| 00:00.16 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 00:00.59 | [TK]D-Fender | basicer, "core show application extenspy" <- |
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| 00:15.12 | muanang | hi there! |
| 00:15.30 | muanang | can anyone help me how to fxotune? |
| 00:15.35 | muanang | thank you! |
| 00:23.56 | ChannelZ | fxotune -i -m 13 -l 1 |
| 00:24.00 | ChannelZ | or something like that. |
| 00:24.22 | ChannelZ | Actually.. disable your echo cancellers, restart DAHDI, then tune. |
| 00:24.45 | ChannelZ | Then re-enable, restart, make sure fxotune got run that loads the config it wrote. |
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| 00:42.49 | muanang | disable the echo? can i disable the echo by disabling the asterisk? |
| 00:47.24 | djgerm | Hello, I am using two asterisk instances as clients to another server, I want to test a TURN/STUN/ICE server, but I can't seem to get asterisk to direct its media to the ICE server for relay. Is there some configuration I am missing to set the ICE server ? |
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| 00:53.37 | ChannelZ | muanang, no turn off the echo cancellers in /etc/dahdi/system.conf (if I remember right) and restart the drivers - with asterisk shut down |
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| 01:44.38 | funtriaco | I'm creating a very simpler dialer... should i spawn the calls via php_agi or are call files still recommended ? |
| 01:49.18 | [TK]D-Fender | how exactly is AGI spawning calls? |
| 01:52.06 | funtriaco | well havent decided yet, maybe via the Asterisk Manager? |
| 01:52.26 | funtriaco | [TK]D-Fender: Asterisk Manager API Action Originate |
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| 01:52.58 | [TK]D-Fender | AMI is a way to Originate a call... |
| 01:53.11 | funtriaco | or, I was going to create a bunch of .call files as need |
| 01:53.13 | [TK]D-Fender | AGI isn't really... because that is an active channel... |
| 01:53.54 | funtriaco | I just havent seen yet a way to get the results back of a call started via .call files, how do i know if was successful or the trunks were all busy, etc. |
| 01:54.19 | funtriaco | I would like to have that kind of information, to know if to attempt to call the number again. |
| 01:54.35 | [TK]D-Fender | Have you read the docs on call files? |
| 01:54.43 | [TK]D-Fender | And AMI originate? |
| 01:54.58 | [TK]D-Fender | Both have signals for when they fail or a place to go if they do.... |
| 01:55.09 | [TK]D-Fender | (inherently) |
| 01:55.20 | [TK]D-Fender | You can do more yourself if the channel is Local. |
| 01:56.10 | funtriaco | been reading docs for both... got to a point that didnt know which one to committ too. |
| 01:56.47 | funtriaco | I think i'm going to attempt .call files as they seeem easier to implement |
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| 02:01.01 | [TK]D-Fender | they are kinda dead-easy lazy.. |
| 02:01.34 | [TK]D-Fender | Also depends how you want to regulate how many you do at a time, etc |
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| 02:11.37 | LemensTS | On Polycom's, if a call comes in you can hit hold and it will put first call on line key (on a 2 line phone). What happens if another line calls and you hit hold? |
| 02:13.22 | [TK]D-Fender | Should be "nothing |
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| 11:11.13 | Zogot | hey there |
| 11:11.34 | Zogot | quick question, is there a page that details some of the textual limitations of say, peer names? |
| 11:11.44 | Zogot | no 'dashes' or a character length or so? |
| 11:40.30 | demarcoz | win 3 |
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| 15:53.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 17:20.02 | linocisco | is Elastix.org down? sorry for off-topic here |
| 17:22.22 | eschmidbauer | Hello |
| 17:25.31 | linocisco | |
| 17:25.31 | linocisco | General Error |
| 17:25.31 | linocisco | SQL ERROR [ mysqli ] |
| 17:25.31 | linocisco | Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) [2002] |
| 17:25.31 | linocisco | An sql error occurred while fetching this page. Please contact an administrator if this problem persists. |
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| 17:30.21 | eschmidbauer | sounds like mysql server is down |
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| 17:59.18 | aruntomar | while upgrading asterisk from 11.x to 13.x series, is it also recommended to upgrade libpri and dahdi along with it? |
| 18:02.45 | rmudgett | Yes you should since Asterisk uses those libraries |
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| 18:03.13 | rmudgett | Though if you aren't using those modules then it doesn't matter. |
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| 18:21.33 | aruntomar | rmudgett: yes, using the libpri and dahdi modules. they are old versions and working, but facing some issues lately. |
| 18:22.04 | aruntomar | so this was one of the thought, that i should upgrade the pri and dahdi libraries as well. |
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| 18:48.03 | overyander | how can i check CID(num) against a pattern? I'm wanting to disallow foreign countries as needed as well as some other weird numbers i'm seeing like 9091111111 |
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| 19:01.19 | mahmoud | hi - is there any DECT-to-VoIP base station that does NOT suck? |
| 19:02.31 | zekoZeko | define not suck |
| 19:02.42 | mahmoud | something that is NOT Gigaset |
| 19:03.02 | zekoZeko | my 3 users using that are happy :) |
| 19:03.09 | mahmoud | which Gigaset? |
| 19:03.28 | zekoZeko | 310, 510, don't know, would have to check |
| 19:03.42 | zekoZeko | i was however told to not use more than one handset per base station |
| 19:03.46 | mahmoud | their website blows my mind. their compatibility page is not made for humans. it's outdated and doesn't list many of their products |
| 19:04.36 | mahmoud | is there any professional DECT thing that is worth my investment? something that gives me peace of mind with lots of compatbility and nice vendor that can answer my emails and treat me like a good pet |
| 19:05.22 | zekoZeko | can't help you there, sorry. not much experience with those. |
| 19:06.06 | mahmoud | Snorm? |
| 19:07.20 | mahmoud | POLYCOM? |
| 19:13.07 | mahmoud | now that after years since freeswitch is made, how does it stack against asterisk? |
| 19:17.50 | *** join/#asterisk aruntomar (~aruntomar@123.201.76.229) |
| 19:20.34 | drmessano | I look Asterisk vs FreeSWITCH as Sendmail vs Postfix |
| 19:21.06 | drmessano | Depending on how hard you look they are either completely different or completely the same |
| 19:21.16 | mahmoud | which one is the sendmail? |
| 19:21.39 | drmessano | Depends which you find better |
| 19:21.46 | mahmoud | which one do you find better? |
| 19:21.50 | drmessano | I prefer Asterisk. I prefer Postfix |
| 19:22.02 | drmessano | Some people swear by Sendmail |
| 19:22.18 | drmessano | Some people still think Windows is kinda cool |
| 19:22.25 | mahmoud | so in that example, you were saying that asterisk to voip is like postfix is to eamil? |
| 19:22.57 | drmessano | Both * and FS are mature, and while they do have overlapping features, FreeSWITCH and Asterisk have a different audience |
| 19:23.57 | drmessano | Freeswitch was never meant to replace Asterisk, it was just meant to do some things better.. Things that Asterisk now does much better.. like from a perfornance standpoint |
| 19:24.36 | mahmoud | dude please don't write one more single word unless you know what you are spewing by your terrible rubber dume keyboard |
| 19:24.45 | drmessano | FreeSWITCH could murder Asterisk when it came to concurrent calls.. back in the Asterisk 1.4 days. But much has changed.. Maybe it still handles calls better, but not as murderously of a difference |
| 19:25.22 | drmessano | Say what? |
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| 19:30.57 | mahmoud | it seems you know nothing about the current state. all you said is boring old history. please don't waste my time and standby until someone who knows speaks |
| 19:31.24 | mjordan | wow. That's pretty rude. |
| 19:31.24 | drmessano | You dont have to be an asshole |
| 19:32.06 | drmessano | Maybe you should learn how to talk to people |
| 19:32.20 | mjordan | mahmoud: perhaps you should go read this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Community+Code+of+Conduct |
| 19:32.22 | *** join/#asterisk ghostlin_ (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 19:32.43 | mahmoud | i remember asterisk being a rude community years ago. is that changed? |
| 19:33.14 | drmessano | Apparently you haven't |
| 19:33.49 | mahmoud | you are mixing things, drmessano -- i'm the good guy who was trying to increase the SNR by educating you to not speak when you don't know |
| 19:33.56 | mjordan | mahmoud: I don't really know or care what happened at any point in the past. For quite some time now, we've taken a stance that trolling people or generally being rude to others isn't something that any of us feel like participating in. |
| 19:34.03 | drmessano | good guy? |
| 19:34.19 | mjordan | mahmoud: I don't take it that way, and I doubt drmessano did either. |
| 19:34.44 | mahmoud | mjordan: noted. i'll obey the law of this channel. apologies |
| 19:35.09 | zamba | haha |
| 19:35.15 | drmessano | Actually mahmoud, I was trying to educate you on the difference and how Asterisk and FreeSWITCH have their strengths.. it's a difficult comparison. One is not better than the other, even CURRENTLY |
| 19:35.19 | drmessano | So STFU |
| 19:35.29 | zamba | what a complete assh*le |
| 19:35.32 | mahmoud | lol now drmessano is no obeying the law |
| 19:35.48 | drmessano | Im not? |
| 19:37.52 | drmessano | STFU = Say, Thanks For Understanding .... I also attempted to educate you and you were rude. But this is silly and pointless. You want a solid answer, fine.. FreeSWITCH is better. ---> /join #freeswitch |
| 19:38.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:38.34 | mjordan | likes Asterisk, but that's just me |
| 19:38.59 | drmessano | mjordan, shushhhhhh.. I was trying to divert |
| 19:39.01 | mjordan | To be frank, I find the constant comparisons tiresome. I'm not sure why you can't have two good open source communications projects. |
| 19:39.05 | eschmidbauer | good to see the flame wars continuing |
| 19:39.08 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 19:39.13 | mjordan | Why would I flame them? |
| 19:39.24 | mjordan | shrugs |
| 19:39.28 | eschmidbauer | no-- FS vs * flame wars |
| 19:39.43 | drmessano | [15:35:15] <drmessano> Actually mahmoud, I was trying to educate you on the difference and how Asterisk and FreeSWITCH have their strengths.. it's a difficult comparison. One is not better than the other, even CURRENTLY <---- |
| 19:39.44 | eschmidbauer | an age old rivalry |
| 19:39.47 | mjordan | that's my point; I don't think there is one, other than people feel like constantly trying to create one |
| 19:39.49 | drmessano | A question was asked |
| 19:39.51 | mjordan | I don't have a rivalry with them |
| 19:39.54 | mjordan | I've got enough things to worry about |
| 19:40.10 | eirirs | isnt it like the typical linux distro flamewars? |
| 19:40.12 | drmessano | The answer is the same as it always has been.. They are different.. |
| 19:40.22 | mjordan | shrugs |
| 19:40.22 | eschmidbauer | i think asterisk was originally designed for a single pbx |
| 19:40.36 | mjordan | eschmidbauer: I'd disagree with that. |
| 19:40.40 | eschmidbauer | of course it's much more than that now |
| 19:40.44 | mjordan | I'd say most people tend to deploy it in that fashion. |
| 19:41.01 | mjordan | But there's been "stuff" in it for over a decade that dealt with clusters, larger systems, routing, etc. |
| 19:41.02 | eschmidbauer | i mean originally, like when what's his name wrote it back in 90s |
| 19:41.08 | mjordan | so, in 1999? |
| 19:41.11 | eschmidbauer | yes |
| 19:41.17 | mjordan | I'll grant that for one year, it might have not been. |
| 19:41.21 | drmessano | lol |
| 19:41.25 | eschmidbauer | <PROTECTED> |
| 19:41.26 | mjordan | I don't know. The source control history is kind of murky. |
| 19:46.56 | drmessano | Personally, i've always stuck with Asterisk because it's familiar, it's continued to grow and mature, and it does everything I need. Same reason I continue to use Apache when all the cool kids use nginx or whatever. Apache has gotten MUCH faster in the last few releases, nulling most of the "Apache is old and sucky" arguments |
| 19:47.39 | drmessano | Asterisk went through a few awesome growth spurts after FreeSWITCH came out and all the cool kids proclaimed Asterisk a deadlock filled mess |
| 19:47.51 | drmessano | But it's better now, and shit still works |
| 19:49.40 | eschmidbauer | which is how FS was spawned |
| 19:49.41 | mjordan | eschmidbauer: yes it was. |
| 19:49.41 | mahmoud | sigh. you guys must have used to the low signal-to-noise ratio. this is so tiring |
| 19:49.41 | mjordan | eschmidbauer: the Asterisk you know today is 2.0. In 10, the media core was rewritten. With 12+, we've pretty much gutted a good chunk of it again. |
| 19:49.42 | eschmidbauer | i agree |
| 19:49.42 | mjordan | It may surprise you, but there's actually quite lot of work that happens in the internals. We just try hard not to break people. |
| 19:49.42 | eschmidbauer | the creator of FS was an asterisk dev |
| 19:49.42 | mjordan | yes he was |
| 19:49.42 | eschmidbauer | he wanted to rewrite it from the ground up |
| 19:49.44 | mjordan | yes he did |
| 19:49.44 | eschmidbauer | asterisk community did not want to, so he started FS |
| 19:49.44 | mjordan | I'd disagree with part of that |
| 19:49.44 | mjordan | Some folks did not want to do the changes he wanted to do at that point in time |
| 19:49.44 | eschmidbauer | the powers that be did not want to |
| 19:49.47 | mjordan | Sure, at that point in time. That doesn't mean that things haven't been changed, rearchitected, or reworked since then. |
| 19:49.48 | mjordan | And as far as I'm concerned, Asterisk is an open source project. |
| 19:49.48 | eschmidbauer | i'm not saying they haven't |
| 19:49.48 | mjordan | If you want to make a change, patches welcome. |
| 19:49.48 | mjordan | And it's been that way for a long time. |
| 19:49.58 | eschmidbauer | i think the original sip stack for asterisk was pretty poor |
| 19:49.59 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ncjffsnjvzmzjctf) |
| 19:49.59 | eschmidbauer | not sure if that was ever rewritten, i know chan pjsip exists but the license for pjsip sucks |
| 19:49.59 | eschmidbauer | FS used the nokia sip stack sofia which is pretty tight |
| 19:49.59 | *** join/#asterisk superscrat (~asanders@23-124-69-52.lightspeed.brhmal.sbcglobal.net) |
| 19:51.27 | drmessano | kisses and hugs mahmoud |
| 19:51.37 | drmessano | You need a hug.. I get it |
| 19:51.58 | mahmoud | that's done. you are in my ignore list. |
| 19:55.05 | *** join/#asterisk infobot (ibot@rikers.org) |
| 19:55.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:55.05 | *** part/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 19:55.24 | eschmidbauer | i still prefer nginx in low volume |
| 19:55.46 | drmessano | To each their own |
| 19:55.49 | eschmidbauer | php-fpm + nginx isso much easier to get setup than apache + php-fpm |
| 19:56.21 | eschmidbauer | feel like apache tries to do way more than it should |
| 19:56.45 | eschmidbauer | nginx is like 1 config file that's 30 lines long, no modules |
| 20:02.42 | [TK]D-Fender | overyander: "core show function REGEX" |
| 20:05.07 | mahmoud | any dect-to-sip recommendations? |
| 20:11.23 | eschmidbauer | mahmoud: i've used the polycom kirk-wireless |
| 20:11.25 | eschmidbauer | it's not bad |
| 20:11.30 | eschmidbauer | the panasonic models are nice too |
| 20:16.53 | mahmoud | any thoughts on how they compare to gigaset? |
| 20:18.02 | eschmidbauer | never used gigaset |
| 20:18.26 | mahmoud | how old is your polycom |
| 20:18.48 | *** join/#asterisk linocisco (~linocisco@111.84.193.131) |
| 20:18.53 | linocisco | hi all |
| 20:19.14 | linocisco | i have configured NTP server but phones are not getting correct time and date |
| 20:19.23 | linocisco | what do I do? |
| 20:22.33 | [TK]D-Fender | Configure your phones properly to actually use it |
| 20:22.33 | tuxd00d | linocisco: time zone? |
| 20:22.34 | mahmoud | why are dect vendors so terrible at making HTML pages that list their products |
| 20:22.48 | *** join/#asterisk PHunter (~Phunter@wsip-68-14-213-118.ph.ph.cox.net) |
| 20:22.50 | linocisco | tuxd00d, yes |
| 20:23.17 | tuxd00d | [TK]D-Fender is likely correct. |
| 20:23.41 | PHunter | Anyone have any tips on resolving sip_autodestruct warnings? I have a ton of them going on right now, and it appears to be impacting calls.. |
| 20:23.56 | [TK]D-Fender | This isn't *'s job |
| 20:23.58 | PHunter | I core set verbose 9 and not seeing other errors. |
| 20:24.21 | tuxd00d | PHunter: Debug? |
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| 20:25.18 | [TK]D-Fender | phIt isn't |
| 20:25.18 | PHunter | just enabled debug @ 5 |
| 20:25.23 | [TK]D-Fender | PHunter: It isn't |
| 20:25.40 | [TK]D-Fender | And that is just a message, not an error |
| 20:25.53 | PHunter | Well something is happening along side that warning.. |
| 20:26.11 | PHunter | because our call center is saying they arent getting calls in the queue |
| 20:26.47 | PHunter | This hasnt always happened. This is new. |
| 20:26.48 | [TK]D-Fender | Go show actual calls going to a queue then, member dumps, queue definitions, etc |
| 20:26.57 | [TK]D-Fender | beacuse that message is not relevant |
| 20:27.41 | PHunter | I have to wait until it stops serving them calls again. I keep bouncing asterisk when it does because something is locking up. |
| 20:27.45 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 20:28.32 | PHunter | http://pastebin.com/ryR0cDun |
| 20:28.36 | PHunter | I dont think that is normal. |
| 20:29.24 | [TK]D-Fender | You aren't looking at SIP debug se we can't tell what comm that's even related to. |
| 20:30.07 | PHunter | Its kind of hard for me to even see the core debug/verbosity because its already scrolling at unreadable speeds. |
| 20:31.00 | [TK]D-Fender | Well if you want to debug things... you have to actually look. That's just how it works... |
| 20:34.18 | PHunter | [TK]D-Fender: I understand that. However when you enable sip debug when you have 80+ active devices and 20-25 active lines, its kind of hard to monitor in SSH |
| 20:35.10 | [TK]D-Fender | May be hard. Still is necessary. |
| 20:35.16 | PHunter | http://pastebin.com/hszgkDBd |
| 20:37.44 | [TK]D-Fender | PHunter: [Apr 14 13:34:21] NOTICE[7655]: chan_sip.c:28984 check_rtp_timeout: Disconnecting call 'SIP/tpllc.204-00000032' for lack of RTP activity in 31 seconds |
| 20:37.51 | [TK]D-Fender | THAT .. is a problem? |
| 20:37.56 | [TK]D-Fender | so, no "?" there. |
| 20:37.59 | [TK]D-Fender | it IS a probl;me |
| 20:38.10 | [TK]D-Fender | sorry* |
| 20:38.18 | [TK]D-Fender | gah... it's late... checkout time here.. |
| 20:38.28 | PHunter | Well I know lack of RTP is a problem. But when I tell my CTO its his brainchild server setup |
| 20:38.31 | PHunter | he says |
| 20:38.33 | PHunter | "no" |
| 20:38.47 | PHunter | He refuses to acknowledge its a RTP issue |
| 20:38.58 | PHunter | hes had me look at CPU, Memory and disk usage |
| 20:39.00 | PHunter | etc |
| 20:39.05 | [TK]D-Fender | bad case of denial |
| 20:39.11 | [TK]D-Fender | That's a call-drop situation |
| 20:39.11 | PHunter | Welcome to my job. |
| 20:39.16 | [TK]D-Fender | With hard proof. |
| 20:39.28 | [TK]D-Fender | ClueBat (tm) time |
| 20:39.38 | PHunter | Yeah, well my boss sent me to get DCAA because he couldnt figure out ASterisk at all |
| 20:39.47 | [TK]D-Fender | You are 2 versions behind... should get that while you're at it |
| 20:40.33 | [TK]D-Fender | ok, gotta head out of here... but that is 1 real issue we can see there... |
| 20:40.41 | [TK]D-Fender | I'd expect a bad reinvite, etc |
| 20:40.52 | [TK]D-Fender | drill it for a bit... |
| 20:40.56 | [TK]D-Fender | heads home |
| 20:41.01 | PHunter | thanks |
| 20:51.22 | *** join/#asterisk ctpdump (~tcpdump@2602:ffda:da:2:216:3eff:feae:d69c) |
| 20:55.50 | mahmoud | can i register any DECT phone to my DECT base? my DECT base has a SIP leg. will any DECT phone able to register AND call SIP via the base station? |
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| 21:17.10 | qakhan | hi all, php script is taking 10sec to execute, while same php script is working fine on other linux machine |
| 21:21.59 | Micc | If I set a small range for rtp ports and I run out of ports in that range, will asterisk tell me, or will things just start not working? |
| 21:23.15 | WIMPy | Both |
| 21:24.53 | Micc | As long as it tells me, I'll know how to fix it. |
| 21:25.25 | Micc | But I'm assuming 400 ports per server is good enough for 80 calls per server. |
| 21:36.11 | [TK]D-Fender | it will tell you it's not working anymore :) |
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