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| 00:19.59 | pensmit | can the VoiceMail app listen for a keypress like Background()? |
| 00:22.29 | ChannelZ | core show application VoiceMail |
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| 00:26.35 | [TK]D-Fender | penIt isn't something you get to tell it to listen for additional things |
| 00:26.52 | [TK]D-Fender | pensmit: It isn't something you get to tell it to listen for additional things |
| 00:28.36 | WIMPy | Apart from 0 and *. |
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| 00:37.05 | pensmit | thanks |
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| 01:03.06 | Technodrome | hey |
| 01:03.14 | Technodrome | good day |
| 01:03.40 | WIMPy | Good night |
| 01:04.04 | Penguin | Good morning, good afternoon, good evening, and good night. |
| 01:04.20 | Technodrome | what is a good soft phone for mac , i can't find one :( |
| 01:04.37 | Penguin | Did you try zoiper? |
| 01:04.42 | WIMPy | See, that's whay we just say moin. Not what. |
| 01:06.07 | Penguin | ~zoiper |
| 01:06.07 | infobot | [~zoiper] Zoiper (formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, Mac OS X, and Linux that can be found at http://www.zoiper.com |
| 01:08.14 | WIMPy | Oh, the evil has become mobile? |
| 01:09.44 | ChannelZ | Blink also |
| 01:09.55 | Penguin | blinks |
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| 01:22.26 | Technodrome | well |
| 01:22.35 | Technodrome | i really still wish they made eye beam for mac |
| 01:22.37 | Technodrome | but they don |
| 01:22.38 | Technodrome | t |
| 01:22.55 | Technodrome | so i can find an old download, but i can't get a serial key for it anymore, they won't even allow you to purchase it |
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| 01:32.51 | Technodrome | Penguin: they all suck pretty bad sadly enough :( |
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| 03:52.15 | pensmit | How can I make a menu selection happen quicker when there is a 2 and a 202 in the context and I need both? |
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| 03:56.15 | Penguin | pensmit: Make another context. |
| 03:57.01 | Penguin | If both 2 and 202 are on the same context, and you use something like BackGround() or WaitExten(), if you press 2, they will wait for 02 to make sure you really mean 2 and not 202. |
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| 04:33.50 | pensmit | So basically someone calls in and they get a menu that has a 1 or 2 option but they also need to be able to dial an en extension that starts with 20x that cannot be changed. Any ideas to improve speed? It's too bad you can't specify how long to wait for dtmf. |
| 04:34.36 | Penguin | If you make it go right after pressing 2, how will they ever be able to dial 202? |
| 04:34.51 | ChannelZ | Bitches gotta be quick |
| 04:35.00 | pensmit | yeah |
| 04:35.03 | pensmit | bitches need to go quick |
| 04:35.09 | pensmit | that's what i want |
| 04:35.13 | ChannelZ | Anyway you can futz with the timeouts, see function TIMEOUT |
| 04:35.57 | pensmit | that is totally my style |
| 04:35.59 | pensmit | futzing |
| 04:35.59 | ChannelZ | like Set(TIMEOUT(absolute)=1) perhaps |
| 04:36.17 | pensmit | what would that timeout be applied ot |
| 04:36.38 | ChannelZ | sorry I meant digit, not absolute |
| 04:36.58 | ChannelZ | core show function TIMEOUT |
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| 04:38.01 | pensmit | Thank you my man. |
| 04:38.08 | pensmit | Am I going about this totally wrong? |
| 04:38.11 | pensmit | by the way |
| 04:38.34 | ChannelZ | well besides making your extensions something else.. |
| 04:39.03 | pensmit | yeah |
| 04:39.04 | ChannelZ | or making people dial something unique before an extension, like # or something |
| 04:39.08 | pensmit | yeah |
| 04:39.13 | pensmit | i thought of both those |
| 04:39.31 | ChannelZ | Well it can't read minds. |
| 04:39.40 | ChannelZ | So not sure what other solution there would be |
| 04:39.42 | pensmit | That is terrible. |
| 04:39.58 | pensmit | I mean reading minds is important |
| 04:39.59 | pensmit | lol |
| 04:40.01 | pensmit | ok thanks a lot |
| 04:40.09 | pensmit | I owe you 2 Internet beers. |
| 04:40.24 | ChannelZ | Thanks but I'm in Internet AA |
| 04:44.24 | Penguin | One way I dealt with that was to only allow dialing a direct extension for a short amount of time, then go into the menu options. |
| 04:45.15 | Penguin | "If you know your party's extension, dial it now." wait 4 seconds ... "For tech support, press 1. For accounting, press 2..." |
| 04:45.26 | Penguin | After the 4 second wait, no more dialing of extensions. |
| 04:49.21 | pensmit | That makes sense Penguin. Thanks |
| 04:49.32 | pensmit | Lol@ChannelZ I just read your comment. |
| 04:49.43 | Penguin | That way the bitches don't necessarily have to be so quick. |
| 04:49.58 | pensmit | Bitches better get up on it now. |
| 04:50.02 | pensmit | :) |
| 04:51.24 | Penguin | If TIMEOUT(digit) will accept floating point values, you could make the bitches have to be extremely quick. |
| 04:51.44 | pensmit | Ok. Last question for the night. Should I change the asterisk password that comes with AsteriskNow or did they take care of the security there. |
| 04:51.52 | Penguin | Set(TIMEOUT(digit)=0.0000001) |
| 04:52.21 | Penguin | You've already asked that question five or six times and been answered at least twice. |
| 04:53.23 | pensmit | shit really |
| 04:53.25 | pensmit | I missed it |
| 04:53.26 | pensmit | damn |
| 04:53.30 | ChannelZ | RLLY |
| 04:53.39 | ChannelZ | the user is probably randomly generated |
| 04:54.38 | pensmit | the password you mean? |
| 04:55.04 | ChannelZ | yes for the asterisk system user which is what I thought you kept asking about |
| 04:55.32 | pensmit | yes i did |
| 04:55.40 | pensmit | damn sorry guys |
| 04:55.41 | pensmit | lol |
| 04:55.41 | Penguin | You said user, so I think he was just making sure you meant password and not what you actually said. |
| 04:56.05 | pensmit | the linux user named asterisk |
| 04:56.09 | ChannelZ | I like to leave out words |
| 04:56.18 | pensmit | on the ASteriskNow default install with bare asterisk |
| 04:56.22 | Penguin | We know the name isn't going to be randomized. |
| 04:56.43 | pensmit | Oh I'm not correcting you |
| 04:56.44 | pensmit | lol |
| 04:56.47 | Penguin | That would make things really hard to administer. |
| 04:57.02 | pensmit | I thought there was some coolness going on that I didn't know about |
| 04:57.24 | ChannelZ | actually if you useradd with no password and don't change it.. what is it? |
| 04:57.25 | pensmit | Ok so I'm not going to worry with it |
| 04:57.34 | Penguin | null |
| 04:57.59 | pensmit | you guys are confusing me with your smartness |
| 04:58.14 | pensmit | so I'm ok without worrying about that password for the asterisk user |
| 04:58.24 | Penguin | pipes /dev/zero into pensmit |
| 04:58.42 | Penguin | And by pipes, I mean redirects |
| 04:58.57 | pensmit | I would think a useradd with no password would be terrible |
| 04:59.13 | pensmit | but then again, I fell on my head a lot as a child |
| 04:59.19 | Penguin | Certain things simply do not work with a null password. |
| 04:59.24 | Penguin | Like ssh, for example. |
| 04:59.28 | pensmit | oh ok |
| 04:59.31 | pensmit | well tehre you have it |
| 04:59.54 | ChannelZ | you can look in /etc/shadow - it might not have a hash in there at all. |
| 05:00.03 | ChannelZ | most 'system' type accounts dont |
| 05:00.34 | Penguin | But if you NEED ssh to work for a user without a password, there's always PermitEmptyPasswords. |
| 05:00.46 | pensmit | oh I see |
| 05:00.48 | pensmit | no hash? |
| 05:00.54 | pensmit | yes |
| 05:00.55 | pensmit | ok |
| 05:00.58 | pensmit | thanks guys |
| 05:01.13 | ChannelZ | hash = what the actual password gets turned into |
| 05:01.18 | pensmit | yes |
| 05:01.19 | pensmit | ok |
| 05:01.21 | pensmit | I'm with you |
| 05:01.23 | ChannelZ | it doesn't actually know your password |
| 05:01.27 | pensmit | yes |
| 05:01.36 | pensmit | I got you |
| 05:01.58 | Penguin | Feel free to change the default password to something extremely secure if you worry about it. |
| 05:02.01 | ChannelZ | Which incidentally is why when you have a website that says "email me my forgotten password" you should run away |
| 05:02.26 | Penguin | apg -a1 -m13 -x26 -E :\;\* -s |
| 05:02.36 | pensmit | what the hell is that |
| 05:02.58 | pensmit | nevermind |
| 05:03.03 | pensmit | enough questions for tonight |
| 05:03.19 | ChannelZ | a password generator that makes gobbily-gook |
| 05:03.38 | Penguin | apg -a1 -n1 -m26 -s |
| 05:03.40 | pensmit | my tiny little chihuahua-like brain is tired |
| 05:03.47 | pensmit | but go on...i'll record this |
| 05:04.22 | Penguin | I use apg for all things password. |
| 05:05.08 | pensmit | Ok..sleepy time. Thanks a ton guys. |
| 05:05.21 | Penguin | For my Cisco phones, apg -a1 -m13 -x26 -MSNCL -E^[]{}\"? |
| 05:18.26 | phix | i use apg to |
| 05:18.28 | phix | o |
| 05:41.44 | *** join/#asterisk Technodrome (~asdfasdf@unaffiliated/technodrome) |
| 05:41.46 | Technodrome | hey guys |
| 05:41.53 | ChannelZ | hey lady |
| 05:42.05 | Technodrome | what is acceptable latency time for calls? |
| 05:42.11 | Technodrome | we really want below 50ms huh? |
| 05:43.28 | ChannelZ | Well up in the 300+s is more annoying |
| 05:47.16 | Technodrome | ChannelZ: what soft phone software do you use? |
| 05:47.34 | ChannelZ | Zoiper or Blink.. CSipSimple on Android sometimes |
| 05:47.43 | ChannelZ | I don't really use softphones much except for farting around |
| 05:47.58 | Technodrome | we used to use the cisco one at work |
| 05:48.07 | Technodrome | ever heard of jets? |
| 05:48.10 | Technodrome | jetsi |
| 05:48.13 | ChannelZ | nope |
| 05:48.15 | Technodrome | jitsi * sorry |
| 05:49.54 | Technodrome | you could be in for a pleasant surprise ChannelZ |
| 05:49.58 | Technodrome | the best i've ever used |
| 05:50.13 | Technodrome | can be connected to 10 different sip networks , and it doesn't flinch |
| 05:50.32 | Technodrome | zoiper products are heh |
| 05:51.08 | Technodrome | blink is cool, but its all in python so kinda slow + to do anything really cool you need the paid version |
| 06:07.54 | phix | hey gang, I am getting some nasty echo when making calls of a tdm400 card. Any auto tuning apps available or is it trial and error? |
| 06:08.16 | phix | s/of/on/ |
| 06:08.58 | phix | thnx infobot! |
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| 06:16.53 | Penguin | It's not the latency that is the most problem; it's the jitter. |
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| 07:31.40 | ChannelZ | phix: have you done fxotune? |
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| 07:44.54 | phix | ChannelZ: Probably ages ago, how does that work again? |
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| 07:50.47 | phix | sudo fxotune -i ? or something else ChannelZ ? |
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| 07:56.29 | phix | ChannelZ: ok so I did fxotune -i, it did stuff, I still have echo, any way I can adjust echo settings / volume, on the fly when I am in a call so I can adjust it manually? |
| 07:58.23 | phix | hmm, I changed txgain and rxgain settings to 0, seems to be a bit better, I can still hear my self though, barely, I have to really concentrate to hear it |
| 08:03.26 | *** join/#asterisk Technodrome (~asdfasdf@unaffiliated/technodrome) |
| 08:03.40 | Technodrome | Penguin: what is the difference between latency and jitter? |
| 08:06.00 | phix | ok hardly any echo but I am soft at the other end :\ |
| 08:06.15 | phix | if I put the volume up I can hear myself |
| 08:06.37 | phix | so looks like the echo canceller isn't working or needs to be adjusted some hoe |
| 08:06.40 | phix | how* |
| 08:06.42 | phix | any ideas ChannelZ ? |
| 08:06.49 | kaldemar | Technodrome: jitter is the variation in latency |
| 08:08.10 | Technodrome | the jitter i think the last i checked on my call was 40ms |
| 08:08.12 | Technodrome | how bad is that? |
| 08:08.14 | phix | I am using the oslec canceller |
| 08:08.55 | Technodrome | something to cancel echo is a headset |
| 08:09.15 | Technodrome | unless its some other kind of echo |
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| 09:33.55 | Technodrome | is there any way to have so little latency that it would be extremely difficult with no echo cancellation on to tell? |
| 09:34.05 | Technodrome | or is that just a reality of the situation? |
| 09:38.48 | ChannelZ | phix: sorry was off watching a movie |
| 09:39.37 | ChannelZ | The first important part is to get your tx and rx gains set right.. which turns out to be fairly difficult unless you can get a milliwatt test number from your telco |
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| 09:41.40 | ChannelZ | But barring that, then yes you run fxotune to let it learn the characteristics of your line. I use fxotune -i -m 13 -l 1 |
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| 09:43.30 | ChannelZ | After it does its tests, it writes out values to fxotune.conf. You then use fxotune -s to load that config and tune the channels... this is usually done for you when DAHDI is started, but you can check your init script |
| 09:44.23 | ChannelZ | I assume you have echocanceller=mg2 or such in your /etc/dahdi/system.conf |
| 09:46.20 | ChannelZ | Technodrome: you'll still hear it even at only a couple of milliseconds so no, not really. |
| 09:46.29 | Technodrome | i see |
| 09:46.37 | Technodrome | ChannelZ: yeah i mean no one has ever sad anything |
| 09:46.39 | Technodrome | said* |
| 09:46.52 | Technodrome | but i did it with my cell phone, muted the cell phone and spoke into the computer |
| 09:46.54 | Technodrome | and yeah there was a little latency |
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| 09:51.04 | ChannelZ | I want to say RTP packets for ulaw are like 20ms so at absolute minimum it's at least twice that |
| 10:07.40 | Technodrome | ChannelZ: my jitter is 19ms up, 1 ms down |
| 10:14.59 | Technodrome | how bad is that jitter? |
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| 11:41.33 | gigiozzo | hi everybody |
| 11:41.58 | gigiozzo | my first time with asterisk and voip in general |
| 11:42.23 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 11:42.31 | gigiozzo | i have a question |
| 11:43.13 | gigiozzo | there is a way to answer a call and wait until callee hangup? |
| 11:44.20 | gigiozzo | i forgot to tell you that i'm using latest asterisk 1.8 |
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| 12:41.54 | KrisB | hello, I just installed Asterisk on my Debian OS. I configured two SIP accounts and two extensions, which I can locally use with SIP clients (I can call from one extension to another). I decided to route calls to my virtual (DID) number (bought from callcentric.com) to sip:extension@my-ip on asterisk server. I get this error though: http://pastebin.com/7Hhg3MZk |
| 13:09.07 | KrisB | hello, can you help me? |
| 13:21.56 | *** join/#asterisk pfui (c4cd9017@gateway/web/freenode/ip.196.205.144.23) |
| 13:23.04 | pfui | ho can I tell asterisk to perform a certain set of actions upon a sip peer's registration/disconnection? |
| 13:24.00 | pfui | *how |
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| 13:31.00 | nextime | pfui: AMI? |
| 13:34.54 | pfui | nextime: seems to be what I'm looking for... |
| 13:35.07 | pfui | is the API documented somewhere? |
| 13:35.33 | WIMPy | manager show ... |
| 13:35.50 | WIMPy | or somewhere on wiki.asterisk.org. |
| 13:37.25 | pfui | WIMPy: https://wiki.asterisk.org/wiki/display/AST/AMI+Manager+Commands |
| 13:37.37 | pfui | not the most user friendly format... |
| 13:39.24 | WIMPy | Allthough you only need the login command and otherwise just listen for events. |
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| 13:41.44 | pfui | I'll give it a shot |
| 13:41.47 | pfui | thanks |
| 13:44.03 | nextime | pfui : there are already a lot of libraries and scripts to talk with the manager interface on an high level of abstraction that make it trivial |
| 13:44.14 | nextime | just use them for the language you prefer |
| 13:44.57 | WIMPy | They might also make it more complicated instead. |
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| 13:48.59 | pfui | okay, slightly related question: can I queue a sip simple message while a peer is offline and have it delivered once the peer registers with asterisk? |
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| 13:54.59 | iq | Hi |
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| 14:28.39 | paule32 | hello |
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| 18:44.32 | oryn | anyone know why context would be ignored in a sip trunk in sip.conf, it keeps looking in context public, its driving me nuts |
| 18:45.05 | oryn | i have tried type = user / peer / friend |
| 18:45.13 | oryn | makes no difference |
| 18:46.52 | ChannelZ | then it's not matching the peer you think it should for some other reason |
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| 19:07.52 | [TK]D-Fender | it simply isn't matching. |
| 19:08.08 | [TK]D-Fender | type alone doesn't make the parameters match |
| 19:08.16 | [TK]D-Fender | just says which set gets looked at first |
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| 19:30.05 | usn_work | Hello folks, I'm stuck with 11.5.1 - can't get GoSub() to work. Maybe I'm missing something: |
| 19:30.16 | usn_work | [TestSub] |
| 19:30.17 | usn_work | exten => ts,1,Ringing |
| 19:30.17 | usn_work | exten => ts,n,dial(SIP/11) |
| 19:30.17 | usn_work | [commons] |
| 19:30.17 | usn_work | exten => 10,1,GoSub(TestSub,1,1) |
| 19:30.43 | usn_work | Error when dialing 10: |
| 19:30.46 | usn_work | [Sep 22 21:29:16] WARNING[4021][C-0000000d]: pbx.c:4621 pbx_extension_helper: No application 'GoSub' for extension (intern, 10, 1) |
| 19:32.17 | usn_work | what am I doing wrong? |
| 19:33.34 | usn_work | module load |
| 19:34.57 | ChannelZ | app_stack |
| 19:42.38 | usn_work | ChannelZ, why? |
| 19:42.48 | ChannelZ | why what? |
| 19:43.10 | usn_work | basically "why do I have to do that" :) |
| 19:43.14 | ChannelZ | Do you have app_stack loaded? |
| 19:43.17 | usn_work | no |
| 19:43.24 | ChannelZ | Because that's where Gosub, Return, and some other things are. |
| 19:43.41 | ChannelZ | Do you not have autoload turned on in modules.conf? |
| 19:44.01 | usn_work | no :( |
| 19:45.28 | ChannelZ | Unless you're on a VERY limited resources system and are controlling the module loading very specifically for a reason, you'd be doing yourself a favor to turn it on.. you'll probably wind up finding lots more missing/broken things otherwise |
| 19:45.28 | usn_work | Ok, loading app_stack helps, and changing autoload does what the name suggests :/ Thanks a lot ChannelZ :) |
| 19:46.03 | usn_work | My pbx does not work yet, but now I'm making first progress :) |
| 19:48.27 | ChannelZ | Cool, have fun. Also when pasting multiple lines of configs, etc. for help. use a pastebin |
| 19:48.33 | ChannelZ | ~pb |
| 19:48.34 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:49.40 | usn_work | I know, sorry |
| 19:50.16 | [sr] | hi WIMPy |
| 19:58.06 | ChannelZ | needs some lunch |
| 20:00.48 | [sr] | i need to go to sleep |
| 20:03.34 | *** join/#asterisk monsterco (~monsterco@64.231.101.21) |
| 20:04.34 | monsterco | Hi everyone - I have an Aastra phone connected to a Bell DSL modem and Asterisk sees it with it's *local ip* when I do a "sip show peers" - Is there any setting in Aastra phones that I should enable to make sure it sends it's public IP address? |
| 20:05.32 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 20:05.52 | ChannelZ | 1. are the two actually remote to each other? 2. do you have nat turned on for that peer in sip.conf? |
| 20:06.35 | monsterco | ChannelZ - yes, they are remote to each other - Asterisk server is in Datacentra and Aastra phone is at home on DSL |
| 20:06.45 | monsterco | 2- I have NAT=yes for that same extension |
| 20:07.05 | monsterco | Reg. Contact : sip:9999@10.10.9.15:5060;transport=udp |
| 20:07.19 | monsterco | it shouldn't show that local IP and should show public IP - right? |
| 20:09.14 | monsterco | maybe the router is changing the SIP header? I thought Asterisk picks the source IP and doesn't care whats in SIP packet |
| 20:11.52 | monsterco | ?? |
| 20:15.17 | ChannelZ | What version of asterisk? |
| 20:15.29 | [TK]D-Fender | How is an Aastra phone HAVING 2 IP's in the first place? |
| 20:15.50 | [TK]D-Fender | And most "bell modems" ... have ROUTERS built in |
| 20:16.17 | [TK]D-Fender | So unless your phone also has a PPPoE client... you're way off the mark |
| 20:16.34 | ChannelZ | Where did he say it has 2 IPs |
| 20:16.48 | [TK]D-Fender | [16:07]monstercoit shouldn't show that local IP and should show public IP - right? |
| 20:17.04 | [TK]D-Fender | by the implication of showing a private.... where there should be no concept of "private: |
| 20:17.09 | [TK]D-Fender | Hidden in the big print |
| 20:17.19 | [TK]D-Fender | It gets ONE IP |
| 20:17.45 | ChannelZ | yes but all he was asking is "shouldn't that be the external IP in order for it to work" |
| 20:18.27 | [TK]D-Fender | the fact is that it ONLY has a private IP, because that "modem" is actual a NAT router like anything else |
| 20:19.11 | [TK]D-Fender | And no, the phone is on the inside of a private LAN, and the contact header it sends... should be completely irrelevant because he should have set the peer to IGNORE it in the first place |
| 20:19.33 | [TK]D-Fender | Bad peer settings make baby Jesus cry.... |
| 20:25.30 | monsterco | Asterisk 1.8xxx |
| 20:26.19 | monsterco | <ChannelZ> yes but all he was asking is "shouldn't that be the external IP in order for it to work" <<<< you are right |
| 20:27.16 | monsterco | [TK]D-Fender - how do I set the peer to ignore private IPs? |
| 20:27.32 | [TK]D-Fender | nat=yes <- |
| 20:27.38 | monsterco | there is no one way audio or anything like that. It works fine but I know that I should see the public ip when I do sip show peers |
| 20:27.42 | [TK]D-Fender | because it is nat'd |
| 20:33.24 | monsterco | I have nat=yes for that specific peer |
| 20:33.28 | monsterco | and restarted the phone |
| 20:33.33 | monsterco | but still showing local ip |
| 20:34.49 | [TK]D-Fender | monsterco: show us the debug |
| 20:35.18 | [TK]D-Fender | 16:07]monstercoReg. Contact : sip:9999@10.10.9.15:5060;transport=udp [16:07]monstercoit shouldn't show that local IP and should show public IP - right? <- NO. |
| 20:35.51 | [TK]D-Fender | it will typically jsut show whatever IP it was bound to and * doesn't ahve to care when you tell it to use the source IP. |
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| 20:58.22 | crumb | ok, i just compiled a codec for ARM, but i keep getting.. Error loading module 'codec_g729.so': /usr/lib/asterisk/modules/codec_g729.so: cannot open shared object file: No such file or directory |
| 20:58.31 | crumb | but that's where i put it |
| 21:00.59 | crumb | i get the same for g723 |
| 21:02.48 | crumb | show translations |
| 21:02.49 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:02.50 | crumb | No such command 'show translations' (type 'core show help show translations' for other possible commands) |
| 21:03.11 | crumb | ooh.. i think i know what the issue is |
| 21:03.23 | crumb | used the wrong headers when i cumpiled |
| 21:03.30 | monsterco | [TK]D-Fender - I have another server and another Aastra phone on a Bell DSL modem which does show the public IP |
| 21:03.42 | monsterco | I am sure it's a setting on phone or Asterisk server which makes this possible |
| 21:05.48 | monsterco | I restarted again and now it shows public ip |
| 21:05.53 | monsterco | probably nat=yes did it |
| 21:05.54 | monsterco | thanks |
| 21:14.10 | [TK]D-Fender | s/probably/definitely |
| 21:14.24 | Penguin | Oh so close! |
| 21:14.46 | [TK]D-Fender | As for "sip show peers" showing the WAN IP... "nat=no" forces * to take that actual IP source address, and NOT the "contact:" header |
| 21:15.17 | [TK]D-Fender | And when not set... the phone could also be set up to try to discover the WAP IP it may be behind |
| 21:15.27 | [TK]D-Fender | This is all up to you knowing what your phone is set to look at |
| 21:18.27 | monsterco | [TK]D-Fender - I don't think Aastra phones have any public ip lookup feature |
| 21:18.53 | [TK]D-Fender | monsterco: And what IP do you see on the phone itself? |
| 21:18.58 | Penguin | They don't have a nat traversale setting? |
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| 21:27.34 | floren | hi all |
| 21:27.51 | floren | any idea when asterisk 11.6 will be released? is on RC1 now |
| 21:28.12 | Penguin | As soon as it is done, it will be released. |
| 21:29.02 | floren | well, most of the time devs have approximate time frames |
| 21:29.28 | Penguin | Good luck with that. |
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| 21:44.40 | [TK]D-Fender | floren: This is open source... it almost never works like that. |
| 21:45.07 | [TK]D-Fender | floren: They don't run on a schedule. While they may have hopes, they don't havce specific pressures for hard deadlines. |
| 21:45.35 | [TK]D-Fender | floren: When whatever bugs they are looking at get addressed satisfactorily they will release it |
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| 21:57.49 | floren | thanks [TK]D-Fender |
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| 23:05.27 | Technodrome | good day guys |
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| 23:45.39 | ChannelZ | is it? |
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