01:21.25 | *** join/#asterisk infobot (ibot@rikers.org) |
01:21.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
01:22.34 | *** join/#asterisk libardi (~libardi@189.61.226.79) |
01:35.19 | *** join/#asterisk zerohalo (~zerohalo@uranium.zerohalo.net) |
01:39.44 | *** join/#asterisk zerohalo (~zerohalo@uranium.zerohalo.net) |
01:41.31 | *** join/#asterisk vince1 (~vince@ip72-202-187-29.ok.ok.cox.net) |
01:49.52 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
01:59.21 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
02:08.30 | *** join/#asterisk vstemen (~vince@ip72-202-187-29.ok.ok.cox.net) |
02:42.03 | *** join/#asterisk vince1 (~vince@ip72-202-187-29.ok.ok.cox.net) |
02:56.57 | *** join/#asterisk stux|work (stux@endurance.xzibition.com) |
03:00.04 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
03:04.33 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
03:06.43 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
04:01.07 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
04:39.08 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
05:01.51 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
05:20.21 | *** join/#asterisk floppy1 (~joeshmojo@101.165.145.178) |
05:46.59 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
06:02.35 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
06:14.58 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
06:37.21 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
07:03.21 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
07:31.41 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
08:04.08 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
08:19.05 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
09:05.02 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
10:00.43 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
10:05.51 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
10:21.52 | *** join/#asterisk sekil (~sekil@cable-89-216-227-161.dynamic.sbb.rs) |
10:32.52 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
11:06.38 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
11:30.14 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
11:32.48 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
11:59.31 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
12:03.33 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
12:07.24 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
12:26.01 | *** join/#asterisk samwierema (~samwierem@82.169.225.211) |
12:28.59 | *** join/#asterisk chendy (~alexc@121.34.144.88) |
12:30.04 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
12:49.00 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
12:52.13 | sarthor | HI, Current Asterisk Version: 13.9.1. here is error https://paste.ubuntu.com/24157459/ I can connect in local lan but can not connect via Internet, Please help need for guidance. thanks in advance. |
13:08.07 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
13:10.23 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
13:49.58 | *** join/#asterisk samwierema (~samwierem@82.169.225.211) |
13:54.18 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
14:08.51 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
14:34.05 | *** join/#asterisk CryptoManiac (~Crypto@unaffiliated/cryptomaniac) |
14:35.39 | CryptoManiac | Can Asterisk run without dahdi if using just sip channels? Or dahdi has to be installed regardless of channel use? |
14:37.32 | WIMPy | Not needed. |
14:38.14 | CryptoManiac | WIMPy: Thanks |
14:39.32 | CryptoManiac | Another issue I'm having... When dialing from my external softphone to one of the extensions which is a desk IP phone on the internal network, there i sno audio. I am pretty sure my NAT settings are correct because I am able to use the external softphone to dial into other applications and echo test has two-way audio. |
14:43.15 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
14:45.32 | Samot | There's no audio completely? No audio on one side? Or one-way audio? |
14:46.28 | CryptoManiac | no audio completely |
14:46.52 | Samot | So neither side can hear each other/ |
14:47.05 | CryptoManiac | Correct |
14:47.16 | Samot | Show a SIP debug on a call this is happening with |
14:47.19 | Samot | asterisk -rvvvvvvvvvv |
14:47.22 | Samot | sip set debug on |
14:47.26 | CryptoManiac | ok |
14:47.30 | CryptoManiac | will pastebin it shortly |
14:47.35 | Samot | pjsip set logger on <-- if you are using that |
14:59.39 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-lgncywhxgucfqgxh) |
15:02.28 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
15:05.28 | CryptoManiac | Samot: May I PM you the pastebin link? (as it contains my public IP's) |
15:09.34 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
15:11.55 | Samot | I guess but then others cant see it and make assessments.. |
15:12.23 | WIMPy | The bots have probably already found you anyway. |
15:17.38 | *** join/#asterisk chendy (~alexc@121.34.144.88) |
15:24.52 | Samot | Well? |
15:29.47 | CryptoManiac | Apologies guys |
15:30.03 | CryptoManiac | I do not mean to waste your precious time |
15:30.22 | CryptoManiac | It's against IT department policy to post our public IP's. :-( |
15:39.05 | Samot | I said you could PM them. |
15:39.08 | CryptoManiac | came up wtith a solution, i replaced the public IP with <PUBLIC IP> |
15:39.14 | CryptoManiac | http://pastebin.com/k7U2g0t6 |
15:40.27 | Samot | Does 11 even support Opus? |
15:41.44 | Samot | Make all sides of the call ulaw. |
15:42.01 | Samot | Like your Bria client, which is only offering Opus from what I can see. |
15:42.45 | CryptoManiac | ah... |
15:44.17 | Samot | #CodecsMatter |
15:45.24 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
15:45.31 | CryptoManiac | I have disallow=all under my sip user settings, and then allow=ulaw allow=alaw allow=g729 |
15:45.43 | CryptoManiac | but I guess the problem is that Bria is only offering Opus |
15:45.55 | CryptoManiac | To be honest I don't even know what Opus is. I'm a Voip newbie |
15:45.56 | CryptoManiac | :-) |
15:47.47 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
15:48.19 | *** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp130-03-74-14-78-25.dsl.bell.ca) |
15:51.22 | CryptoManiac | Samot: I'm afaraid that was not the problem, I disabled Opus in Bria and set ulaw/alaw as priority followed by g7211 g729 and gsm but still the problem persists |
15:51.46 | Samot | There is no g7211 |
15:51.57 | Samot | You really don't need g729 or gsm. |
15:52.12 | Maliuta_ | Samot: it's the upgraded g711 ;) |
15:52.20 | Maliuta_ | CryptoManiac: http://opus-codec.org/ |
15:53.26 | CryptoManiac | Maliuta_: Thanks |
15:53.45 | CryptoManiac | Samot: Noted |
15:53.58 | CryptoManiac | however, still no audio |
15:54.52 | Maliuta_ | Samot: looks like opus is better than g729, I would probably use opus over g729 - if my devices supported it |
15:55.09 | Maliuta_ | CryptoManiac: is NAT involved anywhere? |
15:55.57 | Maliuta_ | CryptoManiac: and if you are terminating a trunk on opus you might have to transcode to a codec that your SIP devices can handle |
15:57.10 | CryptoManiac | Maliuta_: Yes there is NAT. There is two-way audio between the Bria softphone client and asterisk, for example when I dial the demo extensions 500 an 600. However when I dial a desk IP phone which is on the internal network there is no audio at all... |
15:57.13 | WIMPy | Is there any ITSP that supports opus? |
15:57.46 | Maliuta_ | CryptoManiac: try turning of reinvite for the softphone |
15:57.51 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
15:58.13 | Maliuta_ | or just turn it off fullstop |
15:58.31 | Maliuta_ | for smaller setups it's worth keeping * in the loop |
15:58.52 | Maliuta_ | there are even cases for keeping * in the loop in larger setups |
15:59.03 | CryptoManiac | Maliuta_: I actually want Asterisk to be in the loop |
15:59.30 | Samot | Then you want directmedia/reinvite off. |
15:59.42 | Maliuta_ | Samot++ |
15:59.55 | Maliuta_ | WIMPy: if there is not we can create one! |
16:01.51 | *** part/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
16:02.15 | Samot | No ITSP I know of openly admits they support Opus. |
16:03.18 | CryptoManiac | Samot: I turn that off by canreinvite=no ? |
16:03.20 | Samot | But then again a lot are "ambiguous" when it comes to the RTP media. |
16:03.34 | Samot | canrevinite= was replaced by directmedia= |
16:03.46 | Samot | But yes, they should be no |
16:03.52 | CryptoManiac | ok |
16:04.46 | Maliuta_ | Samot: it really is a wideband codec, so passing to/from PTSN is going to create degredation. Probably a good solution for VoIP only, if it has to go to the PTSN use g729 or g711 |
16:05.01 | Samot | I know what Opus is. |
16:05.10 | Samot | Yes. |
16:05.25 | Samot | I know it's why a lot of providers do offer anything outside of g711 or g729 |
16:05.40 | Samot | And don't do STRP/TLS. |
16:05.47 | Maliuta_ | that and laziness |
16:05.49 | Samot | s/STRP/SRTP/ |
16:05.56 | Samot | Perhaps.. |
16:06.08 | Samot | Or the fact it will require a lot more processing power. |
16:06.18 | Samot | Puts a lot more load on the systems. |
16:06.19 | Maliuta_ | think of all the time/money they save by not implimenting it |
16:06.32 | Maliuta_ | CPU is cheap these days |
16:06.33 | Samot | Or the fact that a lot of the PSTN doesn't use it. |
16:07.37 | Maliuta_ | here in .au pretty soon old school PTSN will be dead, it will be all VoIP over the NBN (National Broadband Network) - even if 99.999999% of the populus don't know that |
16:07.40 | Samot | When you're taking thousands of calls per second and have to transcode or decrypt/encrypt them all.... |
16:08.22 | Maliuta_ | most mobile networks are VoIP inside, and I've even heard that most copper PTSN is VoIP from the exchange now |
16:08.47 | Samot | A lot of carriers are IP based. |
16:09.03 | Samot | But the entire PSTN backbone is not and will not be for some time. |
16:09.16 | Maliuta_ | depending on where you are |
16:09.48 | Samot | Which means if an IP based carrier wants to exchange calls from a non-IP based carrier, what happens? |
16:10.14 | Maliuta_ | when you sign up to the NBN here your phone line (if you have one) goes with it. Comes out of an RJ11 port on the their termination box |
16:10.39 | Samot | Yeah that's not uncommon. |
16:10.44 | Samot | ATT, Comcast, they all do it here. |
16:10.55 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
16:11.59 | Maliuta_ | Samot: I'd have to look into it, but I think all of our international calls are going out via fibre links (probably IP based) I guess the problem is always going to be "the other end". Once it's out of your network you have NFI what tech is being used to deliver it |
16:12.52 | Maliuta_ | Most ISP's have been doing it on DSL connections for a while. The just have VoIP enabled modems they give people. Most don't even know it's VoIP |
16:13.03 | Samot | I know. |
16:13.06 | Samot | That's how it is now. |
16:13.24 | Samot | VoIP is becoming the standard delivery method to the end user.. |
16:13.30 | Samot | The *end user* |
16:13.41 | Maliuta_ | I tried to trunk directly with one a while ago (because friends of mine knew it was a VoIP line) and the ISP and firewalled it all off to only their servers |
16:14.24 | Maliuta_ | end users can bite me. They cause more problems than they're worth :D |
16:15.05 | Samot | "If it wasn't for all my end users, my business would be awesome" |
16:15.14 | Maliuta_ | devs are enough of a PITA for me :) |
16:15.32 | Maliuta_ | although I work with only DBA's ATM |
16:15.42 | Maliuta_ | they are almost as bad as devs |
16:16.13 | Maliuta_ | Samot: end users are SEP |
16:17.08 | Maliuta_ | I don't do that shite anymore, cause me to develop a drinking habit. Also the primary cause of my leukaemia :D |
16:17.26 | Maliuta_ | but the caffiene helped treat that |
16:17.31 | Samot | End users are everyone's concern at a provider. |
16:18.23 | Maliuta_ | Not if you work far enough up the food chain. Then they are just accounts, or DB entries :) |
16:18.32 | Maliuta_ | my concern is servers and services |
16:18.48 | Maliuta_ | if they are up and running properly it's SEP |
16:19.07 | Samot | So basically, the end users are your problem. |
16:19.27 | Maliuta_ | only the ones I work with |
16:19.28 | Samot | Servers or services don't work, that impacts the end user and means you have to get involved. So they are your problem. |
16:21.26 | Maliuta_ | "CC all mail from these wordpress installs to this address - I already get them from another address, but add this one anyway" ... "it's not working I haven't seen an email from those addresses" /me provides logs proving delivery of said email, user doesn't look at his filtering rules that are removing duplicate messages |
16:22.22 | Maliuta_ | I only do internal systems, client systems aren't my problem. Our company specifically doesn't offer that service. Only DB design/optimisation (for mysql/mariadb) |
16:23.04 | CryptoManiac | Found this in my console verbose but that URL is not valid |
16:23.06 | CryptoManiac | [Mar 11 19:16:40] WARNING[4161]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission e360524b5c1941bf9ef0236b28ed3821 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
16:23.33 | Maliuta_ | occaisionally I might have to look at a client system to tell the DBA where the bottle neck is, but then it's up to the client to fix it |
16:23.44 | Maliuta_ | CryptoManiac: firewall issue somewhere? |
16:24.02 | Samot | That URL works fine. |
16:24.24 | CryptoManiac | Maliuta_: Have opened SIP port TCP/UDP and RTP ports 10000 to 20000 as configured in rtp.conf |
16:25.06 | Samot | UDP, right? |
16:25.27 | Samot | You have all those ports open on UDP? |
16:26.17 | Maliuta_ | I limit mine to about 10 ports for my small install. If I am doing a 30 phone setup for a campaign I open more for them - but never that many |
16:26.40 | Maliuta_ | also tend to firewall them off so that only the ITSP can access them |
16:26.54 | Maliuta_ | from the outside - inside is a different matter |
16:27.32 | Samot | 10 RTP ports? |
16:27.51 | Maliuta_ | CryptoManiac: you'd need to see which device that was failing with to see. Just having the ports open might not be enough |
16:28.12 | Maliuta_ | Samot: not making a massive number of calls from here |
16:28.26 | Samot | Yeah, 5 is the limit. |
16:28.42 | Samot | RTP requires two ports per call |
16:28.57 | Maliuta_ | Samot: like I said, when I do larger installs I open more |
16:29.04 | Maliuta_ | yup |
16:29.36 | Maliuta_ | which is why I open 10 - allows 10 concurrent calls from here |
16:29.43 | Maliuta_ | sorry 5 |
16:29.58 | Samot | It also means that people can figure out what RTP ports you are using. |
16:30.31 | Maliuta_ | firewalled off to the ITSP |
16:30.45 | Maliuta_ | not allowing just anyone to trunk into my systems |
16:30.58 | Maliuta_ | may be crazy, just not _that_ crazy |
16:31.04 | Samot | OK |
16:32.17 | Maliuta_ | if someone can look at the number of filetered/rejected ports on a scan of my system and figure out which ports are for RTP then they have all the network foo |
16:32.27 | Samot | Not like anything is actually listening on the ports until they are in use. |
16:32.44 | Maliuta_ | this is also true |
16:33.08 | Samot | The other side doesnt even know what ports until you tell them. |
16:33.36 | Samot | There is no unestablished, new incoming RTP requests |
16:34.08 | Maliuta_ | someone at an ISP level could probably look at the netflows and tell |
16:34.25 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
16:34.30 | Samot | RTP ports arent known by the provider until you tell them what ports to use. |
16:34.42 | Samot | They don't "listen" for anything. |
16:34.42 | Maliuta_ | traffic patterns can tell you alot |
16:35.05 | Samot | Right, the ports are assigned randomly based on the range. |
16:35.18 | Samot | So 10000-20000 means you have 10,000 ports to randomly choose from. |
16:35.22 | Samot | 10 means just that. |
16:35.36 | Samot | Which limit makes it easier to guess patterns? |
16:35.38 | Maliuta_ | good point |
16:35.49 | Maliuta_ | yeah yeah |
16:35.51 | *** join/#asterisk TriJetScud (~TriJetScu@69.172.162.252) |
16:35.56 | Maliuta_ | I see what you're getting at |
16:36.17 | Maliuta_ | throw in some random skype/IM activity and it gets harder |
16:36.27 | Maliuta_ | but still |
16:36.29 | Samot | Opening the RTP ports in your NAT rules is more for making sure NAT isn't lost once the audio stream is established. |
16:36.51 | Samot | I rarely have to open SIP ports or RTP ports on my routers. |
16:37.14 | Maliuta_ | I don't NAT my * servers - I got away from that habit ages ago |
16:37.29 | Samot | I'm talking about on premise systems. |
16:37.47 | Maliuta_ | and I don't allow directmedia, so everything inside (on private IP's) goes to * to be sent out |
16:37.53 | Samot | This also includes phones. |
16:37.58 | Samot | I don't allow directmedia either. |
16:39.30 | Maliuta_ | I try not to run VoIP through NAT if I can avoid it |
16:39.49 | Samot | Client premise systems can't avoid it. |
16:39.53 | Maliuta_ | most of the installs I do are to systems that have a real static IP, run the * server there |
16:42.17 | Maliuta_ | RTP is a bigger issue on the Big Blue Button servers we use for training than it is on any of the * installs I do |
16:43.01 | Samot | I'm not sure what that means. |
16:43.15 | Maliuta_ | _if_ we move to a matrix server to replace our company IRC channel it could be an issue there if people want to do voice/video via that |
16:44.02 | Maliuta_ | Big Blue Button is bit of software we use that allows over internet training. With video/voice and desktop sharing |
16:44.46 | Maliuta_ | it's normally used as part of a moodle install. But we don't need all the moodle foo (although for one the bosses other companies it might be needed soon) |
16:46.26 | Maliuta_ | BBB uses RTP for all kinds of things, so having it configured properly, and the firewalls setup to allow it through properly (including the fact the some of the client ends are going to be comming through NAT) is an issue on those servers. |
16:47.11 | Maliuta_ | approaching 3am - I should probably get some sleep |
16:47.34 | Maliuta_ | tonnes of code to write, and other work I need to do later today |
16:56.20 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
17:11.22 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
17:24.14 | CryptoManiac | Samot and Maliuta_ thanks a lot for your help. Fixed my NAT issues... |
17:32.06 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
18:12.10 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
18:52.36 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
19:01.24 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
19:12.52 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
19:21.28 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-kokttxnfyylvptuq) |
20:13.41 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
20:36.57 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
21:14.21 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
21:29.11 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
21:35.58 | *** join/#asterisk zend (0ec87e29@gateway/web/freenode/ip.14.200.126.41) |
21:37.45 | zend | hi! Can someone help me to understand this Warning in Asterisk, Abnormal 'Gosub(ackcall,~~s~~,1(61,00000000056,0437767327))' exit. Popping routine return locations. |
21:39.19 | dan_j | zend: Do you have a 'return' at the end of the gosub? |
21:40.00 | zend | dan_j: yes I do |
21:40.30 | dan_j | zend: http://lists.digium.com/pipermail/asterisk-dev/2013-September/062428.html |
21:40.38 | zend | dan_j: What happens is callee hangs up, while in the macro. I'm doing this https://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall |
21:41.06 | dan_j | If they hang up inside the macro, then return won't get run. I believe you can ignore this warning if that is the case. |
21:41.46 | CryptoManiac | whatever happened to TK Defender? |
21:41.57 | dan_j | [macro-ackcall] in the link you posted does not have a Return |
21:42.38 | zend | dan_j: i modified mine to have the return.... i'm using second example |
21:43.12 | zend | dan_j: THE AEL version |
21:43.27 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
21:43.52 | zend | dan_j: do you think it ever leads to any imbalances in asterisk stacks & leads to Unexplained behaviour... |
21:44.08 | dan_j | I dont know AEL. But if the HANGUP gets executed, then it doesnt hit the RETURN. I dont think it's an issue. |
21:44.40 | dan_j | Maybe ask again during the week when this channel is more active with people who are more 'in the know' |
21:45.44 | zend | dan_j: Ok , thanks |
22:14.01 | *** join/#asterisk j0hnd0e (~Mutter@24.48.86.58) |
22:14.43 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
22:14.53 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
22:15.09 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
22:42.23 | igcewieling | I suspect using a goto to jump out of a gosub (or macro) would cause that sort of error. |
22:57.40 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
23:02.12 | *** join/#asterisk FuriousGeorge (ad3fb7d5@gateway/web/freenode/ip.173.63.183.213) |
23:15.53 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
23:27.31 | *** join/#asterisk sh_smith (foobar@cpe-76-174-26-91.socal.res.rr.com) |
23:38.49 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |