| 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 |
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| 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 |
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| 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. |
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| 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. |
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| 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. |
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| 13:27.57 | Samot | Alright then. |
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| 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 |
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| 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 |
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| 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! |
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| 15:14.00 | davlefou | [TK]D-Fender, it work! Thanks for your help! |
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| 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. |
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| 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 |
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| 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 |
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| 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 |
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| 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 |
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| 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 |
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| 18:59.01 | jrun | Saydigits ? |
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| 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 |
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| 19:41.44 | jrun | file: playback also fails with chan_sip. is this media related? |
| 19:41.57 | file | entirely possible |
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| 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 |
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| 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 |
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| 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. |
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| 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. |
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| 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 |
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| 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 |
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| 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 |
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| 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 |
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