| 00:11.08 | *** part/#asterisk kharwell (kharwell@nat/digium/x-iursycjgxjkrbkjt) |
| 00:50.21 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 01:00.18 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 01:45.41 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 02:31.31 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 02:31.52 | *** join/#asterisk woose (~root@unaffiliated/woose) |
| 02:53.05 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 02:54.25 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 02:57.40 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 15:55.24 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:55.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.1 (2017/11/08), Standard: 15.1.1 (2017/11/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 15:55.58 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:55.59 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:00.40 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 16:04.50 | *** join/#asterisk protonchris (~chris@darkstar.protonlab.net) |
| 16:23.12 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:23.13 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:38.11 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 16:43.35 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:43.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:52.02 | *** join/#asterisk sekil (~sekil@cable-89-216-194-141.dynamic.sbb.rs) |
| 16:54.20 | *** join/#asterisk sekil (~sekil@cable-89-216-194-141.dynamic.sbb.rs) |
| 16:59.31 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 16:59.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:15.33 | *** join/#asterisk jkroon (~jkroon@165.16.204.173) |
| 17:37.05 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ukdlrfduzxccowtz) |
| 17:38.12 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
| 18:00.59 | *** join/#asterisk rangotail (~rangotail@unaffiliated/rangotail) |
| 18:03.05 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 18:28.33 | *** join/#asterisk joshelson (~joshelson@64.55.128.243) |
| 18:29.55 | *** join/#asterisk rangotail (~rangotail@unaffiliated/rangotail) |
| 18:42.50 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 18:44.31 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 18:48.38 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 18:53.13 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
| 19:14.07 | rrittgarn | when doing 'sip show peer' the field: LastMsgsSent has to do with number of read/unread voicemails right? |
| 19:14.26 | rrittgarn | so if i see 4/0 as the value, 4 new / 0 old, right? |
| 19:19.51 | *** join/#asterisk opmrcl (~Thunderbi@78.133.34.141) |
| 19:30.52 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
| 19:43.57 | thiagoc | hi all, i'm trying to use pjsip with the default pjsip.conf from asterisk source, when I try to register the 1107 endpoint I get "SIP/2.0 401 Unauthorized" on tcpdump |
| 19:44.03 | thiagoc | asterisk cli shows nothing |
| 19:44.07 | thiagoc | any hint? |
| 19:44.19 | thiagoc | I'm using linphone |
| 19:44.40 | thiagoc | my sip identity: sip:1107@xx.xx.xx.xx |
| 19:45.00 | thiagoc | sip proxy address: sip:xx.xx.xx.xx |
| 19:45.42 | thiagoc | i'm stuck |
| 19:47.23 | Samot | thiagoc: You need to show some things. Your configs to confirm the are right. The tcpdump would help and the output from the pjsip set logger on in the console. |
| 19:47.25 | Samot | ~pb |
| 19:47.26 | infobot | somebody said pastebin was 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. |
| 19:48.49 | lvlinux | thiagoc: Also, running the cli with some verbosity (asterisk -rvvv) will show you more things than the default. |
| 19:49.28 | thiagoc | lvlinux, verbose is at 3 |
| 19:49.49 | thiagoc | Samot, it's the default pjsip.conf from asterisk source |
| 19:49.56 | Samot | No. |
| 19:50.01 | Samot | That would be impossible. |
| 19:50.13 | Samot | It wouldn't have any of the settings for your setup. |
| 19:50.24 | lvlinux | It wouldn't be impossible, but it wouldn't do anything at all either. |
| 19:50.39 | Samot | You just can't copy and paste the sample configs and be working. |
| 19:51.03 | Samot | Where is the endpoint setup for 1107? |
| 19:51.05 | thiagoc | https://pastebin.com/raw/y0CLexM6 |
| 19:51.28 | thiagoc | it's a working example, isn't? |
| 19:52.02 | thiagoc | 1107 is there |
| 19:52.13 | lvlinux | this is not the sample config :) |
| 19:52.25 | Samot | At all |
| 19:52.33 | Samot | This is a config made by using the sample config. |
| 19:52.39 | Samot | Completely different things. |
| 19:52.54 | Samot | And the tcpdump? |
| 19:53.08 | thiagoc | it's on configs/samples dir, so... :) |
| 19:53.23 | Samot | Because 401 Unauthorized is a proper response. |
| 19:53.32 | Samot | It's how your AUTH is challenged. |
| 19:53.51 | Samot | So now lets see if linphone is sending back the right response/message. |
| 19:54.11 | Samot | And if you got an 401 error, it's making it to Asterisk. |
| 19:55.44 | thiagoc | https://pastebin.com/raw/iYPqVAnt |
| 19:56.19 | Samot | Are you running Chan_SIP? |
| 19:56.28 | thiagoc | no |
| 19:58.19 | Samot | file: Does PJSIP automatically bind to 5060? |
| 19:58.32 | Samot | Unless you specific a port in the bind= setting? |
| 19:58.51 | thiagoc | I'm not in the same location, the server is behind a NAT, just FYI |
| 19:59.08 | Samot | Server: Asterisk PBX certified/13.13-cert7 <-- Asterisk is most definitely responding. |
| 19:59.17 | Samot | How are you trying to watch this in the CLI? |
| 19:59.21 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 19:59.27 | Samot | And where is the linphones response to that 401 |
| 19:59.33 | Samot | There should have been more to that tcpdump |
| 19:59.46 | Samot | The 401 is *supposed* to happen. |
| 19:59.59 | thiagoc | I will send you the pjsip logger |
| 20:00.00 | Samot | 2:53:23 PM <Samot> Because 401 Unauthorized is a proper response. |
| 20:00.00 | Samot | 2:53:33 PM <Samot> It's how your AUTH is challenged. |
| 20:00.13 | Samot | Where is the linphone's response to that 401? |
| 20:00.17 | Samot | Is there one? |
| 20:00.25 | Samot | You should see a second REGISTER come through |
| 20:00.35 | thiagoc | hmm |
| 20:00.42 | Samot | That's how it works |
| 20:00.45 | thiagoc | I have another sip account here |
| 20:00.51 | thiagoc | could be this |
| 20:00.58 | Samot | REGISTER --> 401 Unauthorized -> REGISTER |
| 20:01.18 | Samot | Until you show the linphone responding to the 401 |
| 20:01.19 | thiagoc | my router maybe doesn't handle this |
| 20:01.30 | Samot | Was there more in the tcpdump? |
| 20:01.51 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:01.54 | *** join/#asterisk fblackburn (~fblackbur@modemcable094.94-70-69.static.videotron.ca) |
| 20:01.59 | Samot | Please show the whole tcpdump and the pjsip logger output |
| 20:02.40 | thiagoc | Samot, tcpdump on server or client? |
| 20:02.50 | Samot | On Asterisk. |
| 20:03.08 | Samot | Show me what happened on Asterisk. |
| 20:03.19 | Samot | The console output, whatever. |
| 20:03.34 | thiagoc | asterisk send the unauthorized and it's not reaching linphone |
| 20:03.40 | thiagoc | I think that's the problem |
| 20:03.44 | Samot | Yes. |
| 20:03.47 | Samot | That's a problem. |
| 20:04.07 | Samot | But you "cleaned up" the output |
| 20:04.26 | Samot | So I can't tell if those messages are formatted right or if the information in them is correct. |
| 20:07.09 | *** join/#asterisk w9sh (~dad@c-73-43-86-234.hsd1.ga.comcast.net) |
| 20:08.55 | *** part/#asterisk w9sh (~dad@c-73-43-86-234.hsd1.ga.comcast.net) |
| 20:09.46 | thiagoc | I just change the IP addresses |
| 20:09.59 | Samot | Right |
| 20:10.10 | Samot | But I can't tell if the linphone is sending PRIVATE IP details in the message. |
| 20:10.15 | Samot | Is this over a LAN? |
| 20:10.19 | Samot | Is it over WAN? |
| 20:10.32 | Samot | I can't tell anything about the type of traffic this is. |
| 20:10.43 | Samot | I can't tell if the IP details are correct for the type of message it is. |
| 20:10.51 | Samot | I can't tell if the routes have the right details. |
| 20:11.02 | Samot | I can't tell jack about it. |
| 20:21.36 | rrittgarn | when doing 'sip show peer' the field: LastMsgsSent has to do with number of read/unread voicemails right? |
| 20:21.41 | rrittgarn | so if i see 4/0 as the value, 4 new / 0 old, right? |
| 20:22.04 | rrittgarn | reason I ask is because i have a peer that is showing 4/0 right now, but there are no messages in the user's inbox |
| 20:22.12 | rrittgarn | so the MWI is still flashing because * is sending it the notifications |
| 20:22.19 | rrittgarn | is that cached somewhere like sorcery? |
| 20:23.40 | Samot | Peers have nothing to do with voicemail |
| 20:23.45 | Samot | Voicemail does not require a peer. |
| 20:23.53 | Samot | A peer does not require a voicemail accoun |
| 20:23.55 | Samot | A peer does not require a voicemail account |
| 20:25.28 | Samot | But in this case, you are correct. |
| 20:25.47 | Samot | The peer will show an associated voicemail and it's new/old count. |
| 20:26.21 | Samot | So 4/0 is as you stated, 4 new and 0 old |
| 20:26.34 | rrittgarn | while I understand that it is not required, obviously the system is still using that information (or at least displaying it to me) AND sending sip notify messages to the phone with an MWI count of 4/0 |
| 20:26.41 | rrittgarn | PB of the sip show peer: https://pastebin.com/PMvF38rS |
| 20:26.47 | Samot | Yes. |
| 20:27.11 | Samot | I said you are correct in this case. |
| 20:27.27 | Samot | The output will show the associated voicemail account's new/old count |
| 20:27.28 | rrittgarn | in that it is cached in sorcery? |
| 20:27.36 | rrittgarn | reason being the mailbox is actually empty |
| 20:27.44 | rrittgarn | but the phone is getting MWI messages from * saying its not |
| 20:27.58 | rrittgarn | user is getting annoyed with the blinking light |
| 20:28.09 | Samot | Well then Asterisk is generating notify's for this |
| 20:28.15 | Samot | Which has nothing to do with the actual peer. |
| 20:28.26 | Samot | Since anyone subscribed to the voicemail hint would also get this. |
| 20:29.09 | Samot | voicemail show users |
| 20:30.17 | Samot | What does the NewMsg count show for 4480? |
| 20:30.45 | Samot | And when you say it's empty you mean all the files? |
| 20:30.52 | Samot | INBOX is 100% empty? |
| 20:30.54 | rrittgarn | https://pastebin.com/Lpa7N846 |
| 20:30.57 | rrittgarn | and yes inbox is 100% empty |
| 20:30.59 | rrittgarn | no files |
| 20:31.01 | Samot | The .txt files don't exist? |
| 20:31.21 | rrittgarn | realtime - and that query comes back with a zero count |
| 20:31.34 | Samot | Oh you're using this in realtime? |
| 20:31.48 | rrittgarn | (and the OS level is clear as well - ls -lah /var/spool/asterisk/voicemail/Context/4480/INBOX yields nothing) |
| 20:31.52 | rrittgarn | and yes, its real time |
| 20:32.01 | Samot | But you're storing in the directory |
| 20:32.08 | rrittgarn | no, it stores it with a directory column |
| 20:32.27 | rrittgarn | and does all the queries against that column, so when i run the same count query that * does, i get zero |
| 20:32.32 | Samot | But the physical file is in that directory |
| 20:32.35 | rrittgarn | (trapped the query via mysql logging) |
| 20:32.42 | rrittgarn | no, using the db for storage |
| 20:32.47 | Samot | OK |
| 20:32.53 | Samot | This is all important stuff to know |
| 20:32.57 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:33.14 | Samot | Because 1) I don't do RealTime 2) I've never used DB storage for Voicemail. |
| 20:33.16 | rrittgarn | sorry - i tend to get better answers from this channel when I don't say realtime... most people throw up their hands |
| 20:33.21 | Samot | This is beyond my ability to help. |
| 20:33.28 | *** topic/#asterisk by bford -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.2 (2017/11/10), Standard: 15.1.2 (2017/11/10); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 20:33.28 | Samot | Because it matters. |
| 20:33.35 | Samot | And people that don't use it can't help |
| 20:33.42 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:33.53 | Samot | RealTime does things a bit differently. |
| 20:33.56 | rrittgarn | I agree to a certain extent, the core processes that * goes through are the same programatically, so any extra information on where it caches things is useful |
| 20:34.14 | rrittgarn | s/programatically/logically/ |
| 20:34.17 | Samot | Right but now I never would have said... |
| 20:34.22 | Samot | "Look in the DB" |
| 20:34.33 | Samot | Or though "Well stuff could be in the voicemail DB" |
| 20:34.59 | Samot | My entire line of troubleshooting is flawed if that is the case. |
| 20:35.15 | Samot | I've never used DB voicemail. |
| 20:35.17 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:35.22 | Samot | I have no clue how it functions for things. |
| 20:35.29 | Samot | Or what data is stored in it |
| 20:35.31 | rrittgarn | ok, pretend it was a file issue. What would you recommend checking next? |
| 20:35.51 | rrittgarn | *voicemail stored in flat file |
| 20:36.07 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:36.52 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:37.41 | Samot | Well I would check the output of voicemail show users |
| 20:37.42 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:37.56 | Samot | I would leave myself a voicemail and see what the NOTIFY triggers. |
| 20:38.14 | Samot | I'd delete voicemail via the voicemail app instead of directly on the server. |
| 20:38.32 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:38.48 | Samot | I'd make sure there wasn't a lockfile or some other . file in the directory |
| 20:38.57 | Samot | I'd reload sip to see if that cleared it out |
| 20:39.17 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:39.42 | rrittgarn | ok a couple things i didn't try in that list, so I appreciate the help. I did prune the peer and it still came back with 4/0 |
| 20:39.57 | rrittgarn | so in theory it's getting that info from app_voicemail, correct? |
| 20:40.05 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:40.14 | Samot | In a sense. |
| 20:40.29 | Samot | app_voicemail is doing stuff to get that data and act on it. |
| 20:41.00 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:41.04 | rrittgarn | correct, in this particular case its running a query every X seocnds, where X is=pollfreq in voicemail.conf |
| 20:41.45 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:42.35 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:43.20 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:44.15 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:45.00 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:45.46 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:45.54 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 20:46.36 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:47.30 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:48.10 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:49.00 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 20:54.14 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 20:58.32 | *** join/#asterisk overyander (~jeff@12.49.160.131) |
| 21:03.13 | *** join/#asterisk jeffspeff (~jeff@12.49.160.131) |
| 21:09.34 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 21:10.31 | qakhan | hi all, calls are dropping when agent answer the call. call does not drop on greeting message. here is i found in sip debug https://pastebin.com/Fg8tZPMk |
| 21:57.48 | Yiota | what's the best way to install asterisk on ubuntu? |
| 21:58.14 | Yiota | actually, i managed to install it via apt, but are there any guides for configuration? |
| 21:58.29 | Samot | The wiki |
| 21:58.41 | Samot | The sample configs |
| 21:59.08 | Samot | There's an entire option for loading the sample configs that explain all the options. |
| 21:59.16 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 22:20.39 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 23:01.51 | *** part/#asterisk kharwell (kharwell@nat/digium/x-yncvepudulqsoimg) |
| 23:05.27 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:30.13 | *** join/#asterisk opmrcl (~Thunderbi@78.133.34.141) |
| 23:48.09 | *** join/#asterisk RovingWrityer (~RovingWri@unaffiliated/rovingwriter) |
| 23:50.01 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 23:51.14 | *** join/#asterisk r0ckp3arl_1 (~rock@203-206-251-69.dyn.iinet.net.au) |