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| 00:12.32 | [TK]D-Fender | heads out for a while |
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| 02:50.24 | bhans | Hi guys, I'm currently using asterisk 1.4, I have a problem regarding handle_response_register: Outbound Registration: Expiry for xx.xx.sip. I aldready had the NAT opened and only UDP is allowed for sip registry. What could cause this problem? When it timed-out, I cannot recieve calls anymore. |
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| 05:33.31 | [TK]D-Fender | bhans, What problem?> I don't see an error or anything in full at all |
| 05:40.21 | bhans | [TK]D-Fender, I searched a lot over the web with my problem and it seems that the logs im getting (outbound registration: expiry, re-register) are just normal. But at times, incomming call aren't working, I could do outbound calls anytime with no problem. |
| 05:40.59 | [TK]D-Fender | Nothing tells me that whatever you think you are seeing means anything |
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| 10:55.37 | WinterNight | Hello. :) I have problem with simple install Asterisk on FreeBSD. I created 3 files sip.conf, pjsip.conf and extensions.conf with this guide: https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts . I can't call from 6001 to 6002 and from 6002 to 6001. In CLI I see demo-bob and demo-alice are have unspecified host. When trying to call I getting error "cause 20 - Subscriber absent". Please help. :) |
| 11:15.26 | WinterNight | Anybody. Please. :) |
| 11:27.16 | avb | WinterNight: well, what does sip show peers shows? |
| 11:27.19 | avb | at least |
| 11:27.41 | avb | where does 6001/6002 routes to |
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| 11:53.19 | WinterNight | Thank You for response avb . :) I can't paste. sip show peers shows that demo-alice and demo-bob are "Unspecified" in Host column. |
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| 12:06.17 | WinterNight | Maybe this is FreeBSD problem? Please help. |
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| 13:46.17 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:46.26 | [TK]D-Fender | WinterNight, Show us the actual configs and the call attempt |
| 13:46.30 | [TK]D-Fender | ~pb |
| 13:46.30 | infobot | hmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:46.31 | [TK]D-Fender | ^^^^ |
| 13:46.40 | [TK]D-Fender | along with the status dump |
| 13:55.44 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 13:56.11 | WinterNight | Thank you [TK]D-Fender . :) :) :) I can't paste, Asterisk are installed on separate machine on FreeBSD system. Jitsi and Linphone clients have statuses as registered but in CLI "sip show peers" I have "Unspecified" address in "Host" column. When calling I have error "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)". I created only 3 files sip.conf pjsip.conf and extensions.conf identically from: https://wiki.asterisk.org/ |
| 13:56.47 | [TK]D-Fender | I don't see why you can't pastebin from * CLI.... |
| 13:58.07 | [TK]D-Fender | "Unspecified" means is hasn't registered. |
| 13:58.15 | [TK]D-Fender | Show the configs AND the status dump |
| 13:58.52 | WinterNight | Ok, I will start SSH server. |
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| 14:09.19 | WinterNight | This is my sip.conf: http://pastebin.com/fDqLCRuk pjsip.conf: http://pastebin.com/cacMr1b8 extensions.conf: http://pastebin.com/7K01z2n2 . |
| 14:10.01 | WinterNight | I can give SSH access to server. |
| 14:12.55 | WinterNight | [TK]D-Fender : How I can show status dump? |
| 14:18.18 | Samot | These look pretty incomplete to me. |
| 14:19.01 | *** join/#asterisk nix8n82 (~AndChat58@67-130-74-235.dia.static.qwest.net) |
| 14:19.36 | WinterNight | Samot : What can I do? |
| 14:20.07 | Samot | Well there's nothing in the sip.conf showing the external ip, local networks, etc. |
| 14:20.28 | Samot | Nothing is showing Chan_SIP and CHAN_PJSIP listening on unique ports. |
| 14:20.35 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-ihauuzhpkfhuqypm) |
| 14:20.54 | [TK]D-Fender | indeed |
| 14:21.20 | [TK]D-Fender | And by specifying nothing that would make them use their defaults which I'd suspect are BOTH 5060 and only one is going to get to bind |
| 14:24.25 | WinterNight | I have DNS record A in OVH sip.wn.systems pointing to my Asterisk server 82.177.238.240. This is my sockstat http://pastebin.com/f4baxaN5 The hostname of server is sip.wn.systems |
| 14:25.09 | avb | WinterNight: guys are right. you need to use or chan_sip or chan_pjsip |
| 14:25.17 | avb | or both but on different ports |
| 14:25.37 | avb | you dont need to configure both of them |
| 14:26.17 | avb | remove sip.conf or pjsip.conf depends on what do you want to use |
| 14:26.44 | Samot | Well he's also missing things like modules.conf based on his statement.. |
| 14:26.50 | [TK]D-Fender | At least set them to different ports NOW so they don't screw with each other |
| 14:26.53 | avb | if you want to use pjsip, then in ext.conf you would need to use Dial(PJSIP/...) |
| 14:26.55 | Samot | So I'd actually like to see what modules are actually *loaded*. |
| 14:29.13 | WinterNight | Thank You. :) This is my modules.conf: http://pastebin.com/HMPtTifS |
| 14:30.10 | [TK]D-Fender | Go fix your sip & pjsip configs tpo specify the bindport, localnets, ettc |
| 14:30.19 | [TK]D-Fender | all the basics you should ahve for them to work properly |
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| 14:39.44 | WinterNight | So can I delete pjsip.conf? |
| 14:40.14 | [TK]D-Fender | just fix it... |
| 14:40.28 | [TK]D-Fender | and remomve the entires that make it look like you were trying to set up bob & alice on BOTH |
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| 14:55.08 | WinterNight | I renamed pjsip.conf and WORKING WORKING WORKING, Thank You. :) :) :) I can call but nothing hear. I have this errors: http://pastebin.com/2UMN36iD |
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| 14:55.10 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:56.04 | [TK]D-Fender | WinterNight, You haven't shown us how it is networked (actual IP interfaces), firewall dumps, and you've specified nothing in your sip.conf for the basics |
| 14:56.48 | [TK]D-Fender | "core set verbose 10" , "sip set debug on" <--- set both and show us a new call as you provide the other information |
| 14:57.11 | WinterNight | Ok. :) |
| 14:57.43 | *** join/#asterisk enjoi (~blahkat@unaffiliated/enjoi) |
| 14:57.48 | enjoi | greetings |
| 15:00.11 | avb | WinterNight: quick guess is that you have softphones connected behind the nat add canreinvite=no and nat=yes |
| 15:00.33 | [TK]D-Fender | 1st is a dead parameter |
| 15:00.36 | [TK]D-Fender | LONG since replaced |
| 15:00.52 | [TK]D-Fender | And much more required from what little we've seen so far |
| 15:00.57 | [TK]D-Fender | http://pastebin.com/cacMr1b8 |
| 15:01.05 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 15:01.12 | avb | [TK]D-Fender: canreinvite id dead? |
| 15:01.23 | [TK]D-Fender | That was replaced in *** 1.6 *** |
| 15:01.26 | Samot | Well very deprecated. |
| 15:01.34 | avb | [TK]D-Fender: huh? |
| 15:01.36 | Samot | But yeah, it was replaced. |
| 15:01.52 | avb | hovewer it is still working |
| 15:02.03 | avb | not sure about asterisk13, but it works in 11 |
| 15:02.10 | Samot | That's why I said "deprecated" |
| 15:02.25 | [TK]D-Fender | And should be been removed 4 branches BEFORE that |
| 15:02.33 | [TK]D-Fender | Stop using ancient junk :p |
| 15:02.36 | avb | :) |
| 15:02.48 | avb | im always forgetting about directmedia |
| 15:02.51 | avb | true |
| 15:02.58 | [TK]D-Fender | Actually... 5 branches |
| 15:03.01 | Samot | Isn't 1.6 when username was replaced with defaultuser as well? |
| 15:03.08 | [TK]D-Fender | yup |
| 15:04.25 | avb | [TK]D-Fender: have you seen if you can disable directmedia for a nated clients but keep it for a local extensions in a newer asterisks? |
| 15:04.39 | avb | i know that would screw up recording, but still interesting case :) |
| 15:04.48 | [TK]D-Fender | You can on any version |
| 15:04.56 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
| 15:04.56 | [TK]D-Fender | and it wouldn't screw up recording |
| 15:05.05 | [TK]D-Fender | if you're recording * won't ALLOW a reinvite |
| 15:05.18 | [TK]D-Fender | So itt wouldn't matter what you set |
| 15:05.27 | avb | well, thats what i mean :) |
| 15:05.30 | [TK]D-Fender | Same for DTMF functions, etc |
| 15:05.48 | avb | i think you havent understood a case :) |
| 15:05.53 | avb | 101 is behind the nat |
| 15:06.05 | avb | 102 and 103 are on the same subnet |
| 15:06.37 | avb | i want to 102 and 103 to talk directly |
| 15:06.42 | [TK]D-Fender | I think your description was poorly planned out :) |
| 15:06.46 | avb | but not directly with 101 |
| 15:06.47 | avb | :) |
| 15:06.49 | avb | indeed |
| 15:06.50 | avb | lol |
| 15:07.10 | [TK]D-Fender | If both ends ARE allowed to reinvite and * has no reason to stand in the middle then it will |
| 15:07.14 | WinterNight | This is my sip.conf: http://pastebin.com/sRzARvQU This is call debug from CLI: http://pastebin.com/iuSSSfTt I can't hear nothing. |
| 15:08.04 | [TK]D-Fender | <--- SIP read from UDP:198.27.81.118:5070 ---> |
| 15:08.04 | [TK]D-Fender | INVITE sip:080990046352022154@82.177.238.240 SIP/2.0 |
| 15:08.27 | [TK]D-Fender | Looks like you're getting attacked by someone looking to abuse your system for toll-fraud |
| 15:08.29 | *** join/#asterisk kharwell (kharwell@nat/digium/x-pruiuntvifadqwnh) |
| 15:08.29 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:08.42 | *** join/#asterisk karelk (~karel@31.10.149.26) |
| 15:09.24 | LunaLovegood | Is it possible to make VoiceMailMain() simpler for old people? Like, remove the folder options, the advanced options, the forwarding options, password/busy/name/temporary messages, etc. ? I want it to behave like the old answering maching that recorded messages on tape. |
| 15:09.37 | LunaLovegood | *machines |
| 15:09.37 | [TK]D-Fender | And the one I see failing is the 404 you are giving that user sending those calls |
| 15:09.47 | [TK]D-Fender | Retransmitting #5 (NAT) to 198.27.81.118:5070: |
| 15:09.47 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 15:09.52 | [TK]D-Fender | To: 080990046352022154<sip:080990046352022154@82.177.238.240>;tag=as15392c3b |
| 15:10.07 | igcewieling | LunaLovegood: No. |
| 15:10.24 | WinterNight | This is sip show peers: http://pastebin.com/UWwFhEQn One client is Linphone on my Android phone and second is my laptop with Jitsi client. |
| 15:10.30 | igcewieling | LunaLovegood: you can disable a few of the options, but not many. |
| 15:11.46 | LunaLovegood | oh, are there alternatives then? |
| 15:12.08 | [TK]D-Fender | <--- Reliably Transmitting (NAT) to 37.47.33.32:2034 ---> |
| 15:12.08 | [TK]D-Fender | SIP/2.0 200 OK |
| 15:12.14 | [TK]D-Fender | Contact: <sip:6001@192.168.0.2:5060> |
| 15:12.18 | [TK]D-Fender | o=root 1708820196 1708820196 IN IP4 192.168.0.2 |
| 15:12.24 | [TK]D-Fender | Passing the wrong IP for audio.. |
| 15:12.41 | [TK]D-Fender | you have not setup your externarrd, localnet, nat, etc |
| 15:12.53 | [TK]D-Fender | Go fix your [general] section |
| 15:22.57 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:22.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:35.03 | *** join/#asterisk mmlj4 (1000@ip98-163-252-185.no.no.cox.net) |
| 15:37.38 | *** join/#asterisk WinterNight (~norbert@h238240.net.pulawy.pl) |
| 15:38.40 | LunaLovegood | VoiceMail(1234@default,u) plays both the custom unavailable message, and the default one before recording the message. How do I make it only the custom one? (except when unavail.wav doesn't exist for that mailbox) |
| 15:39.25 | [TK]D-Fender | "core show application voicemail" <- |
| 15:39.31 | [TK]D-Fender | #veryfinemanuals |
| 15:40.15 | LunaLovegood | ok thanks |
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| 15:53.19 | WinterNight | Its working fine only on localnet. But why not with remote? Thank You. :) |
| 15:53.41 | [TK]D-Fender | I just TOLD YOU |
| 15:53.50 | [TK]D-Fender | <[TK]D-Fender> Passing the wrong IP for audio.. |
| 15:53.51 | [TK]D-Fender | <[TK]D-Fender> you have not setup your externarrd, localnet, nat, etc |
| 15:53.51 | [TK]D-Fender | <[TK]D-Fender> Go fix your [general] section |
| 15:53.55 | [TK]D-Fender | FIX YOUR GENERAL SETTINGS |
| 15:54.42 | WinterNight | Ok. :) |
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| 16:06.25 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:13.39 | *** join/#asterisk WinterNight (~norbert@h238240.net.pulawy.pl) |
| 16:18.09 | WinterNight | This is my sip.conf: http://pastebin.com/D1PYVeav Still not working remote connections. What can I do more? |
| 16:19.52 | Samot | localnet=192.168.0.0/255.255.255.0 <-- Completely wrong. |
| 16:20.08 | Samot | You gave a /16 a /24 subnet. |
| 16:26.17 | [TK]D-Fender | Who said it was a 16? |
| 16:26.30 | [TK]D-Fender | 0 is perfectly legit as a FIXED part of a /24 |
| 16:26.54 | [TK]D-Fender | and super-common as the actual default range on a huge amount of el-cheapo residential routers |
| 16:27.03 | *** join/#asterisk WinterNight (~norbert@h238240.net.pulawy.pl) |
| 16:27.54 | [TK]D-Fender | WinterNight, "sip show settings" |
| 16:27.57 | Samot | Yeah, true. |
| 16:30.40 | WinterNight | My sip show settings: https://pastebin.ubuntu.com/23624465/ |
| 16:32.42 | [TK]D-Fender | ok, now what ports exactly have you forwarded to your server? |
| 16:33.34 | WinterNight | Range SIP 5060-6000 and RTP 10000-20000. |
| 16:33.54 | [TK]D-Fender | show itt |
| 16:34.28 | WinterNight | This is Linksys DD-WRT. |
| 16:34.54 | *** join/#asterisk hfb (~hfb@47.139.18.193) |
| 16:36.39 | [TK]D-Fender | show it |
| 16:36.53 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ptnuubjmscnqwcfe) |
| 16:36.53 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:40.40 | WinterNight | How? I check by tcpdump RTP packets. |
| 16:40.53 | avb | you need to have disallow=all first for your allow to work |
| 16:41.05 | [TK]D-Fender | <[TK]D-Fender> show it |
| 16:41.20 | avb | and you are missing directmedia=no :) |
| 16:41.53 | WinterNight | Ok, avb I add this. |
| 16:42.39 | enjoi | just got unlazy and dug a dvd drive out of the closet, kickstart headache solved |
| 16:44.59 | *** join/#asterisk monsterco (~monsterco@CPE000db93cb535-CM1cabc0735ed0.cpe.net.cable.rogers.com) |
| 16:46.58 | avb | WinterNight: do you have rtp.conf btw? |
| 16:47.15 | avb | im not sure what are the default rtp ports |
| 16:47.20 | avb | maybe they are not 10k-20k |
| 16:47.32 | WinterNight | It's working!!! :D Thank You!!! |
| 16:47.46 | avb | general] |
| 16:47.46 | avb | rtpstart=10000 |
| 16:47.46 | avb | rtpend=20000 |
| 16:47.54 | avb | thats what should be in your rtp.conf |
| 16:48.07 | avb | nice |
| 16:48.13 | avb | god bless america |
| 16:49.17 | igcewieling | ..because they need all the help they can get. |
| 16:49.24 | igcewieling | 8-| |
| 16:49.29 | [TK]D-Fender | Pretty sure God has abandoned America... |
| 16:49.38 | [TK]D-Fender | #expat |
| 16:50.58 | WinterNight | Thank You again :) :) :) |
| 16:51.14 | *** join/#asterisk cervajs2 (~cervajs2@178.148.broadband4.iol.cz) |
| 16:52.06 | cervajs2 | is it possible turn off AMI debug? i'm trying logger add channel debug_log_123456 notice,warning,error,debug,verbose,dtmf |
| 16:52.06 | cervajs2 | core set verbose 5core set debug 5core set debug 0 managermanager set debug off |
| 16:52.15 | cervajs2 | but its still in debug log |
| 16:53.24 | Samot | ami set debug off |
| 16:53.56 | cervajs2 | my debug setup more readable http://pastebin.com/50sydwdA |
| 16:55.35 | Samot | Why are you logging debug at 5? |
| 16:55.40 | Samot | That's just a waste of space. |
| 16:55.52 | cervajs2 | its from https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:56.10 | Samot | What are you debugging? |
| 16:56.38 | cervajs2 | doesnt matter? (problem with asterisk without internet) |
| 16:56.58 | cervajs2 | manager set debug off doesnt works |
| 16:58.28 | avb | [TK]D-Fender: you moved away from us? |
| 16:58.34 | avb | USA i mean |
| 16:58.35 | avb | lol |
| 16:59.07 | avb | if god exists he abandoned the whole world :) |
| 16:59.37 | avb | cervajs2: you mean remove 'manager connected blablabla'? |
| 17:01.42 | enjoi | god was never in america ;o |
| 17:01.53 | cervajs2 | avb: does it matter if there is someone connected via AMI? if i dont want AMI debug messages in debug log |
| 17:02.03 | enjoi | he stayed back and hung in the bahamas while shit got cray |
| 17:02.19 | avb | lol |
| 17:02.35 | cervajs2 | i tried disable AMI but i still see ami debug messages :( |
| 17:02.49 | avb | cervajs2: there is a variable in manager.conf which allows you to disable this messsages |
| 17:03.05 | cervajs2 | avb: but its not working |
| 17:03.14 | [TK]D-Fender | avb, Nope, never left Canuckistan |
| 17:03.28 | cervajs2 | avb: manager show settings -> Debug: No |
| 17:03.47 | enjoi | In history when America and others delivered care packages to tribes of 3rd world countries they had no idea at the time about airplanes or civilization, eventually they started building statues worshipping them (the only explaination was they were gods in sky) |
| 17:04.03 | cervajs2 | i still see messages like this |
| 17:04.03 | cervajs2 | [Dec 13 17:59:38] DEBUG[1782] manager.c: Examining AMI event: |
| 17:04.04 | cervajs2 | Event: Newexten |
| 17:04.04 | cervajs2 | Privilege: call,all |
| 17:04.10 | avb | oh |
| 17:04.16 | enjoi | god was never on earth, if there is a creator its off somewhere else far away or everywhere, we're just tribes men worshipping airplanes right now |
| 17:04.27 | [TK]D-Fender | looks like core |
| 17:04.29 | avb | core set debug off |
| 17:04.30 | avb | ? :) |
| 17:04.56 | cervajs2 | but i want debug :) but without ami events |
| 17:07.34 | Samot | So you want a debug but not all the information it offers? |
| 17:07.52 | Samot | And again, why are you logging debugs? |
| 17:07.57 | enjoi | is that like select hearing |
| 17:08.23 | Samot | It shows WAY more information at a level that is rarely needed for average troubleshooting of problems. |
| 17:09.16 | enjoi | yesss install via cdrom sucess |
| 17:09.17 | cervajs2 | Samot: solving some deeper problem with missing internet connectivity |
| 17:09.23 | enjoi | bite me usb |
| 17:09.37 | Samot | What is the issue with the Internet connectivity? |
| 17:11.08 | enjoi | You could take your logs and have a python script put them into a nice pretty html/css layout and then set the python script to run via cron to update daily or w/e else this way you can easier categorize without having to 'selectively hear' just certain things |
| 17:11.22 | avb | core set debug channel ... |
| 17:11.34 | cervajs2 | Samot: :) thats not the point. but the problem is in strange behavior like sometimes phone unregister in call. there is missing audio between two phones in the same lan after some time etc |
| 17:11.34 | enjoi | I want to know about ALL the critters that go bump in the night, not just the happy friendly harmless ones |
| 17:12.25 | Samot | cerajs2: OK, so the phones are remote to the PBX? |
| 17:12.39 | Samot | Or on the same network? |
| 17:13.24 | cervajs2 | Samot: on same network. please stay with the question. i need debug withou AMI events |
| 17:13.35 | Samot | To solve an issue? |
| 17:13.42 | Samot | That may not need the debug at all. |
| 17:14.11 | Samot | So you're spinning your wheels on how to do something that might not even be needed. |
| 17:16.03 | cervajs2 | Samot: it's general mechanism. i often solving problems in app_queue and core set debug 5 app_queue is the best friend |
| 17:16.53 | Samot | So what is the status of a "sip show peer <device>" when a device losses "registration" during a call? |
| 17:17.19 | Samot | Does Asterisk still show it OK or UNREACHABLE or just not connected at all? |
| 17:19.24 | cervajs2 | Samot: i dont know. how this information helps? |
| 17:19.58 | Samot | You have a network issue. |
| 17:20.13 | Samot | These details help figure out what is happening. |
| 17:20.35 | Samot | See if your device has the status of UNREACHABLE it means that it's not responding to Asterisk SIP messages.. |
| 17:21.19 | Samot | And Asterisk will try for a certain amount of time to get an answer from the endpoint and after that time frame is up and it doesn't have a response, it drops the registration. |
| 17:22.02 | cervajs2 | Samot: yeah. i have pcaps for analysis. but its better to have context with asterisk dialplan + chan_sip debug btw |
| 17:22.41 | Samot | core set debug is going to debug *system* level activity. |
| 17:23.04 | Samot | sip set debug on or sip set debug <device> will show the actual SIP messages. |
| 17:25.05 | cervajs2 | back to my original question. it seems to me, its not possible turn off ami debug messages. the solution is turn off ami + kill connected clients |
| 17:35.53 | *** join/#asterisk robmal (r@wporzo.pl) |
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| 17:45.34 | Kobaz | what's the standard name/number for the +1 phone number stuff |
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| 17:52.57 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 17:56.16 | [TK]D-Fender | huh? |
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| 18:17.20 | Kobaz | [TK]D-Fender: there's a standard for it |
| 18:17.26 | Kobaz | like ansi or something |
| 18:17.31 | Kobaz | rfc |
| 18:17.32 | Kobaz | one of those |
| 18:17.33 | [TK]D-Fender | Your wording was a mess |
| 18:18.04 | [TK]D-Fender | So the name for the phone number FORMAT? |
| 18:18.07 | Kobaz | yeap |
| 18:18.12 | [TK]D-Fender | NANPA <_ |
| 18:18.20 | Kobaz | yeah yeah, but other than that |
| 18:18.29 | [TK]D-Fender | North American Number Plan Administration |
| 18:18.31 | Kobaz | specifically the fully formatted +1 .. .. .. |
| 18:18.46 | Kobaz | the +1 has like a subpart standard |
| 18:18.47 | [TK]D-Fender | 1 NXX NXX-XXXX = NANPA |
| 18:18.51 | Kobaz | 1, yes |
| 18:18.57 | Kobaz | + <---- that part |
| 18:19.08 | [TK]D-Fender | + only exists overseas and is effectively "00" |
| 18:19.21 | Kobaz | yeah yeah, i know all that |
| 18:19.26 | Kobaz | there's some sort of standards number |
| 18:22.49 | Samot | E.164 |
| 18:22.58 | Kobaz | yay |
| 18:22.59 | Kobaz | thanks |
| 18:23.01 | Samot | +1NXXNXXXXXX |
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| 18:47.27 | igcewieling | The term is E.164 |
| 18:50.28 | *** join/#asterisk thiagoc_ (~thiagoc@unaffiliated/thiagoc) |
| 18:53.01 | enjoi | Does the current AsteriskNow come with openvpn? |
| 18:59.33 | *** join/#asterisk miralin (~Thunderbi@194.8.128.47) |
| 19:00.24 | [TK]D-Fender | No, but it can be added |
| 19:05.13 | *** join/#asterisk Cuzner (~ccuzner@74.117.140.46) |
| 19:09.15 | enjoi | I was pondering on it.. |
| 19:09.57 | enjoi | I think rather I will just use a dedicated hardware vpn unit and put the box on that to keep it as secluded as I can from the public |
| 19:11.41 | enjoi | I feel that by having the vpn and the pbx on the same box it maybe puts too much of a possible target on you |
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| 19:20.59 | [TK]D-Fender | All your eggs in one basket |
| 19:23.33 | enjoi | there is a good saying for that |
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| 19:46.22 | drmessano | You shouldnt be using your PBX for your firewall |
| 19:49.22 | enjoi | so my common sense spider tingle was correct |
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| 19:59.17 | avb | drmessano: may i know the reason? :) |
| 19:59.46 | avb | lets consider that i have firewalled everything except 5060 and rtp ports |
| 19:59.57 | avb | and ssh is whitelisted |
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| 20:06.28 | drmessano | I would highly discourage using an application server as your firewall in any regard. Single point of failure, higher attack surface, etc |
| 20:06.39 | enjoi | any preference between kamailio and opensips? |
| 20:07.28 | drmessano | avb: You could put a Mikrotik in as your firewall for under $100 and then just use your PBX as your PBX. A novel idea, no doubt |
| 20:08.34 | [TK]D-Fender | avb, We mean as your internet gateway's firewall (first level exposure) |
| 20:10.32 | drmessano | Asterisk is meticulously maintained, no doubt. But most compromises are combinations of vulnerabilities. I would hate to have to explain to the boss how unpatched unprivledged shell access + an unpatched priviledge escalation turned into a mass compromise, because $70 wasn't spent an actual router. |
| 20:13.42 | [TK]D-Fender | Or that I have to reboot the server for reason X ... and that takes down ALL services until everything (hopefully) comes back up. |
| 20:13.59 | drmessano | (hopefully) +1 |
| 20:15.08 | drmessano | "What could go wrong?" - Dead Meat, Hotshots |
| 20:15.47 | [TK]D-Fender | EVERYTHING |
| 20:16.53 | [TK]D-Fender | "Stay on target" - Red Leader, X-Wing attack squadron against the 1st Death Star. |
| 20:18.05 | drmessano | This time it was right, it would work, and no one would have to get nailed to anything. |
| 20:29.12 | enjoi | Would I still need to setup a separate sip server for asterisk? |
| 20:29.39 | [TK]D-Fender | no |
| 20:29.49 | [TK]D-Fender | You only use one in huge deployments |
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| 20:48.01 | *** join/#asterisk Jinxed- (~b0ot@147.177.70.170) |
| 20:48.16 | Jinxed- | Has anyone every tried to encode to P25 standards with asterisk? |
| 20:52.48 | *** join/#asterisk nix8n82 (~AndChat58@63-155-104-69.chyn.qwest.net) |
| 21:07.59 | enjoi | So I am still head deep on my learning curve here just getting the jist of everything. I have installed Asterisk and even setup 4 extensions with SIP/IAX and 2 users. |
| 21:08.42 | enjoi | I know in order to make calls locally the users (if using hard phones) would need to connect to the pbx? (and if softphones e.g. avaya, etc..) |
| 21:09.08 | enjoi | avayax I should say I guess |
| 21:09.20 | enjoi | `yuck` |
| 21:12.29 | [TK]D-Fender | Not sure what the question in there is |
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| 21:19.19 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:25.16 | enjoi | sorry had to tend to dogs |
| 21:25.54 | enjoi | What I was getting at is, how would I connect the users and allow them to then call the outside world and allow outside world to call them? |
| 21:26.39 | enjoi | I assume after I allow outside world to call me, I would then create an IVR so the users can be set into some kind of menu to be directed to a particular queue |
| 21:29.46 | [TK]D-Fender | or just ring them directly. |
| 21:29.51 | [TK]D-Fender | Or anything you want |
| 21:30.09 | [TK]D-Fender | it's your dialplan... calls come from your phones & providers and all get processed however you want |
| 21:32.43 | avb | enjoi: i think you you are talking about obtaining a trunk |
| 21:33.13 | avb | enjoi: didww.com |
| 21:33.31 | avb | you can buy DID numbers from any part of the world |
| 21:33.51 | avb | to dial out, you would need to get an outgoing trunk |
| 21:33.57 | enjoi | ah |
| 21:34.07 | avb | provider depends a lot on what destinations do you want to dial |
| 21:34.19 | enjoi | Could I not setup my own trunk? |
| 21:34.33 | enjoi | so I could split one already owned line into multiples? |
| 21:34.41 | avb | if you have some e1/t1/pri |
| 21:34.51 | avb | you can |
| 21:35.12 | avb | you can configure trunk between asterisk and your existing pbx if you have any |
| 21:36.33 | avb | most simple solution is buy 2 pri cards for asterisk server |
| 21:37.00 | *** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net) |
| 21:37.22 | avb | the configure it in a way pri from telco -> asterisk -> pri port on old pbx |
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| 21:43.26 | igcewieling | The easiest and least expensive way is to signup with an ITSP. I personally use vitelity.net, but there are many others. |
| 21:44.10 | igcewieling | once you are no longer a n00b, then look into getting the far more expensive PRI types of lines. |
| 21:44.39 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 21:45.09 | enjoi | Nice |
| 22:00.55 | enjoi | Right now I am just kind of diving in and starting small getting a feel for it and hands on |
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| 22:45.53 | Samot | I'm not sure how one would make their "own" trunk with a E1/T1/PRI line. |
| 22:47.27 | WIMPy | Nothing special about that. |
| 22:48.08 | Samot | If you have a PRI, you don't have your "own" trunk. |
| 22:48.26 | Samot | You have something from the ISP and you put a conversion device between it and the PBX. |
| 22:48.39 | Samot | But its still their rules. |
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