| 00:01.24 | *** part/#asterisk monsterco (~monsterco@70.50.210.158) |
| 00:01.28 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:22.19 | *** join/#asterisk u0m3 (~u0m3@5-12-141-224.residential.rdsnet.ro) |
| 00:28.50 | *** join/#asterisk danjenkins (danjenkins@gateway/shell/firrre/x-hwopcllampzttmub) |
| 00:30.59 | *** join/#asterisk Cubber (~ronny@mail.ronnybull.com) |
| 00:49.06 | MajesticFudgie | None of my Sip Clients will work now that I have PJSIP .-. |
| 00:49.19 | MajesticFudgie | Is there a way I can stop it from doing this |
| 00:53.51 | MajesticFudgie | Or has it swapped from using sip.conf to pjsip.conf |
| 00:55.05 | *** join/#asterisk elyob (~elyob@host86-134-76-198.range86-134.btcentralplus.com) |
| 00:57.22 | MajesticFudgie | Oh I think I have it working \o/ |
| 01:12.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:33.38 | *** join/#asterisk elyob (~elyob@host86-134-76-198.range86-134.btcentralplus.com) |
| 01:38.19 | MajesticFudgie | Asterisks consoles making no sense |
| 01:38.33 | MajesticFudgie | It's unable to see something I'm staring at |
| 01:41.03 | *** join/#asterisk degan6 (~degan6@c-73-151-71-161.hsd1.ca.comcast.net) |
| 01:44.49 | MajesticFudgie | Extension state subscription failed: Extension 6003 does not exist in context 'from-internal' or has no associated hint |
| 01:44.58 | MajesticFudgie | I have no idea where it's pulling 6003 from |
| 01:45.03 | MajesticFudgie | or what it means ._. |
| 01:45.22 | MajesticFudgie | Could someone enlighten me please? |
| 01:47.12 | Silicon | uh |
| 01:47.24 | Silicon | one of your endpoints have it set as a BLF probably |
| 01:47.41 | Silicon | trying to subscribe to it |
| 01:48.22 | MajesticFudgie | Idk what that means :P |
| 01:48.34 | Silicon | one a phone |
| 01:48.39 | Silicon | u know those little buttons you hit |
| 01:48.39 | MajesticFudgie | https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 01:48.42 | Silicon | :P |
| 01:48.44 | MajesticFudgie | Im working off this as a config lol |
| 01:49.05 | MajesticFudgie | I dont have it set to BLF |
| 01:49.12 | MajesticFudgie | I'd need the pro software to lol |
| 01:49.29 | Silicon | you have no devices hooked to it? |
| 01:49.46 | MajesticFudgie | only thing that can be hooked to it is Zoiper on my PC and SipGate |
| 01:50.40 | Silicon | if you turn on more debugging it should tell you what is trying to subscribe |
| 01:50.45 | Silicon | and making that error |
| 01:51.01 | MajesticFudgie | How else would I increase the debugging |
| 01:51.08 | MajesticFudgie | Im connecting with asterisk -rvvvvvvv |
| 01:51.09 | MajesticFudgie | lol |
| 01:51.12 | Silicon | pjsip set logger on |
| 01:51.23 | Silicon | in ur console |
| 01:51.27 | MajesticFudgie | ah |
| 01:52.07 | MajesticFudgie | done, Ill wait for it to happen again |
| 01:53.04 | Silicon | ya |
| 01:53.09 | Silicon | it should tell you |
| 01:53.37 | MajesticFudgie | http://paste.ubuntu.com/16229455/ |
| 01:54.16 | Silicon | what device has 192.168.1.149 as a ip? :P |
| 01:54.24 | MajesticFudgie | I assume my own PC |
| 01:54.39 | MajesticFudgie | Yup |
| 01:54.44 | Silicon | err wait |
| 01:54.44 | Silicon | 192.168.2.204 |
| 01:54.51 | Silicon | i assume is the device? |
| 01:54.53 | MajesticFudgie | Oh, The asterisk server |
| 01:55.01 | Silicon | ok |
| 01:55.08 | Silicon | so something on your pc |
| 01:55.27 | Silicon | is trying to see if that extension is up |
| 01:55.35 | MajesticFudgie | oh |
| 01:55.42 | MajesticFudgie | X-Lite was open |
| 01:55.45 | MajesticFudgie | I wonder if it was that |
| 01:55.55 | Silicon | if it doesn't happen again |
| 01:56.00 | Silicon | there's your answer :P |
| 01:56.11 | MajesticFudgie | lol I think that solved it |
| 01:56.18 | MajesticFudgie | How familiar with PJSIP are you? |
| 01:56.26 | Silicon | not very |
| 01:56.32 | Silicon | new to asterisk myself |
| 01:56.45 | Silicon | i know more about it now though than i care to :P |
| 01:56.47 | MajesticFudgie | I'm having a rough time tryin to get my setup working with a SIP Provider, SIPGATE in this case |
| 01:57.42 | Silicon | they should provide you with a config that should work |
| 01:57.55 | MajesticFudgie | Not working entirely |
| 01:57.59 | MajesticFudgie | Calls come in |
| 01:58.05 | MajesticFudgie | but nothing really works as intended |
| 01:58.17 | MajesticFudgie | It fails to execute the extension its supposed to |
| 01:58.25 | MajesticFudgie | and outgoing calls just sit quiet |
| 01:58.37 | Silicon | you got your outgoing routes setup |
| 01:58.39 | MajesticFudgie | and the number I dial doesnt ring, in this case my mobile phone |
| 01:58.40 | Silicon | and |
| 01:58.54 | Silicon | set to sipgate as a trunk |
| 01:59.02 | MajesticFudgie | How do I set sipgate as a trunk |
| 01:59.06 | MajesticFudgie | I may already have done it |
| 01:59.32 | Silicon | are you just using config files or using a gui frontend to configure? |
| 01:59.41 | MajesticFudgie | config files |
| 01:59.48 | MajesticFudgie | Good old SSH and such |
| 01:59.48 | MajesticFudgie | lol |
| 01:59.52 | Silicon | ya i dunno how to do it with configs |
| 01:59.55 | Silicon | i'm not that good |
| 02:00.03 | MajesticFudgie | http://basichelp.sipgate.co.uk/hc/en-gb/articles/206404129-Asterisk-PBX is what I followed |
| 02:00.11 | MajesticFudgie | I've done it to the letter |
| 02:03.16 | Silicon | ya |
| 02:03.19 | Silicon | you need an outbound route |
| 02:04.24 | MajesticFudgie | The inbound route is wierd |
| 02:04.44 | MajesticFudgie | If I dont put the right extension in, it'll fail looking for the extension it needs |
| 02:04.48 | MajesticFudgie | In this case my SipGate ID |
| 02:04.55 | MajesticFudgie | If I add it, nothing happens |
| 02:05.00 | MajesticFudgie | No errors or anything |
| 02:05.04 | MajesticFudgie | It just sits there |
| 02:06.41 | MajesticFudgie | Oh |
| 02:06.43 | MajesticFudgie | unless |
| 02:13.09 | MajesticFudgie | Still not fixed it |
| 02:13.12 | MajesticFudgie | No clue anymore |
| 02:27.36 | MajesticFudgie | None of my extensions are working now :< |
| 02:48.53 | MajesticFudgie | yeah my extensions dont work at all |
| 02:49.54 | *** join/#asterisk KerioMorgan (~Adium@us.kerio.com) |
| 02:55.37 | MajesticFudgie | I have the extension '1' when I call it, it should say 123 and hang up |
| 02:55.44 | MajesticFudgie | It just sits silent and doesnt answer |
| 02:56.01 | MajesticFudgie | calling anything other than 1 will refuse to ring as it doesnt exist |
| 02:56.09 | MajesticFudgie | It sees the extension but doesn't process it |
| 03:02.37 | MajesticFudgie | bbl, H/L me if anyone has any idea why that'd happen |
| 03:10.22 | Samot | MajesticFudgie: Do you have a debug of the call? It will tell you what is happening... |
| 03:10.37 | MajesticFudgie | Which debug would I need |
| 03:12.47 | *** join/#asterisk zz_mzb (~mzb@ppp108-88.static.internode.on.net) |
| 03:13.01 | MajesticFudgie | http://paste.ubuntu.com/16230083/ is the output of 'pjsip set logger on' ~ Samot |
| 03:13.25 | Samot | Did you increase the verbosity? |
| 03:13.37 | MajesticFudgie | Not entirely sure how to |
| 03:13.53 | MajesticFudgie | Still learning the commands and such lol |
| 03:14.22 | Samot | asterisk -rvvvvvvvvvv |
| 03:14.25 | Samot | Then do the call |
| 03:14.34 | Samot | Then pb the results. |
| 03:14.36 | Samot | ~pb |
| 03:14.36 | 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. |
| 03:14.42 | MajesticFudgie | kk sec |
| 03:14.47 | MajesticFudgie | I had it with a few v's |
| 03:15.01 | MajesticFudgie | So far nothing logs |
| 03:15.11 | Samot | There's none of that data in the debug |
| 03:15.22 | MajesticFudgie | Nothings logging atm |
| 03:15.28 | Samot | If you're in * CLI just do: core set verbose 10 |
| 03:15.31 | MajesticFudgie | Bearing in mind all I've done is connect with asterisk -rvvvvvvvvvv |
| 03:15.33 | Samot | Then make a call. |
| 03:15.39 | Samot | You need to make a call. |
| 03:15.50 | MajesticFudgie | Still nothing |
| 03:15.53 | MajesticFudgie | I'm making the call |
| 03:16.04 | MajesticFudgie | Nothings logging at all |
| 03:16.14 | Samot | Exit out |
| 03:16.20 | Samot | do asterisk -rvvvvvvvvvvv |
| 03:16.22 | MajesticFudgie | https://surl.im/i/4ydxo.png |
| 03:18.13 | Samot | pjsip set logger on |
| 03:18.16 | Samot | Make the call |
| 03:18.23 | Samot | Grab everything and pb it |
| 03:18.37 | MajesticFudgie | Shall I let the call time out on its own? |
| 03:18.47 | MajesticFudgie | As it'll hang up I think after about a minute |
| 03:19.30 | *** join/#asterisk vader- (~Adium@pool-100-14-222-238.phlapa.fios.verizon.net) |
| 03:19.55 | Samot | Yes. |
| 03:20.03 | MajesticFudgie | kk |
| 03:21.28 | MajesticFudgie | It's not timing out atm |
| 03:22.44 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 03:23.29 | MajesticFudgie | http://paste.ubuntu.com/16230136/ I ended it in the end |
| 03:31.45 | Samot | [May 5 04:21:44] NOTICE[14485]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from '"1" <sip:1@81.135.231.20>' failed for '89.163.242.100:5070' (callid: c680f54407baa66d2cf3b04869ec0826) - No matching endpoint found |
| 03:32.05 | MajesticFudgie | idk what that is |
| 03:32.24 | MajesticFudgie | I don't recognise the 2nd IP At all |
| 03:33.16 | Samot | This one? 89.163.242.100? |
| 03:33.36 | Samot | Your box is wide open. |
| 03:33.44 | Samot | They are trying to make calls to you. |
| 03:33.50 | Samot | They are trying to make calls through you. |
| 03:33.59 | MajesticFudgie | oh? |
| 03:34.27 | MajesticFudgie | atm I have DMZ on as my router refuses to play ball with my port forwards |
| 03:34.37 | MajesticFudgie | namely UDP 10000:20000 |
| 03:34.46 | MajesticFudgie | Other than that idk |
| 03:34.53 | Samot | https://www.irccloud.com/pastebin/KjMZZO3q/ |
| 03:35.01 | Samot | Is that you making that call? |
| 03:35.14 | Samot | DMZ? |
| 03:35.21 | Samot | Fix your stuff. |
| 03:35.34 | Samot | You've opened your Asterisk box to just raped. |
| 03:35.42 | drmessano | 5060 UDP |
| 03:35.47 | drmessano | 10000-20000 UDP |
| 03:35.53 | Samot | You've opened your Asterisk box to just be raped. |
| 03:35.54 | Samot | 5061 UDP |
| 03:35.56 | MajesticFudgie | The 81. IP is mine |
| 03:36.00 | drmessano | Nothing to play ball about |
| 03:36.08 | drmessano | Just open what you need |
| 03:36.23 | Samot | Plus I don't see your inbound call at all. |
| 03:36.35 | Samot | Are you calling from the PSTN into the PBX or from an extension on the PBX? |
| 03:36.48 | MajesticFudgie | I did open what I need but my router downright refuses to open the 10000-20000 UDP |
| 03:36.50 | Samot | Oh and stop using "1" as an extension. |
| 03:36.55 | MajesticFudgie | I'm not |
| 03:37.24 | Samot | 10:55 PM <MajesticFudgie> I have the extension '1' when I call it, it should say 123 and hang up |
| 03:37.31 | MajesticFudgie | that was originally |
| 03:37.33 | MajesticFudgie | It's not 1001 |
| 03:37.35 | MajesticFudgie | now* |
| 03:37.37 | Samot | OK |
| 03:37.40 | Samot | Quit changing things |
| 03:37.44 | Samot | Without telling us. |
| 03:37.53 | Samot | I was looking for ext 1 |
| 03:38.07 | MajesticFudgie | Idk where that call is coming from |
| 03:38.12 | Samot | Right. |
| 03:38.17 | Samot | Because your box is open to the Internet |
| 03:38.19 | MajesticFudgie | It looks like its From me, To Me |
| 03:38.23 | MajesticFudgie | Through something else |
| 03:38.24 | Samot | No. |
| 03:38.26 | MajesticFudgie | or 'via' |
| 03:38.41 | Samot | Ext "1" is a common extension used by sip hacks. |
| 03:38.46 | MajesticFudgie | mkay |
| 03:38.49 | MajesticFudgie | and define Open |
| 03:38.57 | Samot | You have your box open and you have anonymous and guest SIP calls allowed. |
| 03:39.04 | MajesticFudgie | idk how to stop the guest calls |
| 03:39.11 | drmessano | turn off guest |
| 03:39.15 | drmessano | turn off anonymous |
| 03:39.17 | drmessano | Done |
| 03:39.17 | MajesticFudgie | Point me to the right config and I will |
| 03:39.22 | drmessano | SIP Settings |
| 03:39.24 | Samot | Open as in "No condom to stop STDs" |
| 03:39.26 | drmessano | In the GUI |
| 03:39.31 | MajesticFudgie | uh |
| 03:39.37 | drmessano | OOPS |
| 03:39.37 | MajesticFudgie | Im using SSH && N++ |
| 03:39.41 | MajesticFudgie | Theres no GUI :P |
| 03:39.45 | drmessano | Wrong channel lol |
| 03:39.46 | Samot | Wrong channel man. |
| 03:39.47 | drmessano | HAHAHHAHHA |
| 03:39.49 | MajesticFudgie | lol |
| 03:39.56 | drmessano | sip.conf |
| 03:40.16 | drmessano | Not used to seeing Samot in here |
| 03:40.18 | MajesticFudgie | I assume I'll need to add some conf directives |
| 03:40.53 | Samot | Hah. |
| 03:41.01 | Samot | Yes |
| 03:41.01 | drmessano | Youre using PJSIP? |
| 03:41.05 | MajesticFudgie | trying to |
| 03:41.07 | Samot | And take your box off DMZ. |
| 03:41.08 | MajesticFudgie | and WHAT |
| 03:41.15 | MajesticFudgie | allowguest is yes by default |
| 03:41.53 | drmessano | Yeah thats no good |
| 03:42.00 | MajesticFudgie | Done it's off |
| 03:42.22 | MajesticFudgie | I can't take my box off DMZ without losing port range 10000,20000 |
| 03:42.27 | Samot | Make sure your NAT and port forwards are working. |
| 03:42.34 | Samot | Then make sure they work. |
| 03:42.36 | drmessano | Why not? |
| 03:42.37 | MajesticFudgie | As the router I have decides those ports conflict |
| 03:42.47 | Samot | ..... |
| 03:42.52 | Samot | Is something else using them? |
| 03:42.54 | drmessano | What router does that? |
| 03:42.55 | MajesticFudgie | nope |
| 03:43.00 | MajesticFudgie | Apparently a BT Hub |
| 03:43.05 | Samot | Hub? |
| 03:43.27 | Samot | Oh man. |
| 03:43.33 | Samot | This is a DSL modem. |
| 03:43.34 | MajesticFudgie | Modem+Router combo |
| 03:43.36 | MajesticFudgie | Yup |
| 03:43.44 | MajesticFudgie | https://surl.im/i/iz1wr.png |
| 03:43.47 | MajesticFudgie | fking stupid |
| 03:44.00 | Samot | So you have a piece of crap router. |
| 03:44.01 | MajesticFudgie | It apparently conflicts with something yet doesnt know what |
| 03:44.07 | MajesticFudgie | It certainly seems so |
| 03:44.11 | Samot | Probably because BT does VOIP? |
| 03:44.28 | MajesticFudgie | Not sure |
| 03:44.35 | Samot | You should call them. |
| 03:44.45 | MajesticFudgie | I'm still getting someone trying to call |
| 03:44.47 | Samot | Unless you have full access to the router... |
| 03:44.49 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:cd4b:c920:cb93:5605) |
| 03:44.51 | MajesticFudgie | NOTICE[15586]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from '"sipvicious" <sip:100@1.1.1.1>' failed for '188.138.41.20:5115' (callid: 961520489068856196369328) - No matching endpoint found |
| 03:44.53 | MajesticFudgie | Or something |
| 03:45.04 | Samot | Is your box still on DMZ? |
| 03:45.17 | MajesticFudgie | Yes, but it wont change anything |
| 03:45.33 | Samot | So iptable them out. |
| 03:46.04 | drmessano | or just ignore them |
| 03:46.16 | MajesticFudgie | DMZ is off, fallen back to portforwarding everything other than the UDP Range |
| 03:46.21 | drmessano | If youre not actually doing anything with the call, ignore it |
| 03:46.38 | MajesticFudgie | atm im just getting the no matching endpoint stuff |
| 03:46.44 | drmessano | If your auth is strong and your dialplan is sane, forget it |
| 03:46.48 | drmessano | Yep |
| 03:46.51 | drmessano | So ignore it |
| 03:47.01 | drmessano | Welcome to having SIP |
| 03:47.06 | drmessano | Thats basically it |
| 03:47.09 | MajesticFudgie | lol |
| 03:47.23 | MajesticFudgie | I'm used to it with HTTP, IRC, Spigot, FTP etc |
| 03:47.28 | drmessano | Do you run apache on the internet? |
| 03:47.41 | MajesticFudgie | I have apache on my public servers |
| 03:47.45 | MajesticFudgie | This is an internal test server |
| 03:47.48 | drmessano | Surely you get logs with code red attempts, right? |
| 03:48.12 | MajesticFudgie | Probably |
| 03:48.15 | MajesticFudgie | I don't check often |
| 03:48.24 | drmessano | Yep.. Not much they are gonna do to a Linux box |
| 03:48.24 | MajesticFudgie | On another note, The call still doesnt go through |
| 03:48.31 | drmessano | So we ignore them |
| 03:48.54 | *** join/#asterisk zz_mzb (~mzb@ppp108-88.static.internode.on.net) |
| 03:49.10 | MajesticFudgie | I'm spotting a lot of SIP/2.0 401 Unauthorized |
| 03:49.26 | MajesticFudgie | But I don't understand whats causing it to throw that |
| 03:50.10 | MajesticFudgie | I know XLite is connected as dialing an invalid extension works |
| 03:51.14 | Samot | Dial a working extension. |
| 03:51.20 | Samot | How many extensions do you have setup? |
| 03:51.25 | MajesticFudgie | 1 |
| 03:51.29 | MajesticFudgie | sec |
| 03:51.52 | MajesticFudgie | http://paste.ubuntu.com/16230285/ |
| 03:51.58 | MajesticFudgie | Thats literally it |
| 03:52.27 | MajesticFudgie | dialing 1001 sits there quiet logging nothing and doing nothing |
| 03:52.34 | MajesticFudgie | dialing anything else plays the invalid tone |
| 03:52.37 | MajesticFudgie | and hangs up |
| 03:58.20 | Samot | Wait... |
| 03:58.27 | Samot | You only have one extension? |
| 03:58.41 | MajesticFudgie | atm yes |
| 03:58.41 | Samot | How can you test calls? |
| 03:58.44 | MajesticFudgie | I had more |
| 03:58.50 | MajesticFudgie | Well I cant test anything |
| 03:58.53 | MajesticFudgie | As extensions wont work |
| 03:59.02 | Samot | Because of the audio issues? |
| 03:59.05 | MajesticFudgie | All I'm trying is the hello world extension |
| 03:59.09 | MajesticFudgie | It doesnt even answer the call |
| 03:59.17 | MajesticFudgie | Xlite sits calling |
| 03:59.23 | Samot | Are you calling 1001 from 1001? |
| 03:59.40 | MajesticFudgie | Im dialling 1001 from 'thomas |
| 03:59.47 | Samot | What is "thomas"? |
| 03:59.52 | Samot | That means nothing to me. |
| 03:59.53 | MajesticFudgie | My Device, XLite |
| 03:59.58 | Samot | OK |
| 04:00.05 | Samot | How is it registered to the PBX? |
| 04:00.12 | *** join/#asterisk vader- (~Adium@pool-100-14-222-238.phlapa.fios.verizon.net) |
| 04:00.23 | MajesticFudgie | it connects using the credentials in pjsip.conf |
| 04:00.47 | Samot | Huh? |
| 04:01.57 | MajesticFudgie | It's classed as an endpoint in pjsip |
| 04:02.35 | MajesticFudgie | I set it up using the example config on the wiki for pjsip.conf as it refused X-Lites connection until I did |
| 04:03.02 | Samot | OK well XLite is ending the call. |
| 04:03.16 | Samot | XLite is sending the INVITE to the PBX |
| 04:03.26 | Samot | The PBX sends back 100 Trying |
| 04:03.28 | Samot | CANCEL sip:1001@192.168.2.204 SIP/2.0 |
| 04:03.41 | Samot | And XLite immediately sends that back. |
| 04:03.50 | MajesticFudgie | Wierd |
| 04:03.56 | MajesticFudgie | why would it do that |
| 04:04.02 | Samot | I don't know. |
| 04:04.07 | Samot | I can't see any of the call logs. |
| 04:04.15 | Samot | Does anything show up in /var/log/asterisk/full? |
| 04:04.43 | Samot | I see almost no verbosity output in your debugs. |
| 04:04.55 | Samot | That actually show the call being handled by the PBX. |
| 04:05.12 | Samot | So me the dialplan that allows 1001 to accept calls. |
| 04:05.29 | MajesticFudgie | <PROTECTED> |
| 04:05.37 | Samot | Do you have any logs for asterisk? |
| 04:06.10 | MajesticFudgie | Yes but they're not different from the console |
| 04:06.12 | Samot | Show me the dialplan that allows 1001 to accept calls. |
| 04:06.38 | MajesticFudgie | sec |
| 04:06.40 | Samot | Better yet. |
| 04:06.41 | MajesticFudgie | I did put it in here somewhere |
| 04:06.44 | Samot | All of extensions.conf |
| 04:07.29 | MajesticFudgie | http://paste.ubuntu.com/16230459/ |
| 04:07.32 | MajesticFudgie | Thats literally it |
| 04:07.48 | MajesticFudgie | 1 extension with the purpose of saying hello world |
| 04:07.56 | MajesticFudgie | While I try and get it working |
| 04:08.06 | Samot | OK.. |
| 04:08.48 | Samot | Show your pjsip.conf |
| 04:09.03 | MajesticFudgie | sure, sec |
| 04:09.26 | MajesticFudgie | http://paste.ubuntu.com/16230503/ - I'll change the sipgate pass before I go to bed lol |
| 04:10.18 | MajesticFudgie | I just spotted something, although Im unsure how much of a difference it makes |
| 04:10.59 | MajesticFudgie | hm |
| 04:11.01 | MajesticFudgie | line 8 |
| 04:11.20 | MajesticFudgie | uncommenting that makes Zoiper throw 401 |
| 04:11.39 | MajesticFudgie | Ill test with xlite |
| 04:12.03 | MajesticFudgie | oh |
| 04:12.06 | MajesticFudgie | my own fault |
| 04:13.29 | MajesticFudgie | yeah no change |
| 04:13.41 | MajesticFudgie | Line 8 makes no difference |
| 04:15.23 | Samot | Well I'm not see the call execute any dialplan at all. |
| 04:15.43 | Samot | You should see it executing that default dialplan. |
| 04:15.49 | Samot | when you do asterisk -rvvvvvvvvvv |
| 04:16.23 | MajesticFudgie | Nothing logs |
| 04:16.31 | MajesticFudgie | It acts like it executes it |
| 04:16.42 | MajesticFudgie | As any other number than 1001 does come up invalid on the SIP Phone |
| 04:16.52 | Samot | Do you see it log that? |
| 04:16.59 | MajesticFudgie | yes |
| 04:16.59 | MajesticFudgie | [May 5 05:16:47] NOTICE[17108]: res_pjsip_session.c:2020 new_invite: Call from 'thomas' (UDP:192.168.1.149:55323) to extension '123' rejected because extension not found in context 'default'. |
| 04:17.11 | Samot | That's not dialplan. |
| 04:17.18 | MajesticFudgie | hm? |
| 04:17.23 | Samot | Well wait..sorry. |
| 04:19.46 | Samot | K |
| 04:19.48 | Samot | Try again. |
| 04:19.54 | Samot | asterisk -rvvvvvvvvvv |
| 04:20.08 | Samot | pjsip set logger host 192.168.1.149 |
| 04:20.20 | Samot | .149 is your softphone right? |
| 04:20.43 | [TK]D-Fender | extension '123' rejected because extension not found in context 'default'. |
| 04:20.48 | [TK]D-Fender | this means exactly what it says |
| 04:21.00 | [TK]D-Fender | and you should never every have a context named [default] |
| 04:21.08 | [TK]D-Fender | ever* |
| 04:21.14 | MajesticFudgie | It's for testing |
| 04:21.19 | MajesticFudgie | It doesn't matter what I had anyway |
| 04:21.28 | MajesticFudgie | I had it named from-internal or something earlier |
| 04:21.38 | MajesticFudgie | and yes Samot, it is |
| 04:22.22 | Samot | So do that and make a call. |
| 04:22.22 | [TK]D-Fender | Should keep it that way. |
| 04:22.40 | [TK]D-Fender | your [general] section should point it to a black-hole context, and your peers to a real one. |
| 04:22.52 | MajesticFudgie | Samot, same as earlier so I ended it ill pb the log |
| 04:23.18 | MajesticFudgie | http://paste.ubuntu.com/16230568/ |
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| 05:13.34 | Samot | MajesticFudgie: |
| 05:13.52 | Samot | In the * CLI do the following: |
| 05:13.57 | Samot | core set verbose 10 |
| 05:14.01 | Samot | core set debug 10 |
| 05:14.10 | Samot | pjsip set logger host 192.168.1.149 |
| 05:14.13 | Samot | Make the call |
| 05:14.18 | Samot | pb the results. |
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| 11:08.36 | MajesticFudgie | Samot http://paste.ubuntu.com/16235479/ - I ended the call shortly after it failed to do anything |
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| 11:37.45 | Samot | MajesticFudgie: Nothing has changed. |
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| 11:51.00 | Samot | MajesticFudgie: You should try changing it over to Chan_SIP and testing things that way. |
| 11:51.21 | Kunsi | MajesticFudgie: |
| 11:51.23 | Kunsi | <--- Received SIP request (665 bytes) from UDP:192.168.1.149:55323 ---> |
| 11:51.24 | Kunsi | CANCEL sip:1001@192.168.2.204 SIP/2.0 |
| 11:56.41 | MajesticFudgie | Kunsi hm? |
| 11:56.49 | MajesticFudgie | Samot how would I do that |
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| 12:16.11 | Samot | MajesticFudgie: Come on man, you got to google a bit. |
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| 12:18.06 | MajesticFudgie | So far googling has gotten me into a total mess |
| 12:18.24 | Samot | The read the Asterisk Wiki |
| 12:18.28 | Samot | ~book |
| 12:18.28 | 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 |
| 12:18.36 | Samot | Or get the book. |
| 12:18.58 | Samot | It's one thing to *help* you, it's another to be *teaching* you. |
| 12:20.27 | MajesticFudgie | there |
| 12:20.31 | MajesticFudgie | Disabled pjsip |
| 12:20.36 | MajesticFudgie | and its all working again as intended |
| 12:21.02 | Samot | You can actually see the call hitting |
| 12:21.10 | Samot | With asterisk -rvvvvvvvvv |
| 12:21.12 | Samot | ? |
| 12:21.33 | MajesticFudgie | yup |
| 12:21.40 | MajesticFudgie | Everythings working, audio, extensions the lot |
| 12:23.12 | Samot | So this wasnt a case of stuff not working over all |
| 12:23.20 | Samot | You just couldn't do PJSIP. |
| 12:23.45 | MajesticFudgie | Well define couldn't |
| 12:23.51 | MajesticFudgie | Whether Asterisk couldn't |
| 12:23.54 | MajesticFudgie | or whether I couldn't xD |
| 12:24.00 | Samot | The latter. |
| 12:24.16 | Samot | Mostly misconfigured. |
| 12:24.51 | MajesticFudgie | Well no one found an issue with my pjsip |
| 12:27.46 | MajesticFudgie | and I got sipgate working |
| 12:27.56 | MajesticFudgie | Which is wierd, it wasnt working last night using chan_sip |
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| 12:48.56 | mvp | Hi, may someone helpme please? I´m getting the following error "Unable to specify channel 1: No such device or address" after running "dahdi_gen" and "dahdi restart" on CentOS 7, what´s happening and how can I solve that? |
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| 13:01.02 | swimmercol | HI folks....I have this scenario: Cisco_phone --- Asterisk-A ------ Asterisk B ------ SIP_Provider |
| 13:01.02 | swimmercol | And I am trying to have this working in the way that Cisco_phone send direcly the Media(RTP) to the SIP_Provider bypassing the 2 Asterisk |
| 13:01.25 | swimmercol | I tried some combinations of: directrtpsetup,direcmedia,canreinvite,nat settings and didn't have luck |
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| 13:11.56 | Penguin | swimmercol: In addition to having directmedia set to yes, you must also be sure that your extensions to not contain applications or options that will cause asterisk to stay in the media stream. |
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| 13:14.01 | swimmercol | Penguin: got it, I tried to bypass Asterisk B...there...once I receive the call I just executed a Dial that's it...even the Media is going through Asterisk B |
| 13:14.45 | Penguin | swimmercol: Do you have any Dial options? |
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| 13:31.06 | swimmercol | you mean any options in the Dial command? |
| 13:35.07 | Penguin | I thought that was what I asked. |
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| 13:49.32 | ruel | mvp, sounds like your Dahdi is broken possibly? Maybe the correct module hasn't been loaded. |
| 13:51.30 | lvlinux | swimmercol: Does your sip provider entry also have directmedia allowed in your sip.conf? |
| 13:52.43 | swimmercol | I found some commands in Dial which could force Asterisk to stay in the media path |
| 13:52.53 | swimmercol | going to do some tests and then put the feedback here |
| 13:52.55 | swimmercol | thanks 1! |
| 13:56.38 | mvp | Well, it was a permission issue, sorry to bother. Have a pretty nice day. |
| 13:56.59 | Penguin | I hope to have a pretty day. |
| 13:59.18 | [TK]D-Fender | Thanks have a beautiful time! |
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| 14:06.25 | mvp | lvlinux: just read your message, it comes to happen that some of the file permissions under /dev/dahdi were of root ownership, as I´m running a non-root asterisk it could not access the device. A "chown" command solved the problem. Thanks for your time. Hope it can help anyone else with the same problem. |
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| 14:11.57 | lvlinux | mvp: ah, yes. Glad you got it figured out :-) |
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| 14:32.18 | Elwell | hi all, is there an astersk repo available that carriies binaries for debian? especially given it's scheduled for removal in ~10 days from debian |
| 14:32.46 | lvlinux | Elwell: you mean up-to-date binaries?? Or any binaries? |
| 14:33.20 | WIMPy | What's that story? |
| 14:33.44 | Elwell | ideally up to date. see https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=816042 |
| 14:36.20 | lvlinux | I'm not aware of any that are up to date. There may be some, but why not just compile it? It's not difficult. |
| 14:37.21 | Elwell | lvlinux: sure, I probably will, but it'd be nice if there was a 'blessed' release |
| 14:37.40 | file | the project does not currently do packaging |
| 14:38.17 | lvlinux | Problem with packaging is you can't make everyone happy. And somebody will complain about it... |
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| 14:40.15 | lvlinux | "Why isn't the Asterisk package for my S390 architecture machine running Zingbat Linux kept up to date?? Don't you Asterisk peeps do anything besides sit around?" lol |
| 14:41.33 | file | it's just a large time investment starting out as it causes ripples - need to research the best option, keep them up to date, and also integrate into release workflow |
| 14:41.51 | lvlinux | yup |
| 14:44.22 | file | so from a project perspective I won't say they won't ever happen, but it's not a project currently being worked on |
| 14:44.45 | Elwell | sure - I suspect the only way it'd work if it was part of some CI process |
| 14:45.09 | Elwell | and didn't actually suck up too much time |
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| 16:11.42 | igcewieling1 | It appears on Asterisk 11 users can set their password to be empty. Is this something expected or should I file a bug? |
| 16:12.03 | file | set their password... where |
| 16:12.10 | igcewieling1 | set their password to empty using Voicemailmain |
| 16:12.26 | file | dunno! |
| 16:12.58 | igcewieling1 | i.e. log into a mailbox, go into change the password and press # when prompted. Seems like a horrible from a security standpoint. |
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| 16:15.33 | igcewieling1 | file: thanks. I'll post a bounty for a fix. |
| 16:24.32 | igcewieling1 | file: looks like the minpassword= option will do what I need. |
| 16:28.17 | file | options, options, options! |
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| 17:09.45 | moe` | wassup? |
| 17:11.12 | moe` | file, drmessano, [TK]D-Fender... do one of you guys know if the LinkSys PAP2T can be used to turn an analog line into an extension? |
| 17:11.32 | [TK]D-Fender | Obviously not |
| 17:11.39 | moe` | got it working so I can call it, and it can call extensions or landlines via one of three SIP providers |
| 17:11.45 | moe` | ah man, seriously? |
| 17:11.51 | moe` | what's obvious about it that I missed? |
| 17:11.54 | moe` | doh. |
| 17:12.16 | [TK]D-Fender | A basic understanding of phones at all |
| 17:12.21 | [TK]D-Fender | ~fxsfxo |
| 17:12.21 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:12.52 | moe` | ok, are there any magic boxes that can turn an analog line into an extension? |
| 17:12.58 | moe` | such a thing must exist, no? |
| 17:13.24 | moe` | so incoming calls on the analog line can be routed to an arbitrary extension |
| 17:13.24 | [TK]D-Fender | Nothing is oging to make that dumb ATA its a land-line gateway |
| 17:13.29 | [TK]D-Fender | But a proper device |
| 17:13.33 | [TK]D-Fender | buy |
| 17:13.42 | moe` | what would be a proper device? |
| 17:13.52 | [TK]D-Fender | one with an FXO port |
| 17:14.53 | moe` | so crap, I have two of these PAP2T boxes and a PAP2T-NA box on the way, and they're mostly useless. |
| 17:15.24 | moe` | of course they're extensions, sure, but I wanted a box to turn the analog line, so incoming calls, get routed to an asterisk extension. |
| 17:15.26 | moe` | doh. |
| 17:15.58 | [TK]D-Fender | and don't use the word "extensions" |
| 17:16.14 | [TK]D-Fender | those are FXS ATA's |
| 17:16.22 | [TK]D-Fender | they PROVIDE voltage, they do not ACCEPT it. |
| 17:16.42 | [TK]D-Fender | Plug it in and sit back and wait for it to fry. |
| 17:17.31 | moe` | ok, so if I were to connect the analog landline to the PAP2T it would be a bad thing |
| 17:18.15 | [TK]D-Fender | -----> FRY <----- |
| 17:19.13 | moe` | What about an SPA400? |
| 17:19.27 | moe` | or is that meant to be a satelitte to the SPA9000? |
| 17:19.28 | [TK]D-Fender | Can you read the product description and manual? |
| 17:19.36 | [TK]D-Fender | Why kind of ports does it say it has? |
| 17:19.44 | moe` | SPA400 says 4x FXO |
| 17:20.02 | moe` | http://www.cisco.com/c/en/us/products/unified-communications/spa400-internet-telephony-gateway-4-fxo-ports/index.html |
| 17:20.10 | moe` | forgive my stupidity |
| 17:20.26 | [TK]D-Fender | That last question was just laziness |
| 17:20.59 | moe` | well no, just looking for a greybeard to verify. |
| 17:21.45 | [TK]D-Fender | "Verify"? You'd have to have done an appropriate amount of research on your own for it not to be a complete hand-off. |
| 17:22.03 | [TK]D-Fender | And since you only just started after all of this... Nope, doesn't earn it. |
| 17:23.04 | [TK]D-Fender | And last I head for the SPA-400, you couldn't specifically target a given port with it. All were treated the same in round-robin fashoin |
| 17:23.18 | [TK]D-Fender | For which you should always read the manual before buying. |
| 17:27.28 | moe` | can you recommend a Cisco/Linksys box that will turn a single landline to VoIP extension? |
| 17:27.42 | moe` | cuz apparently my searches and what I've ordered have come up failure. |
| 17:27.52 | [TK]D-Fender | Show me the searches |
| 17:33.02 | moe` | google'd ... Cisco LinkSys FXO gateway |
| 17:33.11 | moe` | SPA400 and SPA8000/9000 come up |
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| 17:37.00 | [TK]D-Fender | https://www.google.ca/#q=cisco%2Flinksys+box+to+use+landlines |
| 17:37.08 | [TK]D-Fender | Your exact words gives me MORE |
| 17:42.10 | moe` | as always, dude, I appreciate your tolerance and patience. |
| 17:42.25 | moe` | were it not for your help, my asterisk setup would not be what it is (such as it is) |
| 17:44.05 | *** part/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 17:45.26 | moe` | http://www.cisco.com/c/en/us/products/unified-communications/spa3102-voice-gateway-router/index.html |
| 17:45.42 | moe` | FXO and FXS ports, looks reasonable, no? |
| 17:46.32 | moe` | I only need one port for each of landline/FXO and handset/FXS in this usage case. |
| 17:49.13 | [TK]D-Fender | Then that'll do. |
| 17:49.31 | *** join/#asterisk miralin (~Thunderbi@194.8.128.120) |
| 17:50.09 | moe` | Thank you sir. |
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| 17:58.12 | jennie | Hello friends |
| 17:58.21 | jennie | can someone please help me with a noob questions/ |
| 17:59.37 | jennie | how do I do this thing? setup sshfs to mount 173.192.242.178:/mnt/azure/pbxrecordings/pbx to /mnt/callrecordings |
| 17:59.43 | jennie | I am in control panel of PBX |
| 18:00.17 | jennie | google is not helping or maybe I am searching wrong |
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| 20:20.28 | mvp | exit |
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| 22:59.40 | MajesticFudgie | yay someones attacking my Asterisk install |
| 22:59.46 | MajesticFudgie | by spamming incorrect passwords |
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| 23:18.20 | TandyUK | welcome to the internet |
| 23:18.28 | TandyUK | fail2ban ftw |
| 23:35.06 | *** part/#asterisk kharwell (kharwell@nat/digium/x-usmbihrkmqqvlvai) |
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| 23:55.36 | Samot | So you still haven't locked that box down? |
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