| 12:16.35 | *** join/#asterisk acidfu_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
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| 12:25.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 12:25.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 12:26.46 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
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| 12:28.55 | [sID] | Samot: But how ? |
| 12:29.26 | Samot | Well what is your CPS limit? |
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| 12:30.28 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:31.06 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:43.21 | [sID] | Samot: 10 peer secend |
| 12:43.36 | [sID] | second |
| 12:43.46 | Samot | So you're sending more than 10 calls per second? |
| 12:44.53 | [sID] | So, how do I send it gets block my ip |
| 12:49.30 | Samot | Well why are you sending to many calls at once? |
| 12:52.14 | [sID] | Soft Call center for sending many agents |
| 12:53.11 | Samot | So auto dialing? |
| 12:53.40 | *** join/#asterisk CeBe (~CeBe@a81-14-235-90.net-htp.de) |
| 12:53.49 | [sID] | Yes |
| 12:54.03 | Samot | Then throttle the autodialer. |
| 12:54.32 | [sID] | I have to reduce the level dialplan |
| 12:54.51 | Samot | How are the calls being initiated? |
| 12:55.18 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:56.48 | [sID] | By sip account |
| 12:57.16 | Samot | How is the auto-dialing happening? |
| 12:57.34 | *** join/#asterisk Akuma (~Akuma@AYLMPQ0104W-LP140-01-845441731.dsl.bell.ca) |
| 12:58.02 | Samot | I doubt that all the agents have managed to synchronize themselves to manually send 10+ calls at the exact same second. |
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| 13:07.09 | peacehope | Today when someone call here and press 1, it always goes to one extension. I'd like it distribute between two extensions. One time to extension 'A', another time to extension 'B'. |
| 13:07.12 | peacehope | Is it possible? |
| 13:09.46 | [TK]D-Fender | peacehope, it's yyour dialplan, do whatever you want |
| 13:09.50 | [sID] | somepoortech: by soft call center |
| 13:09.59 | [TK]D-Fender | peacehope, Dial one.. then dial the other. Nothing more to it than that |
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| 13:10.46 | *** part/#asterisk fspy (fspy@unaffiliated/spychalski) |
| 13:11.35 | [sID] | Samot by soft call center |
| 13:11.39 | [sID] | Samot: There are 10 agents and soft is set to 2 peer agent. |
| 13:11.49 | peacehope | [TK]D-Fender: It's support option, and I want to divide between each person here. I've never configured an Asterisk. I suppose it can be done on extensions-in.conf file here. Right? |
| 13:11.54 | [sID] | Samot And soft sends agement for a asterisk and the rings |
| 13:12.18 | Samot | [sID]: Then drop it to one per agent. You shouldn't send more calls than agents anyways. |
| 13:12.39 | [TK]D-Fender | peacehope, call processing = dialplan = extensions.conf |
| 13:12.44 | [sID] | Samot: Regardless of me. |
| 13:13.02 | [sID] | I have to handle this |
| 13:13.15 | Samot | [sID]: Tell the system that is sending the calls to not send more than 10 at a time. |
| 13:13.17 | Samot | That's it. |
| 13:14.33 | [TK]D-Fender | ~book |
| 13:14.33 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:14.35 | [TK]D-Fender | peacehope, ^^^ |
| 13:14.42 | [sID] | Samot: But how does this limit? |
| 13:14.46 | [sID] | I have 3 trunk and each must be a limit one 10 second 7 |
| 13:14.55 | [TK]D-Fender | peacehope, You'll have plenty of reading to do to learn how the dialplan works.... |
| 13:15.15 | [TK]D-Fender | peacehope, Did you just inherit your system from a previous admin? |
| 13:15.18 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:15.26 | Samot | [sID]: You're using a program that is sending the calls, tell the program not to send so many calls. |
| 13:16.13 | [sID] | I can not interfere in this program. |
| 13:16.21 | Samot | Well that's the problem. |
| 13:16.30 | [sID] | These are constants that can not be changed |
| 13:16.36 | Samot | You want to fix the problem, that's the solution. |
| 13:16.48 | Samot | Well there's no way you're going to get your CPS increased. |
| 13:17.02 | Samot | So you have to limit your calls based on your CPS. |
| 13:17.08 | [sID] | It is not a solution. |
| 13:17.26 | Samot | So your auto-dialing solution is causing your provider to block you. |
| 13:17.42 | [sID] | It is not a solution. |
| 13:17.42 | [TK]D-Fender | limit them in the dialplan then... |
| 13:18.04 | [sID] | [TK]D-Fender: yes, It wants to do |
| 13:18.14 | [TK]D-Fender | Go do it then |
| 13:18.28 | peacehope | [TK]D-Fender: There is nobody for manipulate this here. I am a sysadmin with no experience with Asterisk. Every call comes to me, and I want to divide between me and another person, being fair. |
| 13:18.53 | [sID] | [TK]D-Fender: Only I am looking for help has just as you do. |
| 13:18.54 | [TK]D-Fender | peacehope, depends on what you base "fair" on. |
| 13:19.17 | peacehope | [TK]D-Fender: call 1 for me, call 2 for him, call 3 for me, call 4 for him... |
| 13:19.23 | [TK]D-Fender | [sID], Go shove your call data in a DB and check before dialing. |
| 13:19.39 | [TK]D-Fender | [sID], this is up to you tto come up with how you want to measure them. |
| 13:20.26 | [TK]D-Fender | [sID], or track the last X calls with some mroe local storage like AstDB and calc the time between tthe first & last. |
| 13:20.38 | [sID] | [TK]D-Fender: It will be difficult to calculate the connection in real time based on the base |
| 13:21.26 | [sID] | [TK]D-Fender: I was thinking more on the global variables somehow set it up. |
| 13:22.30 | [TK]D-Fender | Globals, AstDB, external DB, whatever. Come up with whatever method you want for the storage and calc the times ebtween 2 points |
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| 13:24.18 | peacehope | [TK]D-Fender: This is my extensions-in.conf: http://pastebin.com/GQUKMzzS . Could you please take a look? It's the option 1. Extensions: 1040 is me, and 1030 is the other person. |
| 13:29.20 | [sID] | [TK]D-Fender: I like sum each call by the trunk |
| 13:34.54 | [TK]D-Fender | [sID], Any way you want , go for it |
| 13:38.51 | [TK]D-Fender | <[TK]D-Fender> peacehope, depends on what you base "fair" on. |
| 13:40.59 | peacehope | [TK]D-Fender: Just alternate. Call 1 for me, call 2 for him, call 3 for me, call 4 for him... I want to divide between us. And when one is busy, ring to the other (it already happens). |
| 13:42.06 | [TK]D-Fender | There are dialplan ways, but those don'ttend to work well if you have multiple simultaneous calls |
| 13:42.19 | [TK]D-Fender | your other option is a queue in that case |
| 13:45.11 | peacehope | [TK]D-Fender: It's only 2 lines today. Maybe it will not be a problem so. |
| 13:46.11 | [TK]D-Fender | You'll have to track which one you called last, and then flip that just before you dial the other. |
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| 13:49.02 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:50.39 | peacehope | [TK]D-Fender: Is there a simple way to do it randomly between two extensions instead tracking? |
| 13:51.13 | [TK]D-Fender | tracking = just flip-flopping a single value in astDB or a global variable |
| 13:51.19 | [TK]D-Fender | better than random |
| 13:51.33 | [TK]D-Fender | either way 1 check, 1 value to flip. 2 lines of dialplan. |
| 13:55.44 | fahmad | i am using grandstream GXW410x Gateway, I have two telephone lines which are allowed to dial 0, i would like asterisk to dial out directly using another channel if 1st channel is busy, i have found only way that we can do it using 991 for port1 992 for port2 but i want both working with single dialing rule |
| 13:57.55 | Samot | fahmad: The answer is still the same. |
| 13:58.04 | [TK]D-Fender | fahmad, If you expect a single Dial() to do it then you need to configure the gateway to pick the channel that way. Read its manual to see how to do that. |
| 13:58.48 | fahmad | hmm |
| 13:59.26 | Samot | See. Answer is still the same. |
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| 14:10.13 | peacehope | What is the term or syntax to identify which extension received the last call? I didn't find any example yet. |
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| 14:11.36 | peacehope | (I am with documentation opened) |
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| 14:12.08 | [TK]D-Fender | There is no "syntax" to find these things out. You set a value yourself and check for it. |
| 14:13.13 | [TK]D-Fender | And finding a sample that really looks like what you want may be difficult. Dialplan is programming, and you should have at least a basic sense of programming logic tto do it. |
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| 14:18.28 | Samot | Well that was interesting |
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| 14:18.58 | peacehope | [TK]D-Fender: About Linux and programming it's fine for me. My problem is being Asterisk. I will read about dialplan so. Thanks. |
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| 15:59.08 | shido6 | espresso shots for "heart attack in a cup" |
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| 16:53.34 | fabio_ | Hello |
| 16:54.30 | mub | fabio_: Your nick makes me think of sexy dark skinned men |
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| 16:59.28 | [TK]D-Fender | Except that Fabio was a famous blond-haired Caucasian model & actor... |
| 16:59.40 | [TK]D-Fender | So why that bring dark-skinned to mind... dunno |
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| 17:20.29 | ziggoto | Hello |
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| 17:36.10 | hdon | hi all :) for outbound sip traffic from asterisk, can asterisk be configured on which address it should bind the local end of the socket? |
| 17:37.21 | hdon | our asterisk box has a LAN IP on our locally routable server subnet, and also on our publicly routable server subnet. but when it connects to a peer on the Internet from its address on the local subnet, it gets nat'd like every other workstation in the building to the wrong address. |
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| 17:43.51 | [TK]D-Fender | * will use tthe best interface based on your routing table and whatt you told it to bind to |
| 17:44.07 | [TK]D-Fender | chan_sip supports binding to all, or one, but not multiple specific |
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| 17:45.41 | ziggoto | Hey guys |
| 17:46.11 | ziggoto | Can MixMonitor record audio from WaitForSilence application? |
| 17:46.55 | ziggoto | Does anyone ever tried? |
| 17:48.56 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 17:59.11 | cusco | yes |
| 17:59.15 | cusco | ah no |
| 17:59.32 | cusco | sorry didn't try with waitforsilence |
| 17:59.33 | cusco | but I don't see why not anyway |
| 18:17.37 | hdon | hi all :) we're using a script to translate old asterisk configuration to new |
| 18:17.51 | hdon | i'm seeing this error in my log file which looks like a parse error: |
| 18:18.49 | hdon | http://ix.io/19Lm ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '>', expecting $end |
| 18:19.19 | hdon | so i grep for '> 0' because that is cited as being the issue. i see lines generated by freepbx like this: exten => 2004,1,Set(__RINGTIMER=${IF($[${DB(AMPUSER/2004/ringtimer)} > 0]?${DB(AMPUSER/2004/ringtimer)}:${RINGTIMER_DEFAULT})}) |
| 18:19.33 | hdon | is it possible that this is no longer valid syntax for asterisk dialplan? |
| 18:19.45 | hdon | or that the configuration translator messed it up? |
| 18:22.40 | ziggoto | Thanks cusco, it's worked :) |
| 18:24.58 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 18:28.51 | [TK]D-Fender | hdon, your astDB value is blank and your expression fails because there is literall nothing on the left side |
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| 18:45.20 | hdon | [TK]D-Fender, do i understand then that a sort of "eval" is being used? i didn't expect to encounter the parser being entered after asterisk fully initialized |
| 18:46.05 | [TK]D-Fender | huh? |
| 18:46.26 | hdon | [TK]D-Fender, well this looks to me like a parser error, which i would expect to encounter during initialization... |
| 18:46.38 | [TK]D-Fender | What "initializattion? |
| 18:46.43 | [TK]D-Fender | that is dialplan |
| 18:46.59 | [TK]D-Fender | You only get an error when the results of its execution fail |
| 18:47.19 | [TK]D-Fender | There is no "load" QC if tthe minimum syntax is valid |
| 18:47.45 | [TK]D-Fender | if you use a funtion wrong or in a way that doesn't actually return a proper value where you need one then it will error out |
| 18:48.14 | hdon | i see |
| 18:48.24 | hdon | ok, i just didn't expect a parser error |
| 18:48.57 | [TK]D-Fender | dialplan is evaluated at run-time, not "compiled" |
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| 19:39.26 | SpaceInvaders | Hi! |
| 19:39.28 | SpaceInvaders | Why would |
| 19:39.30 | SpaceInvaders | same => n,NoOp(File Format = {${FILE_FORMAT(/etc/asterisk/Asterisk-whitelist.txt)}) |
| 19:39.32 | SpaceInvaders | return |
| 19:39.36 | SpaceInvaders | {u |
| 19:39.44 | SpaceInvaders | rather than just the u (per docs)? |
| 19:39.58 | SpaceInvaders | oops |
| 19:40.05 | SpaceInvaders | please disregard |
| 19:40.41 | [TK]D-Fender | Yes, that was a completely extra brace right at the front... |
| 19:40.43 | SpaceInvaders | Isn't it wonderful and amazing how code does exactly what you tell it to? :^) |
| 19:41.04 | SpaceInvaders | yea I've been staring at it for like 15 minutes and didn't see it until I pasted it, here |
| 19:41.20 | [TK]D-Fender | and ISO certification doesn't mean your products are good, it just means their defects are reproduced PERFECTLY. |
| 19:41.59 | SpaceInvaders | ROFL |
| 19:42.07 | SpaceInvaders | I have some ISO9000 exp :D |
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| 19:49.52 | Kunsi | SpaceInvaders: https://en.wikipedia.org/wiki/Rubber_duck_debugging |
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| 19:54.31 | SpaceInvaders | which reminds me of one of my favorite songs!!! https://www.youtube.com/watch?v=cDy4PZPMDwU |
| 19:56.00 | SpaceInvaders | Kunsi that's why I stared at it for 15m lol. I hate finding stuff after I ask. But you bring up a good point. I may start pasting to gedit before I paste in-channel :) |
| 20:08.31 | SpaceInvaders | clearglobalvars=no in [globals] means dialplan reload shouldn't reset a global variable, right? |
| 20:12.52 | TandyUK | you should always use a pastebin of some sort ;) |
| 20:15.03 | Samot | I love when answers like "not many" are actually worse when converted to real stats. |
| 20:15.15 | Samot | "How many audio issues are reported?" |
| 20:15.19 | Samot | "Not many" |
| 20:15.28 | Samot | "Like? 1 in what?" |
| 20:15.39 | Samot | "Oh like 1 in 4 or 5 maybe more" |
| 20:16.05 | Samot | "So like almost 35% of support issues are audio related, you don't see a problem with this?" |
| 20:16.07 | Samot | "No |
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| 20:46.17 | TandyUK | lol |
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| 21:01.53 | pdugas | Remote Polycom IP501 behind a cheap NAT'ing router at home has been working for weeks but stopped recently. Am seeing REGISTER request and 401 response on Asterisk box but not the expected REGISTER follow up with the proper auth info using the nonce from the 401. Am seeing repeated REGISTER and SUBSCRIBE requests after - all 401 unauthorized. Something wonky with that home router is my guess. Connected phone to cable modem |
| 21:01.53 | pdugas | (bypassing router) and it worked. Am I off-base thinking the router is crap? |
| 21:03.14 | Kunsi | pdugas: if it works without router, but doesnt work with, there must be something wrong with your router (also, check nat settings in asterisk) |
| 21:04.06 | pdugas | Kunsi, nothing changed on the * side in over a year :) |
| 21:04.17 | Kunsi | ok |
| 21:04.39 | pdugas | so I don't think the NAT settings are wrong but I'll revisit. Fishing for ideas like that. Thx. |
| 21:05.25 | pdugas | No port forwards or DMZ crap enable on the router. UPnP disabled. No filters. Just can't figure out what could have changed. |
| 21:05.54 | pdugas | It sort of feels like the NAT table in the router is foobar and not remapping the return traffic. |
| 21:06.14 | pdugas | No way to sniff out there so no whay to really know what's up. |
| 21:06.17 | pdugas | Grrr.... |
| 21:08.56 | Kalavera | hey guys I have an issue with incoming calls of one of our sip trunks. I can hear the end who answer but they cant hear me when I call. In the other hand they could her me at some point |
| 21:26.25 | *** join/#asterisk eofster (~eofster@ip5f5af385.dynamic.kabel-deutschland.de) |
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| 21:53.48 | philfry | I have an issue where phones use analog lines and when i try to dial the log shows it going out over g0 but the phone dialing isn't ringing and the destination never rings either. Where would you start? |
| 21:54.08 | philfry | sorry the outbound trunk is analog |
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| 22:32.30 | hdon | hi all :) how do i ask asterisk to validate configuration files? or reload them? is sighup enough? |
| 22:35.58 | hdon | oh ok i see |
| 22:36.05 | [TK]D-Fender | WHICH files? |
| 22:36.08 | hdon | so my init script has "reload" command which just runs asterisk -rx reload |
| 22:36.14 | [TK]D-Fender | * doesn't do much by way of "validation" |
| 22:36.16 | hdon | i see now what those options mean |
| 22:36.29 | hdon | so my question now is: what if my configuration is bad? will existing calls be ended? |
| 22:40.33 | hdon | can i write dialplans from the CLI? |
| 22:41.42 | mub | the asterisk CLI? why not just use nano or vim? |
| 22:42.32 | hdon | mub, well my earlier question was how safe it was to reload asterisk configuration -- do i risk ending current calls by doing this? |
| 22:42.35 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 22:42.40 | hdon | i would think the answer is "no" but i am not very experienced |
| 22:42.54 | mub | hdon: I'm the kind of person that would just reload and see |
| 22:43.05 | mub | then calmly explain to this channel why I don't deserve my position |
| 22:43.11 | hdon | mub, :3 |
| 22:43.26 | mub | I say go for it |
| 22:43.33 | hdon | oh no |
| 22:43.38 | hdon | the whole building has gone quiet |
| 22:43.43 | mub | oh no |
| 22:43.44 | hdon | now everyone is yelling! |
| 22:43.54 | mub | what really? |
| 22:43.57 | hdon | asterisk won't come back up! |
| 22:44.05 | hdon | pastebins error log |
| 22:44.06 | mub | oh shit, copy the old configs back over |
| 22:44.12 | mub | reload |
| 22:44.14 | hdon | i didn't make copies |
| 22:44.23 | mub | Y THO? |
| 22:44.24 | Kunsi | nub |
| 22:44.52 | hdon | http://asterisk.sourcearchive.com/ |
| 22:44.54 | hdon | whoops |
| 22:44.59 | hdon | http://pastebin.com/wwvdjvEj |
| 22:45.02 | hdon | there is my configuration |
| 22:45.22 | mub | That's a perfectly valid configuration AFAIK |
| 22:48.17 | hdon | oh my bad |
| 22:48.19 | hdon | but seriously |
| 22:48.24 | hdon | if i do "dialplan reload" |
| 22:48.27 | hdon | what's the worst case scenario? |
| 22:48.37 | hdon | i mean, asterisk should be able to continue serving existing calls, right? |
| 22:55.57 | [TK]D-Fender | Your next call will flow as it's coded |
| 22:58.12 | Kalavera | hey guys I have an issue with incoming calls of one of our sip trunks. I can hear the end who answer but they cant hear me when I call. In the other hand they could her me at some point |
| 22:59.15 | *** join/#asterisk jr220 (~jamesruss@184-89-242-81.res.bhn.net) |
| 22:59.58 | [TK]D-Fender | You need to be less vague than "some point" |
| 23:00.19 | [TK]D-Fender | Especially when talking about multiple calls over some random period of time |
| 23:00.27 | [TK]D-Fender | And the first step is always "show the call" |
| 23:00.38 | [TK]D-Fender | No body, no autopsy. |
| 23:00.40 | [TK]D-Fender | ~pb |
| 23:00.41 | infobot | A "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:00.42 | [TK]D-Fender | ^^^ |
| 23:00.50 | Kalavera | even thought configuratione didnt change and firewal was opened |
| 23:02.02 | Kalavera | [TK]D-Fender: I am a bit slow I am on the highway |
| 23:04.08 | [TK]D-Fender | Stop texting & driving then and come back when you can safely concentrate on this. |
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| 23:13.08 | [TK]D-Fender | heads out for a bit |
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