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| 00:00.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 00:19.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:48.34 | gusto | so |
| 00:48.43 | gusto | someone still awake? |
| 00:56.16 | dr4c4n | awake? |
| 00:59.02 | gusto | yes |
| 01:00.54 | dr4c4n | I'm awake |
| 01:01.02 | dr4c4n | but it's not actually that late for me |
| 01:01.14 | dr4c4n | i want to setup asterisk |
| 01:01.22 | dr4c4n | but haven't yet, so mostly lurking in here |
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| 01:11.22 | WIMPy | We are all zombies. |
| 01:11.41 | dr4c4n | om nom braiiiiiiiiiiiiins |
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| 01:29.25 | igcewieling | No wonder this customer has so many problems! the asset code is "0666" |
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| 02:11.15 | Khronos | Hi guys. |
| 02:11.49 | Khronos | Trying to hook an asterisk server into a sangoma t1 gateway and the gateway isn't passing the incoming callerid to the t1 jack and also the pbx isn't able to send callerid out. |
| 02:14.40 | Khronos | OUtgoing and incoming calls work just callerid passing is my only issue right now. |
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| 02:42.52 | saint_ | is there a way to have a group of phones setup (like GROUP1=SIP/xxx&SIP/yyy) so I can Dial(${GROUP1}) instead of dialing all of the extensions I need ? |
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| 02:45.31 | saint_ | Ha.. I found dialgroup.. perfect. |
| 02:46.51 | saint_ | ha no.. never mind.. damn it |
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| 06:08.18 | bulkorok | hi |
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| 08:08.22 | zamba | i'm getting lots and lots of these: asterisk[30909]: NOTICE[30923]: chan_sip.c:22546 in handle_request_invite: Sending fake auth rejection for device 200<sip:200@<myownipaddress>;tag=dc614377 |
| 08:08.43 | zamba | allowguest=no |
| 08:08.46 | zamba | alwaysauthreject=yes |
| 08:08.48 | zamba | i have those two set |
| 08:10.33 | kaldemar | that's why you get the notices. |
| 08:11.15 | zamba | well, it's filling up my logs.. anything i can do about it? |
| 08:11.34 | kaldemar | don't log notices. |
| 08:12.03 | kaldemar | or upgrade to 11.3.0 or newer where that notice is not seen anymore. |
| 08:21.37 | *** join/#asterisk xoveruk (~user@196.200.85.77) |
| 08:21.42 | xoveruk | hi |
| 08:22.08 | xoveruk | Why would a server after a while decide to send IAX2 traffic to port 1024 instead of 4569, it seems to happen periodically |
| 08:24.40 | ketas | holy s... provider said that sip login password cannot be changed! |
| 08:25.35 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:25.47 | ketas | "could you describe why you need it" |
| 08:26.28 | ketas | why should user describe the reason for password change... |
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| 08:36.40 | *** join/#asterisk polysics (~Adium@host86-1-dynamic.48-82-r.retail.telecomitalia.it) |
| 08:36.46 | polysics | hello there |
| 08:36.51 | polysics | still toying with CONNECTEDLINE |
| 08:36.58 | polysics | and still seemingly not owrking :-) |
| 08:38.56 | polysics | what should CONNECTEDLINE do on set? send some SIP packet? |
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| 09:12.12 | WIMPy | If it's done on a SIP channel, yes. |
| 09:13.08 | Greenlight | polysics: I think it also depends if the SIP endpoint supports it |
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| 09:13.50 | WIMPy | Isn;t it just an invite? |
| 09:14.10 | Greenlight | If you change it during the call though |
| 09:14.27 | Greenlight | From what I glanced over when I tried it last week, it seemed some softphones didn't support it |
| 09:15.30 | Greenlight | It certainly didn't work when I tried it |
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| 09:23.09 | polysics | yeah, it does not seem to be the case here either |
| 09:23.20 | polysics | I think I will wait until I get my Grandstream phone |
| 09:23.55 | Greenlight | Yea, I didn't spend much time on it, and decided when I've more time to play around I'll check with a proper SIP handset rather than xlite softphone |
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| 09:33.41 | Arhur70 | hi, I have a question about patlooptest. |
| 09:34.27 | Arhur70 | in short it seems ok but I don't see any interrupts on /proc/interrupts |
| 09:34.49 | Arhur70 | it is normal behaviour? |
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| 09:38.15 | StaRetji | Howdy folks |
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| 12:03.08 | polysics | if I know a peer (SIP/300), can I find out if he is in any channels currently and their names? |
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| 12:13.24 | Greenlight | polysics: asterisk -rx "core show channels" | grep SIP/300 |
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| 12:23.38 | polysics | great, thanks |
| 12:25.09 | Arhur70 | anyone on my question? ;) thx |
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| 12:30.00 | kaldemar | polysics: if you want to use the CLI command and always want the full channel name, use "core show channels concise" instead of "core show channels". |
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| 12:38.45 | BorjaGVO | Hi everyone. I need documentation about AMI events and its possible header values (http://pastebin.com/PYzErDYq). Anyone know where can I find it? Thank you! |
| 12:39.20 | Greenlight | The wiki and sourcecode are your best bets |
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| 12:50.34 | Greenlight | BorjaGVO: Or, from my AMI code: http://pastebin.com/qfpXt8hV |
| 12:50.54 | Greenlight | Not an exhaustive list |
| 12:52.05 | BorjaGVO | Greenlight: Where did you get that from? |
| 12:52.47 | Greenlight | From my code that deals with the AMI. I created that a few years back, so not sure where I originally found the defenitions, sorry |
| 12:54.04 | Greenlight | http://www.voip-info.org/wiki/view/asterisk+manager+events#QueueMemberStatusEvent |
| 12:54.28 | BorjaGVO | Greenlight: I see...no problem. However, what kind of managin do you do with AMI? Why do you need that? |
| 12:54.46 | Greenlight | or, better yet, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_QueueMemberStatus |
| 12:54.50 | Greenlight | Google is your friend :) |
| 12:54.54 | BorjaGVO | (just curiosity...want to learn :-)) |
| 12:55.39 | Greenlight | We manage placing and tracking calls via the AMI |
| 12:55.46 | Greenlight | Including tracking calls in queues etc |
| 12:56.08 | *** join/#asterisk serafie1 (~erin@nat/digium/x-bivujgydpfkfnfuq) |
| 12:56.09 | Greenlight | So we've out own internal objects and we keep that in sync with what's happening inside asterisk |
| 12:56.17 | mirela666 | http://goo.gl/f5SUQ :D |
| 12:58.35 | BorjaGVO | mirela666: yeah, google will help in many cases...but thought I had to ask the experts :P |
| 13:01.46 | mirela666 | BorjaGVO: just wanted to use lmgtfy.com, no hard filings :D |
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| 13:04.13 | BorjaGVO | mirela666: lol..sure |
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| 13:52.35 | [gnubie] | waves |
| 13:53.36 | carrar | waves back |
| 13:54.31 | [gnubie] | i just installed asterisk v11.4.0 on my wheezy i386 box and it's my first time to build it from scratch. |
| 13:55.22 | carrar | always the best way |
| 13:55.35 | [gnubie] | the last time i setup asterisk was on squeeze but the asterisk packages were from digium's apt repository. it was on v1.8 |
| 13:56.03 | [gnubie] | now, i am slowly porting my v1.8 configs to v11.4 system |
| 13:58.07 | [gnubie] | first thing i noticed that i can't make voicemail and gtalk does not work. |
| 14:04.12 | [gnubie] | it does not send the voicemail created as urgent via exim4/smarthost to gmail. |
| 14:06.41 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 14:06.42 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:07.05 | izbushka | hi |
| 14:07.53 | izbushka | is it possible to inform user about new voicemail by voice before dial() extension? |
| 14:09.48 | izbushka | so how i could check for new voicemail in dialplan? |
| 14:12.29 | carrar | VM_INFO() ? |
| 14:13.45 | izbushka | carrar, i'll try. thanks |
| 14:16.59 | carrar | there is also vmcount() same but returns less |
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| 14:25.42 | ChrisInSydney | hey all |
| 14:26.12 | ChrisInSydney | Has anyone got Cisco 7912s working OK with a SIP image ?? |
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| 14:45.50 | danfromuk | Is it possible to have more than one periodic announcement in a queue? |
| 14:46.11 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.132) |
| 14:46.52 | Greenlight | danfromuk: Yes |
| 14:47.41 | Greenlight | Comma separate them in periodic-annouunce |
| 14:47.55 | Greenlight | And you can also specify random-periodic-announce=yes if you choose |
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| 15:03.05 | BorjaGVO | Hi people. I know that this is just Asterisk channel, but no one at #freepbx nor forum will help me with this: |
| 15:03.18 | BorjaGVO | Hi all. I have a problem. I have nearly 80 queues. All of them have members (queue agents) belonging to a remote peer (another PBX). When I set penalties for the different agents, everything works fine if I set maximum priority to extensions configured in the same server where queue is configured. However, if a invert the priorities and set to try first the extensions in the other peer, queue scheduling doesn't work...doing thing |
| 15:03.24 | drkat | Hey guys, anyone have any issues with cisco small business sg-300 provisioning voice vlan to polycom 335's? |
| 15:03.30 | BorjaGVO | ...soon |
| 15:03.39 | BorjaGVO | I found this article: http://asterisk.net.ru/en/2010/02/27/heavy-queue-usage-in-freepbx/ |
| 15:03.42 | *** join/#asterisk boscage (boscage@71-12-76-139.dhcp.gnvl.sc.charter.com) |
| 15:03.49 | BorjaGVO | The thing is that I have no information about external extensions state. I think that's the whole reason for the problem. |
| 15:03.57 | BorjaGVO | What do you think? Anything that I'm missing or something that can be done to solve this and I didn't realize? |
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| 15:04.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:07.00 | *** part/#asterisk polysics (~Adium@host86-1-dynamic.48-82-r.retail.telecomitalia.it) |
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| 15:11.23 | [TK]D-Fender | BorjaGVO: http://www.voip-forum.com/asterisk/2011-01/distributed-presence-asterisk/ |
| 15:11.43 | [TK]D-Fender | BorjaGVO: Naturally this would probably be a serious pain to try to bring into FreePBX |
| 15:12.31 | BorjaGVO | Thanks [TK]D-Fender . I'm just realizing now that follow-me module might help |
| 15:12.35 | BorjaGVO | What do you think? |
| 15:15.09 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:15.15 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
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| 15:19.20 | [TK]D-Fender | BorjaGVO: You need presence state. Follow-me does nothign to help that |
| 15:23.26 | f843d0 | Greetings, is it possible to reserve a certain amount of "lines" toward one or more extensions? I'm not talking about "call-limit", let's say I want at least n calls for xxxx, m calls for yyyy, and zzzz would return "busy" if n+m reached a certaing amount... |
| 15:27.34 | *** join/#asterisk boscage (boscage@unaffiliated/boscage) |
| 15:27.45 | Qwell | f843d0: Put your limiting extensions in a group, then on the limited extension, check the group count. |
| 15:27.57 | [TK]D-Fender | f843d0: It's your dialplan. Sure... |
| 15:31.33 | SuperNull | anyone do video calling ? im just curious |
| 15:34.22 | [TK]D-Fender | SupI've done it one or twice just to see, |
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| 15:41.41 | f843d0 | Qwell [TK]D-Fender : ok thanks, I'll try that out |
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| 15:46.26 | Katty | hellllooooooooooooo nurse. |
| 15:46.42 | carrar | WHA |
| 15:46.42 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
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| 15:47.14 | [TK]D-Fender | T?? |
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| 15:48.42 | Katty | i have /no/ idea. |
| 15:48.50 | Katty | i cannot be held responsible for the things i say. |
| 15:49.08 | carrar | vim version 7.3.1005 |
| 15:49.12 | carrar | updated!! |
| 15:49.17 | Katty | wooot! |
| 15:49.24 | Katty | my hair could use an update. |
| 15:49.26 | carrar | love those version numbers |
| 15:50.31 | Katty | i love caffeine. |
| 15:50.44 | Katty | i would sleep with it. |
| 15:50.47 | carrar | HAI! |
| 15:50.52 | carrar | You can |
| 15:50.56 | Katty | no, i can't. |
| 15:50.58 | Katty | i'm up all night. |
| 15:51.03 | carrar | put some in bottles |
| 15:51.08 | carrar | lay them under your pillow |
| 15:51.09 | Qwell | http://www.zazzle.com/molecular_structure_of_caffeine_pillow-189319383844917424 |
| 15:51.11 | Qwell | The hell you say? |
| 15:51.13 | Katty | oh, i suppose i could do that. |
| 15:51.18 | Katty | YES. i will take 2 |
| 15:51.27 | Katty | hugs on Qwell |
| 15:51.29 | Qwell | http://www.etsy.com/listing/69710679/caffeine-pillow-molecular-structure |
| 15:51.32 | Qwell | cheaper, better IMO |
| 15:51.33 | Katty | Qwell: how's the lady? |
| 15:51.37 | Qwell | Katty: quite well |
| 15:51.43 | Katty | carrar: and how's the mrs? |
| 15:51.44 | Katty | misses. |
| 15:51.46 | Katty | mrses? |
| 15:51.52 | carrar | They are good |
| 15:51.56 | gorkish | Hello everyone; I have done a lot of testing w/ Polycom SPIP/VVX phones and TLS+SRTP and believe I have identified a regression bug but wanted to seek some feedback before I file an issue. Does anyone know of anybody successfully using Polycom phones with Asterisk 11 with TLS and SRTP? |
| 15:51.58 | carrar | eating breakfast it seems |
| 15:52.04 | Katty | Qwell: wooot! |
| 15:52.58 | Katty | carrar: sounds tasty. |
| 15:53.05 | Katty | and how is my favorite fender bender. |
| 15:53.14 | [TK]D-Fender | Katty: Mew. |
| 15:53.23 | [TK]D-Fender | Katty: meh (tm) |
| 15:53.24 | *** join/#asterisk Vince-0 (~quassel@196-209-246-90.dynamic.isadsl.co.za) |
| 15:53.27 | Katty | you had a thing yesterday, right? |
| 15:53.41 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.133) |
| 15:53.42 | [TK]D-Fender | Always something |
| 15:53.55 | carrar | a thing? |
| 15:54.02 | Katty | one of thsoe open mic things |
| 15:54.05 | carrar | oh |
| 15:54.09 | carrar | poem night? |
| 15:54.10 | Katty | where he makes funny faces |
| 15:54.13 | carrar | heh |
| 15:54.13 | Katty | and generally sings to himself |
| 15:54.20 | Katty | right? ^_^ |
| 15:54.21 | Qwell | Katty: How's the Mr.? Mister...y? |
| 15:54.28 | [TK]D-Fender | Was supposed to be band practice, and then arriving late to the open mic/ jam I co-host, but I SNAFU'd and band was only for next week so I arrived extra early. |
| 15:54.33 | Katty | Qwell: well it's umm. yeah. |
| 15:54.53 | [TK]D-Fender | Qwell: Kyrie Elaison down the road that I must travel! |
| 15:55.10 | Katty | Qwell: mostly wonderful (= |
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| 16:12.19 | gorkish | anyone re: polycom/srtp? Is there a better place to ask before i submit to the issue tracker? |
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| 16:22.56 | mjordan | gorkish: what version of 11 |
| 16:23.10 | mjordan | gorkish: and what issue are you seeing |
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| 16:24.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:25.14 | gorkish | mjordan: 11.4.0 newest polycom firmware. When polycom puts call on hold then resumes call no more SRTP audio. * shows the error [May 22 22:23:32] WARNING[17884][C-00000000]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10 |
| 16:26.44 | *** join/#asterisk johnwigley (91fff1ec@gateway/web/freenode/ip.145.255.241.236) |
| 16:26.56 | gorkish | tracing/decrypting the session i see the polycom resume SRTP to asterisk but packets are failing SRTP authentication. Packets look good in wireshark; sequence number increments properly and the auth tag appears correct after resume from hold |
| 16:27.34 | mjordan | hm |
| 16:27.44 | mjordan | do you get spammed with the unprotection errors, or is it reasonably constant? |
| 16:27.53 | *** join/#asterisk Freeaqingme (~Freeaqing@2a03:f80:ed15:37:235:56:92:7100) |
| 16:28.06 | gorkish | every 2s (the auth error does not trigger the retry code in res_srtp) |
| 16:28.57 | Freeaqingme | Hi folks. Is anybody aware of a somewhat uptodate debian (ubuntu) repository? I found this one, but it seems kinda out of date.. http://packages.asterisk.org/deb/pool/main/a/asterisk/ |
| 16:29.08 | mjordan | nor should it really |
| 16:29.19 | mjordan | sounds like the unprotect errors are on RTCP packets |
| 16:29.41 | gorkish | mjordan: they are not; they are on SRTP packets |
| 16:29.45 | mjordan | gorkish: go ahead and report the issue. The real trick is to get the polycom models, firmware versions, and a pcap of the invite requests |
| 16:29.56 | mjordan | well, it isn't on the audio |
| 16:30.02 | mjordan | rtcp can be protected as well |
| 16:30.56 | gorkish | mjordan: Yes I know; I have a fully decrypted pcap trace so i am looking at essentially unprotected packets. the test in res_srtp limits the error to display only every 100 frames |
| 16:31.26 | johnwigley | Hi, experiencing a weird issue with an asterisk server, at 1 minute past the hour calls are often going silent but not dropping. Listening to the call recordings shows that one side of the audio drops out but instead of ending abruptly you can hear it fade out on the recording mid sentence and then stay silent until one party ends the call. There are no audible artifacts at all indicating packet loss or breakup. Any ideas as t |
| 16:31.29 | gorkish | mjordan: I have not had time to test other versions but it worked correctly in * 1.8.11 with same libsrtp and same polycom firmware |
| 16:31.57 | gorkish | mjordan: i will do some further testing to attempt to isolate the issue if you think submitting an issue is the correct approach at this point |
| 16:32.20 | johnwigley | When it happens it seems to happen to all calls simultaneously or within a few seconds, but only to one side of the call, on the recording you can hear the other party asking what's happening. |
| 16:32.31 | mjordan | well, it's certainly supposed to handle a basic SRTP call, and there's been some annoyances with SRTP in a few different issues as of late. It wouldn't shock me that the hold/unhold for certain SRTP negotiations got screwed up again. |
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| 16:33.16 | johnwigley | The fade out on the recording is strange it's as if the call volume is being dropped to zero over a second or so rather than an abrupt cut off. |
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| 16:34.51 | gorkish | mjordan: yes i have been through the closed issues and they dont seem to be related; there was formerly an issue with polycoms resetting the srtp sequence number upon resuming from hold but that is not the issue. my current line of thinking is that asterisk is dropping the srtp key on the reinvite when the call comes off of hold but i have not tested this theory yet |
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| 16:39.58 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:43.53 | drmessano | gorkish, https://issues.asterisk.org/jira/browse/ASTERISK-19609 |
| 16:43.54 | LieutPants | [ASTERISK-19609] [Status: Open] [Assigned: xaled] SRTP to RTP bridging with two crypto lines in SDP does not work - https://issues.asterisk.org/jira/browse/ASTERISK-19609 |
| 16:46.43 | gorkish | drmessano: i have configured the polycom UA's to only make one offer in a=crypto: and tried with both AES_CM_128_HMAC_SHA1_32 and AES_CM_128_HMAC_SHA1_80 individually. i will double check that this issue is not related |
| 16:47.20 | igcewieling | gorkish: Are you running the latest polycom firmware |
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| 16:48.22 | gorkish | igceweiling: yes. further, the same configuration works with at least one older version of asterisk (1.8.11) correctly but i have not had time to search for where the regression occured |
| 16:49.52 | gorkish | we try to stay on the LTS releases so I will test the latest from the 10 branch and based on the results there start a poor-mans binary search |
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| 16:50.47 | gorkish | since i skipped that release i have no idea where between 1.8 and 11.4.0 things went south |
| 16:51.23 | gorkish | actually i dont even know if 1.8.22 is OK or not.. so i will do all of that before filing an issue report |
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| 17:21.19 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:23.05 | danfromuk | Hi. We've got a client that receives calls using g729. Calls were working fine. The DID provider sent calls to our servers and we passed them through to the sip peer. However, calls are now failing with SIP error 488 Not Acceptable Here. It appears to be a codec issue but can't see what the problem is. |
| 17:23.22 | danfromuk | The DID provider insists that they havent changed anything, and we havent changed anything. |
| 17:25.28 | igcewieling | danfromuk: pastebin the sip debug of a failed call |
| 17:25.34 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 17:26.50 | danfromuk | From both sides? |
| 17:27.02 | danfromuk | the did provider and the peer? |
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| 17:33.58 | igcewieling | danfromuk: from the place you are seeing the 488 not acceptable |
| 17:45.05 | danfromuk | One moment. Sorry. Was speaking to the client. |
| 17:46.44 | igcewieling | danfromuk: unfortunately you lost your timeslice. |
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| 18:07.20 | danfromuk | Sorry for the delay. I understand if you are no longer available. Heres the pastebin just in case. http://pastebin.com/CSivBUij |
| 18:07.54 | danfromuk | It doesnt show SIP Error 488. I've tried to force the codec so it doesnt even bother contacting the sip peer since it doesnt have anything but g729 in its list. |
| 18:08.33 | danfromuk | It takes the call from the DID provider. The DID provider clearly supports g729. But then asterisk instantly gives up and doesnt even try starting a dialog with the sip peer. |
| 18:08.43 | danfromuk | If i add alaw to the sip peer, everything works fine. |
| 18:09.14 | *** join/#asterisk jpsharp (~jsharp@c-76-101-138-247.hsd1.fl.comcast.net) |
| 18:10.14 | igcewieling | if you can't show us the error it is hard to troubleshoot |
| 18:10.24 | jpsharp | Is there a way to get the lenght of the file recorded using Record()? Or failing that, is there a dialplan function to get Unix epoch seconds? I'm probably just overlooking both in the docs. |
| 18:11.37 | danfromuk | igcewieling: ive pasted the entire output including sip debug. I can't see why asterisk won't connect the call to the sip peer. It just says 'No audio format found to offer. Cancelling call to babbingtoncroft_201' |
| 18:15.11 | jpsharp | Oh. ${EPOCH} |
| 18:19.52 | navaismo | danfromuk, the peer babbingtoncroft_201 support all listed codecs()combined - 0x10e (gsm|ulaw|alaw|g729)? |
| 18:20.41 | danfromuk | navaismo: No. The client only wants g729, so disallow=all and allow=g729 |
| 18:21.45 | navaismo | thats in the asterisk cfg but the peer? |
| 18:22.16 | danfromuk | The peer is also set up to only allow g729. |
| 18:22.31 | navaismo | this is the sdp exchange according your debug-->Capabilities: us - 0x80000008010e (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) |
| 18:22.57 | navaismo | which sip client are you using? |
| 18:23.38 | danfromuk | Acrobits Softphone |
| 18:23.52 | navaismo | have you tried with another client? |
| 18:24.37 | danfromuk | Yes. Same issue. |
| 18:25.01 | danfromuk | I think I may just have to splash out on some g729 licenses to avoid this issue. |
| 18:25.34 | navaismo | You dont have g729 license? |
| 18:25.43 | danfromuk | No. Its supposed to be passthrough |
| 18:26.29 | navaismo | hmmm i never used passthrough so cant tell if that works or not |
| 18:27.43 | navaismo | Or maybe the trunk is using another codec and asterisk needs to transcode |
| 18:28.16 | danfromuk | Yes, but the trunk clearly supports g729. Not sure why asterisk isn't playing nicely. |
| 18:28.52 | igcewieling | danfromuk: passthu works about as well as being "just friends" with an ex. It seldom works out well. |
| 18:29.24 | danfromuk | Last time i tried to install a g729 license and codec, asterisk failed to start properly. it just seemed to be unresponsive even though i could get into the cli. |
| 18:29.33 | igcewieling | danfromuk: regardless of all that, what you think is configured is not what is configured. This tells us: (2:22:31 PM) navaismo: this is the sdp exchange according your debug-->Capabilities: us - 0x80000008010e (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) |
| 18:29.53 | navaismo | yes it support it, but that doesn't mean that the provider its using as preffer or only codec |
| 18:30.00 | danfromuk | Where do you see that the peer is audio=0x10f (g723|gsm|ulaw|alaw|g729) ? |
| 18:30.07 | navaismo | in your debug |
| 18:30.35 | danfromuk | Are you looking at the backup asterisk server, or the peer? 37.128.190.83 is a backup server and should be ignored in the sip debug. |
| 18:30.49 | navaismo | line 60 |
| 18:30.53 | danfromuk | the peer IP is 85.17.87.148 |
| 18:31.45 | [TK]D-Fender | [14:27]danfromukYes, but the trunk clearly supports g729. Not sure why asterisk isn't playing nicely. <- depends what is negotiated on ANSWER... |
| 18:32.01 | drmessano | Force it |
| 18:32.11 | danfromuk | [TK]D-Fender: cant see an ANSWER in the debug. |
| 18:32.20 | [TK]D-Fender | danfromuk: G.729 is LAST on their list. If you answer local first that may agree on something else regardless of a 2nd leg to follow. |
| 18:32.27 | [TK]D-Fender | danfromuk: Look harder :) |
| 18:32.56 | drmessano | If you want to use G729, and only G729, force it. Don't let the provider and Asterisk try to read your mind. |
| 18:33.05 | drmessano | Thats Codecs 101 |
| 18:33.14 | drmessano | , man |
| 18:33.15 | danfromuk | navaismo: on line 60, I took peer as the asterisk server since thats the peer between the did provider and asterisk. |
| 18:33.55 | danfromuk | drmessano: can't force it because I want other DIDs to use alaw. |
| 18:34.16 | drmessano | ehhh ok |
| 18:34.23 | danfromuk | Im going to try installing g729 licenses and codec and see if i get any problems this time. |
| 18:34.25 | navaismo | danfromuk, I guess your best option is buy a g729 license |
| 18:35.02 | danfromuk | yep. i think so too. |
| 18:35.12 | danfromuk | Ok. looks like ive got a bit of work tonight. |
| 18:35.13 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:35.21 | danfromuk | thanks everyone. |
| 18:35.27 | danfromuk | as always |
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| 18:38.22 | igcewieling | puts on is psychic friends network outfit and tries to figure out this report "When dialing out it takes very long until the call finally dials out. It usually starts ringing right away. Not there is a long dead space before the call goes through and it starts ringing. It feels like the system is dead if you don't wait it out." |
| 18:39.30 | DBordello | For a simple home PBX, any advantage of a Polycom 501/550/601 over one another? |
| 18:40.05 | tzanger | igcewieling: sounds like the termination provider is taking its sweet time sending back the TRYING |
| 18:40.18 | drmessano | DBordello, that's one heck of a home phone |
| 18:40.21 | tzanger | that happens with my system occasionally, and almost always with voip.ms |
| 18:40.29 | drmessano | Overkill, I would think |
| 18:40.48 | DBordello | drmessano, I am actually just going to buy 1, to experiment with asterisk. I figure a good handset would be nice to have |
| 18:41.05 | navaismo | igcewieling, or the phone is taking much time to send the invite because the phone's dialpattern timeout |
| 18:41.06 | igcewieling | tzanger: maybe. Could also be a dialplan issue on the phone |
| 18:41.06 | tzanger | DBordello: sure, but you can get smaller and just as great quality/feel polycoms for less |
| 18:41.07 | drmessano | Well, a 335 or something is good too |
| 18:41.24 | tzanger | igcewieling: depends on whether the same number sometimes takes time or always takes time |
| 18:41.28 | DBordello | tzanger and drmessano a good speaker phone is important to me. |
| 18:41.32 | tzanger | I read that as the same number sometimes takes time |
| 18:41.35 | drmessano | Ok |
| 18:41.39 | igcewieling | I did not hate the SPA phone I got in 2004 8-| |
| 18:41.39 | drmessano | So get an IP335 |
| 18:41.45 | bipolar | DBordello, I use the 560 (gigabit version of the 550) for phones at the office. It's a very nice phone with lots of features, but a bit overkill for a home phone. |
| 18:41.57 | tzanger | DBordello: I agree a good speakerphone is essential. I have the 501 at the office, I love it |
| 18:42.15 | drmessano | To me, the 331 speakerphone is as good as the 550 |
| 18:42.16 | igcewieling | bipolar: apparently you've never met any phone geeks |
| 18:42.38 | DBordello | Well, any advantage of the higher-end phone for playing with asterisk? |
| 18:42.55 | drmessano | Yeah, you get to start off spending more money on a hobby that will suck away most of it |
| 18:43.08 | DBordello | drmessano, well, eventually the rest of the house will need phones :) |
| 18:43.19 | bipolar | igcewieling, lol |
| 18:43.25 | drmessano | So buy something you can deploy everywhere else |
| 18:43.38 | drmessano | I again suggest the IP335 |
| 18:43.48 | drmessano | Then when you have that one working, buy 2 more |
| 18:43.51 | drmessano | Save your money |
| 18:43.59 | igcewieling | has a friend with both a VAX and a Norstar in his house |
| 18:44.03 | DBordello | I am checking out the IP 335. On ebay they don't seem any less expensive than the 501 |
| 18:44.12 | tzanger | igcewieling: and apparently cheap electricity |
| 18:45.00 | drmessano | DBordello, of course.. a several hundred dollar phone 8 years down the line is going to cost the same as a new $100 phone |
| 18:45.21 | DBordello | Any advantage of the newer, phone? 501 vs 335 |
| 18:45.38 | drmessano | HD voice. You wanted something to play with |
| 18:45.55 | DBordello | Ah, HD voice |
| 18:51.18 | Scapal | ahhhhh, I'm losing it! The queues on my asterisk 1.8 is behaving as a LIFO and not only a couple of times but all the time!!! |
| 18:51.25 | Scapal | <PROTECTED> |
| 18:51.25 | Scapal | <PROTECTED> |
| 18:51.25 | Scapal | <PROTECTED> |
| 18:51.25 | Scapal | <PROTECTED> |
| 18:53.11 | *** join/#asterisk Alagar (~helpdesk@vsusm16.vernalissystems.com) |
| 18:55.47 | danfromuk | The asterisk website says that the g729 codec are for version 1.4.x. Is there a different download for 1.8.x? |
| 18:56.17 | Qwell | http://downloads.digium.com/pub/telephony/codec_g729/ |
| 18:56.26 | Qwell | actually, one sec |
| 18:56.44 | Qwell | Use this one. http://my.digium.com/en/docs/G729/g729-download/ |
| 19:01.09 | danfromuk | Thanks |
| 19:07.28 | navaismo | igcewieling, which companies to host a dedicated server do you recommend, medium price?? |
| 19:07.52 | igcewieling | navaismo: No idea. |
| 19:08.01 | navaismo | uh |
| 19:08.01 | igcewieling | I think we use XO or something. |
| 19:08.19 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 19:08.45 | navaismo | hmmm |
| 19:08.51 | igcewieling | As far as I'm concerned all our boxes are in a magical place in Manhattan where tech go to trip over cables and take down service. I'm told we rent space from XO, but I've never seen the facility. |
| 19:09.08 | przerull | hello. So when the outbound leg of the dial command hangs up on the inbound leg, is it possible to send the outbound leg to a particular location in the dialplan without using the e option? |
| 19:09.18 | navaismo | i was looking at severbeach but 120 for normal server |
| 19:09.29 | navaismo | too much for my little pocket |
| 19:10.01 | WIMPy | przerull: Use the h extension. |
| 19:11.03 | igcewieling | WIMPy: "h" is only called when the inbound leg hangs up, not the outbound leg. |
| 19:11.17 | igcewieling | przerull: I think you mean the "g" option, not the "e" option? |
| 19:11.51 | WIMPy | No, both. |
| 19:11.57 | danfromuk | The digium guide to installing g729 says that I should issue the command 'asterisk -rx "module load codec_g729a.so"'. Does that persist after restart? Or do i need to put it into modules.conf ? |
| 19:12.06 | igcewieling | WIMPy: is that an asterisk 11ism? |
| 19:12.14 | jpsharp | You're not going to be able to touch dedicated server hosting with any decent level of service for under about $90/mo. |
| 19:12.35 | jpsharp | danfromuk: As long as modules.conf has "autoload = yes", it'll load on restart. |
| 19:12.45 | danfromuk | Thanks |
| 19:12.49 | WIMPy | I'm not aware of any changes. But when B hangs up I even get to h 3 time, wehn A hangs up, twice. |
| 19:12.56 | jpsharp | And doesn't have "noload => codec_g729.so" |
| 19:12.57 | przerull | igcewieling: the g option sends the inbound leg to the next priority in the extension, the e option does send the outbound leg to hangup |
| 19:13.01 | przerull | but |
| 19:13.18 | przerull | it overrides the F option |
| 19:13.46 | przerull | which I need to use as our clients use a feature we call post call digit collection (the outbound leg enters digits after the inbound leg hangs up) |
| 19:15.13 | WIMPy | But then you already know wehere the outbound leg ends up. |
| 19:16.10 | przerull | WIMPy: yeah it works fine when the inbound leg hangs up first but not when the outbound leg hangs up first |
| 19:17.02 | danfromuk | If I want to add 1 license to an existing license, is it simply a case of buying 1 license and running 'register' to add it to the existing license? |
| 19:17.12 | navaismo | yep |
| 19:18.46 | przerull | i get the feeling that I'm just not doing things right because it always feels like I really have to do all kinds of complicated things to bend asterisk to my will |
| 19:19.32 | przerull | is there a reilable way to tell which leg of a call hangs up last? |
| 19:20.31 | przerull | ideally both legs would "register" before they are bridged and each would "deregister" when they hang up. they'd use database locks to ensure that one has to wait until the other deregisters |
| 19:21.17 | przerull | this works great if both legs get sent into an h extension but that doesn't seem to happen when the outbound leg hangs up first |
| 19:28.07 | DBordello | navaismo, are you looking to host your own server, or rent a server? |
| 19:29.01 | navaismo | rent a dedicated server |
| 19:29.15 | gusto | ha |
| 19:29.16 | DBordello | You could probably get away with a VPS on a good node |
| 19:29.35 | DBordello | Asterisk doesn't strike me as very resource intensive |
| 19:29.44 | gusto | i have a virtual one too |
| 19:29.53 | gusto | KVM and VirtIO |
| 19:29.57 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 19:30.10 | gusto | running netbsd 6.1 |
| 19:30.14 | gusto | :-P |
| 19:30.17 | jpsharp | DBordello: Its not, unless you get into heavy transcoding. |
| 19:30.18 | DBordello | gusto, nice. |
| 19:31.12 | jpsharp | You got Asterisk to build under netbsd? |
| 19:31.34 | jpsharp | has tried, but never gotten far, but never really put a lot of work into it. |
| 19:32.39 | DBordello | navaismo, what are you trying to host (more detailed) |
| 19:34.36 | igcewieling | przerull: look at hangup handlers in Asteirsk 11 |
| 19:38.46 | przerull | igcewieling: thanks so to get the outbound leg to do what I want, i'll have to also create a predial handler to push the hangup handler to the outbound channel then right? |
| 19:39.46 | Freeaqingme | Is there any rationale to not maintaining the ubuntu packages anymore? |
| 19:47.00 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 19:48.03 | igcewieling | przerull: I don't know, I've not used them, I've only read about them. Once we have some development time I may see about converting our existing code to using hangup handlers. |
| 19:48.39 | igcewieling | Freeaqingme: I believe that was never an official package. don't be a wuss, install from source. |
| 19:51.06 | WIMPy | przerull: Or just skip the dialplan and use AMI. |
| 19:52.05 | Freeaqingme | igcewieling, oh, I dare installing from sources. But I am trying to puppetize this thingy, and then installing from sources isn't too practical |
| 19:52.11 | Freeaqingme | so I am considering if I should package it myself |
| 19:52.32 | Freeaqingme | hence looking for the reasons the packages are no more maintained (e.g. if there's a technical reason I should be aware of) |
| 19:52.52 | igcewieling | Freeaqingme: because every month you need to build a new package? |
| 19:53.13 | Freeaqingme | I imagine that with the proper scripts, that can be reduced to one command? |
| 19:53.57 | igcewieling | until something changes to break the single command |
| 19:54.10 | Freeaqingme | then the fun begins ;) |
| 19:54.29 | igcewieling | Freeaqingme: it is more fun to do ./configure && make && make install and a lot less work than building a package |
| 19:54.40 | Freeaqingme | yeah, it may be |
| 19:54.48 | Freeaqingme | tnx |
| 19:55.07 | *** join/#asterisk tech_travis (~Travis@174.46.237.98) |
| 19:55.25 | WIMPy | igcewieling: Oh, I didn't know we were allowed to say so these days. |
| 20:02.12 | przerull | thanks everyone |
| 20:07.11 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:17.41 | *** join/#asterisk gatty (ajg@2a01:348:11b:beef:d53:529b:fe93:2d00) |
| 20:22.58 | protocoldoug | What would I get from a commercial SBC over having a firewall and an asterisk box to manage a few trunks? One inbound, and a handful outbound. Just tandeming calls over trunks. I don't need presence for a billion softphones or the like. Just trunks. |
| 20:23.21 | protocoldoug | To me, I don't really see anything. But, I'm curious as to what others may think. |
| 20:23.30 | _Corey_ | protocoldoug: You would have more gear to admire in your rack |
| 20:24.34 | protocoldoug | rofl, that's the last thing i need XD especially one more straw on the camel's back that requires a vendor relationship, haha. |
| 20:24.55 | _Corey_ | In all seriousness, I haven't found many features aside from larger-scaling stuff that would be gained |
| 20:25.20 | _Corey_ | they're selling a lot of them to SMB customers running Microsoft Lync these days because msft doesn't believe SIP should be UDP |
| 20:25.50 | protocoldoug | that's hilarious, seems like one of the best use cases for UDP, haha |
| 20:26.50 | protocoldoug | it's some kind of panacea to my boss, but, I think it's a waste of money, and worse... more gear to manage when i've already standardized on asterisk |
| 20:26.57 | DBordello | Our university recently switched to Lync. It seems to work pretty well |
| 20:27.00 | protocoldoug | the sbc is some kind of panacea, that is |
| 20:27.32 | gatty | commercial SBC = someone to point finger at when something bad happens... |
| 20:27.52 | _Corey_ | Well, it'd be in front of Asterisk, so if you're concerned about very bad SIP traffic hitting Asterisk... there's that. |
| 20:28.23 | _Corey_ | But can it offer any "feature" to the scenario... ? Probably no. |
| 20:28.39 | gatty | many bosses just like to have commercial vendor to point fingers at, there's no sound technical founding in it. |
| 20:29.10 | protocoldoug | I'll know exactly where my traffic should come from, a handful of IPs, so in my thinking... I should be able to tighten down my firewall to those |
| 20:29.17 | protocoldoug | i definitely understand that, good point gatty |
| 20:31.12 | gatty | if Asterisk / DAHDI could talk DPNSS, I wouldn't need to use crappy commercial gateway cards or SBCs.... SBC in SME has its place for survivability, but a well configured Asterisk box can do that just as well if not better. |
| 20:32.14 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 20:34.34 | DBordello | What is a SBC/ |
| 20:34.36 | DBordello | ? |
| 20:34.42 | protocoldoug | "session border controller" |
| 20:35.06 | gatty | session border controller... essential a back-to-back user agent to provide a little security / perimiter control for external-facing connections |
| 20:35.55 | DBordello | interesting |
| 20:36.36 | protocoldoug | one "heard on the playground" thing i've heard is that some (all? any? maybe?) SBCs will open a single port for RTP for each single call, so you don't have to keep huge ranges open. |
| 20:36.50 | protocoldoug | is it possible I could tap into a way to do this with iptables? |
| 20:37.15 | gatty | you can configure Asterisk to do that, think it's the symmetric RTP option, then drop the port range down a bit in rtp.conf |
| 20:37.16 | protocoldoug | like say "hey asterisk, where are you going to open this? 10001? Ok, I'll open it in IP tables" |
| 20:37.30 | gatty | or get a SIP-aware firewall ;) |
| 20:37.54 | *** join/#asterisk CartoonCat (~kvirc@184-100-209-58.ptld.qwest.net) |
| 20:37.57 | protocoldoug | beer worthy advice, thanks :) |
| 20:38.00 | CartoonCat | hello |
| 20:38.28 | gatty | think 'even' devices like Sonicwall are SIP-aware these days |
| 20:40.09 | *** join/#asterisk Elico (~Thunderbi@109.65.205.100) |
| 20:40.27 | Elico | I have a problem that asterisk is rejecting the calls from unknown reason. |
| 20:40.51 | Elico | I will share sip.conf and extention.conf in a pastie?? |
| 20:41.05 | drmessano | Share the debug log |
| 20:41.56 | Elico | how to get it please?? from a -vvvvr i got: |
| 20:41.57 | Elico | [May 23 23:41:24] NOTICE[9267]: chan_sip.c:23272 handle_request_invite: Call from '100' (192.168.10.124:41012) to extension '100' rejected because extension not found in context 'internal'. |
| 20:42.24 | drmessano | Is there an extension 100 in the internal context? |
| 20:43.16 | Elico | I will share the sip.conf and extention.conf |
| 20:43.38 | *** join/#asterisk Rumbles (~Rumbles@31.205.54.123) |
| 20:43.39 | Elico | http://pastie.org/7949838 |
| 20:44.04 | Elico | http://pastie.org/7949841 |
| 20:44.09 | Elico | thanks drmessano |
| 20:45.21 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:49.55 | Elico | drmessano: IS IT ok?? it seems to me like it suppose to work since it worked... |
| 20:51.06 | gatty | Elico: should be extensions.conf (plural) |
| 20:51.15 | Elico | hoo nice thanks... |
| 20:52.19 | Elico | gatty: it's working both ways... the same idea |
| 20:53.51 | gatty | Elico: you def have extn 100 shown in context internal in 'dialplan show'? |
| 20:54.32 | Elico | gatty: no from an unknown reason |
| 20:54.52 | Elico | gaty what do you think? |
| 20:55.49 | gatty | the config file should be called extensions.conf and you should reload the dialplan after renaming it, then check dialplan show again. |
| 20:58.31 | CartoonCat | when starting asterisk im not getting a pid file made and that stoping mt from installing the amp portal. any ideas? |
| 21:00.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:03.33 | Elico | gatty did that and still dosn't show any 100 thing |
| 21:11.59 | Elico | gatty: it's not loading the extentions.conf from unknown reason. where is it configured to load it? |
| 21:12.11 | gatty | how have you spelled the filename? |
| 21:13.10 | Elico | extensions.conf |
| 21:13.32 | gatty | and when you ran 'dialplan reload', did it display any errors or warnings? |
| 21:14.31 | Elico | no |
| 21:15.48 | Elico | gatty: I will post the debug output of extentions.conf |
| 21:16.17 | Elico | http://pastie.org/7950049 |
| 21:18.09 | Elico | gatty: found what the problem wsa.. |
| 21:18.13 | Elico | was |
| 21:18.24 | Elico | the text was in utf-8 instead of ansi.. |
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| 21:47.54 | ketas | ffs, i broke something and can't find out what exactly |
| 21:48.19 | talntid | give us some details, and we'll help ya out |
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| 21:56.24 | ketas | hmm, that didn't make any sense... strange issues with fw |
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| 22:40.50 | cstk421 | evening all |
| 22:42.21 | cstk421 | goal is to build an asterisk server for a business voip phone system. Is the standard asterisk pbx tutorials a good way to start ? not sure if the "PBX" part of the name distinguishes between a voip / sip environment from a standard analog pbx environment. |
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| 22:50.29 | newtonr | ~book |
| 22:50.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:50.40 | newtonr | cstk421: read that book :) |
| 22:51.58 | newtonr | cstk421: Googling you'll find plenty of tutorials. Some may be out of date, but look for ones focusing on Asterisk 1.8 or 11. PBX stands for Private Branch eXchange and is general telecom terminology. |
| 22:52.26 | cstk421 | yes I'm going through one now. just wondered if i needed a voip specific tutorial to build my environment |
| 22:52.35 | cstk421 | thanks good stuff starting now ! |
| 22:52.38 | CartoonCat | anyone know what could cause astrick to compile fine, install, but not make a .pid when ran? |
| 22:52.44 | cstk421 | gonna do it on ubuntu as a xenserver vm |
| 22:53.26 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-wyqfuthiycnewfhc) |
| 22:53.34 | newtonr | cstk421: you need a tutorial specific to whatever environment you want to build... |
| 22:54.02 | newtonr | cstk421: if you want analog, read about analog, if you want voip read about voip... |
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| 22:55.47 | cstk421 | i get that once I'm underway but i assume the basic installation of asterisk on ubuntu is straightforward. Then I would look up config tutorials based on voip correct ? |
| 22:55.52 | gatty | CartoonCat: either not configured to drop a pid file or no permissions on pid file directory |
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| 22:58.00 | CartoonCat | gatty: neither of those are covered by the tutorial so I didnt mess with the default settings. doesnt help I keep losing access to my linode |
| 23:02.38 | CartoonCat | gatty: i see nothing in the configs about a pid |
| 23:04.53 | CartoonCat | and /var/run/asterisk is owned by root |
| 23:05.06 | newtonr | cstk421: yeah that works. Also https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality |
| 23:05.30 | cstk421 | thx |
| 23:05.38 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 23:12.21 | cstk421 | hey newtonr would you say a web guy would be a good idea ? if so which one ? or just cli for all configs ? |
| 23:14.54 | newtonr | cstk421: web guy? I'm guessing you mean GUI ? hehe. I recommend starting to learn by going through the book and learning it all command line and flat text files. Then later on try FreePBX. Digium has a fairly minimal distro that comes with it - http://www.asterisk.org/downloads/asterisknow |
| 23:15.04 | cstk421 | guy yes sorry |
| 23:15.14 | cstk421 | gui |
| 23:15.17 | cstk421 | damn keyboard |
| 23:16.04 | cstk421 | got it will do thanks |
| 23:16.09 | newtonr | GUI's are great but they won't teach you anything about whats underneath |
| 23:17.06 | newtonr | that being said. FreePBX is pretty amazing |
| 23:19.23 | cstk421 | now that i have ubuntu server and asterisk installed I'm going to install freepbx on top of it instead of using the distro |
| 23:19.35 | cstk421 | yes based on some quick reading its pretty cool |
| 23:26.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.237) |
| 23:27.21 | igcewieling | GUI? GUIS ARE EVIL! Er...um..I mean, #FreePBX has its own channel. |
| 23:30.12 | Maliuta | igcewieling: I thought those were the same statement |
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