00:22.29 | *** join/#asterisk infobot (ibot@rikers.org) |
00:22.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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 |
00:25.38 | *** join/#asterisk matrix1233 (~matrix123@41.230.41.233) |
01:02.03 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
01:06.56 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-oadouqwjtvmnlxdo) |
01:26.12 | *** join/#asterisk matrix1233 (~matrix123@41.230.41.233) |
01:36.43 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
01:39.49 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
01:59.00 | *** join/#asterisk scgm11_ (~scgm11@r186-52-153-48.dialup.adsl.anteldata.net.uy) |
02:20.18 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
02:20.38 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
02:56.29 | *** join/#asterisk gswain (uid91227@gateway/web/irccloud.com/x-ftwcdpgwniezusov) |
02:56.58 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
03:08.03 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
03:25.16 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
03:26.36 | *** join/#asterisk matrix1233 (~matrix123@41.230.41.233) |
03:32.48 | bimbo | hello, what is the purpose of listing preferred codecs in order with allow in chan_pjsip (allow=!all,g729,alaw) |
03:34.04 | bimbo | sorry, nevermind, wrong question |
03:34.08 | Samot | Because that's how codecs work. |
03:34.15 | Samot | Regardless of the SIP stack. |
03:35.52 | bimbo | Samot: yeah, I was not thinking about transcoding here, so that was a bad question |
03:56.26 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
04:33.56 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
05:34.35 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
06:06.08 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
06:06.50 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
06:10.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
06:16.31 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
06:17.05 | *** part/#asterisk bimbo (~emerino@201.105.156.74) |
06:22.57 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
06:23.55 | *** join/#asterisk MrMojit0 (~MrMojit0@hoofddorp.cn.nl) |
06:28.32 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
07:05.19 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
07:13.45 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
07:20.37 | *** join/#asterisk jkroon (~jkroon@vc-nat-gp-s-41-13-2-80.umts.vodacom.co.za) |
07:22.39 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
07:49.13 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
07:57.40 | *** join/#asterisk jkroon (~jkroon@vc-nat-gp-s-41-13-14-171.umts.vodacom.co.za) |
08:00.41 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
08:16.00 | *** join/#asterisk tlpresearch (0539061b@gateway/web/freenode/ip.5.57.6.27) |
08:16.42 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
08:18.13 | tlpresearch | Hello, I need some help making calls via Asterisk (on public IP) between soft SIP phones on 3G networks. Call OK but no audio. Thanks. |
08:31.33 | *** join/#asterisk bl3nto (~bl3nto@78.134.210.254) |
08:46.47 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
09:07.27 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
09:11.01 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
09:13.23 | *** join/#asterisk DanB (~DanB@clt-195.192.203.229.ip-anschluss.net) |
09:14.04 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
09:29.37 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
09:57.45 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
10:00.10 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
10:06.12 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
10:44.28 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
11:00.23 | *** join/#asterisk matrix1233 (~matrix123@41.226.241.158) |
11:07.00 | *** join/#asterisk boris_t (~boris_t@363103629.convex.ru) |
11:21.06 | *** join/#asterisk mub (~jub@static-173-53-12-18.rcmdva.fios.verizon.net) |
11:32.01 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-omcmilxvndjozcae) |
11:41.06 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
11:59.52 | *** join/#asterisk scgm11_ (~scgm11@r186-49-62-41.dialup.adsl.anteldata.net.uy) |
12:22.57 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
12:29.24 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
12:36.16 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
12:51.28 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
12:54.43 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
12:54.54 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
12:54.58 | *** join/#asterisk scgm11_ (~scgm11@r186-49-62-41.dialup.adsl.anteldata.net.uy) |
13:03.51 | qakhan | all, i have weird problem with sip trunk registeration. it stop working suddenly. |
13:04.16 | Samot | How did it stop working? |
13:05.05 | qakhan | when i restart the asterisk services, sip trunk registers but after 30 - 60 second it show registration sent |
13:05.23 | Samot | Show it |
13:05.25 | Samot | ~pb |
13:05.25 | infobot | it has been said that 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. |
13:05.56 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
13:08.52 | qakhan | here https://pastebin.com/xEGJ2Ga2 |
13:09.05 | Samot | No. |
13:09.21 | Samot | I mean the actual SIP debug showing the REGISTER attempts. |
13:15.42 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
13:17.02 | *** join/#asterisk pdugas (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
13:27.57 | Samot | Alright then. |
13:32.39 | *** join/#asterisk matrix1233 (~matrix123@41.226.240.93) |
13:38.37 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
13:38.37 | *** mode/#asterisk [+o cresl1n] by ChanServ |
13:39.21 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
13:46.43 | *** join/#asterisk matrix1233 (~matrix123@41.230.47.10) |
13:51.57 | *** join/#asterisk matrix1233 (~matrix123@41.230.47.10) |
14:02.24 | *** join/#asterisk kharwell (kharwell@nat/digium/x-zlwdsjffrcwdeiib) |
14:02.24 | *** mode/#asterisk [+o kharwell] by ChanServ |
14:08.34 | *** join/#asterisk scgm11_ (~scgm11@r186-50-211-14.dialup.adsl.anteldata.net.uy) |
14:14.52 | davlefou | hi, how can i use this numbre : +33.... |
14:15.20 | qakhan | Samot here is sip debug https://pastebin.com/xKR6HfqF |
14:15.59 | [TK]D-Fender | davlefou, Use it. |
14:16.38 | [TK]D-Fender | qakhan, Contact: <sip:s@192.168.10.25:5060> |
14:16.45 | [TK]D-Fender | You're sending them your PRIVATE IP for the contact |
14:16.50 | [TK]D-Fender | screwed up your NAT config |
14:17.41 | qakhan | if i disable all DIDs in the [allinbound] context then there is no problem with sip trunk registration. it register after every 45 sec |
14:18.00 | [TK]D-Fender | no |
14:18.12 | [TK]D-Fender | No peer or dialplan has ANY impact on your registration |
14:18.13 | qakhan | but as i allow single DID to receive calls, registration problem |
14:18.23 | [TK]D-Fender | YOUR NAT SETTINGS ARE SCREWED UP |
14:18.37 | [TK]D-Fender | <[TK]D-Fender> qakhan, Contact: <sip:s@192.168.10.25:5060> <------------------------------------ |
14:18.40 | qakhan | [TK]D-Fender NAT setting on asterisk server? |
14:18.46 | [TK]D-Fender | <[TK]D-Fender> qakhan, Contact: <sip:s@192.168.10.25:5060> <------------------------------------ |
14:19.05 | qakhan | but i never setup nat setting on server. |
14:20.31 | davlefou | [TK]D-Fender, is it possible to use + in dialplan? |
14:21.03 | qakhan | [TK]D-Fender my * server is not over the internet, its local usage only. all LAN users |
14:22.21 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
14:22.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:22.54 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ybmdjkbwvvsyakws) |
14:22.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
14:23.33 | qakhan | here is successfull registration when i disable DIDs in context https://pastebin.com/5MaqcNP2 |
14:24.03 | Samot | Contact: <sip:s@192.168.10.25:5060> <-- Still using a PRIVATE IP |
14:24.59 | qakhan | but how its working then |
14:25.10 | qakhan | i am not using NAT on server |
14:25.53 | Samot | [Jun 8 10:13:33] NOTICE[9644]: chan_sip.c:15347 sip_reg_timeout: -- Registration for 'itcu_rvest@inbound33.vitelity.net' timed out, trying again (Attempt #29) |
14:25.56 | Samot | ^^ First off |
14:26.11 | Samot | I asked you over an hour ago to show a restart and the FIRST attempt after the restart |
14:26.36 | Samot | But that clearly shows there were 28 other attempts to register or wait for a reply from Vitelity |
14:26.58 | Samot | The fact you are sending them a private IP means they are most likely sending replies back to an unroutable IP over the Internet |
14:27.29 | qakhan | no. i was tesing that thing which i told you |
14:27.42 | qakhan | disable DID in context |
14:28.00 | qakhan | if you say NAT where is it ? on there server? |
14:28.02 | Samot | 9:09:28 AM <Samot> I mean the actual SIP debug showing the REGISTER attempts. |
14:28.02 | Samot | 10:15:27 AM Q<qakhan> Samot here is sip debug https://pastebin.com/xKR6HfqF |
14:28.16 | Samot | ^^ At what point did you say you were doing anything? |
14:28.42 | Samot | Your PBX is not behind NAT? |
14:29.05 | qakhan | yes |
14:29.15 | qakhan | behind a router |
14:29.15 | Samot | Yes, it is behind NAT? |
14:29.18 | Samot | OK |
14:29.28 | Samot | So you are sending Vitelity your LAN IP |
14:29.35 | Samot | Instead of your WAN IP |
14:29.52 | qakhan | but there was no change on router side |
14:30.07 | Samot | That is what is happening. |
14:30.16 | Samot | Does Asterisk have the proper network settings? |
14:31.15 | qakhan | i am having an other server with different sip trunk account with vitelity. |
14:31.29 | Samot | I don't care about the other server. |
14:31.30 | qakhan | that sever is on the same network |
14:31.32 | [TK]D-Fender | <qakhan> [TK]D-Fender my * server is not over the internet, its local usage only. all LAN users <- YOU ARE REGISTERING TO A PROVIDER |
14:31.38 | Samot | Does Asterisk have the proper network settings? |
14:31.41 | Samot | THIS box. |
14:31.41 | [TK]D-Fender | NO |
14:31.43 | Samot | Not another box. |
14:31.44 | [TK]D-Fender | it does't |
14:32.11 | Samot | Sigh. |
14:32.12 | Samot | OK |
14:32.25 | Samot | I have a tech onsite about to do a 96 room cut over at a hotel. |
14:32.34 | Samot | I don't have time for this. |
14:33.05 | [TK]D-Fender | <qakhan> if you say NAT where is it ? on there server? <- your SERVER is behind a NAT ROUTER and is telling the PROVIDER the WRONG ADDRESS |
14:33.14 | [TK]D-Fender | FIX YOUR ASTERISK SIP SETTINGS FOR NAT |
14:34.18 | *** join/#asterisk cmendes0101 (~cmendes01@cpe-76-170-253-155.socal.res.rr.com) |
14:35.22 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
14:40.24 | davlefou | is it possible to have an action in this case: rejected because extension not found in context? |
14:41.03 | [TK]D-Fender | davlefou, What kind of action? |
14:42.23 | [TK]D-Fender | <davlefou> [TK]D-Fender, is it possible to use + in dialplan? <- you can MATCH it. It's a charater like any other |
14:42.43 | [TK]D-Fender | character* |
14:43.00 | davlefou | some message ou call en agi to analyse the number. |
14:44.46 | [TK]D-Fender | There is nothing to "analyze" |
14:44.52 | [TK]D-Fender | you aren't accepting the call |
14:45.01 | [TK]D-Fender | What is hard to understand about that messag? |
14:45.12 | [TK]D-Fender | Call came in looking for a match for that number |
14:45.16 | [TK]D-Fender | You don't HAVE a match |
14:46.05 | davlefou | if have _+.,1,AGI..., it should work? |
14:46.18 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-zdgvbduhivdsgorv) |
14:46.18 | *** mode/#asterisk [+o rmudgett] by ChanServ |
14:46.28 | [TK]D-Fender | Did you TRY? |
14:46.47 | davlefou | ('direction','+.','1','NoOp','${EXTEN}') |
14:51.41 | davlefou | [TK]D-Fender, i have no result, did i made an mistake? |
14:52.09 | [TK]D-Fender | Are you showing anything useful? |
14:52.20 | [TK]D-Fender | Where do I see the actual attempt? |
14:52.28 | [TK]D-Fender | Proof that dialplan is there and ready? |
14:53.28 | *** join/#asterisk sh_smith (foobar@cpe-76-174-26-91.socal.res.rr.com) |
14:55.30 | davlefou | it seems works: NoOp("SIP/david-00000001", "Cas +: +33") |
14:55.44 | davlefou | [TK]D-Fender, Thanks! |
15:08.03 | *** join/#asterisk stefanauss (~stefanaus@95.142.177.153) |
15:12.32 | *** join/#asterisk Bitcho (~Bitcho@unaffiliated/bitcho) |
15:14.00 | davlefou | [TK]D-Fender, it work! Thanks for your help! |
15:26.38 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
15:39.27 | davlefou | [TK]D-Fender, is it possible to use mail? |
15:40.17 | [TK]D-Fender | Yes? |
15:40.31 | [TK]D-Fender | I've mailed all sorts of letters and boxes in my life |
15:40.43 | [TK]D-Fender | so ... I think that's a "yes" |
15:41.47 | davlefou | i want to say, use adress mail for launch an call in dialplan asterisk. |
15:42.11 | [TK]D-Fender | Do you know how to " launch an call in dialplan asterisk."? |
15:42.53 | davlefou | not to tape 04xxxx but sabrine@truc.fr to call them! |
15:43.22 | [TK]D-Fender | "tape"? |
15:43.56 | davlefou | in my soft phone et asterisk ll find the good number via an agi. |
15:43.58 | [TK]D-Fender | Nothing you are saying makes any sense |
15:44.26 | [TK]D-Fender | you are presenting broken pieces of sentences and not communicating a clear & complete idea or question |
15:47.21 | davlefou | Normaly we use phone number to call some one. But i have en list of number and mail au people, it should be possible to launch un call with taping courriel and asterisk will be able to change in real number. |
15:49.08 | Kuunsi | so, you want asterisk to do some email-to-number-mapping? |
15:49.19 | Kuunsi | should be possible, go write some agi scripts :) |
15:49.21 | davlefou | i have receive the mail and copie un soft phone to call. Is it more clear my ideas? |
15:49.32 | davlefou | Kuunsi, yes! |
15:49.44 | Kuunsi | wait, why do i havae a second u |
15:50.04 | Kunsi | .. fixed |
15:50.42 | davlefou | Kunsi, Agi script is not difficulte but is possible tout have an dialpan take that? |
15:50.56 | [TK]D-Fender | davlefou, did you TRY/ |
15:51.16 | [TK]D-Fender | daYou seem to have a serious issue just DIALING something and LOOKING |
15:51.17 | davlefou | Yes, i made _.@. |
15:51.38 | Kunsi | in dialplan, you match for s or i (don't remember), then call agi script with called "number", then use returned number to actually dial |
15:51.47 | [TK]D-Fender | <davlefou> Yes, i made _.@. <- nothing after the first "." matters |
15:55.33 | davlefou | @ don't work but the word seems ok. |
15:58.15 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
15:58.30 | *** join/#asterisk sekil (~sekil@cable-89-216-220-219.dynamic.sbb.rs) |
16:00.35 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:01.20 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:02.05 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:02.55 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:04.30 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:05.18 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:12.09 | *** join/#asterisk gravspeed (~gravspeed@207.114.166.146) |
16:12.53 | gravspeed | so it seems like every other morning i get someone grinding on my hosted server... but the logs do not contain the ip address |
16:13.08 | gravspeed | i get lines like these: NOTICE[1265] chan_sip.c: Call from '' to extension '00046812410886' rejected because extension not found in context 'from-sip-external'. |
16:13.45 | gravspeed | how can i make asterisk log the ip so fail2ban can block them |
16:14.18 | gravspeed | currently i am retrieving the ip from voipmonitor pcaps |
16:14.27 | gravspeed | and blocking manually with iptables |
16:14.30 | davlefou | gravspeed, you need to put fail2ban, what linux and asterisk did you use? |
16:14.38 | gravspeed | i'm using fail2ban |
16:15.06 | gravspeed | it's an old hosted box that needs to die... v1.6.2.10 |
16:15.36 | gravspeed | logs for those type of hits dont' contain the ip address, therefore fila2ban can't do anything about it. |
16:15.50 | gravspeed | *fail |
16:16.06 | [TK]D-Fender | 1.6.2 does not include enough to catch them |
16:16.20 | [TK]D-Fender | You have to upgrade to get that info |
16:16.26 | gravspeed | balls |
16:16.43 | gravspeed | i was looking at logger.conf and i didn't see anything... i guess that explains why |
16:16.51 | [TK]D-Fender | or.... |
16:16.58 | [TK]D-Fender | you are LETTING that hit the dialplan |
16:17.06 | [TK]D-Fender | make amatch and deal with it there |
16:17.26 | gravspeed | hmm... you might be on to something there... |
16:17.56 | [TK]D-Fender | or take the next step and stop allowing un-authed calls in the first place |
16:18.11 | gravspeed | really i need to make some time and get the last 3 clients off this pos |
16:18.36 | gravspeed | i didn't design this one.. i inherited it and it is a strange bird... |
16:19.14 | gravspeed | sometimes the numbers that i see are actually valid numbers, but if they aren't authenticated they get the not found in context error |
16:20.26 | gravspeed | i think that's how this one controls who can make long distance calls... |
16:22.39 | gravspeed | it's kinda fun to watch, it's all controlled by sql tables, the queries fly by in the cli |
16:25.16 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
16:35.06 | *** join/#asterisk cmendes0101 (~cmendes01@cpe-76-170-253-155.socal.res.rr.com) |
16:35.32 | *** join/#asterisk jkroon (~jkroon@165.16.204.168) |
16:38.42 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
16:39.07 | jrun | can pjsip's logger be instructed to log by endpoint/contact/* |
16:39.09 | jrun | ? |
16:42.13 | [TK]D-Fender | pjsip<tab> <-------------- |
16:42.28 | [TK]D-Fender | follow the command list... |
16:47.51 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
16:50.47 | jrun | [TK]D-Fender: i have, 'pjsip set logger <tab>' on gives on/off |
16:51.17 | jrun | ...and host |
16:52.07 | jrun | ok that's it; i think :) thanks |
16:54.55 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
16:56.53 | *** join/#asterisk theGoat (~textual@pool-71-162-187-37.phlapa.fios.verizon.net) |
16:58.30 | jrun | what makes pjsip to send some contacts as <sip:ip_addr:port> in some INVITES and <sip:user@ip_addr:port> in others? |
16:59.35 | jrun | chan_sip seems to never send the <sip:ip_addr:port> format |
17:00.50 | file | if it's a registered device, then we use whatever they have told us |
17:01.05 | jrun | we have a phone (with pjsip stack in fact) that seems to ingore those INVITES that have Contact in <sip:ip_addr:port> format |
17:01.09 | file | otherwise we will use the URI provided for a contact in the configuration, or in the Dial string |
17:05.16 | jrun | file: nat is in play so in this cas a registered device is sending Contact: <sip:user@private_ip:port> in its INVITE |
17:06.18 | file | If rewrite contact is enabled we update the host and port portion |
17:06.33 | jrun | rewrite_contact=yes |
17:07.08 | jrun | that's the initial INVITE in fact. device gets an Unauth, hang on... |
17:07.32 | file | Initial invite? |
17:08.11 | jrun | the very first INVITE from device to the server |
17:08.24 | jrun | well, i see that again it send private_ip in the second invite. |
17:08.33 | jrun | with Authorization: |
17:09.30 | file | I don't understand what you are referring to then, I thought you meant an outgoing INVITE |
17:10.15 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
17:14.35 | jrun | file: may i pm you? |
17:15.14 | file | For what reason? |
17:15.22 | jrun | logs |
17:16.22 | file | I don't accept PMs for that. It doesn't allow anyone else to help and the results aren't searchable. |
17:16.43 | jrun | https://gist.github.com/12bc966fd7a129960acb5417f604cc17 |
17:18.38 | jrun | this works, but when same device (293 extension in the logs), with another account though, talks to our other server with chan_sip, outbound calls behave strangly. |
17:19.18 | jrun | in the sense that the called extension rings, call it picked up but on the calling device nothing happens (no ongoing call counter). |
17:19.47 | jrun | the only diff we have seen is the Conctact: header so in a sense i'm not sure if that actually is the source of problem. |
17:19.58 | file | I'm still confused. What am I looking at, and why are you now talking about chan_sip? |
17:20.56 | jrun | we have phone from yealink, Yealink SIP-T58, with two acocunts on it. |
17:21.39 | jrun | on account is on asterisk 11 with chan_sip and other account (the one you're looking at in the logs) is on asterisk 14 with chan_pjsip. |
17:22.25 | jrun | it works with pjsip but not chan_sip in the way i explained above. inbound calls to the phone with chan_sip also work. |
17:23.33 | jrun | sorry explaining it a bit off i guess. does what i said make sense? |
17:25.27 | file | the device acknowledged that we told it it was answered |
17:25.48 | file | the only thing of note is that we sent a re-invite to one with an internal IP address inside |
17:25.53 | file | do you have direct media disabled? |
17:26.00 | jrun | yes |
17:26.22 | file | I don't see anything else of note |
17:26.31 | file | you could really simplify the scenario to isolate it though |
17:26.46 | file | ie: originate a call to the remote side send it to a playback |
17:26.57 | file | and for inbound just send it to a playback |
17:27.04 | file | if both of those work then you've isolated it further |
17:30.43 | *** join/#asterisk gruetzkopf (gruetzkopf@captured-elf.dont-follow-me.eu) |
17:33.12 | *** join/#asterisk scgm11__ (~scgm11@r186-50-189-141.dialup.adsl.anteldata.net.uy) |
17:33.45 | jrun | file: this is the chan_sip logs: |
17:33.50 | jrun | https://gist.github.com/1064af90162617eea977312b1e349360 |
17:34.37 | jrun | note the Contact sent from asterisk in OK |
17:35.37 | file | that is the Contact for Asterisk |
17:35.41 | file | how it wants to be reached for subsequent requests |
17:35.59 | file | with or without username is valid there, doesn't matter |
17:36.31 | jrun | but same step in pjsip Contact: is in <sip:ip_addr_of_server:port> format. no 'user@' portion. |
17:36.39 | file | sure |
17:37.09 | file | it's up to the implementation what it puts there, in the case of PJSIP it doesn't put a user there - and it doesn't need there to be one there |
17:37.50 | file | you need to further isolate things like I said |
17:38.18 | jrun | i was wondering if i could do the same with pjsip (chan_sip is production, rather not touch it). that would confirm the bug in device's firmware. |
17:40.04 | jrun | so device calls in but destination is a playback? i'm not clear what you mean by remote side? all extensions are on the same pbx. |
17:40.22 | file | yes, destination is a playback |
17:40.30 | file | don't call another device |
17:40.37 | file | that discards bridging and the separate call and simplifies the scenario |
17:40.44 | jrun | i see |
17:44.36 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
17:45.17 | jrun | well, Echo doesn't even pick up |
17:45.35 | file | don't do Echo |
17:45.51 | file | you can't do Echo because in order to send through NAT Asterisk has to know the source IP address and port of your media |
17:46.08 | file | it also doesn't Answer if I remember right |
17:46.10 | file | do Playback |
17:46.26 | file | and also provide your pjsip.conf configuration |
18:00.41 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
18:03.48 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
18:27.24 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
18:31.08 | *** join/#asterisk averythomas (~averythom@cpe-72-224-252-57.maine.res.rr.com) |
18:41.06 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
18:46.49 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
18:59.01 | jrun | Saydigits ? |
19:09.02 | *** join/#asterisk ttbakiatwoam (~ttbakiatw@205.122.215.59) |
19:09.39 | ttbakiatwoam | Does anyone have experience with AMD? I am having issues getting it to detect humans it always thinks the call is a machine |
19:13.12 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:ba27:ebff:fee4:a52f) |
19:21.16 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
19:23.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
19:29.28 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
19:41.44 | jrun | file: playback also fails with chan_sip. is this media related? |
19:41.57 | file | entirely possible |
19:42.00 | *** part/#asterisk FrozenFire (~FrozenFir@pdpc/supporter/active/frozenfire) |
19:42.16 | jrun | although, on the device we don't get pass the Calling stage. |
19:44.58 | jrun | https://gist.github.com/f0fc1e2f6b31142171a41f6caa60f7f7 |
19:45.27 | jrun | device sends an ACK after receiving the OK from server though |
19:47.30 | file | the SIP communication is fine |
19:47.53 | file | you'd need to examine the RTP side, confirming the values in the SDP are correct, looking at rtp set debug on |
19:47.53 | jrun | reason for BYE from server is just finishing the playback i guess however there is no media stream |
19:53.24 | *** join/#asterisk miralin (~Thunderbi@91.237.94.8) |
19:54.52 | jrun | rtp debug logs go where? just the connected console? |
19:55.29 | jrun | i'm getting everything but the rtp logs after doing 'rtp set debug IP' |
19:56.32 | file | they go to the console |
19:57.52 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
20:01.45 | jrun | dead silenc ! |
20:02.24 | jrun | i have two devices on this nat and when i enable debug for the public ip i only get packets from the working device |
20:02.42 | jrun | so media session never gets established i guess |
20:03.29 | [TK]D-Fender | You should be looking at the SIp debug from * CLI, not external |
20:03.39 | [TK]D-Fender | It shouldn't be a guess. |
20:05.30 | jrun | i'm looking at it from within the console |
20:05.44 | jrun | old dev works fine, rtp packets come and go. |
20:06.21 | *** join/#asterisk Micc (~micster@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
20:07.04 | Micc | How much info is in the patch files? I'm currently using 14.3.0, could I just apply the 14.4.0 and 14.5.0 patch files and I would have the 14.5.0 complete source? |
20:07.34 | Micc | I've made a few of my own changes, I'd rather not have to make those changes to the new source. |
20:07.53 | Micc | But I understand that patches my not apply because of my changes. |
20:09.36 | *** join/#asterisk phix (~threat@205.209.220.203.dial.dynamic.acc01-hamm-bme.comindico.com.au) |
20:32.46 | Micc | I got it, had to go from 14.3.0 to 14.3.1 then 14.4.0 14.4.1 then 14.5.0 |
20:44.25 | *** join/#asterisk krzee (~k@openvpn/community/support/krzee) |
20:47.56 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
20:47.57 | *** mode/#asterisk [+o cresl1n] by ChanServ |
20:48.38 | *** join/#asterisk CrowX- (~CrowX-@79.141.115.88) |
20:48.44 | CrowX- | Can I ask an ari specific question here? |
20:49.00 | CrowX- | cause there's no one active on asterisk-ari atm |
20:54.40 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
20:55.12 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
20:55.13 | *** mode/#asterisk [+o cresl1n] by ChanServ |
20:58.56 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
21:00.14 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:00.54 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:01.44 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:01.48 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
21:01.48 | *** mode/#asterisk [+o cresl1n] by ChanServ |
21:02.31 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:03.16 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:04.06 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:04.58 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:05.52 | *** join/#asterisk overyander (~jeff@12.49.160.131) |
21:11.42 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
21:11.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
21:12.29 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
21:17.44 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:17.46 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
21:19.13 | *** join/#asterisk phix (~threat@220-244-203-122.tpgi.com.au) |
21:27.18 | *** join/#asterisk newtonr (~newtonr@99-104-129-136.lightspeed.brhmal.sbcglobal.net) |
21:27.18 | *** mode/#asterisk [+o newtonr] by ChanServ |
21:31.33 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
21:31.34 | *** mode/#asterisk [+o malcolmd] by ChanServ |
21:34.00 | *** join/#asterisk phix (~threat@203.63.186.65) |
21:34.38 | *** join/#asterisk nethope (nethope@triton.nivex.net) |
21:34.55 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
22:02.49 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
22:03.36 | rrittgarn | in Dialplan is there a way to unset a variable, or set it's value to null? |
22:04.08 | Samot | var='' |
22:04.40 | rrittgarn | thanks |
22:06.34 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
22:06.34 | *** mode/#asterisk [+o cresl1n] by ChanServ |
22:06.53 | rrittgarn | just tried that and then when i print the value of the variable i get '' (two single quotes) |
22:07.22 | Samot | or var= |
22:08.13 | rrittgarn | that did it |
22:08.14 | rrittgarn | thanks |
22:09.16 | *** join/#asterisk phix (~threat@203.194.41.40) |
22:32.24 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
22:40.44 | *** join/#asterisk phix (~threat@61.68.191.124) |
22:59.47 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
23:10.12 | *** join/#asterisk cmendes0101 (~cmendes01@cpe-76-170-253-155.socal.res.rr.com) |
23:15.55 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zlwdsjffrcwdeiib) |
23:24.49 | *** join/#asterisk cmendes0101 (~cmendes01@cpe-76-170-253-155.socal.res.rr.com) |
23:33.09 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |