| 18:08.41 | *** join/#asterisk infobot (ibot@rikers.org) |
| 18:08.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:11.06 | kahlb | unfortunately, it wasn't. |
| 18:14.53 | drmessano | WOW |
| 18:22.01 | kahlb | okay, it's obviously a NAT issue. |
| 18:22.04 | kahlb | Wow? |
| 18:23.53 | drmessano | Are you really using 192.168.0.x? |
| 18:26.43 | kahlb | No. I have a few subnets which are routed. |
| 18:27.10 | kahlb | hence 255.255.0.0 |
| 18:28.55 | iulhk | i am getting this error at CLI, "chan_sip.c:18362 check_via: Could not resolve socket address for '[2607:fb90:429:5b6a:a0c8:a9bc:c209:323d]:49420'"? |
| 18:29.21 | igcewieling | that looks ipv6'ish |
| 18:32.02 | aandrew | hm intresting |
| 18:32.06 | iulhk | <igcewieling>: getting this issue against 1 sip peer which is using iphone 7, if the same peer register through some other device, then no error. |
| 18:32.28 | aandrew | what is the difference between a SIP registration showing an IP and UNREACHABLE vs none and UNKNOWN? |
| 18:32.51 | aandrew | looks like the difference between "some traffic getting through, enough to partial-register, but not enough to be able to say registered" |
| 18:33.04 | *** join/#asterisk pankid (~quassel@mail.nschoonover.com) |
| 18:33.23 | drmessano | I dont know anything about PJSIP, but why are all your former Chan_SIP [general] settings under a [vodafone] section now? |
| 18:33.32 | igcewieling | aandrew: unreachable means "qualifty failed". unknown means, the phone has not registered at all. |
| 18:36.55 | kahlb | drmessano: That's what this sip_to_pjsip script did to me. I don't know any better. Maybe I put them to global again and try. |
| 18:37.13 | drmessano | That definitely should be cleaned up |
| 18:40.46 | kahlb | and some options, like qualify=no, are gone for good, aren't they? |
| 18:41.05 | kahlb | or is there a substitute I didn't find? |
| 18:41.54 | file | PJSIP uses templates if you want to reuse stuff |
| 18:42.02 | file | And qualify is in the aor type |
| 18:43.46 | iulhk | The complete Error can be viewd "https://paste.linux.community/view/8376c033" ? |
| 18:44.40 | iulhk | my asterisk's on ipv4, if my sip client using ipv6 then is there any change i have to do ? |
| 18:51.42 | Samot | Get an IPv6 block? |
| 18:52.30 | igcewieling | or turn off ipv6 |
| 18:52.54 | Samot | Too obvious. |
| 18:53.11 | igcewieling | iulhk: not many people use IPv6 with Asterisk. The most common hardphones don't support it. |
| 18:54.50 | drmessano | Pfft.. I use IPv6 with Asterisk |
| 18:54.54 | drmessano | I am NOT chopped liver |
| 18:57.12 | Samot | Brian1001: Please stop with the PMs. |
| 18:57.24 | Samot | I can't do anything about what IRC channels are being logged. |
| 18:57.25 | drmessano | Whoa |
| 18:57.31 | igcewieling | I've seen 1 request from our customers for IPv6 in the past 6 years |
| 18:57.35 | iulhk | <igcewieling>: my sip peers registering, making and receiving calls, just getting these Warnings and Errors at asterisk console. when peer registering by using iphone7 specifically |
| 18:58.03 | drmessano | igcewieling: I've only been using it since Bria added it |
| 18:59.15 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 19:04.56 | *** join/#asterisk KNERD (~KNERD@2604:a880:1:20::9b:4001) |
| 19:10.06 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:10.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:11.06 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 19:17.33 | vader- | Anyone in here ever connect a regular IP SIP phone to a microsoft Skype for Business server? |
| 19:28.37 | Samot | Didn't we just go through this last night? |
| 19:28.57 | Samot | This probably not the best channel for that question. |
| 19:31.53 | Samot | http://lmgtfy.com/?q=connect+SIP+phone+to+skye+business+server |
| 19:31.59 | Samot | ^ Will probably get you faster answers. |
| 19:32.26 | [TK]D-Fender | All SIP is just SIP |
| 19:32.28 | [TK]D-Fender | there is no magic |
| 19:32.33 | igcewieling | considering how nasty Skype was about Asterisk support, I'd never use them. |
| 19:32.45 | [TK]D-Fender | We shouldn'tt be answering this over and over |
| 19:33.22 | vader- | tk i don't think Skype For Business's (Lync) SIP is the same... |
| 19:33.43 | Samot | Of course not. |
| 19:33.52 | Samot | Why would MS do anything by the standards? |
| 19:34.04 | [TK]D-Fender | S4B does straight SIP |
| 19:34.26 | Samot | If you put in the host, user and password... |
| 19:34.31 | vader- | ya i can't get the phone to connect to the server |
| 19:34.31 | Samot | That's pretty much it. |
| 19:34.37 | Samot | Why? |
| 19:34.37 | vader- | or even my soft client |
| 19:34.40 | Samot | Bad password? |
| 19:34.43 | Samot | timeout? |
| 19:34.47 | vader- | timeout |
| 19:34.51 | Samot | Well.. |
| 19:34.53 | Samot | Come on.. |
| 19:35.01 | Samot | That means it can't connect to the server... |
| 19:35.14 | Samot | That's not a SIP issue. |
| 19:35.33 | vader- | i know the servers info though and i know it's correct |
| 19:35.45 | Samot | We can't help. |
| 19:35.58 | [TK]D-Fender | Firewalls, routing. USer error |
| 19:36.40 | Samot | Is this your SFB server? |
| 19:36.49 | Samot | Or are you buy service from someone? |
| 19:36.56 | Samot | s/buy/buying/ |
| 19:37.00 | vader- | it's my own hosted Sfb |
| 19:37.08 | Samot | Then look at your logs and debugs. |
| 19:37.17 | Samot | You have the tools in front of you to debug this. |
| 19:37.27 | vader- | i think they use SIP over TLS not TCP |
| 19:37.33 | vader- | thats probably why |
| 19:37.40 | Samot | It's your server. |
| 19:37.42 | Samot | Go turn it off. |
| 19:37.50 | drmessano | They use TLS |
| 19:37.54 | drmessano | I told you that last night |
| 19:37.54 | Samot | Or setup your SIP client to use TLS. |
| 19:38.10 | Samot | Is your client setup to use TLS? |
| 19:38.19 | drmessano | They've used TLS since 2003 |
| 19:38.29 | drmessano | Literally not a suprise |
| 19:38.30 | vader- | the sip clients im using don't seem to have tls as an option |
| 19:38.37 | drmessano | Then you can't do it |
| 19:38.38 | Samot | Are you kidding me?! |
| 19:38.38 | vader- | trying with an IP phone and with x-lite |
| 19:38.47 | Samot | X-Lite does TLS. |
| 19:38.53 | Samot | Most IP phones do TLS. |
| 19:38.56 | drmessano | Zoiper Does |
| 19:39.01 | Samot | Well maybe X-Lite doesn't.. |
| 19:39.02 | drmessano | Blink |
| 19:39.08 | drmessano | Bria does |
| 19:39.23 | Samot | But you are in here asking for help on connecting to this server.. |
| 19:39.26 | Samot | Multiple times.. |
| 19:39.34 | Samot | And you're using UDP to connect to TLS... |
| 19:40.14 | Samot | RTFM. |
| 19:40.39 | drmessano | IIRC clients can only connect with TLS, but you can use UDP on 5060 for connecting to a provider ONLY |
| 19:40.49 | drmessano | Users can register on 5060 |
| 19:41.01 | drmessano | Users CANT register on 5060 |
| 19:41.17 | drmessano | Actually |
| 19:41.21 | drmessano | TCP on 5060 |
| 19:41.41 | drmessano | Same as Exchange |
| 19:41.56 | drmessano | But you can't register a client on 5060 |
| 19:42.47 | *** join/#asterisk Dpunkt (~Dpunkt@p54945127.dip0.t-ipconnect.de) |
| 19:42.56 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 19:43.57 | Samot | I figured taking notes, especially on a computer on the Internet, would be easier in this day and age. |
| 19:44.13 | Samot | You know, so you don't have to ask the same people the same question over and over and over again. |
| 19:46.03 | drmessano | Funny thing is |
| 19:46.05 | drmessano | I googled |
| 19:46.12 | drmessano | Skype for Business Ports |
| 19:46.14 | Samot | I know. |
| 19:46.22 | drmessano | First result, MS article.. all the ports |
| 19:46.56 | drmessano | Same as its been since 2003 |
| 19:47.18 | drmessano | 5060 TCP for server <> server (insecure) or for PSTN |
| 19:47.51 | drmessano | 5061 TLS for server <> server or clients |
| 19:49.04 | Samot | drmessano: Stop. |
| 19:49.09 | Samot | That's too much work. |
| 19:49.39 | Samot | You just ask someone. That's easier. |
| 19:50.01 | vader- | i asked in here because i did read the docs you are mentioning |
| 19:50.11 | vader- | i did google this |
| 19:50.27 | Samot | So you knew that you couldn't use UDP on the sip client? |
| 19:51.09 | vader- | i didn't say anything about UDP i was asking about TLS requirement with skype |
| 19:51.27 | Samot | Well how are you connecting? |
| 19:51.31 | drmessano | The doc clearly states you need TLS |
| 19:51.34 | vader- | im not |
| 19:51.37 | vader- | thats the problem |
| 19:51.43 | Samot | In your client.. |
| 19:51.45 | drmessano | What are you TRYING to use? |
| 19:51.50 | Samot | What is the transport?! |
| 19:52.07 | vader- | i don't see the option for TLS in either the phone or x-lite |
| 19:52.13 | vader- | so i tried TCP and i tried UDP |
| 19:52.19 | Samot | Oh come on. |
| 19:52.20 | drmessano | Neither is TLS |
| 19:52.31 | [TK]D-Fender | So you're nott following the instructtions and using something that speaks the right language at all |
| 19:52.37 | vader- | i know but oyu just got done telling me that 5060 is used for TCP |
| 19:52.39 | vader- | so i tried that |
| 19:52.43 | drmessano | NO |
| 19:52.47 | Samot | That's just a PORT |
| 19:52.47 | vader- | and 5061 is for TLS |
| 19:52.47 | drmessano | SERVERS ONLY |
| 19:52.52 | drmessano | Thats what I told you |
| 19:52.56 | [TK]D-Fender | TLS <------- NOT just "TCP" |
| 19:53.05 | [TK]D-Fender | SECURE CERTIFICATE. |
| 19:53.07 | drmessano | 5060 TCP for server <> server (insecure) or for PSTN |
| 19:53.08 | Samot | And 5061 is not just for TLS |
| 19:53.21 | drmessano | YOU CANT REGISTER A CLIENT ON 5060.. ITS NOT LISTENING FOR YOU |
| 19:53.26 | Samot | Unless it specifically says "TLS port is 5061" |
| 19:53.28 | drmessano | 5061 ONLY |
| 19:53.34 | drmessano | TLS |
| 19:53.56 | Samot | You cannot connect using mis-matched protocols. |
| 19:54.19 | drmessano | This isn't Asterisk.. |
| 19:54.31 | drmessano | You can't just wish a port into working a certain way |
| 19:55.03 | drmessano | NONE of M$'s products use UDP for SIP |
| 19:55.17 | drmessano | None of them permit client connections on 5060 TCP |
| 19:57.18 | drmessano | I am actually surprised about no UDP on 5060.. I thought they added UDP a handful of years ago, to help facilitate PSTN |
| 19:57.28 | drmessano | But I guess that never happened |
| 19:57.32 | vader- | so it looks like there is no way this will directly work |
| 19:57.43 | drmessano | 5061 TLS |
| 19:57.55 | drmessano | You want a client to use it |
| 19:58.04 | drmessano | Just told you how to do it |
| 19:58.09 | Samot | Did you assign a TLS certificate to the server? |
| 19:58.27 | drmessano | Whoa |
| 19:58.30 | drmessano | Slow down.. |
| 19:58.35 | drmessano | He doesnt even have a client yet |
| 19:58.48 | vader- | yes the server has all the certificates assigned |
| 19:58.56 | vader- | and the lync/skype clients work |
| 19:58.56 | Samot | Public or private? |
| 19:59.02 | vader- | public |
| 19:59.04 | drmessano | So go get a friggin TLS softphone |
| 19:59.05 | Samot | OK. |
| 19:59.10 | Samot | Zoiper |
| 19:59.13 | Samot | Polycom |
| 19:59.17 | Samot | Cisco SPA |
| 19:59.18 | Samot | SNOM |
| 19:59.20 | Samot | Yealink. |
| 19:59.29 | Samot | Bria. |
| 19:59.43 | Samot | What I'm saying is, there's some options. |
| 19:59.49 | drmessano | Many |
| 20:00.00 | Samot | Like any core/name brand. |
| 20:00.21 | drmessano | Not sure how you got "no way this will work" from what we told you |
| 20:00.32 | Samot | X-Lite = Free Bria. So TLS isn't included. |
| 20:01.02 | drmessano | Yeah |
| 20:01.25 | drmessano | Blink does TLS |
| 20:01.28 | Samot | Zoiper supports TLS in various ways in it's free version. |
| 20:07.01 | *** join/#asterisk libardi (~libardi@187.64.235.241) |
| 20:13.53 | drmessano | Ok, so basically you're saying this won't work? |
| 20:17.50 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 20:29.33 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-qbtefkhsykvuxrqi) |
| 20:29.33 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:37.40 | *** join/#asterisk hdon (~hdon@68.110.137.138) |
| 20:38.25 | hdon | hi all :) i'm using asterisk 14.2.1 and i can't delete an urgent message. i've found the file, though. is it safe for me to just "rm" the file? |
| 20:39.52 | igcewieling | hdon: delete both the audio and the .txt file. I'd actually move them somewhere else in case something Bad happens. IIRC Asterisk will re-number the messages next time voicemail is called |
| 20:43.27 | hdon | thanks igcewieling |
| 20:45.07 | *** join/#asterisk robinak (~quassel@unaffilated/robink) |
| 21:00.15 | hdon | okay, i have a more interesting question now. if doing an Attended Transfer with a feature code, if i dial the wrong extension, is there any way i can back out of it until the other end hangs up? for example, if i call an extension and get a busy signal, i can't figure out how to back out, and i just have to wait until the busy signal hangs up on me. |
| 21:10.05 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:14.14 | Samot | atxferabort feature map. |
| 21:16.47 | hdon | ahhh thanks Samot ! |
| 21:18.44 | hdon | while i'm on the subject of feature maps... why does my disconnect feature code "**" seem to do nothing when i'm in a call? |
| 21:19.29 | hdon | "features show" says that my Current code for "Disconnect" is "**" but nothing seems to happen when i dial it while in a call |
| 21:21.03 | hdon | hmm maybe i don't have an "h" extension... |
| 21:28.32 | *** join/#asterisk SeiGGy (~zway@74.203.105.194) |
| 21:28.33 | *** join/#asterisk zway (~zway@74.203.105.194) |
| 21:29.32 | *** join/#asterisk SeiGGy (~zway@74.203.105.194) |
| 21:40.23 | rmudgett | hdon: Features have to be enabled to have an effect. Depending on the feature they are controlled by Dial application flags or the DYNAMIC_FEATURES channel variable. |
| 21:40.58 | hdon | oh, hmm... |
| 22:18.33 | *** join/#asterisk bari (~bari@2a01:8c00:ff00:80d5:7f98:4fc6:3e8:d181) |
| 22:20.05 | Samot | hdon: Did you reload things after you made changes? |
| 22:20.07 | hdon | is there a way to assign feature codes from the asterisk shell? |
| 22:20.33 | hdon | Samot, i didn't change my Disconnect code. it's been "**" all day |
| 22:20.40 | *** join/#asterisk Pasha (~Cory@unaffiliated/cory) |
| 22:20.51 | Samot | Show your features.conf |
| 22:20.54 | Samot | ~pb |
| 22:20.54 | infobot | extra, extra, read all about it, 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. |
| 22:21.59 | hdon | Samot, i'm using freepbx so it has like four files which are nominally feature-code related |
| 22:22.19 | bari | Hello, is possible to use asterisk only for securing communication? I have SIP HW phone without TLS/SRTP support. My SIP provider support it. I would like to make something like "proxy". Is it possible with asterisk? What should I look for? :) |
| 22:22.22 | hdon | or six if you count the ones it leaves for you to write |
| 22:22.28 | igcewieling | ~freepbx |
| 22:22.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:22.35 | hdon | yeah i know :3 |
| 22:27.41 | Samot | hdon then those are controlled through the Feature Codes module. |
| 22:31.36 | *** join/#asterisk newtonr (~newtonr@173-17-133-211.client.mchsi.com) |
| 22:31.36 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 22:39.41 | hdon | thanks Samot |
| 22:55.03 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 22:55.06 | *** join/#asterisk putnopvut_ (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:55.07 | *** mode/#asterisk [+o putnopvut_] by ChanServ |
| 23:16.35 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:22.30 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:29.13 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:39.24 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:49.41 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:59.54 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |