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| 00:30.11 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:30.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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| 02:36.12 | Katty | Bloop. |
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| 03:54.18 | alpha_ | @all, have a question on asterisk load testing by using sipp |
| 03:54.37 | alpha_ | what is the meaning of "woken up" while doing sipp testing? |
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| 09:01.27 | TSM | I am running a ISDN port in NT PTMP mode, it seems than when the CPE makes a call I do not receive the number they are dialing, i have done pri debug and seems to be missing Processing IE 112 (cs0, Called Party Number) |
| 09:01.40 | TSM | https://pastebin.com/A5U8jJHa |
| 09:01.51 | TSM | any ideas, is this a setup issue on my end? |
| 09:02.02 | TSM | the CPE is a Siemens Gigaset |
| 09:30.14 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 09:30.40 | TSM | this is driving me nuts, tried two BRI CPE devices and its the same |
| 09:36.55 | *** join/#asterisk StucKman (~mdione@195.200.189.206) |
| 09:37.39 | StucKman | is it normal that each time a user initiates a call a storm of status NOTIFYs ensues? |
| 09:38.19 | StucKman | hint: we're using a third party solution called XiVO built on top of * |
| 09:41.18 | StucKman | [May 11 10:39:44] VERBOSE[21996][C-000711af] pbx.c: [May 11 10:39:44] -- Executing [00465150325@default:3] Gosub("SIP/dies9n-000d82eb", "outcall,s,1(1,)") in new stack |
| 09:41.22 | StucKman | errr |
| 09:41.25 | StucKman | sorry about that |
| 09:42.26 | StucKman | [May 11 10:39:44] VERBOSE[21996][C-000711af] pbx.c: [May 11 10:39:44] -- Executing [s@outcall:2] Set("SIP/dies9n-000d82eb", "XIVO_PRESUBR_GLOBAL_NAME=OUTCALL") in new stack |
| 09:42.29 | StucKman | ... |
| 09:45.40 | *** join/#asterisk sotoz (~sotoz@095-097-255-066.static.chello.nl) |
| 09:48.41 | sotoz | hi, I'm trying to use webrtc with asterisk and I made it to kinda work. The only problem I'm having is that audio is not being played in the browser. At webrtc-internals I can see the packets being received, and graphs are being generated as expected. I believe it is a problem with the double-NAT that my computer is behind from, because if I connect my laptop directly to the router it works. I'm also trying onsip.com and their server is |
| 09:48.41 | sotoz | <PROTECTED> |
| 09:50.49 | sotoz | any ideas? |
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| 10:24.59 | TSM | anyone running BRI in NT mode? |
| 10:34.03 | StucKman | are there any tools to detangle call logs from the full log? I'm tired of raisin'ing my eyes reading lines |
| 10:36.08 | TSM | cdr |
| 10:36.15 | TSM | it can output to DB |
| 10:36.52 | StucKman | link? |
| 10:38.54 | StucKman | this? http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-SysAdmin-SECT-1.html |
| 10:39.20 | StucKman | I want to debug a call through the logs of a live system with 250+ users |
| 10:39.27 | StucKman | (this is a call center) |
| 10:42.23 | file | define "call logs", are you referring to log messages relating to a specific call? |
| 10:43.58 | file | if so https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging can help you grep, provided you first locate the call identifier |
| 10:44.20 | alpha_ | hello, what is the meaning of "woken up" while doing sipp testing? while doing asterisk load testing? |
| 10:48.07 | StucKman | file: not really useful, I miss all the SIP messages because those are not prefixed with the call id |
| 10:48.57 | file | then you'd need to determine the sip call-id and find based on that |
| 10:49.32 | StucKman | file: I noticed that Call-ID headers in the SIP messages changes a lot |
| 10:49.44 | file | it depends on what you are trying to debug |
| 10:49.54 | file | a call will have the same call-id throughout |
| 10:50.19 | file | unless you define a call as two legs - in which case each would have different ones |
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| 10:50.53 | StucKman | two legs? |
| 10:51.10 | StucKman | you mean the logs for the caller and the ones for the callee? |
| 10:51.11 | file | an incoming and outgoing side |
| 10:51.13 | file | yes. |
| 10:51.34 | StucKman | then I don't know what my third party solution is doing |
| 10:51.55 | StucKman | becase the Call-ID fields change |
| 10:52.11 | StucKman | I'm calling from ina Extension to an external number |
| 10:52.36 | StucKman | <PROTECTED> |
| 10:52.54 | file | the caller would have one Call-ID, the callee would have one Call-ID, and that changes for every unique call |
| 10:52.57 | StucKman | but then the number is changed to our external number and I get messages for |
| 10:53.03 | StucKman | <PROTECTED> |
| 10:53.20 | StucKman | I still got more ID related to this call |
| 10:53.33 | file | yes that's normal, each side has its own Call-ID because each are totally separate outside of Asterisk |
| 10:54.07 | StucKman | hmm, the From change too |
| 10:54.25 | file | yes, that are two completely separate calls from a SIP perspective |
| 10:54.27 | file | er they |
| 10:54.38 | StucKman | hmm, I think the third party thing is doing funny things |
| 10:55.00 | StucKman | <PROTECTED> |
| 10:55.00 | StucKman | <PROTECTED> |
| 10:55.01 | StucKman | and |
| 10:55.06 | StucKman | <PROTECTED> |
| 10:55.06 | StucKman | <PROTECTED> |
| 10:55.10 | file | all normal. |
| 10:55.28 | StucKman | it's like I get some NAT'ing there |
| 10:56.17 | StucKman | so that's from my terminal to my sip server, and then from the sip server to the peer? |
| 10:56.25 | file | yes. |
| 10:56.30 | StucKman | all right |
| 10:56.57 | StucKman | so for a call that comes back to me I'll get 4 different from/to/call-id sets |
| 10:57.10 | StucKman | to me==to another user in my sip server |
| 10:57.47 | file | if you mean you call to a provider and they send it back to you... it depends |
| 10:57.50 | file | maybe yes, maybe no |
| 10:58.00 | file | depends on how their system operates |
| 10:58.19 | StucKman | I'm calling a number that's associates to our A* |
| 10:58.35 | file | like I said, maybe yes maybe no |
| 10:58.56 | StucKman | oks |
| 10:59.13 | file | returns to lurking |
| 10:59.25 | StucKman | it's weird noone ever tried to this before and published anything |
| 10:59.45 | file | tried what, and what would they publish? |
| 10:59.59 | StucKman | make a tool to follow in such a way a call |
| 11:00.05 | StucKman | and publish the tool |
| 11:00.15 | StucKman | 'untangle the logs' |
| 11:00.39 | file | probably because it's hard to do so from a logs perspective |
| 11:00.59 | StucKman | I'll keep tryinganyways |
| 11:01.03 | StucKman | thanks for the info |
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| 11:46.36 | TSM | can you have ISDN lines give a dialtone before dialing? |
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| 12:29.49 | sekil | TSM: I think the analog adapter on the NT can provide dialtone |
| 12:29.54 | sekil | TSM: at least here it does |
| 12:33.52 | pawiecki | StucKman: If you were referring to the Asterisk full log - you can grep by the string, that is given to the call, for example "[Mar 10 09:07:46] VERBOSE[31191][C-00000c9d] pbx.c:" - grep for the "C-00000c9d". Also try tcpdump / wireshark. |
| 12:39.26 | TSM | sekil: what analoge adapter, the NT is a Sangoma A500 card and the TE end is a Siemens ISDN telephone, but from what I can see these devices do not transmit the called number on the D-chan |
| 12:40.11 | sekil | TSM: I was thinking about BRI |
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| 12:42.06 | TSM | this is BRI which is ISDN2 |
| 12:42.47 | sekil | TSM: you're in Europe? |
| 12:42.52 | TSM | yup |
| 12:42.54 | sekil | TSM: or in ! US |
| 12:43.09 | TSM | in US its normal for PRI and BRI to give dial tone from the telco I know |
| 12:43.28 | TSM | normally in europe we deal with it on the D channel |
| 12:43.43 | sekil | TSM: well in Europe we have NT adapters to terminate U bus |
| 12:43.53 | sekil | TSM: and to connect to end user equipment |
| 12:44.12 | sekil | TSM: they also have so called AB ports to connect to say analog phones |
| 12:44.30 | sekil | TSM: and it can provide dialtone on those ports |
| 12:44.32 | TSM | I have no analogue phones, these are ISDN phones, I am in germany |
| 12:44.39 | sekil | TSM: yeah .. |
| 12:44.51 | sekil | TSM: well I was talking about that setup |
| 12:44.54 | sekil | TSM: not yours |
| 12:45.07 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:45.45 | TSM | maybe that sangoma A500 cant simulate that |
| 12:46.08 | sekil | TSM: btw...ISDN phone is not sending called party ? |
| 12:46.18 | sekil | TSM: that sounds broken |
| 12:47.16 | TSM | nop ive checked debug output, it worked when it was connected directly to telco |
| 12:47.44 | TSM | I wonder if they have some hardcoded firmware that waits for dialtone then sends DTMF |
| 12:47.49 | StucKman | pawiecki: as I said, grep is not enough, it can't match the SIP messages, or follow when calls are NAT'ed (I don't know what the right term for that) |
| 12:47.57 | *** join/#asterisk gregs (sid160074@gateway/web/irccloud.com/x-btwkvrxplukuscmg) |
| 12:48.52 | TSM | odd it all works when I am going from sangoma A500 in TE mode to the telco though, weird |
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| 12:49.44 | sekil | TSM: so you connect the card to NT adapter in TE mode? |
| 12:50.01 | sekil | TSM: where do you connect the ISDN phone..to the other S port? |
| 12:52.22 | TSM | the card can do either NT or TE, just turn the internal module around, in TE mode they are directly connected to our telco ISDN S0 bus, in NT mode you should be able to connect it directly to CPE |
| 12:54.31 | sekil | TSM: yes...one port should be TE to one S0 port |
| 12:55.01 | sekil | TSM: the other S0 port can be used to connect other gear...also on the different port on the card you could enable NT to the phone |
| 12:57.02 | TSM | port 1/2 is NT and port 3/4 are TE mode, the TE ones work fine |
| 12:57.56 | sekil | TSM: so what's the issue? |
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| 12:58.28 | sekil | TSM: when phone is calling something..you're not seeing signalling in asterisk? |
| 12:59.10 | TSM | the siemens phones do not seem to transmit calling number, asterisk then just dumps them in as 's' if context is from-internal, if set to from-pstn then ends up being caught by our incoming route |
| 12:59.43 | TSM | it seems the siemens is waiting for line to be setup before it will dial the number which comes back to my point about dialtone |
| 13:03.39 | pawiecki | StucKman: depending on your architecture, you can try to capture packets on both sides (or all the way) and compare them in Wireshark. I don't know of any better way to do this. |
| 13:05.11 | sekil | TSM: that makes sense only in overlap dialing |
| 13:05.56 | StucKman | pawiecki: that doesn't show anything doen by the dial plan |
| 13:06.01 | StucKman | don |
| 13:07.11 | StucKman | basically what I want is: something that can parse the full log and give me the different sets of sip messages and dial plan execution trace of a call |
| 13:07.27 | *** join/#asterisk scgm11_ (~scgm11@r186-52-212-1.dialup.adsl.anteldata.net.uy) |
| 13:07.32 | StucKman | and any other message emited by A* |
| 13:07.56 | StucKman | I more or less can do that by hand, I'll try to automatize it |
| 13:08.18 | StucKman | nut i still hadn't finished finding all the data related to a call |
| 13:10.50 | TSM | sekil: it may just be the way things work here, it do find it odd, have an open ticket with sangoma about it |
| 13:11.29 | sekil | TSM: did you check out q931 traces? |
| 13:14.56 | *** join/#asterisk Eloy (~Eloy@5.149.168.66) |
| 13:15.13 | TSM | sekil: if you mean 'pri set debug on span 1' then yes |
| 13:15.42 | pawiecki | I don't know of any automated way of doing this. I personally mostly depend on 3 things to trace and debug a call: 1. tcpdump and Wireshark to analyze SIP flow of a call. 2. Depending on traffic - Asterisk live CLI or full log. 3. CDR database records. Putting it all together you can trace almost every aspect of a call, but probably there are more experienced people, who may have more effective approach/tools. |
| 13:15.51 | pawiecki | StucKman: ^ |
| 13:19.10 | StucKman | pawiecki: my problems are mostly related to a hellish dial plan with no comments and my lack of knowledge of SIP |
| 13:19.55 | TSM | sekil: you mentioned overlapdial, enabled it on the related spans and presto it now works |
| 13:20.25 | sekil | TSM: the phone is probably sending digits overlap |
| 13:20.40 | sekil | TSM: also it should allow some dialtone to the phone |
| 13:21.36 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:21.39 | TSM | yup it does not |
| 13:21.43 | TSM | now |
| 13:21.58 | sekil | TSM: although on Siemens Gigaset phones you can use overlap and enbloc on analog |
| 13:24.06 | TSM | there is no settings for anything, not sure what you saying on analog, this dect base has no analog ports |
| 13:24.56 | pawiecki | StucKman: SIP is a simple thing in it's core. I recommend this article and many more posts on this blog: https://andrewjprokop.wordpress.com/2014/01/16/how-to-debug-sip/ |
| 13:24.59 | sekil | TSM: I'm saying on DECT with analog port |
| 13:25.12 | sekil | TSM: base with analog port |
| 13:26.05 | sekil | TSM: you can press the button ...you get dialtone back from the pstn...then you press keys sent as DTMF |
| 13:26.18 | pawiecki | StucKman: as for the dialplan, try this: https://wiki.asterisk.org/wiki/display/AST/Dialplan |
| 13:26.20 | sekil | TSM: also you can type all keys in the number..and then press button |
| 13:26.32 | sekil | TSM: it will send them after a second or two.. |
| 13:26.42 | sekil | TSM: enbloc |
| 13:27.03 | sekil | TSM: but on ISDN it should be a digital message |
| 13:27.52 | StucKman | pawiecki: I just want a text file with all the log lines related to a call |
| 13:27.53 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 13:28.09 | StucKman | afterwards I can read them more or less without problem |
| 13:28.09 | TSM | I would have thought but not, it does not pickup the 'line' when you press green, it wants you to put number in, I know what you are saying as I have used pure analoge ones but these no, it wants you to put number in first, i guess its a firmware thing |
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| 13:29.40 | sekil | TSM: there should be no line to pickup on ISDN per se |
| 13:29.46 | sekil | TSM: as you put it |
| 13:30.02 | sekil | TSM: or it should not matter |
| 13:30.29 | sekil | TSM: does it work enbloc? |
| 13:35.14 | TSM | it works with overlapdial=1 |
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| 14:13.36 | acidfoo-_ | does there was a way on Asterisk 1.8 to force the sip channel to re-do SRV lookup ? It seems that once it did it once, if the IPs are changed on the dns server... the old IP resolved the first time it did the SRV record is always used. |
| 14:13.39 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:13.46 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:17.31 | Samot | Yes, you restart it. |
| 14:17.36 | Samot | It does the lookup on start up |
| 14:17.44 | acidfoo-_ | ok ! |
| 14:17.46 | Samot | And every so often, I can't remember the time frame. |
| 14:18.02 | acidfoo-_ | mmm what do you mean by "and every so often" ? :) |
| 14:18.05 | acidfoo-_ | that part interest me |
| 14:18.14 | acidfoo-_ | that periodically it should do the lookup again ? |
| 14:18.30 | Samot | There is a time interval in which DNS lookups are done and stored into memory. |
| 14:18.43 | Samot | I can't recall off the top of my head what that is... |
| 14:18.57 | Samot | But a restart will make it look all FQDNs up again. |
| 14:20.59 | acidfoo-_ | good thanks |
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| 14:21.04 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:22.17 | sekil | wasn't there a dnsmngr.conf to set the timers? |
| 14:22.25 | mehale | hi folks, on rasterisk, how can I check why registration is failing? |
| 14:23.26 | [TK]D-Fender | look at the SIP debutg |
| 14:23.28 | [TK]D-Fender | debug* |
| 14:29.21 | acidfoo-_ | sekil: ! nice |
| 14:36.31 | TSM | would ther be any reason for chan_pjsip not being able to register to another server that has chan_sip? |
| 14:38.46 | Samot | Misconfiguration of settings. |
| 14:38.48 | TSM | one server is on a dynamic IP so i have set its trunk to send registration and authentication outbound, other side has a static IP and has the username/password but it does not register, always says wrong password |
| 14:38.49 | Samot | Network issues |
| 14:39.27 | Samot | Show a call with a debug of this. |
| 14:39.28 | Samot | ~pb |
| 14:39.29 | infobot | well, 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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| 14:57.55 | *** mode/#asterisk [+o rmudgett] by ChanServ |
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| 15:05.00 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
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| 15:15.55 | Samot | TSM? |
| 15:19.36 | igcewieling1 | The Super Machine! |
| 15:25.16 | *** join/#asterisk robinak (~quassel@unaffilated/robink) |
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| 16:24.21 | Samot | Well I guess TSM didn't really want help with his problem. |
| 16:24.28 | Samot | or their problem. |
| 16:25.40 | TSM | the call will not show anything as the trunk is not registerd |
| 16:26.09 | TSM | it currently goes straight to no lines available |
| 16:26.24 | Samot | I asked for a debug of a call. |
| 16:26.29 | Samot | Two hours ago. |
| 16:26.53 | Samot | Registration has nothing to do with outbound calls. |
| 16:27.39 | Samot | You auth outbound calls. |
| 16:28.03 | Samot | So PBX A will send an INVITE and PBX B should send back a 401 Unauthorized challenge. |
| 16:28.13 | Samot | Show a call. |
| 16:28.24 | *** join/#asterisk matrix1233 (~matrix123@41.230.61.140) |
| 16:29.26 | TSM | https://pastebin.com/gmELGgQe |
| 16:29.52 | TSM | without registration how will the PBX on the dynamic IP ever receive calls? |
| 16:30.08 | Samot | No debug details at all. |
| 16:30.16 | TSM | which debug level |
| 16:30.28 | Samot | pjsip set logger on |
| 16:31.12 | Samot | Actually... |
| 16:31.27 | Samot | Is this from the PBX making the call? |
| 16:31.52 | Samot | And the other PBX is the one that has to register? |
| 16:32.17 | [TK]D-Fender | This system didnt even TRY |
| 16:32.24 | Samot | no |
| 16:32.29 | Samot | at all |
| 16:32.34 | [TK]D-Fender | There is no Dial |
| 16:32.39 | [TK]D-Fender | it CHOSE not to try |
| 16:32.48 | [TK]D-Fender | This is a config issue |
| 16:32.50 | Samot | But I asked for an actual debug two hours ago. |
| 16:32.56 | Samot | So problem #1, not listening. |
| 16:33.12 | Samot | 10:36:32 AM <TSM> would ther be any reason for chan_pjsip not being able to register to another server that has chan_sip? |
| 16:33.12 | Samot | 10:38:47 AM <Samot> Misconfiguration of settings. |
| 16:33.14 | TSM | problem #1 is that other people in the office took me away from what I was doing |
| 16:33.23 | [TK]D-Fender | Go get it now then |
| 16:33.30 | TSM | I can get the trace from either side, one is pjsip the other its going to is just chan_sip |
| 16:33.51 | [TK]D-Fender | BOTH sides |
| 16:33.53 | Samot | Is the Chan_SIP PBX the one that has to register? |
| 16:34.17 | Samot | Which isn't registered right now? |
| 16:34.21 | TSM | no chan_pjsip is the dynamic one that I think should register |
| 16:34.24 | *** join/#asterisk matrix1233 (~matrix123@41.230.61.140) |
| 16:34.51 | Samot | Then registration from PJSIP based PBX to Chan_SIP based PBX is not involve. |
| 16:34.53 | Samot | Then registration from PJSIP based PBX to Chan_SIP based PBX is not involved. |
| 16:35.02 | TSM | lets say PBXB is dynamic and PBXA is the static IP one, PBXB is trying to register to PBXA |
| 16:35.16 | Samot | But PBXA is the one making the call? |
| 16:35.21 | Samot | Or PBXB? |
| 16:36.02 | TSM | PBXB is making the call to PBXA, i have tried the other way and its not working either |
| 16:36.09 | Samot | OK |
| 16:36.11 | [TK]D-Fender | <TSM> no chan_pjsip is the dynamic one that I think should register <- fine |
| 16:36.12 | Samot | So.. |
| 16:36.27 | [TK]D-Fender | last call attempt shown didn't even try to dial the other side |
| 16:37.21 | Samot | Well this officially has to be moved to #freepbx. |
| 16:37.23 | Samot | -- Executing [2250@from-internal:1] Macro("PJSIP/2808-0000003b", "user-callerid,LIMIT,EXTERNAL,") in new stack |
| 16:37.35 | Samot | Because now we need to see screenshots of the GUI. |
| 16:37.41 | Samot | And there's no GUI support in here. |
| 16:39.56 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:40.02 | TSM | https://pastebin.com/V40XqJu1 |
| 16:40.20 | TSM | the debug is littered with sip messages but that section if just around the call |
| 16:40.56 | TSM | [2017-05-11 17:21:49] NOTICE[2764] chan_sip.c: Registration from '<sip:WENNDE@voip1.wenn.com>' failed for '90.187.10.93:15230' - Wrong password |
| 16:41.17 | TSM | when I reload PBXB that is the message I see on PBXA when it tries to register |
| 16:42.23 | [TK]D-Fender | Forget that message |
| 16:42.41 | [TK]D-Fender | it did not even TRY to dial the other side |
| 16:43.22 | [TK]D-Fender | Your route and/or trunk is screwed up. I am NOT talking about the SIP part |
| 16:43.26 | TSM | let me look at routes |
| 16:43.35 | [TK]D-Fender | the call hanlding and selection process decided not to even try |
| 16:44.30 | *** join/#asterisk newtonr_ (~newtonr@173.21.147.197) |
| 16:44.30 | *** mode/#asterisk [+o newtonr_] by ChanServ |
| 16:44.34 | [TK]D-Fender | Uncheck that "intra-company" flag on it |
| 16:46.05 | TSM | https://pastebin.com/A3ZQVRhi now that route has been corrected, as i removed trunk before and created it I forgot it to setup route again |
| 16:46.51 | TSM | taken intracompany off but still the same |
| 16:47.09 | [TK]D-Fender | no , that is VERY different |
| 16:47.13 | [TK]D-Fender | it TRIED to call |
| 16:47.20 | [TK]D-Fender | <PROTECTED> |
| 16:47.20 | [TK]D-Fender | <PROTECTED> |
| 16:47.21 | [TK]D-Fender | <PROTECTED> |
| 16:47.32 | [TK]D-Fender | and I am not seeing debug for that attempt |
| 16:47.41 | [TK]D-Fender | Where's the SIP DEBUG? |
| 16:48.07 | TSM | which side |
| 16:48.56 | [TK]D-Fender | THIS ONE |
| 16:48.59 | [TK]D-Fender | you just showed a call |
| 16:49.12 | [TK]D-Fender | are you moving the damn target on us? |
| 16:50.01 | [TK]D-Fender | Stick to one end at a time. |
| 16:50.15 | TSM | https://pastebin.com/MhH1W0zH |
| 16:50.42 | TSM | that should have the bits that show the other side responding forbidden |
| 16:51.03 | [TK]D-Fender | <--- Transmitting SIP request (1146 bytes) to UDP:37.157.37.230:5060 ---> |
| 16:51.03 | [TK]D-Fender | INVITE sip:2251@voip1.wenn.com:5060 SIP/2.0 |
| 16:51.12 | [TK]D-Fender | you're calling a WAN IP |
| 16:51.22 | [TK]D-Fender | Contact: <sip:asterisk@192.168.4.10:5060> |
| 16:51.32 | [TK]D-Fender | And sending a PRIVATE IP as the contact |
| 16:51.49 | [TK]D-Fender | if this system is DYNAMIC, PJSIP does not look up IP's dynamically |
| 16:52.03 | [TK]D-Fender | and this will cause issues at best, and likely fail |
| 16:52.09 | TSM | let me fix the IP for now |
| 16:52.28 | [TK]D-Fender | SIP/2.0 403 Forbidden |
| 16:52.35 | [TK]D-Fender | But at the very least it IS trying to call |
| 16:52.48 | [TK]D-Fender | Show the OTHER side for that call |
| 16:53.35 | TSM | ive set the external IP to what it is currently and its made no difference, ile get a trace from other side |
| 16:54.42 | *** join/#asterisk newtonr (~newtonr@173-21-147-197.client.mchsi.com) |
| 16:54.42 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:54.54 | TSM | https://pastebin.com/G6KfTiGd |
| 16:55.20 | TSM | this line stands out but not sure what it means No matching peer for '2808' from '90.187.10.93:15230' |
| 16:55.26 | [TK]D-Fender | No matching peer for '2808' from '90.187.10.93:15230' |
| 16:55.35 | [TK]D-Fender | Mean "I don't know who this caller is" |
| 16:55.40 | [TK]D-Fender | it doesn't match a trunk in way |
| 16:55.47 | [TK]D-Fender | Show us the trunk you made on this side |
| 16:56.43 | TSM | https://pastebin.com/pzwvQdPV |
| 16:57.13 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:57.31 | [TK]D-Fender | "sip show peers" |
| 16:57.34 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 16:57.48 | *** part/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 16:58.23 | TSM | de2uk-trunk/WENNDE (Unspecified) D Yes Yes 0 UNKNOWN |
| 16:59.15 | TSM | is the registration not meant to join up the peer with its current IP? |
| 17:01.46 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 17:01.48 | [TK]D-Fender | other side didn't register. Names didn't match. |
| 17:01.52 | [TK]D-Fender | there is no match for this call |
| 17:02.28 | [TK]D-Fender | several wrong setting on the chan_sip side for this, and on the pjsip on as well. |
| 17:02.46 | [TK]D-Fender | in addition to not being regitered like you're supposed to have for a dynamic peer |
| 17:03.37 | *** join/#asterisk newtonr (~newtonr@173.21.147.197) |
| 17:03.37 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:04.04 | TSM | on the pjsip side, i just use the FPBX gui to create the trunk, enter username/secret, sip server hostname, set to outbound authentication and registration send |
| 17:04.31 | TSM | on the sip side, should I be using PEER details or USER Details? |
| 17:05.30 | [TK]D-Fender | PEER. type=peer, "directmedia=no", canreinvite should never be used. that's old crap. Also you should be sending the CID in RPID, not as the FROM on the sending side |
| 17:05.55 | [TK]D-Fender | and should explicitly have "nat=no" for them since they should be assumed as being honest with what they offer |
| 17:05.56 | igcewieling1 | sounds like someone has been reading voip-info.org |
| 17:06.21 | [TK]D-Fender | username is ALSO gone and should be written as defaultuser |
| 17:06.41 | [TK]D-Fender | And the PJSIP being the dynamic side is another major mistake |
| 17:06.50 | [TK]D-Fender | I recommend going chan_sip on both ends for this |
| 17:07.07 | [TK]D-Fender | And actually using valid parameters |
| 17:08.39 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:08.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:09.30 | *** join/#asterisk newtonr_ (~newtonr@173-21-147-197.client.mchsi.com) |
| 17:09.30 | *** mode/#asterisk [+o newtonr_] by ChanServ |
| 17:10.28 | TSM | switched to sip on both sides |
| 17:14.12 | [TK]D-Fender | both peers should also be "sendrpid=yes", "trustrpid=yes", "fromuser=WENNDE" |
| 17:17.52 | *** join/#asterisk jkroon (~jkroon@197.96.224.13) |
| 17:23.05 | TSM | seems to work better, calls now go though but on the other side responds back that number cannot-complete-as-dialed, dont hear it, just sounds like its ringing then says all circuits are busy |
| 17:23.20 | TSM | ile have to deal with this another time, need to leave office |
| 17:23.22 | TSM | pain |
| 17:25.51 | TSM | i may be back later |
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| 21:50.48 | TSM | I am trying to register a trunk on one asterisk server onto another but am having issues with calls, I see called ID as 2271%40de2uk-trunk@voip1.wenn.com, it seems to be putting my trunk name into it but not sure why |
| 21:51.22 | *** join/#asterisk matrix1233 (~matrix123@41.230.61.140) |
| 21:52.55 | igcewieling1 | trunk names should not have spaces. |
| 21:53.27 | TSM | its not a space |
| 21:53.54 | eric_hill | @ sign? |
| 21:54.00 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 21:54.14 | igcewieling1 | I just checked, yup, that looks like a stray @ |
| 21:54.38 | igcewieling1 | do you have @ in your trunk name or username or anything like that? |
| 21:54.47 | Samot | I don't understand why this is so hard. |
| 21:54.50 | Samot | Also, FreePBX |
| 21:54.57 | TSM | no, that is what I see when I look at the SIP debug |
| 21:55.03 | Samot | I told you to take this to FreePBX so we can look at your GUI and the settings. |
| 21:55.29 | igcewieling1 | Oh freepbx? I need to remember to ask first. You're on your on your own. Perhaps #FreePBX can be of more help. |
| 21:56.18 | Samot | This is a configuration issue. |
| 21:56.48 | Samot | But since the GUI controls the configuration in FreePBX, this is not the channel to be asking in. |
| 21:57.10 | Samot | Because all the answers will be about editing files directly that the GUI will overwrite |
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| 23:46.53 | KNERD | I am getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) and I have no idea why. If none of the modules load, then I can connect to Asterisk console . Here is output to asterisk -cvvvvv http://pastebin.ca/3811253 |
| 23:49.21 | Samot | Well, does it exist? |
| 23:49.32 | KNERD | Of course it does |
| 23:49.38 | Samot | You confirmed? |
| 23:49.42 | KNERD | of course |
| 23:49.50 | Samot | Asterisk running under the right user? |
| 23:49.53 | KNERD | if it was that simple, I would not be asking here :-) |
| 23:50.04 | Samot | No. |
| 23:50.06 | KNERD | this machien been runnign nearly a year |
| 23:50.59 | Samot | You mean the Asterisk service or just it's uptime? |
| 23:51.39 | KNERD | the whole PBX |
| 23:52.02 | Samot | So Asterisk is running |
| 23:52.10 | Samot | But you can't do asterisk -r |
| 23:52.13 | KNERD | it is , butis isn't |
| 23:52.26 | Samot | ps -ef | grep asterisk |
| 23:52.28 | Samot | ~pb |
| 23:52.29 | infobot | somebody said pastebin was 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. |
| 23:52.29 | KNERD | service shows it is runnung, but I cannot connect |
| 23:53.54 | KNERD | http://pastebin.ca/3811258 |
| 23:54.31 | Samot | How was Asterisk started? |
| 23:55.08 | KNERD | it starts as a service when the machien starts, but I have been doing "service asterisk start|stop..." |
| 23:55.22 | Samot | And this is FreePBX? |
| 23:55.31 | KNERD | no |
| 23:55.40 | KNERD | wel..FreePBX was put on it |
| 23:55.46 | Samot | OK. |
| 23:55.54 | Samot | Which means... |
| 23:56.07 | Samot | Asterisk should be running under the user "asterisk" |
| 23:56.12 | Samot | root 9534 1 0 16:45 pts/0 00:00:00 /bin/sh /usr/sbin/safe_asterisk |
| 23:56.16 | Samot | ^^^ It's not. |
| 23:56.41 | KNERD | let me restart the machine and see what happenes |
| 23:56.45 | Samot | root 9797 1 0 May06 ? 00:00:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 23:56.45 | Samot | asterisk 9799 9797 0 May06 ? 00:34:18 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 23:56.53 | Samot | That's what you should see. |
| 23:56.59 | Samot | Asterisk should not start on it's own |
| 23:57.13 | Samot | FreePBX _needs_ to start it, it the proper way. |
| 23:57.28 | Samot | "fwconsole start" <-- that's the proper way in FreePBX |
| 23:58.57 | KNERD | Same result http://pastebin.ca/3811259 |
| 23:59.27 | Samot | fwconsole stop |
| 23:59.41 | KNERD | okay dine |
| 23:59.43 | Samot | killall -9 asterisk |