samot | That first. |
| 04:13.26 | FarhaadN | ok |
| 04:13.32 | FarhaadN | when explane to u |
| 04:13.35 | Samot | ~pb |
| 04:13.35 | infobot | hmm... pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:13.37 | FarhaadN | sry for my bad langouage |
| 04:13.45 | FarhaadN | i use asterisk for 5 years |
| 04:13.49 | [TK]D-Fender | PASTEBIN IT <----- Asterisk pumps out LINES including the COPYRIGHT |
| 04:13.55 | Samot | Yes. |
| 04:14.01 | FarhaadN | i recently update asterisk version 13 to 14.5 |
| 04:14.02 | Samot | Show the command and the cli connection... |
| 04:14.05 | Samot | Show it |
| 04:14.08 | FarhaadN | ok |
| 04:14.10 | FarhaadN | wait |
| 04:15.13 | FarhaadN | https://pastebin.com/tMt9qsvz |
| 04:16.43 | [TK]D-Fender | So what calls did you place after that? |
| 04:17.29 | FarhaadN | after that , with any calls, register ... |
| 04:17.36 | FarhaadN | nothing show me |
| 04:17.55 | FarhaadN | but in this situation , in full log, i can see logs |
| 04:18.06 | [TK]D-Fender | I asked what calls you places afterwards |
| 04:18.08 | [TK]D-Fender | DETAILS <- |
| 04:18.39 | [TK]D-Fender | You showed connecting to CLI. Didi you place calls AFTER that? |
| 04:18.44 | FarhaadN | i still in the cli |
| 04:18.45 | [TK]D-Fender | what KIND of calls? |
| 04:18.53 | FarhaadN | and make a call |
| 04:19.13 | Samot | OK. So you made a call and that's all you saw on the console? |
| 04:19.17 | FarhaadN | internal calls |
| 04:19.19 | Samot | OK |
| 04:19.20 | [TK]D-Fender | ----> WHAT KIND OF CALL <---- |
| 04:19.23 | FarhaadN | and inbound call |
| 04:19.32 | [TK]D-Fender | ----> WHAT KIND OF CALL <---- |
| 04:19.34 | FarhaadN | Samot: yes |
| 04:19.48 | FarhaadN | [TK]D-Fender: what do you mean? any call i try |
| 04:19.48 | drmessano | Do it again |
| 04:19.58 | [TK]D-Fender | WHAT FUCKING KIND OF CALL |
| 04:20.02 | [TK]D-Fender | PROTOCOL |
| 04:20.08 | [TK]D-Fender | TECHNOLOGY |
| 04:20.09 | [TK]D-Fender | DEVICE |
| 04:20.17 | [TK]D-Fender | what KIND of call? |
| 04:20.21 | [TK]D-Fender | what CHANNEL DRIVER?\ |
| 04:20.31 | FarhaadN | drmessano: i try many times , compile asterisk many time |
| 04:20.37 | Samot | FFS |
| 04:20.39 | drmessano | MAKE A CALL |
| 04:20.40 | Samot | Show your logger.conf |
| 04:20.41 | FarhaadN | [TK]D-Fender: sip calls |
| 04:20.50 | FarhaadN | with new rock device gw |
| 04:20.52 | [TK]D-Fender | Where are you enabling SIP DEBUG in there? |
| 04:20.55 | drmessano | lol |
| 04:20.57 | [TK]D-Fender | You did NOT enable it |
| 04:21.06 | [TK]D-Fender | I have NO proof you are even looking at the right box |
| 04:21.36 | FarhaadN | https://pastebin.com/HZeqgf72 > logger.conf |
| 04:21.42 | FarhaadN | in full log i can see log |
| 04:21.47 | Samot | Huh |
| 04:21.59 | FarhaadN | [TK]D-Fender: with core set debug 10 |
| 04:22.06 | FarhaadN | still i cant show anything |
| 04:22.06 | Samot | 12:12:38 AM <Samot> And your logger.conf |
| 04:22.06 | Samot | 12:12:48 AM <Samot> Let's make sure you have it configured to show stuff in the console. |
| 04:22.10 | Samot | Yes |
| 04:22.14 | Samot | Because you don't have it configured to |
| 04:22.20 | Samot | Like I said 10 minutes ago |
| 04:22.39 | [TK]D-Fender | <FarhaadN> [TK]D-Fender: with core set debug 10 <- that is NOT SIP DEBUG |
| 04:22.41 | FarhaadN | i dont change anything of asterisk setting from my update |
| 04:22.46 | Samot | Guys |
| 04:22.54 | Samot | It's f'ing commented out in the logger.conf |
| 04:22.56 | [TK]D-Fender | https://pastebin.com/tMt9qsvz <- NO CORE DEBUG THERE |
| 04:22.57 | Samot | It's not happening |
| 04:23.04 | Samot | OK. |
| 04:23.31 | [TK]D-Fender | ;console => notice,warning,error,debug <- yup, NOTHING |
| 04:23.42 | Samot | Why I said check that... |
| 04:23.44 | Samot | 10 minutes ago |
| 04:24.18 | FarhaadN | tanx u alllll |
| 04:24.31 | FarhaadN | tanx all of u |
| 04:24.58 | Samot | Np. |
| 04:26.04 | drmessano | np |
| 04:26.16 | FarhaadN | O:-) |
| 04:27.00 | FarhaadN | i have question |
| 04:27.10 | FarhaadN | with this configuration |
| 04:27.19 | drmessano | hands FarhaadN some batteries |
| 04:27.19 | FarhaadN | i dont have any problem with asterisk 11 or 13 |
| 04:27.57 | FarhaadN | and in asterisk -r |
| 04:28.04 | FarhaadN | when i change verbose to 10 |
| 04:28.29 | Samot | It's possible it overwrote your .conf when you compiled 14 |
| 04:28.31 | Samot | Don't know. |
| 04:28.35 | FarhaadN | Should not work? |
| 04:28.39 | Samot | No. |
| 04:28.48 | Samot | If you have console => commented out |
| 04:28.55 | Samot | It means don't show _anything_ there |
| 04:29.01 | [TK]D-Fender | [TK]D-Fender> ;console => notice,warning,error,debug <- yup, NOTHING |
| 04:29.11 | [TK]D-Fender | You have multiple lines, al commented |
| 04:29.13 | drmessano | I thought we just solved this |
| 04:29.14 | [TK]D-Fender | all |
| 04:29.21 | FarhaadN | [TK]D-Fender: its done |
| 04:29.26 | Samot | He wants to know why in 14 it changed. |
| 04:29.30 | [TK]D-Fender | So no, this config is a failure for all versions |
| 04:29.34 | Samot | I'm guessing it overwrote his other logger.conf file |
| 04:29.43 | FarhaadN | i have a old version |
| 04:29.50 | FarhaadN | in another server |
| 04:29.54 | Samot | Compare them then. |
| 04:29.57 | FarhaadN | and i check that configuration |
| 04:30.02 | FarhaadN | and configuration as same |
| 04:30.18 | Samot | Then perhaps it was a bug that was fixed in 14 |
| 04:30.19 | FarhaadN | and in another server work fine |
| 04:30.20 | Samot | And there you go. |
| 04:30.28 | drmessano | FarhaadN: There may have been defaults and now theres not |
| 04:30.32 | Samot | If console => is commented out, it shouldn't work |
| 04:30.36 | Samot | If it did, then it was a bug |
| 04:30.37 | drmessano | or the default is nothing |
| 04:31.00 | FarhaadN | tank you for informations |
| 04:31.06 | drmessano | np |
| 04:41.49 | *** part/#asterisk FarhaadN (~Farhad@82.99.206.194) |
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| 07:31.51 | FarhaadN | i install Wildcard AEX2400 24-port analog card |
| 07:32.13 | FarhaadN | and configure chan_dahdi.conf in asterisk |
| 07:32.25 | FarhaadN | but no incoming call |
| 07:32.29 | FarhaadN | what can i do? |
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| 07:32.41 | drmessano | Read and fix it |
| 07:32.51 | drmessano | Your config is wrong |
| 07:33.20 | FarhaadN | i read manual , and exacly config file with that |
| 07:38.17 | FarhaadN | drmessano: how can i trubelshootin analog line? |
| 07:38.32 | FarhaadN | in asterisk -r > dahdi show channels |
| 07:38.36 | FarhaadN | show me channels |
| 07:38.49 | FarhaadN | i think configuration is ok |
| 07:39.46 | drmessano | Pastebin all of this |
| 07:39.53 | drmessano | Show a failed call |
| 07:39.53 | FarhaadN | ok |
| 07:39.56 | drmessano | Show the configs |
| 07:39.59 | drmessano | You know the drill |
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| 07:48.47 | drmessano | FarhaadN: Today |
| 07:51.40 | FarhaadN | https://pastebin.com/gJEW4e9h |
| 07:51.51 | FarhaadN | sry for delay |
| 07:53.50 | FarhaadN | and in asterisk -r , core set verbose 10 , core set debug 10 , and nothing show me |
| 07:57.09 | drmessano | This is 2 cards? |
| 07:57.59 | FarhaadN | yes |
| 07:58.10 | FarhaadN | one e1 and one analog 24 port |
| 08:13.30 | FarhaadN | :-( |
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| 08:21.15 | lorsungcu | FarhaadN: what does asterisk -cvvvvvvvvv return |
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| 08:30.03 | FarhaadN | lorsungcu: nothing |
| 08:30.35 | lorsungcu | What do you mean |
| 08:30.50 | lorsungcu | Pastebin exactly what you see, including the command itself |
| 08:31.28 | lorsungcu | I've got about 5 minutes before I'm asleep |
| 08:35.25 | lorsungcu | Alright. Going to sleep. |
| 08:35.37 | FarhaadN | i told u |
| 08:35.56 | FarhaadN | asterisk dont show anyhthing |
| 08:36.10 | FarhaadN | full log dont show anything |
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| 08:47.47 | FarhaadN | ok i ask you tommorow |
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| 09:13.06 | Rac-on | FarhaadN: no way that 'asterisk -cvvvvvvvvv' or 'asterisk -crvvvvvvvvv' doesnt return anything |
| 09:21.52 | FarhaadN | Rac-on: dont show anythings |
| 09:22.08 | FarhaadN | you read configuration? |
| 09:22.22 | FarhaadN | is that correct? |
| 09:26.01 | Martin` | Ah I was thinking I need to change config for enabling the BLF on the phones, but I only need to enable it on the phones :D |
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| 12:26.34 | wyoung | o/ |
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| 12:33.37 | [TK]D-Fender | FarhaadN, Show use that chan_dahdi is even loaded... |
| 12:34.51 | wyoung | hai [TK]D-Fender! long time |
| 12:44.08 | tzafrir | FarhaadN, what is the output of lsdahdi #? |
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| 16:04.53 | hdon_ | hi all :) OT question: so i've noticed that my telephone hooked up to my spa2102 rings a little funny when on hook. the first ring sounds fine, but the second ring kind of stutters. any idea which settings i should be looking at? |
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| 16:20.00 | hdon_ | i think it might just be the physical phone, but who can say |
| 16:20.16 | hdon_ | i don't know anything about diagnosing an old telephone bell |
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| 16:29.27 | TandyUK | hdon_: in the spa you configure how the ring sounds |
| 16:30.03 | TandyUK | eg, for a 'normal' UK ring, it would be 350@-19,440@-22;10(*/0/1+2) |
| 16:31.01 | hdon_ | TandyUK: thanks. i'm looking at it more closely though and it looks like the mechanism's timing may just be a little off. |
| 16:31.57 | TandyUK | check a different phone :) |
| 16:32.56 | hdon_ | TandyUK: i plan to, but unless i want to cut my only cat1/rj11 cable, i won't be able to until i get some more (the other two telephones i have do not have rj11 connectors or ports) |
| 16:34.22 | hdon_ | hmm... how do i get asterisk to tell the spa2102 to use a distinctive ring when i call.. |
| 16:34.54 | hdon_ | indications.conf seems to be about this |
| 16:36.28 | hdon_ | maybe |
| 16:36.37 | hdon_ | it looks like it's specifying tones in this file |
| 16:36.48 | hdon_ | which leads me to believe that these are waveforms synthesized to the rtp stream |
| 16:37.18 | hdon_ | but for a sip phone or sip/pots adapter, i assume it's just a sip flag that tells it what class of distinctive ring to use |
| 16:38.31 | hdon_ | oh i guess for certain channel types where asterisk is very intimate with the FXO signal then it would be responsible for generating the ring signal to a pots phone |
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| 16:48.42 | [TK]D-Fender | that has nothing to do with how your phone sounds while ringing on-hook |
| 16:52.01 | salviadud | [TK]D-Fender, I have cisco phones that are missing "native" tones, and asterisk doesn't have problems finding out what's going on. That might matter on dahdi if you are connecting to the pstn. |
| 16:55.51 | [TK]D-Fender | huh? |
| 16:56.36 | [TK]D-Fender | rephrase that... |
| 17:01.16 | salviadud | What I mean is, my cisco phones using sip generate tones for different countries, in my case, I don't use the native mexican tones on sip. But I do have dahdi setup to my country. |
| 17:02.09 | salviadud | So, asterisk doesn't really care what type of tone I use on the sip chan (I think) |
| 17:08.07 | [TK]D-Fender | indications.conf = INBOAND ONLY |
| 17:08.18 | [TK]D-Fender | saying "SIP" alone without context means nothing. |
| 17:08.47 | [TK]D-Fender | If You place a call and the call is rteated as UNANASWERED and the phone is told to right, the PHONE is what generates tones based on ITS rules. |
| 17:09.06 | [TK]D-Fender | If you DID t4reat the call as answered then the indications are INBAND AUDIO generated by * |
| 17:09.16 | [TK]D-Fender | that's just the cALLING leg |
| 17:09.52 | [TK]D-Fender | if the CALLED leg is treated as answered they could be sending YOU ringing while they are technically possibly OOB themselves which is the REMOTE end. |
| 17:11.33 | salviadud | Mmm, would that explain the fact that my sip phones never return a busy signal? |
| 17:11.42 | salviadud | Or is that some sip feature that you can have multiple calls? |
| 17:12.29 | [TK]D-Fender | The answer is LOOK AT YOUR CALLS |
| 17:12.36 | [TK]D-Fender | A lot of things "could be". |
| 17:12.48 | salviadud | Yeah, you're right, I might need to debug it. |
| 17:13.02 | salviadud | Take a closer look. |
| 17:13.09 | [TK]D-Fender | <salviadud> Mmm, would that explain the fact that my sip phones never return a busy signal? <- can you see another call come in when you're on the phone? |
| 17:13.22 | [TK]D-Fender | Ever heard of "call waiting" before? |
| 17:13.51 | salviadud | Definetely |
| 17:14.14 | salviadud | And yes, calls come in and I can place the active call on hold, and answer another one. |
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| 17:15.13 | salviadud | So, it would be a nice trick to disallow that and throw out some busy/congestion signals. |
| 17:15.55 | salviadud | I've only been able to do that with DND. |
| 17:18.28 | [TK]D-Fender | Then stop buying phones that support lots of calls |
| 17:18.35 | salviadud | lol |
| 17:18.53 | salviadud | That's a concrete solution right there. |
| 17:18.58 | [TK]D-Fender | And you can go back to living like it's the early 1900's |
| 17:19.51 | salviadud | I'm happy with my setup, just wondering about the inner workings of * and whatnot. |
| 17:21.28 | [TK]D-Fender | Wondering is what you do when you should be looking. |
| 17:25.00 | salviadud | Well! I've had no problems so far, I might take a closer look just to satisfy my curiosity. |
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| 18:05.33 | drmessano | hdon_: Does your phone have a mechanical bell? |
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| 19:10.15 | samwierema | Is it possible to configure Asterisk (when setting up calls) to not do the RTP (instead rely on suppliers in/out)? |
| 19:10.31 | samwierema | direct_media = yes? (using pjsip.conf) |
| 19:11.24 | Samot | You can but it's not highly recommended. |
| 19:12.08 | samwierema | Samot: ok, there's downsides to it? |
| 19:12.37 | Samot | Of course. |
| 19:13.29 | samwierema | How would I set it up (so I can test)? |
| 19:13.39 | Samot | As well, Asterisk really never leaves the media path it just "proxies" it. Asterisk will get in the media path when required. |
| 19:14.04 | Samot | You would use the setting you just asked about. |
| 19:14.31 | samwierema | Ok |
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| 19:38.28 | skirmisha | guys.. i am back |
| 19:38.40 | skirmisha | back to yesterday issue |
| 19:39.21 | skirmisha | according to my vendor if record-route contains lr parameter at the end, that means UAC must use contact header according to rfc3261 |
| 19:39.32 | skirmisha | do you think that make any sense? i don't think so |
| 19:42.17 | skirmisha | loose route |
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| 21:40.40 | hiyo | Hello, can someone tell me how to have clearer internal calls (like when 2 clients are using CSipSimple or other capable softphone)? |
| 21:43.04 | igcewieling | Hardphones improve audio quality. |
| 21:43.44 | hiyo | igcewieling: Well I don't have any of those sadly. Just trying to make the best of what I have at the moment |
| 21:44.18 | igcewieling | good luck. |
| 21:46.02 | drmessano | hiyo: Which codec are you using? |
| 22:19.55 | hiyo | drmessano: right now its defaulting to PCMU |
| 22:23.09 | lorsungcu | hiyo: what constitutes "better" |
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| 22:33.03 | drmessano | hiyo: Thats as good as it gets |
| 22:43.04 | [TK]D-Fender | no, it can be better |
| 22:43.28 | [TK]D-Fender | but we haven't qualified what factor is being judged in this definition of "poor" |
| 22:48.44 | Samot | Well the term "poor" wasn't used. It was asked how to have "clearer" internal calls. Could be transcoding since most people never muck with the codec settings in softphones and they could be out of order or not matching and transcoding can be happening... |
| 22:48.54 | Samot | Or it's using g711 and wants it to be clearer. |
| 22:49.41 | Samot | So I guess we should get a definition of "clearer" instead of "poor". |
| 22:50.00 | Samot | hiyo: Can you please clarify what you mean by "clearer" calls? |
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| 23:25.01 | drmessano | They're using CSipSimple |
| 23:25.34 | drmessano | 17:43:05 <igcewieling> Hardphones improve audio quality. |
| 23:25.34 | drmessano | 17:43:45 <hiyo> igcewieling: Well I don't have any of those sadly. Just trying to make the best of what I have at the moment |
| 23:25.38 | drmessano | The context is clear |
| 23:26.38 | [TK]D-Fender | Depends on expectations and actual conditions |
| 23:26.52 | drmessano | So what variable would you change? |
| 23:26.53 | [TK]D-Fender | is his line having jitt/PL? We don't know |
| 23:27.09 | [TK]D-Fender | Is it that the packets are constant but the hardware is pretty bad on top of that? |
| 23:27.27 | [TK]D-Fender | Is the hardware fine and his expectations higher than G.711? |
| 23:27.31 | [TK]D-Fender | We don't know any of this |
| 23:28.16 | hiyo | okay I didn't know this would start a debate lol, I just thought that the call quality could be improved (since my phone will sound better when I make a call to someone else on the same carrier, but that's with my actual phone service) |
| 23:28.53 | drmessano | Is it the overall audio quality (tone, clarity) or are there issues like dropouts? |
| 23:29.31 | hiyo | there aren't dropouts, yeah its like the clarity of the calls |
| 23:29.59 | drmessano | Well, you can maybe try a different (PAID) softphone, like Bria.. But overall, not so much |
| 23:30.28 | drmessano | You're already using a non-shitty codec |
| 23:30.38 | drmessano | G722 from Handset to Handset MAY sound better |
| 23:30.54 | drmessano | If these are softphone to softphone |
| 23:31.01 | hiyo | Oh I didn't think PCMU was that great, but I know nothing of codecs |
| 23:31.09 | drmessano | But softphone <> PSTN is still going to get downsampled to G711 |
| 23:31.30 | drmessano | Well as I said |
| 23:31.30 | drmessano | G722 is better |
| 23:31.30 | drmessano | If these are softphone to softphone |
| 23:31.30 | drmessano | But softphone <> PSTN is still going to get downsampled to G711 |
| 23:31.52 | hiyo | Ah thank you for the info |
| 23:32.09 | hiyo | Well I must get going now! |
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| 23:32.32 | [TK]D-Fender | <drmessano> G722 from Handset to Handset MAY sound better <- I can't imagine any time it shouldn't |
| 23:32.51 | drmessano | Because that variable is dependent on implementation and hardware |
| 23:33.09 | drmessano | The hardware may just be lacking |
| 23:33.19 | drmessano | Slapping G722 on it wont fix that |
| 23:33.31 | drmessano | So I can imagine that being a problem |
| 23:47.32 | lorsungcu | [TK]D-Fender: any time there's latency issues |
| 23:47.52 | lorsungcu | Any time there's bandwidth constraints |