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00:08.32 | [TK]D-Fender | pramsky, "sip show registry" |
00:08.48 | pramsky | 0 registrations |
00:10.09 | [TK]D-Fender | Show the full config |
00:10.26 | [TK]D-Fender | again, masking only secrects |
00:10.34 | [TK]D-Fender | that includes things you don't feel may be important |
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00:19.04 | pramsky | got it |
00:22.15 | pramsky | http://pastebin.com/HMVudrG3 |
00:23.19 | [TK]D-Fender | Thats the problem |
00:23.29 | [TK]D-Fender | 1395 <- you started defining other sections |
00:23.40 | [TK]D-Fender | 1578 = your register |
00:23.53 | [TK]D-Fender | Register statements have to come after [general] and before any other section |
00:23.59 | [TK]D-Fender | You cannot just mix them anywhere |
00:25.07 | pramsky | ahh k |
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01:58.31 | jeffspeff | is 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-Fender | Native has fallen out of favour years ago |
02:00.52 | jeffspeff | [TK]D-Fender by native, i'm referring to res_config_mysql. are you referring to the same? |
02:01.09 | [TK]D-Fender | And anything like it |
02:01.15 | jeffspeff | oh |
02:01.21 | [TK]D-Fender | that isn't the ONLY file that is type specific |
02:01.34 | jeffspeff | why did it fall out of favor? |
02:01.56 | [TK]D-Fender | Because who wants to maintain and link code to multiple backends? |
02:02.02 | [TK]D-Fender | that's the entire point of abstraction. |
02:02.15 | jeffspeff | true. |
02:02.29 | [TK]D-Fender | Want to use MySQL? Pass through ODBC like the guy complaining that there was never a native PgSQL module at all |
02:02.59 | jeffspeff | lol |
02:04.00 | jeffspeff | thanks for the info |
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02:29.11 | pramsky | [TK]D-Fender, thanks for your help earlier =) |
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05:11.20 | drmessano | lol |
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06:41.58 | andycol_500 | hi guys |
06:42.13 | andycol_500 | i am seeing a lot of warnings saying ast_prod: Prodding channel failed |
06:42.16 | andycol_500 | what does it mean? |
06:42.23 | andycol_500 | i am running asterisk 13.11.2 |
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07:10.57 | sacoetzee | hi |
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07:19.15 | andycol_500 | hi |
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10:32.10 | andycol_500 | hi all |
10:32.11 | andycol_500 | i am seeing a lot of warnings saying ast_prod: Prodding channel failed |
10:32.16 | andycol_500 | does anyone know what causes it |
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10:48.40 | Samot | Show the warning. |
10:48.44 | Samot | And when are you getting them? |
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10:59.23 | andycol_500 | i am getting them all the time |
10:59.24 | andycol_500 | the warning is |
11:00.09 | andycol_500 | Prodding channel 'SIP/xxxxx-000487f8' failed |
11:06.14 | Samot | So when you make or receive a call? |
11:06.29 | andycol_500 | this is when someone makes a call |
11:06.39 | andycol_500 | sorry i mean receiving a call |
11:06.41 | andycol_500 | so inbound |
11:06.43 | Samot | OK so make a call and show it. |
11:06.48 | Samot | asterisk -rvvvvvvvvvv |
11:06.49 | Samot | !pb |
11:06.52 | Samot | ~pb |
11:06.52 | infobot | hmm... 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. |
11:06.55 | andycol_500 | ok |
11:07.20 | andycol_500 | greping for it now as there is +- 300 concurrent calls |
11:07.57 | Samot | No don't grep. |
11:08.00 | Samot | Make a call |
11:08.03 | Samot | Live data. |
11:08.11 | andycol_500 | ok |
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11:10.10 | andycol_500 | http://pastebin.ca/3720250 |
11:15.11 | Samot | That can't be all of it. |
11:15.24 | Samot | Oh awesome, a2billing. |
11:15.37 | Samot | So you need to make another call this time with "agi set debug on" |
11:15.48 | Samot | So the a2billing steps can be seen. |
11:17.03 | andycol_500 | ok |
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11:21.57 | andycol_500 | http://pastebin.ca/3720253 |
11:21.59 | charles__ | Hi all |
11:22.47 | Samot | These all say the channel is busy. |
11:23.00 | Samot | How can a call produce so little logs? |
11:23.31 | Samot | OK, another call...this time with "sip set debug peer <peer>" with peer being the endpoint/extension the call is going to... |
11:23.57 | andycol_500 | its hard to do that as there is a lot of calls so its not all customer |
11:24.00 | andycol_500 | customers |
11:24.53 | Samot | Well this output is only a few lines, it's not the whole call from what I can see. |
11:25.00 | Samot | What version of Asterisk? |
11:25.03 | andycol_500 | what does the warning actualy mean |
11:25.03 | andycol_500 | 13 |
11:25.13 | Samot | It means there's no channel. |
11:25.24 | Samot | There was a channel but there's not anymore. |
11:25.34 | Samot | And in all these errors the result is "Busy" |
11:25.51 | Samot | So something it telling the call the destination is busy. |
11:25.59 | andycol_500 | ok |
11:26.00 | Samot | So something is telling the call the destination is busy. |
11:26.06 | andycol_500 | so that might be it |
11:26.11 | Samot | What release of 13? |
11:26.14 | andycol_500 | 13.11.2 |
11:26.15 | Samot | Hard to say. |
11:26.27 | Samot | IF 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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11:26.36 | andycol_500 | i wonder if this related to the fact asterisk is sending the wrong cause code mapping |
11:26.52 | Samot | How is it sending the wrong cause code? |
11:27.20 | andycol_500 | let me show u example 1 sec |
11:28.12 | andycol_500 | http://pasteboard.co/5lfKAMB4Z.png |
11:28.34 | andycol_500 | so its 503 unavailable but asterisk sending user busy |
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11:29.57 | andycol_500 | any ideas why asterisk is doing that |
11:30.47 | Samot | Cause Code 17 = User Busy |
11:31.08 | Samot | What is the status of the user(s) that these calls are going to? |
11:31.11 | andycol_500 | yes but look 503 service unavailable at the top |
11:31.19 | Samot | That's a SIP message. |
11:31.26 | andycol_500 | yes |
11:31.35 | Samot | Cause Codes are ISDN codes. |
11:31.41 | Samot | That are translated into SIP codes. |
11:31.50 | Samot | There's not a 1 to 1 relationship. |
11:31.59 | andycol_500 | ok |
11:32.26 | Samot | Do these DIDs route to extensions? |
11:32.54 | andycol_500 | no they route to sip trunks |
11:32.59 | andycol_500 | which goes to a pabx |
11:33.05 | andycol_500 | some asterisk |
11:33.07 | andycol_500 | some aastra |
11:33.10 | andycol_500 | some samsung etc |
11:33.16 | Samot | What? |
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11:33.46 | andycol_500 | so did comes from a2billing to a sip trunk which is customer registering to a2billing |
11:33.59 | andycol_500 | but that incorrent q850 code is also on outbound calls |
11:34.35 | Samot | So the DID that you are testing with... |
11:34.38 | Samot | It goes to a trunk? |
11:35.03 | andycol_500 | yes |
11:35.15 | Samot | And that trunk is up? |
11:35.36 | biomorph | Hi. 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? |
11:35.39 | andycol_500 | yes |
11:35.48 | andycol_500 | as i said though its not on all calls |
11:35.59 | Samot | biomorph: You can just not allow the device to make calls via dialplan. |
11:36.07 | Samot | To the destinations you don't want to to make calls. |
11:36.18 | Samot | andycol_500: OK so now make another test call... |
11:36.28 | Samot | This time with "sip set debug ip <ip of trunk>" |
11:36.36 | andycol_500 | but that prodding channel isnt a major issue, just something i have noticed in the cli |
11:36.43 | Samot | Then make a call. |
11:36.44 | andycol_500 | my major issue is the cause code mapping incorrect |
11:36.58 | Samot | Show where it's incorrect. |
11:37.04 | Samot | On this outbound call.. |
11:37.17 | andycol_500 | my issue is similar to |
11:37.18 | andycol_500 | https://issues.asterisk.org/jira/browse/ASTERISK-23484 |
11:37.31 | andycol_500 | ok 1 sec |
11:38.30 | andycol_500 | http://pasteboard.co/5lqC5xi0F.png |
11:38.49 | andycol_500 | it was declined by asterisk showing 16 which is normal clearing |
11:39.20 | Samot | Who is sending the 603? |
11:39.22 | andycol_500 | but asterisk not by |
11:39.25 | Samot | Asterisk or the endpoint |
11:39.33 | andycol_500 | endpoint |
11:39.39 | andycol_500 | which is also an asterisk system? |
11:39.51 | Samot | OK so the endpoint is denying the call. |
11:39.59 | Samot | Is is a box you manage? |
11:40.02 | andycol_500 | im not worried about it denying the call |
11:40.12 | andycol_500 | im worried why its saying cause 16 when it was denied? |
11:40.13 | Samot | q.850 are cause codes. |
11:40.17 | andycol_500 | yes |
11:40.35 | Samot | Beause 16 is normal clearing. |
11:40.44 | andycol_500 | yes but that wasnt normal clearing? |
11:40.53 | Samot | You made the request to the endpoint, the endpoint rejected it. Call clears normally. |
11:41.08 | Samot | Yes, it did. |
11:41.18 | Samot | Call -> Response -> Clear Call. |
11:41.44 | andycol_500 | ok let me see if i can find another example |
11:42.10 | Samot | So this doesn't happen on every call? |
11:42.13 | Samot | Just some calls |
11:42.15 | Samot | To some users |
11:42.23 | Samot | But not the same users, just random users? |
11:42.36 | andycol_500 | no the incorrect cause code seems to be on 90% of calls |
11:42.39 | andycol_500 | let me explain my setup |
11:42.51 | Samot | Why do you think the code is incorrect?! |
11:43.31 | andycol_500 | because it is different to the top code |
11:43.31 | andycol_500 | 1sec |
11:43.54 | Samot | Stop comparing ISDN codes to SIP codes |
11:43.58 | Samot | THEY DON'T SYNC UP |
11:44.01 | andycol_500 | ok |
11:44.36 | biomorph | Samot: I've been hunting for an example but haven't found one. |
11:44.53 | Samot | If the endpoints are returning these errors you need to look at the endpoints. |
11:45.16 | Samot | But so far this looks like calls are being sent to your endpoints and they aren't accepting the calls for various reasons. |
11:46.28 | Samot | 603 Decline means that the endpoint understood the SIP request, can process the SIP request but refuses to do so. |
11:48.39 | Samot | 503 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. |
11:49.32 | Samot | biomorph: When I pick up a call and dial digits and send those to Asterisk... |
11:49.58 | Samot | Asterisk 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. |
11:50.46 | Samot | Or route them to another destination in Asterisk like an announcement/recording playback. |
11:51.39 | biomorph | Samot: 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) |
12:07.04 | Rasputin3711 | Hello, |
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12:10.02 | Rasputin3711 | How 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: |
12:10.03 | Rasputin3711 | [2016-09-20 01:10:23] NOTICE[28233] chan_sip.c: Peer '200' is now UNREACHABLE! Last qualify: 7 |
12:10.03 | Rasputin3711 | [2016-09-20 01:11:02] NOTICE[28233] chan_sip.c: Peer '200' is now Reachable. (10ms / 2000ms) |
12:11.21 | Rasputin3711 | qualify=yes;qualifyfreq=60;keepalive=0 |
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12:38.14 | Samot | That's a NAT issue with the endpoint. |
12:38.47 | Samot | Going unreachable means the endpoint isn't responding the to the options messages being sent by Asterisk. |
12:39.40 | Samot | So no NAT on the endpoint? Still means the same thing. It's not responding to the messages being sent by Asterisk. |
12:41.52 | Rasputin3711 | sip.conf -> peer [200] nat=no |
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12:44.47 | Rasputin3711 | nat=never |
12:45.47 | mirela666 | nat=devil |
12:45.55 | mirela666 | :D |
12:48.03 | Rasputin3711 | Samot: Is it good idea to set qualify=no? |
12:48.29 | Samot | Are the endpoints local? |
12:48.52 | Rasputin3711 | yep |
12:49.05 | Rasputin3711 | I have no issue with isp trunk |
12:49.24 | Samot | You set qualify to no, then Asterisk won't send option messages. |
12:49.37 | Samot | It still won't solve the issue of the endpoint not receiving messages from Asterisk. |
12:50.02 | Samot | The peer/endpoint will show as unmonitored in "sip show peers" |
12:50.22 | Samot | But again, if the endpoint isn't accepting messages it won't receive calls. |
12:51.01 | Samot | So figure out what is causing the issue. qualify=no is like turning up the car radio to not hear that "weird" sound. |
12:52.30 | Rasputin3711 | What should i do? |
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13:08.50 | Rasputin3711 | Samot: Is nat=never equal nat=no? |
13:09.36 | Samot | No. |
13:10.09 | Rasputin3711 | It should be nat=no for my config? |
13:10.17 | Samot | Generally. |
13:11.20 | Samot | You need to confirm the sip message is getting to the endpoint. |
13:13.41 | Rasputin3711 | qualify=yes;qualifyfreq=60;rtpkeepalive=0 for endpoint and nat=no in main config? |
13:16.32 | Samot | Why are you not looking at the actual network and endpoint. |
13:16.51 | Samot | It's clear that the endpoint can send messages to Asterisk. |
13:17.06 | Samot | But it doesn't respond to the messages sent by Asterisk. |
13:17.12 | Rasputin3711 | All endpoint ~7ms |
13:17.33 | Samot | When the endpoint goes goes unreachable it means asterisk couldn't reach it. |
13:17.41 | Samot | When it goes reachable it means Asterisk could reach it. |
13:17.56 | Rasputin3711 | It happens one or two times per day for ~40 endpoints at once |
13:18.17 | Samot | That generally happens because the endpoint sent a message, generally a register that "refreshes" the connection. |
13:18.24 | Samot | Network. |
13:18.40 | Samot | Why do they stop responding to the messages from Asterisk? |
13:18.56 | Rasputin3711 | I want to know too ) |
13:19.00 | Rasputin3711 | How to debug |
13:19.06 | Samot | I bet when it happens if you resync or reboot the phones they start to work again. |
13:19.09 | Samot | I've told you. |
13:19.26 | Rasputin3711 | [2016-09-20 01:10:23] NOTICE[28233] chan_sip.c: Peer '200' is now UNREACHABLE! Last qualify: 7 |
13:19.27 | Rasputin3711 | <Rasputin3711> [2016-09-20 01:11:02] NOTICE[28233] chan_sip.c: Peer '200' is now Reachable. (10ms / 2000ms) |
13:19.30 | Samot | You need to look at your network and what is causing the messages from Asterisk not make it to the end points. |
13:19.46 | Samot | Stop looking at the Asterisk box for all the answers. |
13:19.50 | Samot | Look at your network. |
13:19.53 | Rasputin3711 | is now UNREACHABLE! 3-5 seconds and all Reachable. |
13:21.27 | Rasputin3711 | In 90% is a network hardware problem? |
13:22.35 | Samot | I don't know. |
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13:22.57 | Samot | That is something you will need to figure out. |
13:23.05 | Samot | Look at the switch, look at the logs. |
13:23.15 | Samot | Do you see issues with interfaces? |
13:23.20 | Samot | Do they go down? |
13:23.25 | [TK]D-Fender | Or it is jsut as likely that the sever itself had some sort of hangup |
13:23.36 | [TK]D-Fender | Which explains why the phones didnèt respond all at the same time. |
13:24.10 | Samot | Well that would be part of looking at the network instead of the logs. |
13:24.22 | andycol_500 | Samot: do you work at digium? |
13:24.28 | Samot | Nope. |
13:24.36 | Samot | Or I would have a cute @ |
13:24.47 | andycol_500 | i see :) |
13:24.52 | file | people who don't work at Digium can also have an @ |
13:24.56 | file | Qwell doesn't work for us for example |
13:25.31 | Samot | Well he does a lot more for you than I probably do. |
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13:25.41 | Samot | I mean, which is basically nothing. |
13:25.41 | Rasputin3711 | Network interface was fine that this happened |
13:25.53 | Samot | What about the traffic? |
13:26.06 | Samot | Does the machine have a high load when this happens? |
13:26.32 | andycol_500 | do u guys recommend migrating to pjsip from chan_sip? |
13:26.41 | Rasputin3711 | CPU: < 1 %, Ram: 80 % free, Net: ~ 5mbps |
13:26.50 | Samot | Are all the endpoints down? |
13:26.53 | Samot | unreachable? |
13:26.54 | Rasputin3711 | All |
13:27.05 | Rasputin3711 | except isp |
13:29.07 | Samot | Traffic goes different paths between WAN and LAN. |
13:29.21 | Rasputin3711 | yep, two nics |
13:29.46 | Samot | Is there traffic going out on the LAN interface? |
13:30.17 | Rasputin3711 | WAN - ISP, LAN - switch - tels |
13:30.32 | Samot | If all your LAN devices are unreachable... |
13:30.54 | Samot | Is there traffic on the LAN interface of Asterisk? |
13:31.10 | Rasputin3711 | may be 3-5 calls |
13:31.24 | Samot | Is Asterisk sending packets/sip messages to the network and are those packets getting where they need to go? |
13:31.42 | Samot | OK so even though they are unreachable they are making calls? |
13:32.04 | Rasputin3711 | maximum 3 are making calls |
13:32.16 | Samot | So that means they can communicate with server, when the endpoint initiates the call. |
13:32.44 | Rasputin3711 | yes, all calls via asterisk server |
13:33.12 | Samot | So something it stopping the packets initiated by Asterisk to the endpoints. |
13:33.24 | Rasputin3711 | directmedia=no |
13:34.09 | Samot | How often is DHCP refreshing on the network? |
13:34.19 | Rasputin3711 | No dhcp, static |
13:34.32 | Rasputin3711 | Only ntp |
13:34.43 | Samot | Are the devices listening on unique ports or all on 5060? |
13:35.50 | Rasputin3711 | 200 10.10.10.20 D No No A 5060 |
13:35.55 | Rasputin3711 | 5060 all |
13:37.40 | Samot | Change the devices to listen on unique ports. |
13:37.44 | Samot | See if that helps. |
13:37.57 | Samot | Change a few and see if those still go unreachable with the rest. |
13:40.18 | Rasputin3711 | Samot: Thank you for you help. I will try it. |
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13:50.17 | *** mode/#asterisk [+o cresl1n] by ChanServ |
13:50.50 | linjan | hello! 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-Fender | If your call... flag it however you want |
13:51.51 | [TK]D-Fender | CDR offers you the usual fields |
13:53.45 | jrun | i 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... |
13:53.48 | jrun | any idea? |
13:53.56 | jrun | and 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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14:40.17 | andycol_500 | has any of you guys had issues running asterisk 13 on debian 8 |
14:40.28 | andycol_500 | i notice as soon as i push more then 80 calls i get a segfault and it crashses |
14:40.31 | andycol_500 | i notice as soon as i push more then 80 calls i get a segfault and it crashes |
14:41.05 | andycol_500 | 80 concurrent |
14:42.13 | [TK]D-Fender | Step 1 : get specific on versions |
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14:44.42 | *** mode/#asterisk [+o newtonr] by ChanServ |
14:45.25 | andycol_500 | 13.11.2 |
14:45.30 | andycol_500 | debian 8.5 |
14:51.22 | [TK]D-Fender | \collectdebug |
14:51.27 | [TK]D-Fender | ~collectdebug |
14:51.30 | infobot | well, 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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14:57.18 | jjrh | andycol_500: did you make sure you have enough file descriptors open? |
14:57.25 | jjrh | I ran into that issue when testing 200+ calls. |
14:57.28 | andycol_500 | yes i set it to 99999999 |
14:57.40 | andycol_500 | it only seems to happen on debian 8 |
14:57.41 | andycol_500 | 7 is fine |
14:58.22 | jjrh | I'm setting up a prod server for this which is duvian (the systemdless debian fork) - my previous test was on ubuntu server |
14:59.10 | jjrh | i'll try to remember to mention here when I get to the smoke test |
15:01.08 | andycol_500 | cool thanks |
15:04.56 | scv | who's ready for astricon! |
15:13.04 | jrun | [TK]D-Fender: SIPAUA is the user-agent. it's sending 'OPTIONS sip:ip_addr:port' |
15:14.12 | [TK]D-Fender | I'm sure there are a lot more lines in that packet |
15:14.24 | jrun | on 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: |
15:16.27 | jrun | https://gist.github.com/96b5be5df554f0c497a0b388075c7f2f |
15:16.49 | jrun | that's coming from the SIPAUA |
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15:26.21 | [TK]D-Fender | We're not looking at the full picture |
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15:35.10 | jrun | you want the back and forth with the server? |
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15:37.35 | jrun | or the config files? |
15:42.16 | jrun | https://gist.github.com/257/96b5be5df554f0c497a0b388075c7f2f |
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15:45.36 | jrun | config: |
15:45.38 | jrun | http://bpaste.net/show/f253847d7583 |
15:46.34 | jrun | client <--> srv: |
15:46.35 | jrun | Your paste can be seen here: https://gist.github.com/e59e8aab009c0a15255ad1b51de80f63 |
15:47.10 | jrun | sorry, ignore 5 lines above. |
15:48.46 | [TK]D-Fender | Looks like masking on the users, etc |
15:48.56 | [TK]D-Fender | At which point I canèt see or trust typos, etc |
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15:56.13 | jrun | all done by script so typo is out |
15:56.18 | *** join/#asterisk prudolf (~prudolf@ip58865e59.dynamic.kabel-deutschland.de) |
15:56.40 | prudolf | hi there.... maybe a little question: |
15:56.43 | jrun | seems like changing identify_by from auth_username to username stops the Notice msg |
15:59.02 | prudolf | i 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 ) |
15:59.36 | prudolf | the our coworkers see the real number from the extention and not the comapany number |
15:59.41 | prudolf | is it possible? |
16:00.11 | Samot | Are they using a phone connected to the Asterisk box? |
16:00.18 | Samot | Or is this going to their cell number? |
16:00.24 | Samot | cell/mobile.. |
16:01.20 | prudolf | we have VOIP to asterisk (freepbx over easybell). the external coworkers are connected over easybell or over cellnumber |
16:01.32 | prudolf | mobilphone i mean |
16:01.59 | prudolf | the coworker need to know who is calling from the company |
16:02.02 | [TK]D-Fender | ~freepbx |
16:02.02 | infobot | [~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 | ^^^ |
16:02.19 | Samot | If you present your company's number as the caller ID most like the mobile carrier is going to present the CNAM of the number. |
16:03.11 | Samot | And what TK said, go to #freepbx for help with FreePBX |
16:04.17 | prudolf | ok... thank you very much |
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21:08.41 | Gaster | How 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 |
21:08.41 | Gaster | <PROTECTED> |
21:09.50 | Gaster | They say that this error is due to 'domain' misconfiguration, but nothing I did't to the corresponding sip.conf section helped. |
21:10.47 | Gaster | TL;DR - A forwarded SIP registration fails with wrong password, althought the password is 100% correct. |
21:11.57 | Gaster | How do I check that both Asterisk and SIP peer use the same 'domain' for password hashing? |
21:18.04 | Gaster | ATM 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-Fender | Show 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 |
21:24.09 | infobot | extra, 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-Fender | And the config |
21:26.03 | Gaster | I've solved the issue. I set "realm" option in sip.conf to my Asterisk's external NAT IP address. |
21:26.27 | Gaster | Apparently, this is used for password hashing 0_0 |
21:27.22 | Gaster | Why 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-Fender | It's your job to put in the settings for your working environment |
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21:29.21 | Gaster | Yeah, 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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21:30.30 | Gaster | How exactly does "realm" option work? |
21:30.58 | [TK]D-Fender | I've never even used it before myself |
21:31.04 | Gaster | O_O |
21:31.10 | [TK]D-Fender | Never needed to touch it |
21:31.16 | Gaster | So did I |
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22:29.14 | Gaster | The 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 |
22:29.14 | Gaster | sing 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) |
23:03.22 | twitchnln | Greetings 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) |
23:51.29 | equilibrio | hello 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 |
23:51.38 | equilibrio | Anybody could tell me how I can fix this? |
23:51.48 | equilibrio | Everything still works, but I do get this warning |
23:51.51 | voipmonk | Homer? |
23:52.30 | equilibrio | Simpson? |
23:52.36 | equilibrio | :P |
23:52.53 | voipmonk | https://github.com/sipcapture/homer/wiki/Examples%3A-Asterisk |
23:53.38 | equilibrio | hmm to start I can see its not the good subnet configure in there |
23:54.01 | equilibrio | voipmonk: is it critical if I disable this module or it is really needed in general |
23:54.03 | equilibrio | just asking |
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