| 00:26.19 | *** join/#asterisk rangotail (~rangotail@unaffiliated/rangotail) |
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| 02:42.30 | *** join/#asterisk _marshy (~a@173-31-160-187.client.mchsi.com) |
| 02:42.50 | _marshy | hey guys, i use my cell phone to connect to my asterisk server |
| 02:43.11 | _marshy | up until a few days ago it worked fine..but now when the call connects there's no audio |
| 02:43.28 | _marshy | connecting to WiFi, it works. but on verizon... no sound (though I can hear the ring) |
| 02:43.40 | Samot | Can they hear you? |
| 02:43.56 | _marshy | hmmm, not sure. let me test |
| 02:46.52 | _marshy | yes |
| 02:46.57 | _marshy | they can hear me |
| 02:55.02 | *** join/#asterisk raden (~Raden@206.8.80.67) |
| 02:58.54 | raden | Can someone tell me how to park a call from the dialplan if someone doesnt pick up because there on a line ? |
| 03:09.56 | *** join/#asterisk themayor (~themayor@unaffiliated/themayor) |
| 03:21.15 | Samot | Get the dialstatus |
| 03:22.05 | Samot | And then based on that park call if needed |
| 03:22.10 | Samot | Park() does the trick |
| 03:34.45 | davidbowlby | your external IP probably changed |
| 03:34.54 | davidbowlby | _marshy |
| 03:35.42 | davidbowlby | _marshy, I have that when a switch IP changes and isn't in the firewall access rule. |
| 04:03.58 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 04:11.30 | raden | Samot, I want to directly from my dialplan park a call so something like exten => 1099,n,GOTO(700,1) |
| 04:11.30 | raden | <PROTECTED> |
| 04:21.53 | Samot | OK. |
| 04:22.06 | Samot | So first you're going to Dial() the device you want to call. |
| 04:22.19 | raden | here lemee paste what I have just for testing |
| 04:22.24 | Samot | That device is going to return a DIALSTATUS.. |
| 04:22.36 | Samot | Based on that DIALSTATUS you can handle the call as needed. |
| 04:22.47 | raden | exten => 9999,n,DIAL(SIP/1000,10) |
| 04:22.47 | raden | exten => 9999,n,DIAL(SIP/700,1) |
| 04:22.54 | Samot | If you want to Park a call, you need to use the Park() app. |
| 04:22.55 | raden | I dial this from 1001 |
| 04:23.00 | Samot | OK. |
| 04:23.07 | raden | when it comes out of the call park it dials 1001 not 1000 |
| 04:23.16 | raden | oh that i did not know |
| 04:23.34 | Samot | I don't see anything regarding Park() in those to lines of dialplan. |
| 04:23.45 | raden | lemee try with that |
| 04:23.56 | Samot | Again, Dial() is going to return a DIALSTATUS |
| 04:24.10 | Samot | Based on that you can Park() the call, send it to VoiceMail() |
| 04:24.30 | Samot | Throw it to someplace else in the dialplan. |
| 04:24.31 | raden | how do i get dial status ? |
| 04:25.14 | Samot | core show application Dial |
| 04:25.45 | raden | well i used park |
| 04:25.48 | raden | same behavior |
| 04:25.54 | raden | it was returned to the extension I called from |
| 04:25.57 | Samot | Show what you have. |
| 04:26.04 | Samot | Did you read how Park works? |
| 04:26.12 | Samot | Do you have the parking.conf configured? |
| 04:26.23 | Samot | http://wiki.asterisk.org <-- Start there. |
| 04:26.25 | Samot | Or |
| 04:26.28 | Samot | ~boo |
| 04:26.28 | infobot | for heaven's sake, samot, don't do that! |
| 04:26.28 | raden | features.conf |
| 04:26.30 | Samot | ~book |
| 04:26.30 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:26.34 | raden | I have the book |
| 04:26.34 | Samot | No. |
| 04:26.38 | raden | all the books |
| 04:26.42 | raden | I have no parking.conf |
| 04:26.45 | Samot | core show application Park |
| 04:26.53 | Samot | You need to configure a Parking Lot |
| 04:27.02 | Samot | With slots and all that good stuff. |
| 04:27.08 | Samot | There's an entire section on it. |
| 04:27.23 | Samot | features.conf is just a feature code to PARK a call DURING a call. |
| 04:27.36 | Samot | It's for DTMF features when the call is connected. |
| 04:27.44 | raden | ok |
| 04:27.55 | raden | this is starting to make more sense, I was very confussed |
| 04:28.44 | raden | Note: As of 7/17/04 CVS the parking.conf file has been renamed to features.conf in order to be a more general config file. |
| 04:28.59 | raden | so everything is in features.conf ? |
| 04:29.06 | raden | oi |
| 04:30.45 | Samot | I guess you can put it there too. |
| 04:31.20 | Samot | I have mine broken into a few files so I only have to update the files for the specific lot. |
| 04:31.34 | raden | I gotcha it would make more sense |
| 04:34.32 | raden | how the hell do i get it not to announce the parking position ? |
| 04:34.37 | raden | 701 then music |
| 04:34.42 | raden | I just want it to go to music |
| 04:35.44 | Samot | https://github.com/mojolingo/asterisk/blob/master/configs/features.conf.sample |
| 04:36.00 | Samot | ; Asterisk 12 Note - All parking lot configuration is now done in res_parking.conf <-- Important line |
| 04:36.12 | raden | yup im looking at that on my screwen |
| 04:36.19 | Samot | https://github.com/mojolingo/asterisk/blob/master/configs/res_parking.conf.sample |
| 04:36.43 | raden | i dont have a res_parking im on old asterisk :( |
| 04:36.49 | raden | need to update system soon |
| 04:36.59 | Samot | Well you didn't say that. |
| 04:37.13 | raden | sorry man wasnt thinking this is one of our boxes from 2011 |
| 04:37.24 | raden | old e5200 intel LOL |
| 04:37.25 | Samot | So what is it running? |
| 04:37.35 | Samot | What version of Asterisk? |
| 04:38.03 | raden | Asterisk 1.6.2.9-2+squeeze12 built by pbuilder @ pungenday on a x86_64 running Linux on 2013-12-20 19:06:10 UTC |
| 04:38.13 | Samot | Uhm. |
| 04:38.21 | raden | <PROTECTED> |
| 04:38.32 | raden | lol |
| 04:38.39 | raden | uhm ? |
| 04:38.51 | Samot | Well. |
| 04:39.02 | Samot | That box was built in 2013 |
| 04:39.13 | raden | yea I saw that i thought it was the original one |
| 04:39.15 | Samot | Asterisk 1.6.x was *dead* for two years at that point. |
| 04:39.16 | raden | so 4 years |
| 04:39.32 | raden | well its the most stable machine we have for asterisk sadly |
| 04:39.34 | Samot | You are serious out of date. |
| 04:39.43 | Samot | Well that could be the issue. |
| 04:39.44 | raden | I know I know were changing things over next month |
| 04:39.51 | Samot | 1.6.2 was not that great. |
| 04:40.04 | Samot | So if it's working now and it's the most stable.. |
| 04:40.07 | Samot | Don't mess with it. |
| 04:40.14 | raden | LOL |
| 04:40.22 | raden | I need to get this stupid parking working :( |
| 04:40.34 | raden | which I now have thanks to you |
| 04:40.44 | raden | just would like to get rid of the parking position announcement |
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| 09:32.34 | miralin | Hi all. Can anyone help me in one question? Is there any replacement for function SIPPEER(peername, status) in Aterisk's PJSIP channel driver? |
| 09:51.31 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 09:57.51 | |stefan| | Does Asterisk LTS 13.17.2 PJSIP have support for DNS SRV ? Been searching the internet for some information but i only find "Improved DNS SRV support in Asterisk 14" |
| 10:00.37 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 10:07.24 | file | it does. |
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| 11:16.11 | |stefan| | does it also support NAPTR records ? didn't find a good specification anywhere i looked. |
| 11:16.34 | file | 13 doesn't because it uses the PJSIP implementation which only does SRV |
| 11:16.46 | file | 14 and above uses our own implementation which supports SRV, NAPTR, IPv6, and failover |
| 11:16.56 | |stefan| | ah ok. |
| 11:17.22 | |stefan| | so then i'll have to enter the _sip._udp.domain.ext as contact then. |
| 11:17.33 | |stefan| | i tried it but didn't work fantastic. but i'll give it another go =) |
| 11:17.44 | |stefan| | thanks =) |
| 11:17.54 | file | nope, it automatically tries to resolve that using the domain name |
| 11:20.50 | |stefan| | ok so then i would just enter domain.tld ? and it will try to resolve _sip._udp.domain.tld ? |
| 11:21.04 | file | yes. |
| 11:21.08 | |stefan| | ok. thanks =) |
| 11:27.50 | |stefan| | will load balancing actually work ? or will it just pick the first record ? |
| 11:28.56 | file | it load balances |
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| 11:33.56 | |stefan| | nice =) |
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| 12:11.24 | stefan27 | Hey |stefan|. Hope that works out well for you, we wrote some external code that manages the NAPTR stuff and updates the asterisk sip peers dynamically but that was all a dirty hack. |
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| 13:18.28 | alexandre9099 | hi, i finally understood what the contexts are (why i took so long for that... anyway), i can forward a call that is recieved on chan_mobile to chan_sip, but i get no sound on each end, what can it be? both asterisk server and sip client are on same network (192.168.0.101 for the server and 192.168.0.105 for the client)) |
| 13:19.13 | voipmonk | ~pb |
| 13:19.14 | infobot | pastebin is probably a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:19.21 | voipmonk | can we see the call? |
| 13:19.50 | alexandre9099 | what you mean by "see the call"? wait a sec, i'll paste extensions.conf |
| 13:20.38 | [TK]D-Fender | THE CALL |
| 13:20.43 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:20.48 | [TK]D-Fender | as in the CALL DEBUG from * CLI |
| 13:21.06 | [TK]D-Fender | with proper debug enabled for BOTH sides of the call |
| 13:21.16 | alexandre9099 | https://pastebin.ca/3902431 |
| 13:21.18 | [TK]D-Fender | dialplan should have no impact on audio |
| 13:21.33 | alexandre9099 | and how should i do that? (i'm noob on asterisk, sorry :( ) |
| 13:21.40 | alexandre9099 | how can i enable debug? |
| 13:22.02 | Samot | sip set debug on |
| 13:22.08 | Samot | ^^ That |
| 13:22.40 | alexandre9099 | i'll check what the output gives me after enabling debug |
| 13:28.01 | alexandre9099 | strangely enough i'm earing the sound on the device (the one that is the "trunk") here is the log https://pastebin.ca/3902433 |
| 13:32.19 | alexandre9099 | [TK]D-Fender: |
| 13:34.54 | Samot | Reliably Transmitting (NAT) to 192.168.0.105:5060: <-- You're on a local LAN nothing is behind NAT |
| 13:35.56 | *** join/#asterisk gerhard7_ (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:36.09 | alexandre9099 | exactly, both computers are on the same network, so what is the cause of the problem? |
| 13:36.14 | alexandre9099 | might it be with chan_mobile? |
| 13:36.15 | [TK]D-Fender | NAT isn't an issue in this case as the IP's are local |
| 13:36.26 | [TK]D-Fender | go prove your softphone side isn't faulty |
| 13:36.46 | [TK]D-Fender | make an exten that does a Record() and then Playback that file you record |
| 13:36.48 | Samot | Well, it could be chan_mobile. |
| 13:37.01 | alexandre9099 | [TK]D-Fender: i'll see how to do that, thanks |
| 13:37.06 | Samot | Go confirm that two Chan_SIP accounts can handle calls without audio issues. |
| 13:37.06 | [TK]D-Fender | to verify that firewalls aren't blocking audio, that your sound card is working, etc |
| 13:37.30 | [TK]D-Fender | He should only need to prove the single chan_sip device involved here irect to *. |
| 13:37.34 | Samot | Sound card? |
| 13:37.36 | Samot | What? |
| 13:37.47 | [TK]D-Fender | if that works just to/from those dilaplan apps that'll pin it on chan_mobile |
| 13:37.59 | [TK]D-Fender | User-Agent: Ekiga/4.0.1 |
| 13:38.14 | [TK]D-Fender | ^ looks like PC soft-phone |
| 13:38.19 | Samot | So we're going straight to the sound card on the PC? |
| 13:38.23 | Samot | OK. |
| 13:38.50 | [TK]D-Fender | so the sound devices it's confingured to use, the hardware itself (incl headset, etc) are all suspect |
| 13:38.54 | [TK]D-Fender | every little piece... |
| 13:39.02 | alexandre9099 | it is :) i tried with another softphone but got same result |
| 13:39.20 | [TK]D-Fender | prove it with the test I requested and we'll move forward from there |
| 13:39.31 | *** join/#asterisk BoBeR182 (~BoBeR182@gateway/tor-sasl/bober182) |
| 13:39.47 | Samot | So you call from the Chan_Mobile device to the Chan_SIP device and have this issue. |
| 13:39.51 | Samot | Do you have it the other way? |
| 13:39.58 | Samot | Chan_SIP to Chan_Mobile? |
| 13:40.20 | alexandre9099 | i'm just trying one more thing, the bluetooth device was set with a class indicating it was a computer, i'm trying to set it as a headset now |
| 13:41.48 | alexandre9099 | can i just add the record line in the middle of the incoming-house context? |
| 13:43.32 | *** join/#asterisk ashutoshkrchaube (673399be@gateway/web/freenode/ip.103.51.153.190) |
| 13:45.42 | [TK]D-Fender | it'll be another exten entirely. |
| 13:45.49 | [TK]D-Fender | just don't mix those lines with your other one |
| 13:49.35 | alexandre9099 | so it should be like this? "exten => 103,1,Record(myrecording,gsm)" |
| 13:49.46 | alexandre9099 | if i call 103 on sip it should record the call, right? |
| 13:50.05 | [TK]D-Fender | if it hits there, yes |
| 13:50.19 | [TK]D-Fender | then do a Playback of it as the enxt step |
| 13:50.24 | [TK]D-Fender | next |
| 13:51.26 | alexandre9099 | and when asterisk knows when to finish the recording? |
| 13:53.34 | alexandre9099 | well, asterisk said "-- <SIP/alexandre-00000001> Playing 'beep.gsm' (language 'en')" and the softphone played nothing |
| 13:53.48 | alexandre9099 | i'll retrieve the recording |
| 14:00.29 | [TK]D-Fender | Playback() <----- |
| 14:00.52 | [TK]D-Fender | it knows when yuo press # |
| 14:01.31 | Samot | I thought that was Background() |
| 14:01.35 | *** join/#asterisk kharwell (kharwell@nat/digium/x-tkizpropnlbxjksv) |
| 14:01.35 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:01.36 | alexandre9099 | hmm, so # is the "next" command? |
| 14:01.48 | Samot | I thought Background was the one that can accept DTMF to interrupt it. |
| 14:02.07 | [TK]D-Fender | no, when recording the audio use # to STOP recording so you can continue to the Playback |
| 14:02.35 | Samot | Oh |
| 14:02.50 | Samot | We're going the route of setting up Record() to see if there is audio? |
| 14:03.12 | alexandre9099 | ok, is it possible to set another "key" to stop the recording (curiosity :) ) |
| 14:03.12 | [TK]D-Fender | <[TK]D-Fender> make an exten that does a Record() and then Playback that file you record |
| 14:03.15 | Samot | Wouldn't an RTP debug to see where the hell it's going and if it's going work? |
| 14:03.30 | [TK]D-Fender | record+playback = rock-solid proof |
| 14:03.37 | Samot | OK. |
| 14:03.46 | Samot | So it rtp debug. |
| 14:04.04 | [TK]D-Fender | 2 commands, no funny scrolling nothing required to interpret success.... |
| 14:04.04 | Samot | But hey, the longer more complicated route works too. |
| 14:04.20 | [TK]D-Fender | Proves your FS to. |
| 14:04.30 | [TK]D-Fender | every kind of proof in 2 simple lines |
| 14:04.40 | Samot | No true. |
| 14:05.01 | alexandre9099 | no sound, just some noise |
| 14:05.01 | Samot | Re-writing dialplan to do it is a bit of a messy troubleshooting step. |
| 14:05.09 | alexandre9099 | i'll try another softphone |
| 14:05.13 | Samot | Stop. |
| 14:05.14 | Samot | FFS. |
| 14:05.17 | Samot | Now.. |
| 14:05.29 | Samot | Show a debug of this call. |
| 14:05.35 | Samot | Where you tried to record and playback. |
| 14:06.03 | alexandre9099 | i forgot to enable debug, let's do it again |
| 14:07.28 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:07.28 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:09.00 | alexandre9099 | here is the log https://pastebin.ca/3902460 |
| 14:09.10 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:09.10 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:13.30 | alexandre9099 | hmm, or there is a problem with the softphone or i'm doing something wrong with the config, the only thing it hear is noise |
| 14:16.29 | [TK]D-Fender | So audio is no good? |
| 14:16.50 | alexandre9099 | i think it is not working :( i'm trying another softphone |
| 14:17.54 | [TK]D-Fender | If you hear "noise" then that sounds like like a soundcard/mic issue |
| 14:18.40 | alexandre9099 | i'll try to install a sip client on my android phone |
| 14:20.51 | Samot | INFO sip:205@192.168.0.101:5060 SIP/2. |
| 14:21.08 | alexandre9099 | that's the ip of the server |
| 14:21.12 | Samot | DTMF is being done via INFO? |
| 14:21.17 | Samot | That's not standard. |
| 14:21.34 | alexandre9099 | not sure? how was it suposed to be done? |
| 14:21.50 | Samot | rfc2833 for Chan_SIP |
| 14:21.59 | Samot | Have you tried another softphone? |
| 14:22.13 | Samot | Because this looks like the exact same one from before. |
| 14:22.23 | Samot | Go download X-Lite or Zoiper or something. |
| 14:22.24 | alexandre9099 | that's what i'm trying now, i'll try on my phone, maybe it a problem with the pc mic |
| 14:23.21 | alexandre9099 | great, zoiper works on linux :D |
| 14:24.01 | Samot | On the same system? |
| 14:24.06 | Samot | Or are you trying different systems? |
| 14:24.23 | Samot | Put Zoiper on the same system as the one you're having an issue with. |
| 14:24.31 | *** join/#asterisk DanB_ (~DanB@clt-195.192.206.125.ip-anschluss.net) |
| 14:25.15 | alexandre9099 | i was going to install zoiper on android, but i remembered that android has sip support built in, i'll try zoiper on the computer with problem, my linux laptop |
| 14:26.20 | alexandre9099 | yup, the problem is with the laptop softphone, i just tested the android sip client and it works without a problem |
| 14:33.23 | alexandre9099 | i think the problem was with codecs on softphone, now ekiga works fine(i enabled every codec it supported)... i'll try again the chan_mobile to chan_sip redirection |
| 14:39.29 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hssomeitcqipmzkk) |
| 14:39.30 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:41.33 | alexandre9099 | calling from sip to chan_mobile only gives me the beeps and then it stays silent :/ |
| 14:43.07 | alexandre9099 | i get this line from asterisk "[Nov 1 14:41:40] WARNING[1691][C-00000002]: res_format_attr_siren7.c:54 siren7_parse_sdp_fmtp: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring." not sure if relevant |
| 14:46.49 | [TK]D-Fender | restrict your peers to a single codec that is appropriate to your usage. |
| 14:47.03 | [TK]D-Fender | ulaw/alaw would be best |
| 14:47.44 | alexandre9099 | how can i do that? is it a asterisk setting? |
| 14:52.17 | [TK]D-Fender | disallow=all |
| 14:52.21 | [TK]D-Fender | allow=ulaw |
| 14:52.22 | [TK]D-Fender | etc... |
| 14:52.25 | [TK]D-Fender | in your SIP peers |
| 14:57.14 | alexandre9099 | still no sound :( |
| 14:57.22 | alexandre9099 | only getting the beeps |
| 14:57.33 | alexandre9099 | maybe it is a chan_mobile problem |
| 14:57.52 | [TK]D-Fender | is the record+playbak now working perfectly? |
| 14:58.31 | alexandre9099 | after the codec thing? i will try |
| 14:59.14 | alexandre9099 | yes, after the codec it is still working |
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| 15:04.38 | alexandre9099 | hmm, maybe it can be a bluetooth adapter problem, i don't have any other bluetooth adapter to try |
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| 15:12.59 | Stokkeland | exit |
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| 17:26.58 | alexandre9099 | hmm, maybe it can be a bluetooth adapter problem, i don't have any other bluetooth adapter to try |
| 17:27.24 | alexandre9099 | didn't want to send this again... |
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| 18:12.13 | alexandre9099 | well, i think i give up :/ |
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| 20:22.47 | RovingWriter | <PROTECTED> |
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| 20:33.11 | lvlinux | hi RovingWriter! |
| 20:34.30 | lvlinux | I know this is probably a longshot, but I'll throw it out in case someone knows: |
| 20:35.16 | lvlinux | Does asterisk support t.38 passthrough with chan_mgcp? |
| 20:36.53 | lvlinux | I didn't find any reference to t.38 in the sample config file. |
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| 20:37.39 | file | no. |
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| 21:00.29 | danielyk | Hi all, Asterisk is listening on following two ports: 0.0.0.0:38517 and :::47748. For which are they used? |
| 21:04.25 | danielyk | Is there someone? |
| 21:09.24 | lvlinux | thanks file. I'm sure t.38 with MGCP is a hot topic with the devs and high up on the list to get implemented. lol |
| 21:09.43 | lvlinux | danielyk: Are you getting that from netstat or what? |
| 21:09.45 | file | people implement what they want and desire, there is no list per se |
| 21:10.19 | file | danielyk: if you have PJSIP loaded then it could be the DNS support |
| 21:10.29 | lvlinux | file: Of course, my statement was tongue in cheek. I'm probably the first person that's even thought about T.38 with MGCP in the last year lol |
| 21:10.37 | file | or ever. |
| 21:10.42 | lvlinux | true hehe |
| 21:10.59 | danielyk | lvlinux: Yes: netstat -tulpn | grep asterisk |
| 21:11.09 | danielyk | file: yes, I use dnsmgr.conf |
| 21:11.17 | file | dnsmgr is different |
| 21:12.16 | danielyk | file: Okay, I have migrated now completely to pjsip. I use dns hostnames for the external addresses. |
| 21:12.29 | file | that doesn't matter, the PJSIP library itself has its own DNS Client |
| 21:12.46 | file | in order for that to work it has to listen on UDP ports so it can get DNS replies, those ports could be it |
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| 21:13.24 | danielyk | That makes sense. Should I disable dnsmgr if I do not use chan_sip any more? |
| 21:13.51 | file | dnsmgr is used to refresh the external addresses regularly if you have configured it to do so and are using a recent enough version |
| 21:16.00 | danielyk | I use 15.1.0... |
| 21:16.37 | file | then it has that functionality |
| 21:17.52 | danielyk | But I do not know where I use an external address resolving except from pjsip |
| 21:18.12 | file | so, the external_* configuration options in pjsip.conf use dnsmgr |
| 21:18.27 | file | if you want them regularly resolved (such as if the address is for a dyndns host) you must enable dnsmgr |
| 21:18.32 | file | and there is no harm in doing so |
| 21:18.51 | danielyk | Okay, then all fine :) |
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