IRC log for #asterisk on 20160920

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00:08.32[TK]D-Fenderpramsky, "sip show registry"
00:08.48pramsky0 registrations
00:10.09[TK]D-FenderShow the full config
00:10.26[TK]D-Fenderagain, masking only secrects
00:10.34[TK]D-Fenderthat includes things you don't feel may be important
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00:19.04pramskygot it
00:22.15pramskyhttp://pastebin.com/HMVudrG3
00:23.19[TK]D-FenderThats the problem
00:23.29[TK]D-Fender1395 <- you started defining other sections
00:23.40[TK]D-Fender1578 = your register
00:23.53[TK]D-FenderRegister statements have to come after [general] and before any other section
00:23.59[TK]D-FenderYou cannot just mix them anywhere
00:25.07pramskyahh k
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01:58.31jeffspeffis an ODBC connection the only way to get a realtime mysql connection working for sip, extensions, etc. or is there something else i'm missing with the native realtime mysql driver?
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01:59.21[TK]D-FenderNative has fallen out of favour years ago
02:00.52jeffspeff[TK]D-Fender by native, i'm referring to res_config_mysql. are you referring to the same?
02:01.09[TK]D-FenderAnd anything like it
02:01.15jeffspeffoh
02:01.21[TK]D-Fenderthat isn't the ONLY file that is type specific
02:01.34jeffspeffwhy did it fall out of favor?
02:01.56[TK]D-FenderBecause who wants to maintain and link code to multiple backends?
02:02.02[TK]D-Fenderthat's the entire point of abstraction.
02:02.15jeffspefftrue.
02:02.29[TK]D-FenderWant to use MySQL?  Pass through ODBC like the guy complaining that there was never a native PgSQL module at all
02:02.59jeffspefflol
02:04.00jeffspeffthanks for the info
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02:29.11pramsky[TK]D-Fender, thanks for your help earlier =)
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06:41.58andycol_500hi guys
06:42.13andycol_500i am seeing a lot of warnings saying ast_prod: Prodding channel failed
06:42.16andycol_500what does it mean?
06:42.23andycol_500i am running asterisk 13.11.2
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07:10.57sacoetzeehi
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07:19.15andycol_500hi
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10:32.10andycol_500hi all
10:32.11andycol_500i am seeing a lot of warnings saying ast_prod: Prodding channel failed
10:32.16andycol_500does anyone know what causes it
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10:48.40SamotShow the warning.
10:48.44SamotAnd when are you getting them?
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10:59.23andycol_500i am getting them all the time
10:59.24andycol_500the warning is
11:00.09andycol_500Prodding channel 'SIP/xxxxx-000487f8' failed
11:06.14SamotSo when you make or receive a call?
11:06.29andycol_500this is when someone makes a call
11:06.39andycol_500sorry i mean receiving a call
11:06.41andycol_500so inbound
11:06.43SamotOK so make a call and show it.
11:06.48Samotasterisk -rvvvvvvvvvv
11:06.49Samot!pb
11:06.52Samot~pb
11:06.52infobothmm... 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.
11:06.55andycol_500ok
11:07.20andycol_500greping for it now as there is +- 300 concurrent calls
11:07.57SamotNo don't grep.
11:08.00SamotMake a call
11:08.03SamotLive data.
11:08.11andycol_500ok
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11:10.10andycol_500http://pastebin.ca/3720250
11:15.11SamotThat can't be all of it.
11:15.24SamotOh awesome, a2billing.
11:15.37SamotSo you need to make another call this time with "agi set debug on"
11:15.48SamotSo the a2billing steps can be seen.
11:17.03andycol_500ok
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11:21.57andycol_500http://pastebin.ca/3720253
11:21.59charles__Hi all
11:22.47SamotThese all say the channel is busy.
11:23.00SamotHow can a call produce so little logs?
11:23.31SamotOK, another call...this time with "sip set debug peer <peer>" with peer being the endpoint/extension the call is going to...
11:23.57andycol_500its hard to do that as there is a lot of calls so its not all customer
11:24.00andycol_500customers
11:24.53SamotWell this output is only a few lines, it's not the whole call from what I can see.
11:25.00SamotWhat version of Asterisk?
11:25.03andycol_500what does the warning actualy mean
11:25.03andycol_50013
11:25.13SamotIt means there's no channel.
11:25.24SamotThere was a channel but there's not anymore.
11:25.34SamotAnd in all these errors the result is "Busy"
11:25.51SamotSo something it telling the call the destination is busy.
11:25.59andycol_500ok
11:26.00SamotSo something is telling the call the destination is busy.
11:26.06andycol_500so that might be it
11:26.11SamotWhat release of 13?
11:26.14andycol_50013.11.2
11:26.15SamotHard to say.
11:26.27SamotIF it's not happening on all the calls then it must be related to the users the calls are going to.
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11:26.36andycol_500i wonder if this related to the fact asterisk is sending the wrong cause code mapping
11:26.52SamotHow is it sending the wrong cause code?
11:27.20andycol_500let me show u example 1 sec
11:28.12andycol_500http://pasteboard.co/5lfKAMB4Z.png
11:28.34andycol_500so its 503 unavailable but asterisk sending user busy
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11:29.57andycol_500any ideas why asterisk is doing that
11:30.47SamotCause Code 17 = User Busy
11:31.08SamotWhat is the status of the user(s) that these calls are going to?
11:31.11andycol_500yes but look 503 service unavailable at the top
11:31.19SamotThat's a SIP message.
11:31.26andycol_500yes
11:31.35SamotCause Codes are ISDN codes.
11:31.41SamotThat are translated into SIP codes.
11:31.50SamotThere's not a 1 to 1 relationship.
11:31.59andycol_500ok
11:32.26SamotDo these DIDs route to extensions?
11:32.54andycol_500no they route to sip trunks
11:32.59andycol_500which goes to a pabx
11:33.05andycol_500some asterisk
11:33.07andycol_500some aastra
11:33.10andycol_500some samsung etc
11:33.16SamotWhat?
11:33.38*** join/#asterisk biomorph (~charles@185.59.125.22)
11:33.46andycol_500so did comes from a2billing to a sip trunk which is customer registering to a2billing
11:33.59andycol_500but that incorrent q850 code is also on outbound calls
11:34.35SamotSo the DID that you are testing with...
11:34.38SamotIt goes to a trunk?
11:35.03andycol_500yes
11:35.15SamotAnd that trunk is up?
11:35.36biomorphHi.  I have one extension that I do not want to be able to make external calls.  Do I have to set up a dummy trunk for that?
11:35.39andycol_500yes
11:35.48andycol_500as i said though its not on all calls
11:35.59Samotbiomorph: You can just not allow the device to make calls via dialplan.
11:36.07SamotTo the destinations you don't want to to make calls.
11:36.18Samotandycol_500: OK so now make another test call...
11:36.28SamotThis time with "sip set debug ip <ip of trunk>"
11:36.36andycol_500but that prodding channel isnt a major issue, just something i have noticed in the cli
11:36.43SamotThen make a call.
11:36.44andycol_500my major issue is the cause code mapping incorrect
11:36.58SamotShow where it's incorrect.
11:37.04SamotOn this outbound call..
11:37.17andycol_500my issue is similar to
11:37.18andycol_500https://issues.asterisk.org/jira/browse/ASTERISK-23484
11:37.31andycol_500ok 1 sec
11:38.30andycol_500http://pasteboard.co/5lqC5xi0F.png
11:38.49andycol_500it was declined by asterisk showing 16 which is normal clearing
11:39.20SamotWho is sending the 603?
11:39.22andycol_500but asterisk not by
11:39.25SamotAsterisk or the endpoint
11:39.33andycol_500endpoint
11:39.39andycol_500which is also an asterisk system?
11:39.51SamotOK so the endpoint is denying the call.
11:39.59SamotIs is a box you manage?
11:40.02andycol_500im not worried about it denying the call
11:40.12andycol_500im worried why its saying cause 16 when it was denied?
11:40.13Samotq.850 are cause codes.
11:40.17andycol_500yes
11:40.35SamotBeause 16 is normal clearing.
11:40.44andycol_500yes but that wasnt normal clearing?
11:40.53SamotYou made the request to the endpoint, the endpoint rejected it. Call clears normally.
11:41.08SamotYes, it did.
11:41.18SamotCall -> Response -> Clear Call.
11:41.44andycol_500ok let me see if i can find another example
11:42.10SamotSo this doesn't happen on every call?
11:42.13SamotJust some calls
11:42.15SamotTo some users
11:42.23SamotBut not the same users, just random users?
11:42.36andycol_500no the incorrect cause code seems to be on 90% of calls
11:42.39andycol_500let me explain my setup
11:42.51SamotWhy do you think the code is incorrect?!
11:43.31andycol_500because it is different to the top code
11:43.31andycol_5001sec
11:43.54SamotStop comparing ISDN codes to SIP codes
11:43.58SamotTHEY DON'T SYNC UP
11:44.01andycol_500ok
11:44.36biomorphSamot: I've been hunting for an example but haven't found one.
11:44.53SamotIf the endpoints are returning these errors you need to look at the endpoints.
11:45.16SamotBut so far this looks like calls are being sent to your endpoints and they aren't accepting the calls for various reasons.
11:46.28Samot603 Decline means that the endpoint understood the SIP request, can process the SIP request but refuses to do so.
11:48.39Samot503 Service Temporarily Unavailable generally means the destination got the call but can't process it because there's not available service at the time to handle it. I.e. no channels available.
11:49.32Samotbiomorph: When I pick up a call and dial digits and send those to Asterisk...
11:49.58SamotAsterisk has to process those digits. So if you don't want them to make outbound calls, then don't accept 10/11 digits from the user.
11:50.46SamotOr route them to another destination in Asterisk like an announcement/recording playback.
11:51.39biomorphSamot: Thanks for your help.  You have pointed me in the right direction.  Sorry it's a long time since I looked at this system.  Getting my brain back in gear.
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12:06.46*** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66)
12:07.04Rasputin3711Hello,
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12:10.02Rasputin3711How can i debug - chan_sip.c:handle_response_peerpoke? I have a network without nat, all tels connect via switch. Time to time all my tels:
12:10.03Rasputin3711[2016-09-20 01:10:23] NOTICE[28233] chan_sip.c: Peer '200' is now UNREACHABLE!  Last qualify: 7
12:10.03Rasputin3711[2016-09-20 01:11:02] NOTICE[28233] chan_sip.c: Peer '200' is now Reachable. (10ms / 2000ms)
12:11.21Rasputin3711qualify=yes;qualifyfreq=60;keepalive=0
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12:38.14SamotThat's a NAT issue with the endpoint.
12:38.47SamotGoing unreachable means the endpoint isn't responding the to the options messages being sent by Asterisk.
12:39.40SamotSo no NAT on the endpoint? Still means the same thing. It's not responding to the messages being sent by Asterisk.
12:41.52Rasputin3711sip.conf -> peer [200] nat=no
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12:44.47Rasputin3711nat=never
12:45.47mirela666nat=devil
12:45.55mirela666:D
12:48.03Rasputin3711Samot: Is it good idea to set qualify=no?
12:48.29SamotAre the endpoints local?
12:48.52Rasputin3711yep
12:49.05Rasputin3711I have no issue with isp trunk
12:49.24SamotYou set qualify to no, then Asterisk won't send option messages.
12:49.37SamotIt still won't solve the issue of the endpoint not receiving messages from Asterisk.
12:50.02SamotThe peer/endpoint will show as unmonitored in "sip show peers"
12:50.22SamotBut again, if the endpoint isn't accepting messages it won't receive calls.
12:51.01SamotSo figure out what is causing the issue. qualify=no is like turning up the car radio to not hear that "weird" sound.
12:52.30Rasputin3711What should i do?
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13:08.50Rasputin3711Samot: Is nat=never equal nat=no?
13:09.36SamotNo.
13:10.09Rasputin3711It should be nat=no for my config?
13:10.17SamotGenerally.
13:11.20SamotYou need to confirm the sip message is getting to the endpoint.
13:13.41Rasputin3711qualify=yes;qualifyfreq=60;rtpkeepalive=0 for endpoint and nat=no in main config?
13:16.32SamotWhy are you not looking at the actual network and endpoint.
13:16.51SamotIt's clear that the endpoint can send messages to Asterisk.
13:17.06SamotBut it doesn't respond to the messages sent by Asterisk.
13:17.12Rasputin3711All endpoint ~7ms
13:17.33SamotWhen the endpoint goes goes unreachable it means asterisk couldn't reach it.
13:17.41SamotWhen it goes reachable it means Asterisk could reach it.
13:17.56Rasputin3711It happens one or two times per day for ~40 endpoints at once
13:18.17SamotThat generally happens because the endpoint sent a message, generally a register that "refreshes" the connection.
13:18.24SamotNetwork.
13:18.40SamotWhy do they stop responding to the messages from Asterisk?
13:18.56Rasputin3711I want to know too )
13:19.00Rasputin3711How to debug
13:19.06SamotI bet when it happens if you resync or reboot the phones they start to work again.
13:19.09SamotI've told you.
13:19.26Rasputin3711[2016-09-20 01:10:23] NOTICE[28233] chan_sip.c: Peer '200' is now UNREACHABLE!  Last qualify: 7
13:19.27Rasputin3711<Rasputin3711> [2016-09-20 01:11:02] NOTICE[28233] chan_sip.c: Peer '200' is now Reachable. (10ms / 2000ms)
13:19.30SamotYou need to look at your network and what is causing the messages from Asterisk not make it to the end points.
13:19.46SamotStop looking at the Asterisk box for all the answers.
13:19.50SamotLook at your network.
13:19.53Rasputin3711is now UNREACHABLE! 3-5 seconds and all Reachable.
13:21.27Rasputin3711In 90% is a network hardware problem?
13:22.35SamotI don't know.
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13:22.57SamotThat is something you will need to figure out.
13:23.05SamotLook at the switch, look at the logs.
13:23.15SamotDo you see issues with interfaces?
13:23.20SamotDo they go down?
13:23.25[TK]D-FenderOr it is jsut as likely that the sever itself had some sort of hangup
13:23.36[TK]D-FenderWhich explains why the phones didnèt respond all at the same time.
13:24.10SamotWell that would be part of looking at the network instead of the logs.
13:24.22andycol_500Samot: do you work at digium?
13:24.28SamotNope.
13:24.36SamotOr I would have a cute @
13:24.47andycol_500i see :)
13:24.52filepeople who don't work at Digium can also have an @
13:24.56fileQwell doesn't work for us for example
13:25.31SamotWell he does a lot more for you than I probably do.
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13:25.41SamotI mean, which is basically nothing.
13:25.41Rasputin3711Network interface was fine that this happened
13:25.53SamotWhat about the traffic?
13:26.06SamotDoes the machine have a high load when this happens?
13:26.32andycol_500do u guys recommend migrating to pjsip from chan_sip?
13:26.41Rasputin3711CPU: < 1 %, Ram: 80 % free, Net: ~ 5mbps
13:26.50SamotAre all the endpoints down?
13:26.53Samotunreachable?
13:26.54Rasputin3711All
13:27.05Rasputin3711except isp
13:29.07SamotTraffic goes different paths between WAN and LAN.
13:29.21Rasputin3711yep, two nics
13:29.46SamotIs there traffic going out on the LAN interface?
13:30.17Rasputin3711WAN - ISP, LAN - switch - tels
13:30.32SamotIf all your LAN devices are unreachable...
13:30.54SamotIs there traffic on the LAN interface of Asterisk?
13:31.10Rasputin3711may be 3-5 calls
13:31.24SamotIs Asterisk sending packets/sip messages to the network and are those packets getting where they need to go?
13:31.42SamotOK so even though they are unreachable they are making calls?
13:32.04Rasputin3711maximum 3 are making calls
13:32.16SamotSo that means they can communicate with server, when the endpoint initiates the call.
13:32.44Rasputin3711yes, all calls via asterisk server
13:33.12SamotSo something it stopping the packets initiated by Asterisk to the endpoints.
13:33.24Rasputin3711directmedia=no
13:34.09SamotHow often is DHCP refreshing on the network?
13:34.19Rasputin3711No dhcp, static
13:34.32Rasputin3711Only ntp
13:34.43SamotAre the devices listening on unique ports or all on 5060?
13:35.50Rasputin3711200 10.10.10.20                               D  No         No          A  5060
13:35.55Rasputin37115060 all
13:37.40SamotChange the devices to listen on unique ports.
13:37.44SamotSee if that helps.
13:37.57SamotChange a few and see if those still go unreachable with the rest.
13:40.18Rasputin3711Samot: Thank you for you help. I will try it.
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13:50.50linjanhello! im using asterisk 14 and doing some king of callcenter with multiple hotlines and queues. for me everything is clear about AddQueueMember etc. but how to track outgoing calls? how to link outgoing call with the hotline in database?
13:51.21[TK]D-FenderIf your call... flag it however you want
13:51.51[TK]D-FenderCDR offers you the usual fields
13:53.45jruni keep getting 'OPTIONS failed from <some_device> because endpoint was not found' but phone _is_ registered and in cli 'pjsip show endpoint my_endpoint' returns with info for that endpoint. any idea? i'm running 13.11.2 and iirc there was a somewhat similar bug that triggered an emergency release for that version...
13:53.48jrunany idea?
13:53.56jrunand are those two are related?
13:57.25[TK]D-Fender"returns with info for that endpoint" <- Yes.  You have an endpoint.  However it does not MATCH the OPTIONS request that is coming in.
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14:40.17andycol_500has any of you guys had issues running asterisk 13 on debian 8
14:40.28andycol_500i notice as soon as i push more then 80 calls i get a segfault and it crashses
14:40.31andycol_500i notice as soon as i push more then 80 calls i get a segfault and it crashes
14:41.05andycol_50080 concurrent
14:42.13[TK]D-FenderStep 1 : get specific on versions
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14:45.25andycol_50013.11.2
14:45.30andycol_500debian 8.5
14:51.22[TK]D-Fender\collectdebug
14:51.27[TK]D-Fender~collectdebug
14:51.30infobotwell, collectdebug is a method of collecting logs allowing others help troubleshoot an issue.  Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
14:51.35[TK]D-Fender^
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14:57.18jjrhandycol_500: did you make sure you have enough file descriptors open?
14:57.25jjrhI ran into that issue when testing 200+ calls.
14:57.28andycol_500yes i set it to 99999999
14:57.40andycol_500it only seems to happen on debian 8
14:57.41andycol_5007 is fine
14:58.22jjrhI'm setting up a prod server for this which is duvian (the systemdless debian fork) - my previous test was on ubuntu server
14:59.10jjrhi'll try to remember to mention here when I get to the smoke test
15:01.08andycol_500cool thanks
15:04.56scvwho's ready for astricon!
15:13.04jrun[TK]D-Fender: SIPAUA is the user-agent. it's sending 'OPTIONS sip:ip_addr:port'
15:14.12[TK]D-FenderI'm sure there are a lot more lines in that packet
15:14.24jrunon the same subnet i also have UniFi user-agent (ubnt's uvp) which doesn't send such a request. in any case, my understanding was that endpoint's name was what would matter in terms of matching which is to be matches agains To:
15:16.27jrunhttps://gist.github.com/96b5be5df554f0c497a0b388075c7f2f
15:16.49jrunthat's coming from the SIPAUA
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15:26.21[TK]D-FenderWe're not looking at the full picture
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15:35.10jrunyou want the back and forth with the server?
15:36.09*** join/#asterisk mirela666 (~mirkob@89.184.168.162)
15:37.35jrunor the config files?
15:42.16jrunhttps://gist.github.com/257/96b5be5df554f0c497a0b388075c7f2f
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15:45.36jrunconfig:
15:45.38jrunhttp://bpaste.net/show/f253847d7583
15:46.34jrunclient <--> srv:
15:46.35jrunYour paste can be seen here: https://gist.github.com/e59e8aab009c0a15255ad1b51de80f63
15:47.10jrunsorry, ignore 5 lines above.
15:48.46[TK]D-FenderLooks like masking on the users, etc
15:48.56[TK]D-FenderAt which point I canèt see or trust typos, etc
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15:56.13jrunall done by script so typo is out
15:56.18*** join/#asterisk prudolf (~prudolf@ip58865e59.dynamic.kabel-deutschland.de)
15:56.40prudolfhi there.... maybe a little question:
15:56.43jrunseems like changing identify_by from auth_username to username stops the Notice msg
15:59.02prudolfi want to call out with all internal phonesnumbers  with a companynubmer. so we change the DID on the Trunc to our Company number.... Now we want that our external Coworker knows, where will call him when we dial there number.... so we need a solutions that when we call known numbers (from a list )
15:59.36prudolfthe our coworkers see the real number from the extention and not the comapany number
15:59.41prudolfis it possible?
16:00.11SamotAre they using a phone connected to the Asterisk box?
16:00.18SamotOr is this going to their cell number?
16:00.24Samotcell/mobile..
16:01.20prudolfwe have VOIP to asterisk (freepbx over easybell). the external coworkers are connected over easybell or over cellnumber
16:01.32prudolfmobilphone i mean
16:01.59prudolfthe coworker need to know who is calling from the company
16:02.02[TK]D-Fender~freepbx
16:02.02infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
16:02.03[TK]D-Fender^^^
16:02.19SamotIf you present your company's number as the caller ID most like the mobile carrier is going to present the CNAM of the number.
16:03.11SamotAnd what TK said, go to #freepbx for help with FreePBX
16:04.17prudolfok... thank you very much
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21:08.41GasterHow can I solve an issue with "wrong password" being shown for SIP friend, although the password is correct (it's a single letter password which is totally correct on both sides)? Asterisk is in local subnet behind NAT, listens to UDP 5060, which is forwarded to it from router. SIP peer is my home PC, also behind NAT. I have no network connectivity
21:08.41Gaster<PROTECTED>
21:09.50GasterThey say that this error is due to 'domain' misconfiguration, but nothing I did't to the corresponding sip.conf section helped.
21:10.47GasterTL;DR - A forwarded SIP registration fails with wrong password, althought the password is 100% correct.
21:11.57GasterHow do I check that both Asterisk and SIP peer use the same 'domain' for password hashing?
21:18.04GasterATM checking the ALG hypothesis - I have no access to the NAT router in front of Asterisk and it might have the dreaded feature.
21:23.26[TK]D-FenderShow us the full debug
21:23.54[TK]D-Fender"sip set debug on" <- chan_sip
21:24.00[TK]D-Fender"pjsip set logger on" <- pjsip
21:24.09[TK]D-Fender~pb
21:24.09infobotextra, 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.
21:24.11[TK]D-Fender^^^
21:24.16[TK]D-FenderAnd the config
21:26.03GasterI've solved the issue. I set "realm" option in sip.conf to my Asterisk's external NAT IP address.
21:26.27GasterApparently, this is used for password hashing 0_0
21:27.22GasterWhy doesn't this work by default? A clean installation with Asterisk behind NAT doesn't work with external SIP peers :C
21:27.39[TK]D-Fender"a clean install" doesn't eman anything
21:27.50[TK]D-FenderIt's your job to put in the settings for your working environment
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21:29.21GasterYeah, but I worked with ast before and I bump into the "realm" option issue just now - so it is used only if a port forward is between a peer and pbx.
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21:30.30GasterHow exactly does "realm" option work?
21:30.58[TK]D-FenderI've never even used it before myself
21:31.04GasterO_O
21:31.10[TK]D-FenderNever needed to touch it
21:31.16GasterSo did I
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22:29.14GasterThe real reason why password auth didn't work was ALG in Asterisk network's router. I've sniffed both good registration SIP packets (with 'realm' set to Asterisk's public ip in sip.conf) and bad registration packets (with 'realm' unset), and I did it on both peer's PC and Asterisk's - now I clearly see that packets are messed up (damn ALG) when pas
22:29.14Gastersing through Asterisk's gateway router, and the only reason setting 'realm' option helps is that it messes up the router's messing up :)
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23:02.07*** join/#asterisk twitchnln (~twitch@107.130.213.25)
23:03.22twitchnlnGreetings all, I was wondering if anyone had any experience with setting up openvpn on grandstream handsets?
23:23.12*** part/#asterisk kharwell (kharwell@nat/digium/x-cyqvdtcwdwpntqif)
23:51.29equilibriohello there asterisk gurus, I am getting this information in my console: res_hep.c:466 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination
23:51.38equilibrioAnybody could tell me how I can fix this?
23:51.48equilibrioEverything still works, but I do get this warning
23:51.51voipmonkHomer?
23:52.30equilibrioSimpson?
23:52.36equilibrio:P
23:52.53voipmonkhttps://github.com/sipcapture/homer/wiki/Examples%3A-Asterisk
23:53.38equilibriohmm to start I can see its not the good subnet configure in there
23:54.01equilibriovoipmonk: is it critical if I disable this module or it is really needed in general
23:54.03equilibriojust asking
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