IRC log for #asterisk on 20160913

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00:09.29mighty44Hello!
00:09.39mighty44Help please! Outgoing calls work , but do not work incoming calls , no beeps
00:09.51mighty44I'm use chan_sip
00:10.42mighty44debug log does not write anything
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01:30.32[TK]D-Fendermighty44, if you register to your provider, go prove that has happened : "sip show registry"
01:30.43[TK]D-Fendermighty44, And then show us a call attempt.
01:30.49[TK]D-Fender"sip set debug on"
01:31.05[TK]D-Fenderlog files don't mean much here vs live CLI
01:31.06[TK]D-Fender~pb
01:31.09infobotextra, extra, read all about it, 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.
01:31.10[TK]D-Fender^^^^
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02:33.02wyoungpastebin.com is great if you like reading Ads
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07:54.34shuttleduckAnyone here familiar with Bria and SIP Keepalive?
07:54.38todwetsprockhi ! on linux, Asterisk 11.17.1, there was a phone call this morning. i like to see the logfiles  (like in asterisk -rvvvvv). how can i do that ?
07:55.34shuttleduckIf i disable all firewall traversal options but enable SIP Keepalive in Bria, Bria starts sending out UDP packets to port 5060 of the registrar
07:55.57shuttleduckIs that the only thing Bria does for hole punching or is there a mechanism involved for media as well?
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08:34.04chl_hi, whats the command to turn off debugging for a single peer?
08:34.10todwetsprockhi ! on linux, Asterisk 11.17.1, there was a phone call this morning. i like to see the logfiles  (like in asterisk -rvvvvv). how can i do that ?
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09:09.18alamarso... I'm new to asterisk and there is varying documentation on how to implement fax - is utilizing spandsp still the way(or a viable) to go?
09:09.37alamar(fax via sip is the usecase)
09:11.46wyoungtodwetsprock: have you setup a CDR?
09:12.11wyoungif not you may be lucky to have one enabled by default in your distro.  Should be under /var/log/asterisk/cdr
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09:34.15Dovidhi. If I have Exten => _X,1,Noop and Exten => 1,1,Noop() Asterisk will preffer the Exten => 1,1 since its more defined. Is there any way in Exten => 1,1 To tell it to go to Exten => _X,1 ?
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09:44.34KValchev~pb
09:44.37infoboti guess 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.
09:45.15KValchevDovid: shhow context with pastebin
09:48.21DovidKValchev: http://pastebin.ca/3715499
09:50.29KValchevDovid: move last 3 lines (14,15,16) to up .(before line 1)
09:54.58DovidKValchev: I had a typo. See this PB http://pastebin.ca/3715502
09:56.10Dovidit wont work since Once it reaches (CONFIRMING_CART_DONATION=1) it will be 1 and the next time around it will be true. Unless the goto at the end does not start at pri1 but pri 5
09:56.17Dovidlet me give that a shot
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10:06.01KValchevwith that code is not impossible. If you want line 11 to jump to line 14 , you must change line 14 with extent diff from 1 to 9. For example with latter A.
10:07.38*** join/#asterisk lsrgr_ (02540b90@gateway/web/freenode/ip.2.84.11.144)
10:07.42lsrgr_hello to all
10:08.09lsrgr_can anyone help me about an problem with the setup of an fxs?
10:08.12KValchevIn you case digits 1 is part from expression [1-9]
10:10.51KValchevDovid: read this http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting
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10:14.10lsrgr_does someone have instructions on how to setup an fxs for an analog phone?
10:15.48KValchevlsrgr_: what hw you use ?
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10:18.01lsrgrhello again
10:18.03lsrgrsorry
10:18.07lsrgrim using
10:18.08lsrgrWildcard TDM410P
10:18.31lsrgrwith two fxo and two fxs
10:18.39lsrgrmodules
10:19.07KValchevdo you read manual from manufacture ?
10:20.56lsrgri havent search for it yet. Im trying from intructions I find on the web but I dont find anything good
10:22.15lsrgrIts also an old card
10:22.28lsrgrI dont think they will still have the manual
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10:48.15lsrgrDoes the FXS modules needs to be setup like an Trunk or an extension? (like FXO are setup)
10:52.10SamotWhat's connecting to the FXS ports?
10:52.14SamotAnalog phones?
10:59.06lsrgrcorrect
11:00.16SamotSo they would most like like extensions/peers.
11:00.22SamotSo they would most likely be extensions/peers.
11:04.09fileyay internet access
11:06.08SamotHey file, is DUNDi still a thing?
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11:09.19lsrgrsamot how do I setup an fxs module in to an extensions/peers?
11:09.49SamotI don't use DAHDi so I don't know what you need to do for it
11:10.21lsrgrI see
11:11.26SamotI don't use FXS/FXO cards. I used adapters.
11:11.31SamotI don't use FXS/FXO cards. I use adapters.
11:11.41fileSamot, some companies are using it internally in creative ways but there's no global network
11:11.56SamotSo it's pretty much died?
11:13.16WIMPyWell, due to the way Asterisk works, its use is a little limited.
11:15.16fileit's not something we work on or receive issue reports for, it generally just works for those who are using it themselves
11:17.18WIMPyLoosing digits while dialling doesn't really meet my definition of "works".
11:17.23SamotI figured since dundi.com is now redirecting to a digium user login page.
11:24.25SamotI remember back in like 2006 people calling the ITSP I was at asking if we did DUNDi peering.
11:32.00PenguinIs anyone else getting what seems to be a new line of SIP attack?  Their callerid is 442030930112 and they're trying to pass calls to 441134018000 with various prefixes.
11:32.40*** join/#asterisk mighty44 (~mighty44@victoriya-mebel.com)
11:34.16SamotYou mean someone is trying to make international calls through your PBX on the Internet?
11:34.26SamotNo, that's totally a new line of SIP attack.
11:34.28WIMPyAnd what's new about it?
11:36.31mighty44Hello!
11:36.43mighty44plz help!
11:37.01SamotWe tried twice yesterday. Are you going to actually follow up with our questions today?
11:37.05mighty44<mighty44> Help please! Outgoing calls work , but do not work incoming calls , no beeps
11:37.05mighty44<mighty44> I'm use chan_sip
11:37.17mighty44<[TK]D-Fender> mighty44, if you register to your provider, go prove that has happened : "sip show registry"
11:37.17mighty44<[TK]D-Fender> mighty44, And then show us a call attempt.
11:37.27SamotYup.
11:37.30mighty44Asterisk*CLI> sip show registry
11:37.30mighty44Host                                    dnsmgr Username       Refresh State                Reg.Time
11:37.30mighty440 SIP registrations.
11:37.32SamotWe both asked for a call attempt.
11:37.38TandyUKuse a damn pastebin
11:37.45TandyUKso youre not registered
11:37.50TandyUKno wonder calls wont work
11:37.51SamotWell you have no registrations.
11:37.58SamotYup/
11:38.29mighty44please write without abbreviations I translate into your language
11:38.54SamotYour SIP trunk is not registered.
11:39.05SamotIt will not accept incoming calls.
11:39.19mighty44attempts to send the call in private ?
11:39.27TandyUKpastebin
11:39.32TandyUKand give us the url
11:40.21mighty44What is pastebin?
11:40.26Samot~pb
11:40.26infobotpastebin is probably 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.
11:40.36TandyUKhttp://lmgtfy.com/?q=pastebin
11:41.15mighty44lol ty, im understand, one minute
11:41.35TandyUKbut the point is, your sip provider is not registered
11:41.45TandyUKa call trace is pointless - it wont work because you arent registered
11:41.55TandyUKso the call has nowhere to go
11:41.56SamotVery true.
11:43.37mighty44http://pastebin.ca/3715670
11:44.15mighty44but outgoing calls work
11:44.21TandyUKyou need to fix your registration first
11:44.50mighty44how? idont know :(
11:44.55TandyUKyou also have firewall issues
11:45.01TandyUKPacket timed out after 32000ms with no response
11:45.10TandyUKthats very bad, no idea how you think outgoing calls work
11:45.21mighty44I just do not know where and what to dig, I am confused
11:46.10mighty44i am have firewall
11:46.39mighty44can firewall problem?
11:46.53KValchevyes
11:47.31mighty44<KValchev> русский да?
11:49.37PenguinI'm not suggesting this is a new concept of SIP attack.  I'm asking if THIS seems to be a new attack that's "out there."  I've been seeing hundreds of IP adddresses making this exact call: the callerid I listed is calling a variation to the number I listed.  Usually it's a random callerid to a random number and not from IPs on bunches of different subnets.
11:49.43KValchevmighty44: не бългаски . :)
11:50.47PenguinBut I've been working on the firewall and I was curious if this is a new botnet or if it's been around for a while and I was avoiding it.
11:52.50mighty44KValchev, :)
11:52.58mighty44KValchev, кому пингвин пишет?
11:53.02mighty44Причем тут атаки
11:53.12PenguinEnglish!
11:53.18mighty44Или он о своей теме?
11:53.24mighty44Sorry
11:54.34mighty44Penguin, im sorry, I just do not understand, and then attack?
11:54.47PenguinI keep blocking these as they come in.  It just seems like a new botnet.
11:56.23SamotPenguin: Are the calls getting through?
11:57.05SamotPenguin: You also have allowguest=no, correct?
11:57.27mighty44firewall in my company is not set to asterisk, the firewall is set for show
11:57.51TandyUKmighty44: that makes no sense
11:58.09Samotmighty44: Do you see any attempts for REGISTER messages being sent to your provider?
11:58.10mighty44yeap
11:58.35SamotIf you see the REGISTER attempts go out, are you getting a response?
11:58.46TandyUKsngrep is your friend here imho
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11:59.28mighty44I think that yes, outgoing work
11:59.57mighty44Samot, I think that yes, outgoing work
12:00.04SamotThat's because your AUTH on outgoing each call.
12:00.17SamotSo that means your username/password are correct.
12:00.38SamotBut you need to REGISTER to accept inbound calls.
12:00.48SamotThey need to know the location of your PBX.
12:01.26Penguinsamot: No, I don't allow the calls to pass through.  I do allow guests for certain reasons, but only to very specific extens, and those extens that are valid are only inbound.
12:01.57SamotPenguin: You have told your box to accept calls from bascially anyone.
12:02.04SamotPenguin: You have told your box to accept calls from basically anyone.
12:03.21mighty44Samot, Join our provider occurs without authorization login
12:04.05Samotmighty44: You are either doing IP AUTH or your are registering with a username/password. Which is it?
12:05.01mighty44IP Auth
12:05.35Penguinsamot: Yes, yes I have.  That is correct.
12:05.59Samotmighty44: OK so there would be no registry for that.
12:06.03SamotUgh.
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12:06.27mighty44sorry
12:06.33PenguinBut those "calls" that I was asking if it's a new botnet are useless for the purpose they are intending them to serve.  I wasn't asking about if I'm accepting their calls or doing anything with them.  I was curious if it's a new botnet or just seems like one to me.
12:07.52mighty44Samot, can show configuration?
12:08.33WIMPyPenguin: You should watch them. It's quite entertaining what kind of prefixes they try.
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12:14.54mighty44Samok, TandyUK, What should I do?
12:24.25mighty44Samok, TandyUK, i am accept the ports from provider to asterisk TCP 5060 5081 UDP 23814-23823
12:25.33mighty44and from asterisk to web CRM TCP 443 1935 5060 5081 UDP 1935 3478 5060 8000-48000
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12:26.46[TK]D-Fendermighty44, Show us the outgoing call
12:26.53[TK]D-Fender~pb
12:26.55infobotmethinks 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.
12:26.56[TK]D-Fender^^^
12:28.24mighty44<PROTECTED>
12:34.30[TK]D-FenderDon't mask IP's, I can't tell what's what in there
12:36.55mighty44Ok
12:37.12mighty44i try again
12:37.23mighty44im set sip set debug Rostelecom
12:38.10mighty44this scheme: Rostelecom - My asterisk - Bitrix24 (WebCRM)
12:39.21[TK]D-FenderWhat are each of those?
12:41.50mighty44Russia
12:42.00mighty44http://pastebin.ca/3715747
12:42.23[TK]D-Fender....
12:42.26[TK]D-FenderWHAT are they.
12:42.28[TK]D-Fendernot WHERE are they
12:44.53[TK]D-Fender<--- Transmitting (NAT) to 80.73.85.134:5081 --->
12:44.54[TK]D-FenderSIP/2.0 100 Trying
12:44.58[TK]D-FenderContact: <sip:460644@192.168.7.144:5060>
12:45.04[TK]D-FenderPacket at line 53
12:45.06[TK]D-FenderBAD
12:45.34[TK]D-FenderYou are getting a call from a PUBLIC IP, and they were targeting YOUR public IP, but this response is sending your PRIVATE IP in the contact.
12:45.43[TK]D-FenderYou have done your NAT configuration wrong.
12:45.46[TK]D-Fendergo fix it.
12:46.16[TK]D-FenderThese --->  localnet, externaddr, nat, directmedia
12:47.03mighty44:(
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13:00.21mighty44Can you help me step by step? I have a whole week I fight with him
13:00.30[TK]D-FenderWith whoÉ
13:00.41[TK]D-FenderFix those settings
13:00.47[TK]D-Fenderthey are documented in the sample config
13:00.53[TK]D-Fender* has to know that you are behind NAT.
13:01.14[TK]D-FenderTThey have to know which peers they can trust for what they offerm, and which one's they can't
13:01.42KValchevmighty44: http://www.voip-info.org/wiki/view/Asterisk+FWD+NAT+Config+Example
13:02.04[TK]D-Fenderno
13:02.10[TK]D-Fendertaht guide is ancient garbage
13:02.17[TK]D-FenderTWELVE years old
13:02.24[TK]D-Fenderthe parameter names have mostly changed
13:03.29[TK]D-FenderRead the SAMPLE CONFIG
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14:45.18DanQuinneyGeneric question, what are people's methods of testing the dial plan, a) Manually make a couple of test calls and check the output in console. b) Have a script that does A. c) Testing? What's that? d) Something else
14:48.32newtonrYou don't typically need a script unless you are testing performance.
14:49.55newtonrJust make some test calls to run through your dialplan if you need to verify basic function
14:50.33newtonrHowever its always a good idea to test performance and load before you put a new system into production
14:51.14newtonrYou can use tools like SIPp to generate SIP traffic
14:51.16newtonrhttp://sipp.sourceforge.net/
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15:06.26[TK]D-FenderDanQuinney, "A"
15:07.11[TK]D-FenderDanQuinney, You could alternately do a path check with "dialplan show extension@context"
15:07.34[TK]D-Fenderwhich will show you what it should process if it were to go there (somewhat)
15:08.57DanQuinneyah, interesting
15:09.08DanQuinneyI didn't know that
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15:34.46igcewielingtyping "help" in the Asterisk CLI and pressing enter gives all sorts of "I didn't know that" information.
15:42.01drmessanoAs does IRC
15:42.04drmessanohelp
15:43.31igcewielingdrmessano: best kept secret on the internet!
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19:25.29*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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20:23.35igcewielingidles wonders if there is a parallel universe where SIP is a simple and elegant protocol.
20:25.00PenguinThat sounds like a pair of ducks.
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20:52.29Kobaz[2016-09-13 16:14:16.542] WARNING[12360]: res_musiconhold.c:695 monmp3thread: poll() failed: Interrupted system call
20:52.32Kobazmusiconhold no workey
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23:09.23ruben23hi there guys, my asterik has a none working dtmf, i mean when i try to press digits it does not recognized it by dtmf, how do i troubleshoot this somehow..? any idea...thanks
23:12.28[TK]D-FenderConfirm what the other side is expecting ... and set that on your peer
23:25.57ruben23[TK]D-Fender: can you explain a bit futher, what expectation like that.?
23:26.49ruben23when i call it ask me for an extensions but when i dial that extension with dtmf, it does not accept causing my calls not to move forward.
23:34.17[TK]D-FenderBecause you set the wrong mode
23:34.28[TK]D-FenderWhat is hard to understand about this?
23:34.34[TK]D-FenderThe DTMF mode is a setting.
23:34.40[TK]D-FenderIf you get it WRONG it will not work.
23:34.48[TK]D-FenderPick the right one that each side is expecting
23:48.00ruben23<PROTECTED>

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