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01:54.17 | elvisthedj | Hello - Is it possible to call an agi from within an agi? e.g., I have a script (perl) and I need to use google's asr and would rather not have to integrate the speech recognition into my existing, but simply call it. |
01:54.52 | Samot | Why would you call an AGI from an AGI? |
01:55.18 | Samot | You're using AGI to execute a script that does dialplan functions... |
01:55.35 | Samot | If you're already calling a file via AGI, call the other file via the main file called. |
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01:58.44 | elvisthedj | I have an IVR that uses tts and voice recognition as opposed to DTMF's. My menu.agi that I wrote used googles old api and there is a new working perl agi already written that I'd like to use instead of rewriting my script. My agi now will prompt a user and then go to a sub &getinput() or whatever... My subs don't work with the current api, so instead of rewriting the routines, I'd just like to call |
01:58.50 | elvisthedj | to the already written agi from gethub (also in perl) |
01:59.51 | Samot | If you want to call on other scripts via AGI you either need to have AGI calls for them all.. |
02:00.06 | Samot | Or call a single master file that includes/calls on the other files. |
02:01.16 | elvisthedj | gotcha.. I'm not a perl guru, but I'm guessing I'll just #include the file and then do something->like->this. I'll stop googling for agi help and start googling for perl help :) |
02:01.21 | elvisthedj | thank you, sir |
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07:18.41 | LiuYan | using asterisk-java to GetVar in HangupEvent (), I can't get the variable value in this event. But I can get value in NewState event while State is Ringing. Does variables get cleaned in Hangup? |
07:19.34 | LiuYan | I tried put '_' or '__' infront of variable name, but I got same result. |
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08:19.07 | _8bits | exten => _out-X.,3,Set(CALLERID(all)=${IF($[${CALLERID(all)} = 37037]?370371234:${CALLERID(all)})}) |
08:19.10 | _8bits | whis this is not working? |
08:19.35 | _8bits | [Oct 20 11:15:37] WARNING[49818][C-00005fdd]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting '-' or '!' or '(' or '<token>'; Input: |
08:19.35 | _8bits | "" <3705489465> = 37037 |
08:19.54 | *** join/#asterisk UOL (~king@111.68.105.235) |
08:19.56 | UOL | hello |
08:22.59 | _8bits | <_8bits> exten => _out-X.,3,Set(CALLERID(all)=${IF($[${CALLERID(all)} = 37037]?370371234:${CALLERID(all)})}) |
08:23.00 | _8bits | <_8bits> whis this is not working? |
08:23.00 | _8bits | <_8bits> [Oct 20 11:15:37] WARNING[49818][C-00005fdd]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting '-' or '!' or '(' or '<token>'; Input: |
08:23.00 | _8bits | <_8bits> "" <3705489465> = 37037 |
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08:59.15 | UOL | sip trunk unreachable |
08:59.22 | UOL | sometimes its reachable |
08:59.35 | UOL | after some mins its shows un-reachable |
09:12.25 | UOL | Received response: "Forbidden" from |
09:16.38 | TandyUK | the other end doesnt like you for some reason |
09:17.34 | UOL | the other end is VOIP providers |
09:17.37 | UOL | sip providers |
09:18.03 | UOL | no restriction from their side |
09:18.34 | UOL | Received response: "Forbidden" from '<sip:924232560865lbo@10 |
09:18.45 | UOL | 924232560865lbo is my providers username |
09:18.59 | TandyUK | right so youre being told "forbidden" |
09:19.06 | UOL | and am trying to call with extension 7001 |
09:19.09 | TandyUK | ask provider why they are giving you a forbidden response |
09:19.48 | TandyUK | or do a trace |
09:19.54 | UOL | how? |
09:19.59 | TandyUK | analyse the sip packets and what they contain |
09:20.08 | TandyUK | wireshark / tcpdump |
09:20.16 | UOL | method pleas |
09:20.21 | UOL | am totally new in asterisk |
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09:21.07 | TandyUK | or i nasterisk, sip set debug on (for everything) or sip set debug peer <extension or ip> to limit to just one peer |
09:21.32 | TandyUK | of if newer asterisk, pjsip ...... |
09:26.28 | UOL | if i create pjsip trunk |
09:26.40 | UOL | how can i check its reachable or not? |
09:33.03 | *** join/#asterisk Phil-Work (~Phil-Work@office.cs.lei.uk.cloudcall.com) |
09:34.02 | Phil-Work | asterisk Ringing() seems to time out "Spawn extension exited non-zero" after 180 seconds |
09:34.07 | Phil-Work | any idea how to raise this value? |
09:50.15 | wdoekes | _8bits: you'll want to test CALLERID(num), not CALLERID(all) |
09:50.50 | wdoekes | _8bits: the expression parser gets this: '$[' '"" <3705489465> = 37037' ']' |
09:51.04 | wdoekes | when you want it to get this: '$[' 3705489465 = 37037' ']' |
09:51.21 | wdoekes | er: '$[' '3705489465 = 37037' ']' |
09:55.43 | UOL | [2017-10-19 08:36:13] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (35ms / 2000ms) |
09:55.44 | UOL | [2017-10-19 08:38:17] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 35 |
09:55.45 | UOL | [2017-10-19 08:46:09] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (39ms / 2000ms) |
09:55.46 | UOL | [2017-10-19 08:47:13] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 39 |
09:55.47 | UOL | [2017-10-19 08:56:15] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (40ms / 2000ms) |
09:55.48 | UOL | [2017-10-19 08:59:19] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 37 |
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10:14.46 | UOL | how to fix it |
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10:42.53 | pchero_work | Seems like firewall, network or sip device problem. |
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11:07.50 | sibiria | UOL: do you happen to have high packet loss to that "trunk"? |
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11:22.16 | _8bits | try to ping to that trunk ip |
11:22.27 | _8bits | to check if it is online when it goes unreachable |
11:30.33 | UOL | firewall is off |
11:30.52 | UOL | i dont know about packets |
11:30.57 | UOL | how can i check them? |
11:33.13 | pawiecki | UOL: I missed your post, what's the problem? |
11:33.45 | UOL | NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (36ms / 2000ms) |
11:33.58 | UOL | chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 38 |
11:34.46 | UOL | that's the issue |
11:40.23 | pawiecki | UOL: It's a SIP Trunk, right? Is it a remote host? Is your internet connection stable? |
11:41.50 | file | providing a log with "sip set debug on" will show if the remote is actually not responding or not |
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11:46.49 | UOL | yes its a sip trunk with voip providers |
11:48.07 | UOL | i can ping voip providers ip, proxy , gateway from my asterisk machine which i created on vmware |
11:49.25 | UOL | my sip trunk settings are |
11:49.26 | UOL | type=peer |
11:49.26 | UOL | domain=multinet.pri |
11:49.26 | UOL | fromdomain=10.99.34.216:5083 |
11:49.26 | UOL | username=xxxxxxxxxxxxxxxx |
11:49.26 | UOL | fromuser=xxxxxxxxxxxxxxxx |
11:49.26 | UOL | secret=xxxxxxxxxxxxxxxx |
11:49.27 | UOL | host=125.209.93.196 |
11:49.28 | UOL | port=5083,5060, 5160 |
11:49.29 | UOL | insecure=port,invite |
11:49.29 | UOL | outboundproxy=125.209.93.196:5083 |
11:49.29 | UOL | disallow=all |
11:49.30 | UOL | allow=ulaw,alaw,gsm |
11:49.30 | UOL | qualify=yes |
11:49.48 | pawiecki | UOL: as file mentioned above, set debug on that trunk and show us what's happening when trunk goes unreachable |
11:53.08 | UOL | how to on the debug ¿ |
11:53.14 | UOL | sip set debug on ¿ |
11:53.19 | UOL | that command? |
11:53.49 | pawiecki | UOL: yes, and you should know that if you administer asterisk server. |
11:54.15 | pawiecki | uhh I mean no, set the debug only on that trunk |
11:54.19 | UOL | am new for asterisk |
11:56.04 | UOL | sip set debug ip 125.209.93.196:5083 peer multinet-outgoing |
11:56.07 | UOL | am trying this |
11:56.09 | UOL | but failed |
11:56.26 | pawiecki | 'sip set debug peer multinet-outgoing' should be ok |
11:56.29 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
11:57.11 | UOL | Retransmitting #2 (no NAT) to 125.209.93.196:5083: |
11:57.12 | UOL | OPTIONS sip:125.209.93.196 SIP/2.0 |
11:57.12 | UOL | Via: SIP/2.0/UDP 10.99.34.216:5060;branch=z9hG4bK24264759 |
11:57.12 | UOL | Max-Forwards: 70 |
11:57.12 | UOL | From: "Unknown" <sip:924232560865lbo@10.99.34.216>;tag=as17ae315f |
11:57.13 | UOL | To: <sip:125.209.93.196> |
11:57.13 | UOL | Contact: <sip:924232560865lbo@10.99.34.216:5060> |
11:57.13 | UOL | Call-ID: 33a354ff2aa6d7e32d125cf07249ecd7@10.99.34.216:5060 |
11:57.15 | UOL | CSeq: 102 OPTIONS |
11:57.16 | UOL | User-Agent: FPBX-13.0.192.19(13.12.1) |
11:57.16 | UOL | Date: Fri, 20 Oct 2017 11:57:00 GMT |
11:57.16 | UOL | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE |
11:57.17 | UOL | Supported: replaces, timer |
11:57.18 | UOL | Content-Length: 0 |
11:57.25 | UOL | 5060? |
11:57.38 | UOL | i have set this to 5083 |
11:57.45 | UOL | then why 5060 |
11:57.50 | UOL | how can i change it? |
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11:58.37 | pchero_work | How did you set to 5083? |
11:59.19 | pchero_work | type sip show settings |
11:59.32 | UOL | fromdomain=10.99.34.216:5083 |
11:59.37 | UOL | like this |
12:00.05 | pchero_work | type sip show setting |
12:00.18 | file | those options would not have an impact on getting a response to that OPTIONS |
12:00.32 | file | Asterisk is sending it out and not getting a response back, but only sometimes |
12:00.47 | UOL | Global Settings: |
12:00.48 | UOL | ---------------- |
12:00.48 | UOL | UDP Bindaddress: 0.0.0.0:5060 |
12:00.48 | UOL | TCP SIP Bindaddress: Disabled |
12:00.48 | UOL | TLS SIP Bindaddress: Disabled |
12:00.48 | UOL | Videosupport: No |
12:00.48 | UOL | Textsupport: No |
12:00.50 | UOL | Ignore SDP sess. ver.: No |
12:00.51 | UOL | AutoCreate Peer: Off |
12:00.51 | UOL | Match Auth Username: No |
12:00.51 | UOL | Allow unknown access: Yes |
12:00.51 | UOL | Allow subscriptions: Yes |
12:00.52 | UOL | Allow overlap dialing: Yes |
12:00.53 | UOL | Allow promisc. redir: No |
12:00.54 | UOL | Enable call counters: No |
12:00.54 | UOL | SIP domain support: No |
12:00.54 | UOL | Path support : No |
12:01.03 | pchero_work | Okay.. |
12:01.04 | file | and use something like pastebin.net instead of copy/pasting tons of output into here |
12:01.16 | UOL | opss |
12:01.17 | UOL | ok |
12:01.20 | UOL | sorry |
12:02.00 | pawiecki | file: hmm i'm not sure, but is that setting valid: "port=5083,5060, 5160" ? |
12:02.12 | file | it's not |
12:02.19 | file | it parsed it as '5083' |
12:03.37 | pawiecki | UOL: correct that setting - set a single port for SIP commnication. |
12:04.01 | UOL | ok |
12:04.10 | UOL | well |
12:04.13 | UOL | now am trying to call |
12:04.18 | UOL | but not happening |
12:04.26 | UOL | nor any error on debug mode |
12:05.14 | pawiecki | UOL: maybe, let's start from the begining - Did you manage to setup your SIP Trunk that you could call in and out? |
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12:06.25 | UOL | yes |
12:06.28 | UOL | i created |
12:06.54 | pawiecki | UOL: no private messages please. |
12:09.48 | UOL | okey |
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12:17.39 | UOL | pawiecki |
12:17.39 | UOL | https://pastebin.com/FrFPkVCn |
12:17.46 | UOL | trunk is ok now.. but calling issue |
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12:21.05 | pawiecki | UOL: did you get instructions on how to configure trunk to your ITSP? |
12:22.13 | *** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
12:24.27 | UOL | i configured the trunk |
12:24.43 | UOL | ultinet-outgoing/9242325 125.209.93.196 No No 5083 OK (18 ms) |
12:24.50 | UOL | trunk is reachable now |
12:25.01 | UOL | 2017-10-20 17:24:05] WARNING[1948][C-00000067]: chan_sip.c:23862 handle_response_invite: Received response: "Forbidden" from '<sip:924232560865lbo@10.99.34.216:5083>;tag=as2d9ea1c8' |
12:25.02 | UOL | Scheduling destruction of SIP dialog '597bb423575ff82c4049f1d462cbf847@10.99.34.216' in 6400 ms (Method: INVITE) |
12:29.33 | pawiecki | UOL: Is '924232560865lbo' a valid number? |
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13:49.41 | *** mode/#asterisk [+o bford] by ChanServ |
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14:27.07 | FarhaadN | i have to elastix pbx , in one of them(elastix1) i have pstn card and config , i want to configure this elastix(elastix1) as a pstn gateway , and send call to other elastix(elastix2) , and calls from elastix2 out from pstn over elastix1 |
14:28.38 | FarhaadN | i create trunk in 2 pbx , incoming calls is ok |
14:28.46 | FarhaadN | but outgoing call not working |
14:28.59 | FarhaadN | chan_sip.c:24052 handle_response_invite: Failed to authenticate on INVITE to '"9998" <sip:9998@192.168.200.2>;tag=as0ef8cf3b' |
14:29.20 | FarhaadN | this error accourd |
14:29.32 | FarhaadN | anyone can help me? |
14:29.48 | Samot | There's no elastix support here. |
14:30.04 | Samot | 1) It's a GUI |
14:30.07 | Samot | 2) It's dead. |
14:30.43 | FarhaadN | i now |
14:30.52 | FarhaadN | i know |
14:31.03 | FarhaadN | but i need help |
14:31.07 | *** join/#asterisk RovingWriter (~RovingWri@unaffiliated/rovingwriter) |
14:31.09 | Samot | You always do. |
14:31.17 | Samot | You need to hire someone that knows this stuff. |
14:34.12 | FarhaadN | so u cant help? |
14:34.30 | Samot | IRC/Forums should not be the defacto support channel for voip providers. |
14:34.38 | Samot | I don't get how this has become of thing lately. |
14:34.50 | Samot | You have an auth failure. |
14:35.00 | Samot | It's like any other auth failure with a peer. |
14:35.03 | fauxalliance | ^^small time issue |
14:35.20 | Samot | This is like Asterisk 101/SIP 101. |
14:35.40 | fauxalliance | auth failures and one way audio... 80% of the requests. |
14:35.54 | fauxalliance | and filing a bug report will not help. |
14:36.07 | Samot | It's not even that. |
14:36.16 | fauxalliance | provider.. |
14:36.17 | fauxalliance | i know |
14:36.25 | Samot | It's the 80% of those request mainly come from 90% "tech support people" |
14:36.26 | FarhaadN | why authentication is need? |
14:36.31 | fauxalliance | commercial entities looking for _free_ handholding and spoonfeeding. |
14:36.43 | Samot | Because you didn't configure the system to not use auth. |
14:36.55 | [TK]D-Fender | Except when you do. |
14:36.58 | FarhaadN | this eroor accourd in elastix2 |
14:37.19 | Samot | Then start looking there. |
14:37.25 | [TK]D-Fender | FarhaadN, your SIP peers do not agree. |
14:37.30 | Samot | And get someone that knows Asterisk |
14:37.31 | [TK]D-Fender | you can't jsut look a one side |
14:37.36 | FarhaadN | i just need outcall from elastix2 to elastix1 |
14:37.41 | Samot | OK |
14:37.52 | Samot | You just need to fix the auth issue for that to happen. |
14:38.07 | Samot | This isn't the first time we've had to help you peer two PBX systems. |
14:38.20 | Samot | I'm not sure why you need that help more than one or two times. |
14:38.41 | Samot | You should have taken notes, remembered some of the things we've walked you through. |
14:38.55 | FarhaadN | [TK]D-Fender : i config every trunk with this doc : https://www.elastix.org/docs/elastix-pstn-card-gateway-3cx/ |
14:39.33 | FarhaadN | Samot: another time, i was call 2 extentions in 2 pbx |
14:39.41 | FarhaadN | i want* |
14:39.51 | FarhaadN | and use iax peer |
14:39.56 | Samot | Doesn't matter. |
14:40.00 | [TK]D-Fender | That guide has some stupid stuff in there |
14:40.11 | Samot | Well that guide is also for the other side being 3CX |
14:40.30 | Samot | So completely irrelevant if you don't understand that core concepts to begin with. |
14:40.51 | FarhaadN | Samot: i just config elastix side from doc |
14:40.53 | Samot | So a lack of knowledge and a guide that isn't even for this type of setup. |
14:41.04 | Samot | You need to configure the OTHERSIDE PROPERLY |
14:41.28 | Samot | You can't just take a doc for A and Z and then make two A's that will connect. |
14:41.49 | FarhaadN | no, config two side |
14:41.51 | Samot | Specially when Z isn't the same as A so how A connects to Z would not be how A connects to another A |
14:41.58 | FarhaadN | but only incoming work |
14:42.14 | Samot | Asterisk <--> 3CX NOT THE SAME AS Asterisk <--> Asterisk |
14:42.21 | Samot | They are two DIFFERENT PBX systems. |
14:42.30 | Samot | They handle crap completely different. |
14:42.47 | Samot | How you send a call from Asterisk to a 3CX peer is not going to be the same as Asterisk to Asterisk. |
14:43.01 | Samot | There will be differences. |
14:43.28 | [TK]D-Fender | Depending how the * end was intended to work work in that first scenario |
14:43.43 | davidbowlby | feels for Samot |
14:43.46 | davidbowlby | lol |
14:44.04 | FarhaadN | wait i screen shot that |
14:44.40 | davidbowlby | Have you tried turning it off and back on again? |
14:45.12 | Samot | Again.. |
14:45.29 | davidbowlby | lol |
14:45.33 | Samot | This is basically "Be my support team" for someone working at a VoIP provider. |
14:45.50 | Samot | Some provider is paying FarhaadN to be a support tech. |
14:46.00 | davidbowlby | I'm sitting here writing my library to talk to AMI and this mofo is using you guys to support his platform |
14:46.00 | Samot | Who ends up in here for about 90% of what needs to be handled. |
14:46.48 | Samot | I'm writing an Asterisk/FreePBX module for WHMCS right now. |
14:46.54 | Samot | So yeah, I get that feeling. |
14:47.08 | [TK]D-Fender | If he can't deal with a basic SIP peer setup between 2 servers he shouldn't be administering a PBX for anyone, let alone a company providing those services to others |
14:47.17 | Samot | Yup. |
14:47.23 | Samot | But that seems to be a thing now. |
14:47.38 | davidbowlby | worse, he can't even google his way into a solution or example |
14:47.40 | Samot | "Tech support" people just using IRC/Forums as their source of knowledge. |
14:47.59 | davidbowlby | it's just as ugly in the language channels |
14:48.14 | davidbowlby | ppl begging for help with their homework |
14:48.16 | Samot | Yeah... |
14:48.21 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-maprkrtdjuawxego) |
14:48.22 | *** mode/#asterisk [+o rmudgett] by ChanServ |
14:48.24 | Samot | But writing a bad PHP for a website.. |
14:48.33 | FarhaadN | http://ibb.co/dvQ3Y6 |
14:48.34 | FarhaadN | http://ibb.co/gLpxt6 |
14:48.37 | davidbowlby | or "what's the best practice" turns into we're doing it for you |
14:48.45 | Samot | Isn't having crappy 911/emergency routing that breaks calls for life and death needs. |
14:48.57 | Samot | VoIP != Web Hosting |
14:48.57 | davidbowlby | Samot, very true! |
14:49.02 | NirS | Guys, everybody starts somewhere - the guy maybe a little clueless right now - but every newbie is |
14:49.20 | Samot | Having Asterisk+Billing software does not make you a VoIP provider. |
14:49.24 | davidbowlby | NirS, but you play with it, you READ, you GOOGLE, you TEST |
14:49.35 | Samot | NirS: FarhaadN is NOT A NEWBIE |
14:49.39 | Samot | That's the point. |
14:49.51 | Samot | Repeat offender is the right term. |
14:50.18 | NirS | oh, in that case, off with his head and RTFM |
14:50.21 | Samot | 1) Not a newbie 2) Has a job as VOIP SUPPORT TECH |
14:50.37 | davidbowlby | wow lol |
14:50.47 | Samot | This isn't Gary who just got stuck or wants to learn Asterisk. |
14:51.03 | Samot | This is a person who went out and is getting paid to be a support tech with Asterisk knowledge. |
14:51.16 | Samot | Who doesn't appear to having a lot because they are always in here asking for help. |
14:51.17 | NirS | I can point him to packt publishing and read the two books I published, but that would be self promoting, wouldn't it ? |
14:51.24 | davidbowlby | FarhaadN, cook up some ViMs and play, bro |
14:51.36 | davidbowlby | *VMs |
14:51.57 | [TK]D-Fender | NirS, Technically, but if it's valid then I couldn't fault the recommendation |
14:52.01 | FarhaadN | i cant underestand every your word |
14:52.17 | Samot | FarhaadN: We're saying get someone that knows this stuff. |
14:52.45 | Samot | FarhaadN: You need to either seriously start learning or tell your bosses this is beyond you and they need to get someone that knows Asterisk. |
14:52.49 | FarhaadN | can u help me? or only to make fun of |
14:53.03 | Samot | Can't we do both? |
14:53.17 | Samot | There is a *cost* for my time, you know. |
14:53.19 | [TK]D-Fender | FarhaadN, Fix your peers |
14:53.33 | Samot | Being the butt of the jokes is rather a cheap cost. |
14:53.48 | FarhaadN | Samot: every time i enter this channel you did that |
14:53.51 | [TK]D-Fender | and look at the actual call. You showed nothing which implies you didn't REALLY look in the first place. Especially based on that single line you considered relevant |
14:54.05 | Samot | FarhaadN: Because the point is still the same. |
14:54.05 | FarhaadN | [TK]D-Fender: what peer? |
14:54.12 | Samot | Oh FFS. |
14:54.14 | [TK]D-Fender | BOTH |
14:55.16 | [TK]D-Fender | It's a bad sign when a guy shows up applying to be to be a commercial truck driver ... and then asks "what's this steering wheel you speak of?" |
14:55.56 | FarhaadN | i dont know whats the meaning of fix your peers |
14:56.16 | FarhaadN | whats wrong |
14:56.25 | FarhaadN | context, type |
14:56.30 | FarhaadN | or other thing |
14:56.37 | Samot | How do we know? |
14:56.41 | Samot | And how should we know? |
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14:56.45 | Samot | They aren't OUR BOXES |
14:56.50 | Samot | You have shown NOTHING |
14:57.00 | Samot | After all the times you've been helped.. |
14:57.08 | Samot | We still have to DRAG DETAILS FROM YOU |
14:57.19 | Samot | Yeah, I'm done. |
14:57.24 | Samot | You guys have fun with this. |
14:57.59 | FarhaadN | Samot: i dont know what you need to can help |
14:58.13 | Samot | I don't need anything. |
14:58.19 | Samot | I'm not helping. |
14:58.33 | FarhaadN | Samot: yea you just for make fun |
14:58.53 | Samot | I've helped you in the past. |
14:59.16 | FarhaadN | after any word you want to tell me |
14:59.19 | Samot | You've made zero improvements over the last year or so I've helped you. |
14:59.43 | Samot | It's always the same steps over and over again. Having to explain each of them all the time. |
14:59.46 | Samot | You're not learning. |
14:59.54 | Samot | I'm not enabling that anymore. |
15:00.31 | FarhaadN | Samot: so you silence when i ask a question |
15:00.37 | FarhaadN | and let other help me |
15:01.07 | Samot | I'm not stopping others. |
15:01.12 | Samot | They are free to help you. |
15:01.52 | voipmonk | Good luck |
15:04.09 | FarhaadN | Smote: tanx for be quiet in the future |
15:04.26 | *** part/#asterisk FarhaadN (~Farhad@82.99.206.194) |
15:04.31 | Samot | LOL |
15:04.45 | voipmonk | He smote himself |
15:04.59 | Samot | Yup. |
15:05.07 | Samot | He realized no one was going to do his job. |
15:05.33 | Samot | In the next 12 hours, watch. |
15:05.49 | Samot | He'll be back hoping to ask when "fresh" people are around. |
15:06.15 | davidbowlby | Hi group, can you just put money in my pocket? Thanks |
15:06.17 | davidbowlby | lol |
15:06.25 | Samot | I just don't get how this is a thing now? |
15:07.02 | davidbowlby | I think it's been like this since the early days of IRC |
15:07.06 | Samot | With some many advances in tech I don't get how these providers can't get qualified people in the door. |
15:07.06 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
15:07.13 | voipmonk | Yes you do, it's pure laziness |
15:07.15 | Samot | Naw.. |
15:07.22 | Samot | It's gotten worse. |
15:07.36 | Samot | My main client base *is* providers. |
15:07.42 | voipmonk | Reading is becoming a super power like sense that used to be common |
15:07.50 | davidbowlby | I assume it's gotten worse in this channel with so many trunk providers being out there |
15:07.51 | Samot | When then end up with guys like FarhaadN |
15:07.55 | Samot | *they |
15:08.06 | Samot | I usually end up getting guys like FarhaadN fired. |
15:08.12 | davidbowlby | but it is global, I blame the 140 character society |
15:08.31 | Samot | One place had a guy that did all his setups via copy and paste what he found on google. |
15:08.39 | Samot | If he found a blog that said X could be done. |
15:08.47 | Samot | He would tell customer they could do X |
15:08.52 | Samot | Without actually testing it. |
15:08.54 | Samot | Trying it |
15:08.55 | davidbowlby | lol |
15:09.00 | davidbowlby | yeah, I can see that |
15:09.15 | davidbowlby | hell, I worked for a managed service provider and their sales ppl screwed us all the time like that |
15:09.19 | Samot | Then the owners are wondering why clients are bitching them out.. |
15:09.28 | davidbowlby | heheh yeah |
15:09.32 | Samot | Why they are spending sooo much money on support... |
15:09.45 | Samot | ^^ Biggest question providers have. |
15:09.50 | Samot | "Why is support so costly?" |
15:10.02 | Samot | "Because your team takes 2 hours to do 15 minute tasks" |
15:10.23 | [TK]D-Fender | That's a gnerous round-upt to 15... |
15:10.29 | Samot | Yes. |
15:10.29 | [TK]D-Fender | a peering setup shouldn't take 5 |
15:10.37 | Samot | But that includes updating the ticket. |
15:10.40 | davidbowlby | haha |
15:10.47 | Samot | I mean as a support tech |
15:10.52 | Samot | From open to close 15 minutes. |
15:11.05 | Samot | And yeah, that's more a high average for most tasks. |
15:11.16 | [TK]D-Fender | Still quite generous, but he does type kinda slow... |
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15:11.30 | davidbowlby | sounds like this guy has been in here a lot, you'd think he would document some of the solutions you folks have given him |
15:11.39 | Samot | But if you have to google or log into IRC each time a client can't register their endpoint.. |
15:11.44 | Samot | You're bleeding money. |
15:11.45 | voipmonk | Just search the log |
15:11.52 | Samot | Hemorrhaging. |
15:12.20 | Samot | But then again.. |
15:12.33 | [TK]D-Fender | We've got a number of people who've been around here for YEARS (some 7+) who are still absolutely clueless |
15:12.39 | Samot | I spent over 15 years being a one man tech support team to 50K+ end users. |
15:12.41 | [TK]D-Fender | the most basic stuff |
15:13.12 | Samot | So being efficient kinda became required. |
15:15.05 | Samot | The ideal tech support team should be spending the majority of their time working on revenue generating stuff. |
15:15.15 | Samot | And handling very few real issues. |
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15:50.30 | davidbowlby | was talking to a coworker about how computers used to come with programming books |
15:50.50 | davidbowlby | and now they come with icons and pictures to show you how to plug it in |
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18:54.37 | nafg__ | Hi |
18:55.13 | nafg__ | Does asterisk still prefer audio files to be in elm? |
18:55.22 | nafg__ | Err autocorrect |
18:55.28 | voipmonk | elm? |
18:55.32 | nafg__ | Sln |
18:55.51 | nafg__ | Phone autocorrected it |
18:56.00 | nafg__ | :D |
18:57.01 | nafg__ | Does asterisk still prefer sln |
18:57.48 | [TK]D-Fender | * prefers audio files to be in the codec of the call |
18:57.55 | nafg__ | It says that online but most asterisk information online is really old |
19:00.00 | voipmonk | T/P https://wiki.asterisk.org/wiki/display/AST/Asterisk+Audio+and+Video+Capabilities Transcode / Pass through |
19:01.35 | nafg__ | I'm using flowroute for did. Docs say "Supported media codecs are G.711-ulaw and G.729" |
19:02.38 | [TK]D-Fender | <[TK]D-Fender> * prefers audio files to be in the codec of the call <--- |
19:03.26 | nafg__ | Right so does that mean it should be one of those two? |
19:03.51 | [TK]D-Fender | You should know what codec you rstricted your trunk to and have files to match |
19:04.04 | [TK]D-Fender | If yuo allw multiple then yuo should have those files in ALL of those formats |
19:05.20 | nafg__ | I don't want to buy g729 |
19:05.32 | davidbowlby | the buy bandwidth |
19:05.33 | davidbowlby | hehe |
19:05.38 | davidbowlby | *then |
19:05.55 | davidbowlby | flowroute uses G.711-ulaw |
19:06.01 | davidbowlby | that's what I save mine in |
19:06.07 | nafg__ | Valid point, but I don't need too much bandwidth for this |
19:06.27 | [TK]D-Fender | So use the otehr |
19:06.43 | nafg__ | What is g711 ulaw in ffmpeg flags and file extension terms? |
19:06.49 | davidbowlby | and I paid for 729, just don't use it a lot for inbound |
19:07.04 | davidbowlby | I dunno, I use Audacity and upload the files |
19:07.28 | davidbowlby | cause I'm lazy |
19:07.37 | nafg__ | Let me Google it :) |
19:07.46 | davidbowlby | there ya go |
19:07.49 | davidbowlby | a scholar! |
19:09.09 | *** join/#asterisk DanB (~DanB@clt-195.192.201.144.ip-anschluss.net) |
19:10.52 | nafg__ | https://superuser.com/q/670561/146949 |
19:11.47 | nafg__ | A commenter there says: @meda If you use .wav as an extension, ffmpeg automatically guesses that you want a WAV container wrapping your PCM audio. If you do not want that, and instead need raw audio data in a .ulaw file, you need to use -f mulaw to force ffmpeg to use the PCM mu-law output format. â slhck Nov 7 '13 at 6:20 |
19:12.33 | nafg__ | So to be clear, g711-ulaw is codec AND a container format? |
19:12.52 | nafg__ | And does asterisk care about its extension? |
19:13.18 | nafg__ | How much does the container format matter to asterisk? |
19:15.48 | [TK]D-Fender | nafg__, "asterisk convert audio files" <- GOOGLE IT |
19:16.20 | [TK]D-Fender | And be open to using other common tools for this job, including SOX, and if * can already read yuor base file format, ussing * itself |
19:17.00 | nafg__ | What does using asterisk itself mean |
19:19.56 | [TK]D-Fender | use ASTERISK to convert the files |
19:20.24 | [TK]D-Fender | if it can read the source you can convert to any other codec it supports |
19:20.41 | nafg__ | You mean on the fly? |
19:20.51 | [TK]D-Fender | No. |
19:20.57 | [TK]D-Fender | that is what we are trying to AVOID |
19:21.08 | [TK]D-Fender | convert it as a file. |
19:21.11 | nafg__ | Right. So, then how |
19:21.16 | [TK]D-Fender | via CLI |
19:21.20 | [TK]D-Fender | look at your command list |
19:22.03 | nafg__ | That's not an option. I need something my prompt management app can automate |
19:23.23 | [TK]D-Fender | We don;t knwo what app you're using or what it can do. EVERYTHING sounds to be external from it right now |
19:23.29 | [TK]D-Fender | since you're already lookingat ffmpg |
19:23.33 | [TK]D-Fender | to do the work. |
19:23.46 | nafg__ | It's my own app |
19:24.37 | [TK]D-Fender | So coding it to use * doesn't sound any different than any other method really |
19:24.41 | nafg__ | I can just spawn ffmpeg and pipe its stdin and stdout |
19:24.52 | [TK]D-Fender | So pick whatever you want |
19:25.43 | nafg__ | I can have ffmpeg in my docker container, asterisk is in a different one |
19:26.28 | nafg__ | I already picked one, and I asked a question before, and you just gave me an unhelpful phrase to google |
19:27.15 | nafg__ | In quotes it returns one result about digium online converter |
19:27.25 | [TK]D-Fender | "ffmpeg ulaw conversion" <- tried that? |
19:27.35 | [TK]D-Fender | I see a pile of * related links right away |
19:27.45 | nafg__ | And without, I get all the ancient voip-info pages I've gone through etc etc |
19:28.11 | [TK]D-Fender | <PROTECTED> |
19:28.28 | nafg__ | Of course the are * related links. Let me repost my question |
19:28.45 | nafg__ | If you don't know the answer, just say so, telling me to google is not helpful |
19:28.47 | [TK]D-Fender | If syntax changed maybe nobody's documenting it perfectly for you anymore and you'll just have to read up on the minor tweak. |
19:29.02 | nafg__ | 3:11 PM <nafg__> A commenter there says: @meda If you use .wav as an extension, ffmpeg automatically guesses that you want a WAV container wrapping your PCM audio. If you do not want that, and instead need raw audio data in a .ulaw file, you need to use -f mulaw to force ffmpeg to use the PCM mu-law output format. â slhck Nov 7 '13 at 6:20 |
19:29.12 | nafg__ | 3:12 PM <nafg__> So to be clear, g711-ulaw is codec AND a container format? |
19:29.19 | nafg__ | That's my first question |
19:29.38 | nafg__ | 3:12 PM <nafg__> And does asterisk care about its extension? |
19:29.44 | nafg__ | Was the 2nd |
19:29.57 | nafg__ | 3:13 PM <nafg__> How much does the container format matter to asterisk? |
19:30.06 | nafg__ | Was the 3rd question |
19:30.23 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
19:31.28 | nafg__ | These are things I've spent lots of time googling and there is tons of conflicting, outdated (or likely outdated), or just plain unclear information |
19:32.17 | nafg__ | I come to IRC after I'm exhausted from googling, hoping to actually talk to experts |
19:32.29 | FarhaadN | Samot: for your information , this problem is not my fault, and all of my configuration is that correct, and problem acourd becuase in 2 pbx there is same extention |
19:33.13 | FarhaadN | please when you dont know about somthing , no tell any word from that |
19:33.19 | FarhaadN | tnx |
19:33.45 | *** part/#asterisk FarhaadN (~Farhad@82.99.206.194) |
19:33.51 | nafg__ | [TK]D-Fender: it has nothing to do with syntax. Do you know the answer to any of those three questions? |
19:35.26 | [TK]D-Fender | I don't see anything referring to ulaw as being in a container |
19:36.03 | seanbright | nafg__: 1) no, it is not a container format. 2) if it's ulaw, it should have an .ulaw, .mu, .ul, .ulw, or .pcm extension. 3) n/a |
19:37.06 | nafg__ | [TK]D-Fender: then how do you interpret what the commenter is saying? |
19:37.36 | seanbright | wav is a container format |
19:37.58 | [TK]D-Fender | they were talking about wav when that word came up |
19:38.12 | [TK]D-Fender | they then said if you wanted a raw ulaw do X .... |
19:38.25 | [TK]D-Fender | so they drew a clear separation between the two |
19:38.33 | Samot | ROFL |
19:38.45 | Samot | I love when those happen. |
19:39.11 | nafg__ | seanbright: right. The answerer already had the flag for the ulaw codec but the commenter said that's insufficient |
19:39.30 | nafg__ | Because then it will be ulaw codec in wav container |
19:39.57 | nafg__ | Did you guys read the context? |
19:40.14 | seanbright | no, because i don't care |
19:40.16 | nafg__ | https://superuser.com/q/670561/146949 |
19:40.22 | seanbright | is your question "how to convert an audio file to ulaw?" |
19:40.29 | seanbright | "... with ffmpeg?" |
19:40.54 | nafg__ | Partially |
19:41.20 | nafg__ | I'm also trying to understand things better |
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19:42.26 | nafg__ | seanbright: could just look at the superuser page (very short) and tell me if you're disagreeing with the commenter or how I'm misunderstanding him |
19:42.53 | nafg__ | could *you |
19:43.35 | seanbright | "ffmpeg -i input -codec:a pcm_mulaw -f mulaw output.ulaw" |
19:43.50 | seanbright | that looks right-ish, but i don't know anything about ffmpeh |
19:43.56 | seanbright | ffmpeg* |
19:44.55 | nafg__ | seanbright: so if mulaw isn't a container format what's the -f for |
19:46.04 | nafg__ | -f mulaw |
19:46.12 | seanbright | i dunno? |
19:46.17 | seanbright | try 'man ffmpeg' |
19:46.22 | seanbright | or /join #ffmpeg |
19:46.38 | nafg__ | -f is for the container format |
19:46.44 | seanbright | ok? |
19:46.52 | seanbright | again - i don't care |
19:47.03 | seanbright | for once i agree with [TK]D-Fender |
19:47.05 | seanbright | use google |
19:47.07 | seanbright | try it out |
19:47.59 | [TK]D-Fender | seanbright, c'mon we agree a lot more than that! |
19:48.19 | nafg__ | Anything I'm asking here, I've tried googling etc already |
19:48.51 | [TK]D-Fender | Have you tried taking one of the provided ulaw files and using one of your tools to look at it and give you that answer? |
19:49.32 | nafg__ | Unless "container" is the wrong word. But there's definitely codec vs format |
19:49.48 | seanbright | nafg__: what is your source file format? |
19:50.11 | nafg__ | wav... Whatever recorder.js produces |
19:50.26 | seanbright | so you don't know? |
19:50.33 | seanbright | because as you know... wav is a container format |
19:50.53 | nafg__ | Not sure offhand |
19:51.10 | seanbright | ok, let me install ffmpeg real quick so that i can answer this question |
19:51.43 | nafg__ | You can view the man page at https://linux.die.net/man/1/ffmpeg |
19:51.51 | seanbright | lol |
19:51.52 | nafg__ | Thanks |
19:51.58 | seanbright | this is trolling level 1000 |
19:52.05 | davidbowlby | use audacity |
19:52.06 | nafg__ | ? |
19:52.06 | davidbowlby | lol |
19:52.23 | seanbright | nafg__: why are you unable to use the man page and figure this out yourself? |
19:53.18 | nafg__ | I don't know. Do you think I'm too dumb? |
19:53.52 | [TK]D-Fender | -f fmt |
19:53.52 | [TK]D-Fender | <PROTECTED> |
19:53.59 | [TK]D-Fender | well this seems to force format |
19:54.08 | nafg__ | Really now |
19:54.10 | [TK]D-Fender | that doesn't imply "container" to me.... |
19:54.22 | nafg__ | Ok, it's not codec though |
19:54.24 | seanbright | [TK]D-Fender: no no no, that is clearly the container flag |
19:54.28 | seanbright | as indiciated by nafg__ |
19:54.30 | [TK]D-Fender | So forcing a format for your output file ... seems sane enough so far |
19:54.39 | [TK]D-Fender | What do you feel is suspcious there? |
19:54.51 | nafg__ | 3:49 PM <nafg__> Unless "container" is the wrong word. But there's definitely codec vs format |
19:55.05 | davidbowlby | I like pizza |
19:55.08 | nafg__ | Who said anything is suspicious |
19:55.11 | [TK]D-Fender | So.... what's your concern with it? |
19:55.14 | [TK]D-Fender | Does it work? |
19:55.14 | davidbowlby | since we're trolling |
19:55.24 | davidbowlby | pizza always works |
19:55.30 | nafg__ | I think I have a misunderstanding about audio file binary layout |
19:55.43 | davidbowlby | 11pepperoni011 |
19:55.53 | seanbright | the the binary layout of the audio file make any difference? |
19:55.57 | seanbright | does the* |
19:56.05 | nafg__ | Are the 3 things? Codec, format, container format? |
19:56.20 | davidbowlby | whispers "audaaaaaaacityyyyy" |
19:56.21 | nafg__ | That's what I'm trying to learn about right now |
19:57.10 | nafg__ | davidbowlby: can you make me a docker image that I can script audacity headless? |
19:57.27 | davidbowlby | nafg__ you said headless |
19:57.33 | davidbowlby | no one wants to live like that |
19:57.45 | davidbowlby | sets trolling = 0 |
19:57.55 | nafg__ | Servers apparently do :) |
19:58.05 | [TK]D-Fender | No-one GETS to live like that.... for more than a handful of seconds |
19:58.31 | davidbowlby | so you're gonna sit here and tell us you are creating a docker image to do fancy stuff and can't try/fail your way through ffmpeg |
19:58.44 | [TK]D-Fender | None of us are going around container-ing our solutions in bite-sized pieces using docker and insisting on ffmpeg. |
19:58.48 | seanbright | nafg__: this is what i just tested: |
19:58.54 | seanbright | ffmpeg -i input.wav output.au |
19:59.05 | [TK]D-Fender | The list of people we can say have been in that camp so far .... = 1 person |
19:59.26 | seanbright | nafg__: so you said that was partially your question. what is the rest? |
20:00.57 | nafg__ | And what was the result? |
20:01.11 | seanbright | a ulaw file named output.au |
20:01.14 | seanbright | nafg__: so you said that was partially your question. what is the rest? |
20:01.50 | nafg__ | So apparently yes, there are 3 different things: https://en.m.wikipedia.org/wiki/Audio_file_format#Format_types |
20:02.29 | nafg__ | > It is important to distinguish between the audio coding format, the container containing the raw audio data, and an audio codec. |
20:03.15 | seanbright | nafg__: so you said that was partially your question. what is the rest? |
20:03.36 | seanbright | trying to wrap up this painful interaction |
20:03.42 | nafg__ | So I guess ffmpeg's -f is about the "audio coding format," which I confused with container formats |
20:05.05 | nafg__ | seanbright: hence my perceived contradiction between you saying ulaw is not a container format, and the commenter implying it's not only a codec |
20:05.06 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
20:06.02 | nafg__ | In any case it's still two things, ulaw the codec and ulaw the format and those are not necessarily hand in hand? |
20:06.38 | nafg__ | It's possible to have a file with ulaw codec and other format, and vice versa? |
20:07.19 | nafg__ | And is it only the codec that matters to asterisk? |
20:07.32 | davidbowlby | nafg__ I dunno, try it and find out |
20:08.21 | nafg__ | Yeah... Was hoping to find people here at a deeper understanding of how asterisk works |
20:08.33 | davidbowlby | experience is the best understanding |
20:08.53 | davidbowlby | won't you know after testing? |
20:09.14 | nafg__ | Also because I think asterisk makes assumptions based on a files extension |
20:09.30 | davidbowlby | and two years later when talking to someone who is having the issue, be like, "oh yeah, I had to try this, that, and the other thing, but it turns out.." |
20:09.40 | nafg__ | davidbowlby: yeah you know you're right, who needs IRC, let everyone using their own experience |
20:10.01 | davidbowlby | you've asked and you've gotten the answers available |
20:10.04 | davidbowlby | why keep proddin |
20:10.35 | nafg__ | No, I was misunderstood because I used the wrong word, |
20:10.49 | nafg__ | so I asked the corrected question |
20:11.09 | davidbowlby | the folks who responded suggested they don't use ffmpeg, so you're on your own there. |
20:11.26 | nafg__ | The question was not about ffmpeg sir |
20:11.34 | nafg__ | I don't know why you think that |
20:11.41 | davidbowlby | it was how to use ffmpeg to achieve your goal |
20:11.45 | nafg__ | No |
20:12.00 | nafg__ | It was about what rules asterisk follows |
20:12.13 | nafg__ | What aspects of files it cares about |
20:12.36 | davidbowlby | to the end of using ffmpeg... |
20:12.37 | nafg__ | I know ffmpeg syntax. That was never the question |
20:12.43 | davidbowlby | ok |
20:13.13 | nafg__ | No. To the end of learning asterisk more deeply |
20:13.31 | davidbowlby | if you wanna get deep with asterisk, read the code :P |
20:13.33 | davidbowlby | haha |
20:14.03 | nafg__ | Yeah, who doesn't love reading C |
20:14.19 | davidbowlby | docker boys, I assume |
20:15.07 | davidbowlby | hey man, it's Friday, lift your spirits and drop your packets |
20:15.30 | file | Asterisk does not support containers and determines the underlying codec based on the file extension, with the files themselves being raw uncontainerized frames or payloads |
20:15.45 | davidbowlby | damn, that was sexy talk right there |
20:15.51 | davidbowlby | answered your question like *boom* |
20:16.14 | file | drops mic |
20:16.51 | drunkdavidbowlby | damn nick char limit |
20:16.57 | seanbright | 15:40 <@seanbright> is your question "how to convert an audio file to ulaw?" |
20:16.57 | seanbright | 15:40 <@seanbright> "... with ffmpeg?" |
20:16.58 | seanbright | 15:40 < nafg__> Partially |
20:17.04 | seanbright | 16:11 < nafg__> The question was not about ffmpeg sir |
20:17.04 | seanbright | 16:11 < nafg__> I don't know why you think that |
20:17.14 | seanbright | good times |
20:17.21 | drunkdavidbowlby | [16:15:30] <@file>Asterisk does not support containers and determines the underlying codec based on the file extension, with the files themselves being raw uncontainerized frames or payloads |
20:17.25 | drunkdavidbowlby | I can do that too, see |
20:17.26 | seanbright | is exhausted all of a sudden |
20:17.33 | file | gives seanbright a cookie |
20:17.41 | drunkdavidbowlby | takes another sip |
20:17.57 | drunkdavidbowlby | I'm gonna have some bugs to fix tomorrow! |
20:18.24 | nafg__ | seanbright: if you can't reconcile that, maybe sleep on it |
20:20.49 | seanbright | anything else or are you good now? |
20:20.56 | nafg__ | file: ok thanks. But what about what Wikipedia calls "audio file coding" (distinct from codec; IIUC corresponds to ffmpeg -f flag) -- how much does that matter to asterisk? |
20:21.32 | seanbright | the only thing that matters to asterisk is the extension |
20:21.52 | seanbright | if it's .ulaw, it will assume it is ulaw data and try to read it as such |
20:22.00 | nafg__ | Ok thanks |
20:22.21 | drunkdavidbowlby | you're gonna sit there and tell me if the ext is .ulaw, it's going to *ASSUME* ulaw?!!?! |
20:22.45 | drunkdavidbowlby | and I guess next you're gonna tell me if it is .mp3 it's going to assume mp3 |
20:22.55 | file | a codec is the name of the encoding and the process, the file coding is the result of the process |
20:22.56 | drunkdavidbowlby | is shook |
20:23.05 | seanbright | mp3 is a container format |
20:23.26 | nafg__ | Interesting observation (IIUC) - asterisk correlates extension to codec, while ffmpeg correlates extension to format (by default) |
20:26.51 | drunkdavidbowlby | needs another drink after reading that |
20:30.07 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
20:34.19 | drunkdavidbowlby | why is there a new account code event? |
20:34.27 | drunkdavidbowlby | I assume for billing? |
20:34.59 | drunkdavidbowlby | ahh nm, imagine that, looked it up my damn self. |
20:35.07 | drunkdavidbowlby | see how that's done? |
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20:54.49 | voipmonk | clones drunkdavidbowlby |
20:55.32 | drunkdavidbowlby | looks at clone... good lookin guy |
20:56.13 | voipmonk | was gonna send him to Jules but he would come back and bslap me for thinking of that as a solution... Clones lives matter |
20:56.55 | drunkdavidbowlby | I heard there was a whole war about it |
20:57.28 | voipmonk | [J]oules |
20:58.26 | voipmonk | manifests food inside Restaraunt, ready for pickup - bbl |
20:59.08 | voipmonk | Restauhhh hell |
20:59.45 | drunkdavidbowlby | lol |
20:59.54 | drunkdavidbowlby | lots of folks have problems with that one |
21:00.09 | drunkdavidbowlby | even if sober |
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