00:09.07 | *** join/#asterisk nix8n82 (~AndChat62@2600:100e:b04e:249b:3516:56fc:7263:40b2) |
00:11.26 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
01:01.01 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
01:28.27 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
02:00.41 | *** join/#asterisk AndChat|620489 (~AndChat62@191.sub-97-43-192.myvzw.com) |
02:11.02 | *** join/#asterisk nix8n82 (~AndChat62@2600:100e:b02e:5be6:1d81:8671:678d:7e7e) |
02:13.11 | *** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net) |
02:41.20 | *** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net) |
02:44.30 | *** join/#asterisk nighty- (~nighty@b157153.ppp.asahi-net.or.jp) |
04:15.05 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
05:26.36 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
05:35.32 | *** join/#asterisk ^Gecko^ (~asdjhasd@71-131-60-54.lightspeed.hstntx.sbcglobal.net) |
05:37.04 | ^Gecko^ | hey I'm having trouble setting up a Cisco 7941G with asterisk. I got the phone sort of working, but it keeps restarting itself during registration. the only errors in the log that show up are error updating locale, and no trust list installed. |
05:37.32 | ^Gecko^ | load file is SIP41.9-4-2sr3-1s |
05:40.20 | ^Gecko^ | here's the phone config: https://pastebin.com/nZcE4DCG |
06:21.51 | *** join/#asterisk Alblasco1702 (~Alblasco1@ip5456b46b.speed.planet.nl) |
06:35.53 | *** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b027:f662:a98f:899b:3c01:6acc) |
06:39.52 | *** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net) |
06:41.37 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
07:02.04 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
07:40.06 | *** join/#asterisk jkroon (~jkroon@41.114.74.43) |
07:51.24 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
08:01.19 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
08:53.49 | *** join/#asterisk duo_ubuntu (b4fbc359@gateway/web/freenode/ip.180.251.195.89) |
08:54.33 | *** join/#asterisk yteltom (~textual@2601:2c4:c600:2635:3d85:256:ad61:ec68) |
08:55.47 | *** join/#asterisk Downlots (~Downlots@hq.modulus.gr) |
09:23.52 | *** join/#asterisk rpifan (~rpifan@p578D247C.dip0.t-ipconnect.de) |
10:05.41 | *** join/#asterisk jkroon (~jkroon@41.114.74.43) |
10:53.53 | *** join/#asterisk pchero_work (~pchero@dhcp-077-249-058-090.chello.nl) |
10:58.08 | *** join/#asterisk pchero_work (~pchero@51.247.195.35.bc.googleusercontent.com) |
11:22.43 | *** join/#asterisk techquila (~techquila@122-62-61-4-adsl.sparkbb.co.nz) |
11:23.22 | *** join/#asterisk techquila (~techquila@122-62-61-4-adsl.sparkbb.co.nz) |
11:30.23 | *** join/#asterisk adnidor (~adnidor@p200300E59716E9000000000000000001.dip0.t-ipconnect.de) |
11:46.58 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
11:59.22 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@91.240.65.102) |
12:09.52 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-umdvykalkhcroord) |
12:09.52 | *** mode/#asterisk [+o bford] by ChanServ |
12:10.03 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
12:11.55 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
12:19.23 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
12:21.55 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.212.49) |
12:30.13 | *** join/#asterisk dobson (~dobson@68.ip-149-56-14.net) |
12:48.53 | *** join/#asterisk MrMojit0 (~MrMojit0@87.213.99.78) |
12:52.54 | *** part/#asterisk MrMojit0 (~MrMojit0@87.213.99.78) |
12:54.12 | *** join/#asterisk Downlots_ (~Downlots@94.70.72.106) |
13:02.30 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
13:04.39 | *** join/#asterisk adnidor (~adnidor@p200300E5971878000000000000000001.dip0.t-ipconnect.de) |
13:11.14 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
13:11.51 | Janos | got an asterisk server that sometimes just hangs, by this i mean, i can log into the asterisk console and do things like `core show calls`, but no phone can register and nothing gets displayed on the console |
13:12.01 | Janos | logs are also quite |
13:12.22 | Janos | when this happens i have to restart the asterisk service |
13:12.52 | Samot | And what is the status of the server? |
13:12.55 | Samot | How much load, etc? |
13:12.58 | Janos | is there anything i can do at this point figure out what's going on ? |
13:13.07 | Janos | load 0 |
13:13.12 | Janos | disk not full |
13:13.32 | Janos | everything else looks normal |
13:14.51 | Janos | i have enabled tcp support on pjsip and i can telnet the 5060 port on the asterisk |
13:15.08 | Janos | and ssh into the server |
13:15.10 | file | what version of Asterisk? enabling SIP logging shows nothing? |
13:15.33 | Janos | 13.14.1v |
13:16.32 | Janos | pjsip set logger on show a bunch of REGISTER request from all my phones |
13:22.07 | Janos | it's a lot but there seems to be no answer from the server |
13:22.38 | Samot | So the server isn't sending back 401 messages? |
13:23.40 | Janos | none so far |
13:24.05 | file | that version is old so I don't recall the things from that era |
13:24.06 | Janos | all i can see is the 'Received SIP request' with the REGISTER request from each phone |
13:24.12 | Janos | but no reply |
13:24.33 | file | there were ways the processing could get deadlocked in some scenarios |
13:25.42 | Janos | tcpdump shows a bunch of incoming SIP: REGISTER to the udp port of the asterisk but not a single response |
13:26.11 | Samot | Well I would recommend updating. |
13:26.31 | Samot | On to of the fact that pre-13.17.1 has an RTP bug and security holes. |
13:26.44 | Samot | There have been a few security patches since that version. |
13:27.42 | Samot | Janos: At this point you need to upgrade because you are sitting on a version that has known bugs that have been fixed in later releases. |
13:28.50 | Janos | this is a debian 9 I would hope that they backport security patches, so security wise it "should be ok" but yeah this looks like a nasty bug somewhere on the pjsip stack, so update does look like the only thing i can do here |
13:31.39 | Samot | That's not how that works. |
13:31.56 | Samot | They just don't "backport" 13.25 into 13.14.1 |
13:32.41 | Samot | Not when you can just update 13.14 to 13.25. However, it is up to the package maintainer to do that. |
13:33.15 | Samot | And Digium does not maintain or release OS packages/RPMs. |
13:33.19 | Samot | They release source. |
13:36.37 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
13:41.53 | Janos | I think that's pretty much what they do, they backport the security relevant bits from newer versions to older versions, for example https://downloads.asterisk.org/pub/security/AST-2018-004.html was fixed in 13.19.2 and yet I have that security bug corrected in my 13.14.1 version |
13:43.18 | Janos | but again they only do that for security related bugs, not any other kind of bug |
13:44.22 | Samot | It's two years old. |
13:44.29 | Samot | 13.14.1 is two years old. |
13:44.35 | Samot | There have been other bugs fixed. |
13:45.07 | Samot | So you're basically on an LTS version of Asterisk that due to the package you are using is Security Fixes Only. |
13:45.13 | Samot | Don't you see something off on that? |
13:47.07 | Janos | this looks like a bug on the pjsip stack (that is hopefully corrected in newer versions), when in the problematic state i can do `core show calls` and get an answer, but `pjsip show endpoints` shows no output |
13:47.34 | Janos | that's exactly what I'm saying, it's a security fixes only |
13:48.06 | Janos | so yeah, i guess it's time to upgrade |
13:48.53 | Janos | and hope |
13:56.15 | Janos | would the latest 13 release work or should I jump to 16 ? |
13:56.59 | Janos | or rather what would you recommend between those two ? |
13:58.12 | [TK]D-Fender | 16 |
13:58.36 | [TK]D-Fender | It's LTS and 13 is going SFO shortly |
13:59.25 | [TK]D-Fender | Fighting to stay on an old version is setting you up for another cleanup before long |
14:00.26 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
14:00.46 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
14:08.58 | Janos | kk thanks |
14:10.28 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for timeframes |
15:00.04 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
15:00.04 | *** mode/#asterisk [+o cresl1n] by ChanServ |
15:06.46 | *** join/#asterisk rpifan (~rpifan@p578D247C.dip0.t-ipconnect.de) |
15:15.44 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
15:25.47 | *** join/#asterisk kharwell (kharwell@nat/digium/x-gkwbsuopafbyewce) |
15:25.47 | *** mode/#asterisk [+o kharwell] by ChanServ |
15:50.36 | *** join/#asterisk WizJin (~WizJin@103.250.161.192) |
15:50.49 | igcewieling | Carrier instructions are so clear! "Overseas vs. Caribbean Indicator (if To# starts with 011, then enter a 6. If To# starts with 011, then enter a 9.)" |
15:53.25 | sibiria | instructions unclear, placed 9 hour call to somalian high toll hotline |
16:01.23 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
16:09.26 | *** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net) |
16:12.24 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
16:14.05 | igcewieling | sibiria: I'm not trying to get fired. |
16:17.04 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
16:26.17 | Samot | There are about 8-10 Caribbean countries that are not part of the NANP. |
16:26.44 | Samot | So some Caribbean calls can start with 01 |
16:53.08 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
16:55.36 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
17:05.24 | *** join/#asterisk duo_kali (24454aa8@gateway/web/freenode/ip.36.69.74.168) |
17:14.19 | *** join/#asterisk rpifan_ (~rpifan@p578D247C.dip0.t-ipconnect.de) |
17:23.28 | *** join/#asterisk pchero_work (~pchero@51.247.195.35.bc.googleusercontent.com) |
18:38.29 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
18:39.59 | Posterdati | hi |
18:41.51 | Posterdati | please help, I have problem with incoming calls, if they occurr within some minutes after starting asterisk they are ok, after some minutes the number does not respond and I have no sip log of it! |
18:41.54 | Posterdati | thanks |
18:42.08 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
18:43.27 | robink | is having some difficulty configuring Asterisk 16.2.1 after running autoreconf in the sourcedir (with AT_M4DIR set to "autoconf third-party third-party/pjproject"). |
18:43.51 | robink | I'm faced with this: ./configure: line 14100: syntax error near unexpected token `else' |
18:44.17 | [TK]D-Fender | Posterdati, Show an actual call in & out including SIP debug so we can see if your current state looks sane at all. |
18:44.33 | file | robink: if you want to regenerate configure then you should use bootstrap.sh... but why are you doing it? |
18:44.44 | Posterdati | there aren't incoming call logs |
18:44.52 | Posterdati | they were not registered at all |
18:44.53 | robink | ...and indeed, there is an 'else' statement without an opening 'if' (right before 'PBX_JANSSON=1'). |
18:45.33 | [TK]D-Fender | Posterdati, Show us the calls as requested |
18:45.42 | Posterdati | [TK]D-Fender: ok |
18:46.43 | robink | file: It's the default behavior for the Gentoo ebuild I'm using for installation. |
18:46.43 | Posterdati | [TK]D-Fender: https://paste.debian.net/1071690/ |
18:47.06 | robink | file: Especially since one of the patches in the set modifies configure.ac (and yes, I've checked, not applying the patchset makes no difference whatsoever). |
18:47.08 | Posterdati | [TK]D-Fender: if I reload it works again |
18:47.13 | file | I see |
18:47.28 | robink | file: I'll compare the behavior of the ebuild to bootstrap.sh, thanks for pointing that out. |
18:47.52 | [TK]D-Fender | Reliably Transmitting (NAT) to 156.54.82.96:5060: |
18:47.59 | [TK]D-Fender | Contact: <sip:asterisk@10.0.0.2:5060> |
18:48.11 | [TK]D-Fender | You are seding OPTIONS packets to your provider |
18:48.17 | Posterdati | [TK]D-Fender: ah |
18:48.21 | [TK]D-Fender | These show tath you have not configured that peer correctly |
18:48.27 | [TK]D-Fender | your provider is NOT behind NAT |
18:48.38 | Posterdati | [TK]D-Fender: so? What shall I do? |
18:48.39 | [TK]D-Fender | and have that settings wrong |
18:48.55 | [TK]D-Fender | it is also showing you are sending them your PRIVATE IP as the contact for them sending you calls which is wrong |
18:49.11 | [TK]D-Fender | and you have not done the basics to set up your system to work behind the NAT it appears to be behind |
18:49.27 | [TK]D-Fender | this should have been addressed days ago when you came in with this |
18:49.37 | *** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:91e8:e8bb:a932:ba2d:d528) |
18:49.43 | [TK]D-Fender | EXTERNADDR, LOCALNET, NAT, DIRECTMEDIA |
18:49.58 | [TK]D-Fender | set these up properly between your [general] and peer sections. |
18:50.04 | [TK]D-Fender | where each belongs |
18:50.21 | Posterdati | for externaddr I have a non static ip |
18:50.30 | [TK]D-Fender | And then you need to ensure you have forwarded your SIP port from your router to your server along with your RTP port range as specified in rtp.conf |
18:50.47 | Posterdati | [TK]D-Fender: forward is done |
18:50.48 | [TK]D-Fender | <Posterdati> for externaddr I have a non static ip <- you need this to get fixed |
18:51.03 | [TK]D-Fender | You can use a DynDNS service to do a LOOKUP if your IP is dynamic |
18:51.09 | [TK]D-Fender | But this has to be correct |
18:51.33 | Posterdati | [TK]D-Fender: my vDSL is a dynamic ip one |
18:52.23 | Posterdati | [TK]D-Fender: what about the outbound proxy the provider gave to me? |
18:52.44 | [TK]D-Fender | kWhat about it? |
18:52.57 | [TK]D-Fender | You need to fix your settings |
18:53.08 | Posterdati | [TK]D-Fender: yes, how? |
18:53.10 | [TK]D-Fender | Do not waste time on anything else until you have done the basics properly |
18:53.26 | [TK]D-Fender | [TK]D-Fender> EXTERNADDR, LOCALNET, NAT, DIRECTMEDIA <_ FILL THESE IN PROPERLY |
18:53.35 | Posterdati | ok |
18:53.52 | Posterdati | is externaddr the ip that provider gave to me? |
18:54.05 | Posterdati | localnet=10.0.0.0/255.255.255.0 |
18:54.14 | Posterdati | nat=force_rport,comedia |
18:54.22 | Posterdati | directmedia=no |
18:55.24 | [TK]D-Fender | Posterdati> is externaddr the ip that provider gave to me? <-, no it's YOUR IP |
18:55.36 | [TK]D-Fender | Well.. "yes' if you're referring to your INTERNET provide.... |
18:55.42 | Posterdati | yes |
18:56.17 | Posterdati | ok |
18:58.22 | robink | file: Fixed it, added third-party/jansson to AT_M4DIR. |
19:04.29 | Posterdati | [TK]D-Fender: now the incoming call is passing like before, we have to wait |
19:05.19 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
19:05.31 | file | robink: cool |
19:05.51 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
19:09.53 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
19:14.49 | Posterdati | [TK]D-Fender: now I have this: https://paste.debian.net/1071694/ |
19:15.03 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
19:22.12 | [TK]D-Fender | Reliably Transmitting (NAT) to 156.54.82.96:5060: |
19:22.12 | [TK]D-Fender | OPTIONS sip:telecomitalia.it;user=phone SIP/2.0 |
19:22.17 | [TK]D-Fender | Contact: <sip:asterisk@10.0.0.2:5060> |
19:22.19 | [TK]D-Fender | Still bad |
19:22.23 | Posterdati | yes |
19:22.27 | [TK]D-Fender | you've put things wrong, or in wrong places |
19:22.39 | Posterdati | I put all inside [general] |
19:23.34 | Posterdati | I removed context=incoming from [general] too |
19:26.00 | Posterdati | sip.conf -> https://paste.debian.net/1071719/ |
19:26.08 | *** join/#asterisk MrMojit0 (~MrMojit0@2001:1c00:1c00:ed00:7107:86f1:1c18:8f1a) |
19:28.08 | Posterdati | it is all screwed now |
19:32.33 | Posterdati | :( |
19:34.01 | Posterdati | now, if I define an externaddr I cannot receive calls... |
19:49.32 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
20:42.42 | Posterdati | now I have a sip.conf with only EXTERNADDR, LOCALNET, NAT and DIRECTMEDIA configured |
20:42.48 | Posterdati | nothing happens |
20:43.51 | Posterdati | it is an hell |
20:45.07 | Posterdati | I lived with analog lines with no issue, now this system is totally against logic! |
21:05.21 | *** join/#asterisk pchero (~pchero@dhcp-077-249-058-090.chello.nl) |
21:09.19 | wdoekes | if I may just chime in: setting externaddr may have adverse effects too. many SIP providers will detect you being behind (hopefully symmetric) NAT and update your Contact and SDP addresses accordingly. that NAT detection can fail if you already send the "correct" external IP |
21:16.07 | [TK]D-Fender | It's now been 2 hours since you said "doesn't work" and you shown nothing |
21:27.12 | *** join/#asterisk yteltom (~textual@2601:2c4:c600:2635:3d85:256:ad61:ec68) |
21:43.55 | *** join/#asterisk sicelo (~sicelo@Maemo/community/ex-council/sicelo) |
22:32.41 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
22:34.10 | *** join/#asterisk gusto (~gusto@2a01:c844:101c:602:eb58:1b0b:1b4f:d652) |
23:06.46 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
23:18.59 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
23:30.05 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |