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| 03:16.21 | Reinhilde | Pathetic crank call, unintelligible music. +1303 219 39 09 |
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| 09:31.53 | cusco | hey... just configured pgsql as a cdr backend and I notce that asterisk is writting 2 rows per call |
| 09:31.58 | cusco | with the same linkedid... |
| 09:32.33 | cusco | in the dialplan, we're using: Set(CHANNEL(hangup_handler_push)=.... |
| 09:33.01 | cusco | so it runs another piece of dialplan in the end - I believe this should not cause another CDR line, right? |
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| 09:39.39 | velix | One of my peers is "UNREACHABLE". Can I restart it somehow to make it re-register? |
| 09:47.46 | velix | What does this means "Peer 'student01' is now UNREACHABLE!" ? |
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| 09:52.53 | velix | Oh no, nobody in here :-( |
| 09:53.37 | velix | It works from my normal SIP phone, but not from Asterisk. |
| 09:53.44 | velix | Anyone with an idea, when Kobaz might come back today? |
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| 09:55.20 | velix | asterisk really lacks of debug informations. |
| 09:55.39 | velix | "sent to invalid extension but no invalid handler: context,exten,priority=student01-out" |
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| 10:01.20 | velix | Anyone with a second for me please? |
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| 10:08.25 | velix | It's just take a few minutes :( |
| 10:15.04 | velix | it's totally fine for me to pay for help, but there is no help |
| 10:16.29 | velix | This really is nonsense: " sent to invalid extension but no invalid handler: context,exten,priority=student01-out" |
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| 10:20.01 | velix | Damn, since Friday I'm working on this and nothing works at all. |
| 10:20.11 | velix | 90% of the tutorials about Asterisk as outdated or wrong. |
| 10:21.18 | velix | Why is Asterisk such a magic the community wants to keep? |
| 10:21.30 | velix | Is it straight out of hell? |
| 10:26.47 | file | well, it's early morning in North America so I wouldn't expect many people from that region to be present and with the current state of the world I'd expect people to be extra busy |
| 10:27.02 | file | generally the community forums are more active since people can reply at their leisure |
| 10:27.33 | file | and the message states what is going on, you sent a call to an invalid extension |
| 10:28.17 | file | if using Goto then the wiki has a page talking about the arguments along with an example showing the different results https://wiki.asterisk.org/wiki/display/AST/Goto+Application+and+Priority+Labels |
| 10:29.26 | *** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl) |
| 10:34.15 | velix | file: Goto? |
| 10:34.22 | velix | I'm not using Goto :-) |
| 10:34.26 | file | then what are you using? |
| 10:34.41 | file | something sent the call to an invalid place |
| 10:34.56 | velix | file: Do you have some seconds to have a look over my config? |
| 10:35.12 | file | if you post the console output of a call attempt then it will show what is going on. |
| 10:36.00 | velix | Okay, let me get the console output. In the meanwhile, this is my complete ocnfig with the command at bottom: https://bpaste.net/6GYQ |
| 10:37.38 | velix | And this is the error from console: https://bpaste.net/raw/IFLA |
| 10:38.18 | file | the extension you've provided doesn't match an extension in the given dialplan context |
| 10:38.52 | velix | file: hmm, the number looked like 0049xxxxxxxx normally this should work? |
| 10:38.56 | file | "dialplan show <extension>@student01-out" will show you what matches if the extension is tried |
| 10:39.08 | file | and "dialplan show student01-out" shows you the complete extensions for the context |
| 10:39.28 | velix | "There is no existence of 5020@student01-out extension" :D |
| 10:39.42 | file | for example, in your call file... |
| 10:39.49 | file | "00049xxxxxxxxxxxxx" |
| 10:39.54 | file | you have three zeroes at the front |
| 10:40.09 | file | in the dialplan, you have an entry for two zeroes at the front |
| 10:40.15 | velix | file: Yeah, this was a bad example, I've tried different things. |
| 10:40.23 | velix | The last one was 00049xxxxxx ;) |
| 10:40.58 | velix | But even with a valid number, I'm getting "There is no existence of 5020@student01-out extension" |
| 10:41.04 | velix | That doesn't sound good |
| 10:41.06 | file | that's not a valid number |
| 10:41.09 | file | in your context. |
| 10:41.35 | file | student01-out does not include the "default" context so 5020 would not be valid |
| 10:42.11 | velix | Interesting. [student01-out]\ninclude => default ? |
| 10:42.21 | file | yes, then it would become valid. |
| 10:43.15 | velix | Wow, 1 sec. |
| 10:44.56 | velix | Nice, Asterisk now tells me, I've called an invalid number. |
| 10:46.18 | velix | It redirects me to the internal demo |
| 10:46.29 | velix | ast_streamfile failed on SIP/5020-00000007 for demo-instruct |
| 10:47.12 | file | then use the tools at hand to understand why |
| 10:47.33 | file | for example if you've installed the sample config files, then perhaps pbx_ael loaded its sample configuration and is part of your dialplan |
| 10:47.59 | file | so you'd use "dialplan show" to see what the actual flow is and where it came from |
| 10:48.15 | velix | file: I've used "Sipgate Basic" up to now with the same backup and it worked out of the box. I really wonder, why it stopped working after changing the SIP account. |
| 10:48.34 | file | I have no idea, Asterisk ultimately does what it is told to do so presumably configuration related |
| 10:54.50 | velix | file: Okay, I've removed all the preconfigured configurations now. |
| 10:55.47 | velix | Now I'm getting this again: "[Mar 17 10:55:00] WARNING[29646][C-00000004]: pbx.c:4481 __ast_pbx_run: Channel 'SIP/5020-00000003' sent to invalid extension but no invalid handler: context,exten,priority=student01-out," |
| 10:58.41 | velix | That makes no sense, really. |
| 11:01.29 | cusco | I can't figure why cdr is writting two records per call :/ |
| 11:01.32 | file | if it isn't matching an extension, then it makes perfect sense |
| 11:03.42 | velix | file: But you made it match an extension some seconds ago :) |
| 11:04.01 | file | sure, but I don't know what the current state of your dialplan configuration is or what you may have changed |
| 11:04.18 | file | so use "dialplan show" to see what the configuration actually is |
| 11:04.18 | velix | file: Let me try, if I can export this somehow |
| 11:04.53 | file | I'm trying to get you to the point where you can figure it out yourself |
| 11:05.41 | velix | file: I understand this, but I _really_ can't see the problem. I don't want to let you work for me, but I'm really blind on this. |
| 11:05.55 | file | do you have an understanding of dialplan contexts/extensions/pattern matching? |
| 11:06.33 | file | and you don't need to see the problem immediately, you just need to break down the problem |
| 11:06.47 | velix | file: Actually I thought so. After doing many readings, I've set it up with Sipgate Basic account and it worked as expected. Then I got SIP accounts from University to setup the users. I didn't change anything in my extensions. |
| 11:07.00 | file | if the assumption is "this extension should match" then you use the tools at hand to figure out why that assumption is not true |
| 11:09.02 | velix | file: That's my current dial plan. Okay, seems like 5020 still isn't matching. https://bpaste.net/ITRA |
| 11:09.08 | velix | Let me look up the Asterisk book |
| 11:10.01 | file | so what does dialplan show 5020@student01-out show? |
| 11:10.52 | velix | '5020' => 1. Dial(SIP/5020) [extensions.conf:3] |
| 11:11.08 | file | okay, and what did you do in the call file... |
| 11:12.11 | velix | https://bpaste.net/BHAA |
| 11:12.38 | file | so, you weren't trying to dial 5020 at all? |
| 11:13.02 | velix | ah, got ya. |
| 11:13.15 | velix | Actually, I do. |
| 11:13.31 | velix | I want to call 5020 and when he receives the call, the external number should be dialed. |
| 11:13.35 | velix | Click-By-Call ? |
| 11:13.38 | velix | click to call? |
| 11:13.39 | file | ok |
| 11:13.47 | velix | call by click? |
| 11:14.04 | file | your dialplan for matching that still won't work though |
| 11:14.14 | velix | file: Would _this_ help? same => 5020,Dial(SIP/${EXTEN}@student01,,rWT) |
| 11:14.38 | file | this is giving me a headache |
| 11:15.04 | file | no, because your problem isn't that |
| 11:15.08 | *** join/#asterisk electronic_eel (~quassel@HSI-KBW-46-223-65-185.hsi.kabel-badenwuerttemberg.de) |
| 11:15.13 | file | the outgoing call is being made and answered |
| 11:15.31 | file | upon answer it is being sent into the dialplan, however your dialplan is not correct for these 0049 things |
| 11:15.45 | velix | I can remove it of course. |
| 11:15.59 | file | Goto(0${EXTEN:4},1) that means - Strip the first 4 digits off the extension, prefix a 0, and goto that extension |
| 11:16.06 | file | according to your dialplan there is no matching extension for that |
| 11:16.30 | file | the other extensions are similar |
| 11:16.30 | velix | I need to prefix a "0" to get outside university. |
| 11:18.54 | velix | I've strippd it down to this for testing: "exten => 5020,1,Dial(SIP/0${EXTEN}@student01,,rWT)", but I'm getting exactly the same message as before. |
| 11:19.49 | file | you don't have an understanding of dialplan, so explaining why things are... doing what they're doing... is difficult |
| 11:21.12 | velix | file: That's why it's completly fine for me to pay for help. I'm trying this since Friday now. But it's nearly impossible to get help on Asterisk :( |
| 11:21.23 | file | exten => _0049X.,1,Dial(SIP/0${EXTEN:4}@student01,,rWT) |
| 11:22.18 | velix | Sorry for being such a noob. I'm really hard to try not to be. |
| 11:23.14 | velix | Wow, this gives me "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)" |
| 11:23.32 | file | then the student01 SIP peer is not set up correctly or something |
| 11:24.45 | velix | The university gave me SIP accounts for this. I can connect from my SIP phone perfectly (and softphone on Windows and Zoiper). |
| 11:25.13 | file | that doesn't change the above |
| 11:25.19 | velix | sip show peer is empty, interesting. |
| 11:25.20 | velix | 1 sec |
| 11:27.16 | velix | ah, peers! |
| 11:27.22 | velix | Yeah, it says "unreachable". |
| 11:27.51 | velix | Do I need "register =>" only for incoming calls or always? |
| 11:28.08 | file | register informs the remote side of your IP address/port for where to send incoming calls |
| 11:28.21 | file | any additional requirements (such as needing to be registered to place outgoing calls) is a policy of the remote side |
| 11:28.32 | velix | Okay, I don't need incoming calls, but I wonder why it has registration problems. |
| 11:28.41 | velix | ok, then I'll better keep it ;) |
| 11:29.18 | velix | Registration for '...' timed out, trying again... It's no NAT or Firewall problem, since I'm connected by the Sipgate account. |
| 11:31.29 | velix | Yeah, I just tried the outgoing call using sipgate and it works. So my line is fine. That's so annoing :( |
| 11:32.44 | velix | file: Thanks, you've helped me half the way! <3 |
| 11:34.43 | velix | Do I need to quote realms? "student01@192.168.103.10" ? |
| 11:43.00 | velix | Interesting, it tries to connect "Via: SIP/2.0/UDP" ... but it's TCP. |
| 11:43.20 | velix | I've set transport=tcp |
| 11:48.49 | velix | Hmm, It always tries to connect via UDP |
| 11:58.12 | *** join/#asterisk dacod (~dacod@201.47.74.146) |
| 12:03.26 | velix | It seems to be a bug in Asterisk 16. let me build v17 |
| 12:14.08 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:5037:518b:d158:69a2) |
| 12:18.28 | velix | Anyone with an idea, how to force TCP on Asterisk 16 ? |
| 12:19.11 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:5037:518b:d158:69a2) |
| 12:33.33 | velix | Yeah, it really seems to be a bug: Via: SIP/2.0/UDP |
| 12:36.13 | velix | yeah, tcp:// fixed it. |
| 12:40.39 | velix | Ywah, I'm getting CHANUNAVAIL now |
| 12:46.54 | velix | Anyone with an idea, how to work with REALM ? |
| 12:50.42 | Kobaz | sooooo |
| 12:50.45 | Kobaz | offhand |
| 12:50.53 | Kobaz | how well does asterisk handle fragmented UDP |
| 12:53.16 | Kobaz | more like chan_sip |
| 12:53.35 | Kobaz | if my stack gets fragmented packets, tcpdump is fine, but asterisk doesn't "see" anything |
| 13:08.43 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:15.43 | *** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl) |
| 13:29.08 | Samot | 8:50:53 AM <Kobaz> how well does asterisk handle fragmented UDP <- That's an OS thing not an Asterisk thing. |
| 13:29.17 | Kobaz | not so sure about that |
| 13:29.28 | Samot | No? |
| 13:29.28 | Kobaz | actually, that's definitely an application layer thing |
| 13:29.30 | Samot | OK. |
| 13:29.46 | Kobaz | because i've written low level UDP applications and had to do my own UDP fragmentation handling |
| 13:30.06 | Kobaz | UDP is connectionless, so you basically get another packet with a snip of data |
| 13:30.58 | file | it can be... both |
| 13:31.09 | file | UDP packets can be fragmented based on MTU at a lower level, of which you never see |
| 13:31.21 | file | or you can fragment at an application level based on application boundaries which is more reliable |
| 13:31.32 | Kobaz | correct, proper boundaries |
| 13:32.26 | Kobaz | this is at improper boundaries i'm seeing fragmentation |
| 13:32.38 | Samot | True but RFC's state that if this is a problem, use TCP. |
| 13:32.46 | Kobaz | yeah |
| 13:32.51 | Kobaz | that was my next thought |
| 13:32.55 | Kobaz | but i'm just sticking in a VPN |
| 13:32.58 | Samot | 1300 MTU or within 200 of the MTU. |
| 13:33.03 | Kobaz | fsck this public voip crud |
| 13:33.45 | Kobaz | customer routers tehsux |
| 13:34.02 | Samot | Heh. |
| 13:36.55 | Kobaz | jimminy jillikers |
| 13:37.15 | Samot | I remember having NAT issues like 15 years ago with customers. |
| 13:37.42 | Kobaz | every single customer so far is... what's your router? oh it's that one, turn off SIP ALG |
| 13:37.54 | Kobaz | even fios home routers have it now, and it's on by default and cannot be disabled |
| 13:38.05 | Samot | Weird. |
| 13:38.10 | Samot | I have home level Fiber. |
| 13:38.16 | Kobaz | https://forums.verizon.com/t5/Fios-Internet/Fios-Gateway-Router-G1100-and-VoIP-Issues/td-p/859640 |
| 13:38.19 | Samot | I've got 3 phones and multiple BLFs going. |
| 13:38.31 | Samot | And my router only supports DMZ. |
| 13:38.39 | Samot | Can't have a true pass through. |
| 13:41.21 | Kobaz | Samot: basically this user on the thread fixed it by bypassing the fios router |
| 13:41.23 | Samot | Then again, I've used proxies+Asterisk for the last 15 years too. |
| 13:41.30 | Samot | Makes a big difference. |
| 13:41.35 | Kobaz | which makes sense since fios routers are retarded |
| 13:42.30 | velix | Unbelieable. After working 5 hours on the SIP account of the University, I tried a SIP trunk demo... it worked immediatley. |
| 13:42.35 | velix | It's NOT my setup. |
| 13:42.43 | velix | The TCP realm-Stuff seems to be broken somewhy. |
| 13:43.45 | Samot | IS the demo from the same place as the other 4 accounts? |
| 13:43.58 | velix | No, it's a commercial SIP provider. |
| 13:44.20 | velix | I've debugged the communication between Asterisk and the university's accounts: some things are different. |
| 13:44.28 | velix | It doesn't register because FROM and VIA are different. |
| 13:44.37 | velix | It works from my Windows Softphone |
| 13:44.46 | velix | But the debug looks different. |
| 13:47.53 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-lgtalaaabkcapweu) |
| 13:47.53 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 13:49.05 | file | such things have not been touched in years, it's most likely configuration in some way |
| 13:53.17 | velix | file: Sure... like the tcp:// flag |
| 13:53.23 | Samot | '_+49X.' => 1. Goto(0${EXTEN:3},1) <- Why are you doing that? |
| 13:53.41 | Samot | Why are you making it 0+49 |
| 13:53.44 | velix | Samot: that's old. file has fixed it ;) |
| 13:53.54 | Samot | Oh I see it stripped. |
| 13:53.55 | Samot | NM. |
| 13:54.14 | velix | It worked _immediately_ with the commercial SIP trunk, that's incredible. |
| 13:54.29 | Samot | Well since we've never seen any debug output between the two... |
| 13:54.38 | Samot | We can't tell you why it worked and what was different. |
| 13:54.46 | Samot | All we know is what you've been telling us. |
| 13:54.53 | Samot | It's doesn't work here but works here. |
| 14:07.51 | velix | yeah, I'll publish them soon. Sorry, I need to make the trunk work as expected as a back fall ;) |
| 14:07.55 | velix | fallback |
| 14:09.25 | Samot | well if you need to make it work then you should be showing us what is not working. |
| 14:09.43 | Samot | Otherwise we're just getting a play by play of what doesn't work from you. |
| 14:13.53 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-pnqffssqfcxpvewk) |
| 14:13.54 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:14.53 | velix | No, I mean, I'm setting up the trunk first, which works :) |
| 14:15.00 | velix | Then I'll copy all the log files together |
| 14:20.58 | Kobaz | people dont understand that calling me to ask status and then tell me how important it is, doesn't make things go any faster |
| 14:21.29 | drc | But Kobaz, why aren't you done yet? This is important! |
| 14:21.39 | drc | You should really prioritize it! |
| 14:57.50 | Kobaz | there you go |
| 14:57.52 | Kobaz | updated our ivr |
| 15:00.48 | igcewieling | "Welcome to Comedian Mail! *pause* asshole *pause* is unavailable, please leave a message" |
| 15:01.10 | Kobaz | just put everyone into TORTUTE |
| 15:01.13 | Kobaz | FORTURE |
| 15:01.17 | Kobaz | sdfkjasdfkjhsadfasdf |
| 15:01.20 | Kobaz | you know what i mean |
| 15:01.22 | Kobaz | the thing with the T |
| 15:07.03 | *** join/#asterisk Guest5194 (~s98259@unaffiliated/iveeee) |
| 15:29.38 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 15:35.27 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 15:53.00 | *** join/#asterisk salviadud (~rgonzalez@189.207.212.148) |
| 16:11.03 | velix | Hmm... Did I setup the dialplan correctly? I'm sending variable KNDR to CALLERID in extension. |
| 16:13.56 | Samot | Again, not seeing a thing from you. |
| 16:14.10 | Samot | Still waiting for those good vs bad calls from your setups. |
| 16:15.48 | velix | oooohh |
| 16:15.58 | velix | https://bpaste.net/raw/W4ZA |
| 16:16.22 | velix | This works, phone rings on 0049xxxx but it doesn't show the CALLERID :( |
| 16:16.46 | velix | The functionality has been activated. |
| 16:16.50 | Samot | OK but that doesn't mean it wasn't sent out properly |
| 16:17.15 | Samot | Telecom Rule #541: You do not have control over what the accepting carrier will do with your CallerID. |
| 16:17.40 | Samot | They could honor it. They could do their own lookup. They could not even present the Name to the end user. |
| 16:18.09 | velix | Samot: I can set it up in the trunk's customer portal. Then the number gets set, but it's fixed. |
| 16:18.18 | velix | The provider told me, I can do it in Asterisk |
| 16:18.26 | Samot | Sure. |
| 16:18.40 | Samot | But when that call hits, let's say my network, I could ignore your callerID |
| 16:18.40 | velix | But the way I've done it looks fine? |
| 16:18.44 | Samot | Yes. |
| 16:18.54 | velix | Yes, I understand what you mean ;) |
| 16:19.05 | velix | Okay, I just thought, I did it wrong again. |
| 16:33.27 | *** join/#asterisk DanFromUK (sid21651@gateway/web/irccloud.com/x-ojfedpuvcjmtgiyd) |
| 16:39.42 | *** join/#asterisk jkroon (~jkroon@165.16.203.100) |
| 17:10.09 | velix | ah, got it working. I am stupid. |
| 17:10.18 | velix | sendrpid=yes |
| 17:10.18 | velix | trustrpid=no |
| 17:19.17 | igcewieling | If you had pastebin'd the CLI output, someone might have noticed that. |
| 17:21.42 | Samot | Crazy. |
| 17:22.25 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:5037:518b:d158:69a2) |
| 17:22.55 | Samot | Are you actually saying that just do things like "This doesn't work with X but works with Y, do you know why?" doesn't get you far? |
| 17:23.10 | Samot | s/do/doing/ |
| 17:25.31 | velix | :( SORRY |
| 17:25.58 | velix | But as you can see, I'm trying to figure it out for myself and not demand on the channel's help :D |
| 17:28.35 | Samot | Ahh, thats the rub. |
| 17:28.49 | Samot | You've spent most of your time giving us play by play and asking questions. |
| 17:29.33 | Samot | Remember, we spent quit a bit dealing with just the networking side of things. |
| 17:29.38 | Samot | VPNs and all. |
| 17:37.46 | velix | Samot: Oh, that part isn't fixed ... next step |
| 17:39.41 | velix | I've build Asterisk 17 earlier today. "sip.conf" wasn't used in the example configuration, but "pjsip.conf" was. Has it been changed? I thought, one was for channel? |
| 17:42.33 | Samot | Chan_SIP is deprecated. |
| 17:43.19 | Samot | It is now noload as of Asterisk 17 and future version. It _could_ be removed from Asterisk after 2023. |
| 17:43.38 | velix | I see, thanks. |
| 17:43.55 | Samot | Chan_SIP hasn't seen development in almost 6 years and any fixes/updates on it are community driven. |
| 17:52.53 | DanFromUK | When an inbound call is diverted to a number of targets which have macros for the recipient to dial a DTMF code to accept the call, is there any way to play a message to the recipient if the channel was answered elsewhere? |
| 18:03.54 | *** join/#asterisk mducharme (uid303982@gateway/web/irccloud.com/x-emzxthfzhnxlggqt) |
| 18:14.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:31.47 | *** join/#asterisk scampbell (~scampbell@mail.scampbell.net) |
| 18:40.11 | *** join/#asterisk rrittgarn (~rrittgarn@199.48.92.159) |
| 18:40.49 | rrittgarn | Afternoon/Evening all. Question, is a features.conf entry the only way to mute a call while you're on it via DTMF? (not using confbridge/meetme at present) |
| 18:49.14 | rrittgarn | <PROTECTED> |
| 18:55.42 | Samot | velix: No PMs. I don't read them. |
| 18:56.17 | velix | Okay, I didn't want to break into rrittgarn's question. |
| 18:56.22 | velix | This is my complete setup with the university's accounts: https://bpaste.net/raw/67FA |
| 18:56.27 | velix | I'm currently copying the logs. |
| 19:00.25 | velix | These are the logs from a Windows SIP client, which works: https://bpaste.net/3DPA |
| 19:00.48 | velix | As you can see, "Via: SIP/2.0/TCP 192.168.178.23" <-- interesting stuff |
| 19:04.54 | velix | And that's the unsuccessful try from Asterisk: https://bpaste.net/raw/TKBQ |
| 19:05.22 | file | that was successful |
| 19:06.35 | velix | file: No, it says "UNREACHABLE". |
| 19:06.47 | velix | tcptls.c:553 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.103.10:5060: Connection timed out |
| 19:06.57 | velix | don't know what 103.10 comes from :( |
| 19:07.19 | file | the outbound registration was successful |
| 19:07.40 | file | unreachable is separate. |
| 19:08.25 | velix | But I'm getting this every few minutes: "chan_sip.c:24836 handle_response_register: Outbound Registration: Expiry for pbx.univer.city is 120 sec (Scheduling reregistration in 105 s)" |
| 19:08.41 | file | that's how registration works |
| 19:09.00 | file | there is an expiration time, and then before that you re-register |
| 19:09.26 | velix | "Via: SIP/2.0/TCP 192.168.178.23" is from the Windows client, but this one does "Via: SIP/2.0/TCP my-hosts-ip:5060" |
| 19:09.44 | velix | Does it register to my asterisk box? :D |
| 19:09.51 | file | that doesn't change the fact that the registration is successful |
| 19:10.02 | velix | Yeah, but perhaps it registeres on itself? :D |
| 19:10.05 | file | it doesn't. |
| 19:10.07 | velix | ok |
| 19:10.14 | velix | Then let's try to make a call. |
| 19:10.26 | file | as I said, the unreachable state is unrelated to outbound registration |
| 19:10.32 | file | so if it's unreachable then the problem is elsewhere |
| 19:12.06 | *** join/#asterisk imcdona (~imcdona@65-100-46-166.dia.static.qwest.net) |
| 19:12.39 | *** join/#asterisk imcdona (~imcdona@65-100-46-166.dia.static.qwest.net) |
| 19:12.56 | velix | Yeah, the channel is unavailable: https://bpaste.net/raw/M22A |
| 19:13.41 | velix | file: Is there a way to see that it's registered? |
| 19:13.57 | file | registration and outbound calling are unrelated in Asterisk |
| 19:14.11 | velix | sip show registry |
| 19:14.17 | file | and you can tab complete the CLI, "sip" and then tab complete to find relevant things |
| 19:14.30 | velix | Yah, I just wanted to have a debugging breakpoint :D |
| 19:15.06 | velix | file: Could you please give me a hint, where to look at else? |
| 19:15.27 | file | I haven't touched chan_sip in years, 'nor used it with an outbound proxy so I don't remember anything |
| 19:15.41 | velix | file: Can I do it with pjsip? |
| 19:16.03 | file | people use PJSIP, yes |
| 19:16.23 | velix | Could you give me a hint, how to do it in PJSIP? |
| 19:18.11 | file | if you find and follow examples and have issues people may provide help |
| 19:19.00 | velix | ok |
| 19:19.34 | Samot | fromuser=student20@192.168.103.10 |
| 19:19.36 | Samot | ?? |
| 19:20.49 | velix | The admin said, I need to use student10@192.168.103.10 and student20@192.168.103.10 as user names because of the REALM. |
| 19:22.08 | Samot | That's what the fromdomain is for. |
| 19:22.14 | Samot | fromuser is the USER |
| 19:22.32 | Samot | fromdomain is the DOMAIN together they make USER@DOMAIN |
| 19:22.55 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 19:23.39 | velix | okay. |
| 19:24.15 | *** join/#asterisk netman (~netman@185.94.249.222) |
| 19:27.55 | velix | Samot: But the host=tcp:// is the other PBX's IP and outboundproxy=tcp://192.168.103.10 ? |
| 19:30.32 | *** join/#asterisk alexandre9099 (~alexandre@unaffiliated/alexandre9099) |
| 19:31.24 | *** join/#asterisk hfb (~hfb@47.139.19.93) |
| 19:37.45 | velix | I don't understand, why it's connected to itself. "Via: SIP/2.0/TCP myip:5060" |
| 19:38.02 | file | it's not. |
| 19:38.33 | velix | I'm getting this every few seconds now: Really destroying SIP dialog '3d9f3a4f1a569b540a524ce131a39dcd@...:5060' Method: OPTIONS |
| 19:40.41 | velix | It still says "unreachable". Damn :-( |
| 19:41.43 | velix | I don't even know where to start debugging this. I've tried about all the menus in asterisk |
| 19:42.57 | *** join/#asterisk alexandre9099 (~alexandre@unaffiliated/alexandre9099) |
| 19:43.01 | velix | Wow, 9 hours gone on this problem now. |
| 19:48.53 | velix | I really don't understand it |
| 19:51.30 | *** join/#asterisk NirS_ (~nirs@40.67.196.251) |
| 19:53.49 | *** join/#asterisk fructose (~fructose@unaffiliated/fructose) |
| 20:11.39 | velix | No, I'm giving up. It simply is impossible |
| 20:18.51 | *** join/#asterisk scampbell (~scampbell@mail.scampbell.net) |
| 20:26.28 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 20:33.42 | velix | I really wonder, why I am getting thousands of "Really destroying SIP dialog ... OPTIONS" messages. |
| 20:35.47 | velix | ah, it's a bug: http://lists.digium.com/pipermail/asterisk-bugs/2009-July/050904.html |
| 20:40.06 | *** join/#asterisk Guest5194 (~s98259@unaffiliated/iveeee) |
| 20:42.14 | velix | Does anyone else have an idea? |
| 20:49.46 | Samot | That's 11 years old. |
| 20:50.42 | velix | I'm trying PJSIP now, but it's a complete new world. |
| 20:53.33 | velix | Samot: Could you have a look at my last question please? |
| 20:54.10 | Samot | what question? |
| 20:54.38 | velix | Samot: But the host=tcp:// is the other PBX's IP and outboundproxy=tcp://192.168.103.10 ? |
| 20:57.25 | Samot | Don't think you need tcp:// in the host |
| 20:57.34 | Samot | I guess in the outbound proxy. |
| 20:58.43 | velix | file: Is this PJSIP? https://bpaste.net/Q4IA |
| 21:00.15 | velix | Samot: So like this? https://bpaste.net/IAFQ |
| 21:02.15 | *** join/#asterisk sekil (~sekil@178-223-3-39.dynamic.isp.telekom.rs) |
| 21:04.09 | velix | Wow, 12 hours on this right now today :) |
| 21:05.29 | velix | I can't understand, what "UNREACHABLE" means. |
| 21:05.33 | velix | It's online. |
| 21:05.40 | velix | The firewall isn't a problem. |
| 21:06.09 | Samot | It means that Asterisk didn't get replies back from the other side. |
| 21:06.47 | velix | Interesting: "Unreachable generally means you have qualify=yes, but the peer is ignoring OPTIONS requests." |
| 21:06.59 | velix | perhaps that's why I got thousands of OPTIONS requests? |
| 21:08.00 | Samot | Or it's not getting a reply. |
| 21:08.15 | Samot | The device should send back a 200 OK reply. |
| 21:08.33 | Samot | If Asterisk doesn't get it, it tries again for about 6-7 attempts. |
| 21:08.42 | Samot | Then it marks the endpoint unreachable. |
| 21:09.47 | velix | waaaaaaaaaaait. I've set "qualify=no" and "keepalive=yes" |
| 21:09.57 | velix | it's "unmonitored" now, but I can dial through |
| 21:10.04 | velix | But it doesn't reach the device. |
| 21:10.09 | velix | But hey, 90% |
| 21:11.15 | velix | Samot: exten => _0049X.,1,Dial(SIP/0${EXTEN:4}@student01,,rWT) -> this adds a "0" before the normal number, doesn't it? |
| 21:11.55 | [TK]D-Fender | it adds a 0 after stripping the first 4 off |
| 21:12.40 | Samot | Yes,that would add a 0 before regardless if stripping or not happens. |
| 21:12.56 | Samot | In this exact case it would do what TK said. |
| 21:13.03 | velix | yeah. I need to add a "0" from the university network to call externally. |
| 21:13.47 | velix | Unbelieveable. I'm getting the normal "beeeeep .... beeeeep", but my phone doesn't ring :D |
| 21:14.59 | Samot | Are there actual replies to the request? |
| 21:16.00 | velix | oh wait. SIP/163xxxxxxx@student01 <-- shouldn't this be 00163? I've dialed 0049163... |
| 21:16.27 | velix | Samot: Which kind of replies? |
| 21:16.40 | Samot | To the INVITEs. |
| 21:17.03 | Samot | Again, we're getting back to NAT issues it seems. |
| 21:17.19 | Samot | Devices going unreachable, incoming calls not working but outgoing does |
| 21:17.38 | velix | Nope, 100% no nat problem. The PBX is directly on the internet on the server. |
| 21:17.45 | velix | My iPhone is directly on LTE |
| 21:17.47 | Samot | Where the PHONES ARE |
| 21:17.55 | Samot | Which is BEHIND NAT. |
| 21:18.07 | Samot | They aren't giving everyone phone in the world a public IPv4 address. |
| 21:18.17 | velix | I'm not using SIP on the phone. |
| 21:18.28 | velix | I'm calling by normal ... how do you call it? by air? |
| 21:18.29 | velix | dunno |
| 21:18.36 | Samot | So you're calling outbound. |
| 21:18.40 | velix | Yes. |
| 21:18.57 | velix | When I do it from the SIP tool in Windows, the call goes through |
| 21:19.05 | velix | There is 100% no nat problem on the server. |
| 21:19.08 | Samot | I don't know what that means. |
| 21:19.24 | Samot | Then if you're calling your mobile phone and you hear ringing but your phone never rings... |
| 21:19.28 | velix | I've installed a softphone on Windows, entered the university's SIP accound and called my phone. It works. |
| 21:19.36 | Samot | Either the request never made it to the provider or it didn't go through. |
| 21:19.40 | Samot | OK |
| 21:19.45 | velix | I've set the SIP account data in Asterisk, called my phone and it doesn't ring. |
| 21:19.51 | Samot | so then this is a config issue with the PBX. |
| 21:19.55 | velix | But when using another SIP provider (a real one), it works. |
| 21:20.06 | Samot | Stop. |
| 21:20.08 | velix | It worked after 23 seconds |
| 21:20.37 | Samot | If you keep setting up SIP clients with the University or other Providers and IT WORKS.... |
| 21:20.52 | Samot | Then you try to use the PBX and it doesn't. Then the issue is the PBX. |
| 21:21.13 | velix | Yes, that's why I am here :) |
| 21:21.17 | Samot | It's not configure right or the networking isn't done right or any combination of things. |
| 21:21.30 | Samot | Then start showing things. |
| 21:21.34 | Samot | Actual debugs. |
| 21:21.39 | Samot | We cannot keep guessing. |
| 21:21.41 | velix | Tell me how and I'll do. |
| 21:22.05 | velix | sip set debug peer student01 |
| 21:22.14 | velix | core set verbose 13 |
| 21:38.53 | velix | Samot: Just give me a quick hint what you need. |
| 21:40.30 | Samot | Well.. |
| 21:40.33 | Samot | those are the right commands. |
| 21:40.45 | Samot | I was waiting for the followup |
| 21:41.11 | Samot | The part where you made a call, copy and pasted all the output to a pastebin service and then give us the link. |
| 21:45.40 | velix | 1 sec |
| 21:46.09 | joepublic | looks more like 12 hours actually |
| 21:47.19 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1019:6eef:e18c:18e8:3a:af08) |
| 21:48.26 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1019:6eef:2157:f1c7:2346:c233) |
| 21:49.03 | velix | Samot: https://bpaste.net/F6UA |
| 21:49.27 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1019:6eef:b424:9735:56b0:bfd8) |
| 21:54.24 | velix | That looks interesting: "chan_sip.c:30398 sip_send_keepalive: sip_send_keepalive to UNIVERSITY:5060 returned 0: Broken pipe" |
| 21:56.18 | Samot | That was the only call happening? |
| 21:56.20 | velix | I've turned off keepalive. |
| 21:56.32 | velix | Samot: No, always. I think, the PBX of university cannot handle timeouts. |
| 21:56.35 | velix | eeeh keepalives. |
| 21:57.16 | velix | Actually, Asterisk dials somewhere over the universiy's SIP account. I don't see an error here. |
| 21:57.33 | velix | oh hello! "tcptls.c:553 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.103.10:5060: Connection timed out" |
| 21:57.36 | velix | That's a new one. |
| 22:04.56 | velix | Yeah! got it working. |
| 22:05.23 | velix | fromdomain=192.168.103.10 |
| 22:05.23 | velix | host=192.168.103.10 |
| 22:05.23 | velix | outboundproxy=tcp://UNIVERSITY |
| 22:20.23 | *** join/#asterisk tafa2 (~tafa2@185.115.101.208) |
| 22:30.26 | fructose | I am interested in building a helpline for use during a pandemic. Ideally I'd like to be able to have a system accept incoming calls on one phone line and forward them to volunteers who themselves may only have access to personal phone lines or an Internet connection. Is that achievable with Asterisk or its progeny? |
| 22:35.55 | salviadud | fructose: you might need some hardware |
| 22:36.14 | salviadud | That phone line, would that be a regular landline you already have? |
| 22:37.57 | fructose | salviadud: I'd at least need to avoid the volunteers needing specialized hardware. Presumably we could get whatever line was needed from a phone company. |
| 22:38.08 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1019:6eef:296c:bd68:dfbd:56d1) |
| 22:41.30 | salviadud | If you want to accept calls from a regular FXS port and it is only one line, I'd recommend an ATA adapter |
| 22:41.59 | salviadud | Then after that, you can register it via sip on asterisk |
| 22:42.08 | salviadud | And do as you please. |
| 22:43.29 | fructose | salviadud: Can you accept multiple calls on only one line that way? |
| 22:48.20 | joepublic | I am not recommending ringcentral, but you might want to look at their model. publish one number, calls forward to your volunteers. |
| 22:56.45 | fructose | joepublic: Okay, thanks |
| 23:02.31 | sibiria | why not build it with the most available solutions there are? e.g. a DID for the incoming call -> your asterisk setup running anywhere -> SIP provider terminating to volunteers' landline / cell phone number |
| 23:03.08 | sibiria | leave out the idea of using an incoming landline on your end. save yourself the trouble of dealing with adapters and concurrency problems |
| 23:15.03 | velix | Sorry, what am I missing here to initiate calls via CLI? https://bpaste.net/raw/MTEA |
| 23:16.45 | velix | ah, got it working. |
| 23:23.00 | fructose | sibiria: Thanks, I'll look into that as well |
| 23:28.41 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:36.01 | *** join/#asterisk allizom (~Thunderbi@unaffiliated/allizom) |
| 23:41.57 | velix | Hmm. Can I directly call outside from Asterisk and playback a file? "Extension" string in the callfile is used already to set the playback extension. |
| 23:47.49 | sibiria | yeah you can run BackGround and PlayBack directly from a call file, without ever entering a dial plan |
| 23:48.19 | sibiria | Playback* |
| 23:51.07 | velix | ah, got it working with a callfile. |
| 23:53.01 | velix | Interesting. With "Channel: SIP/...@student01", the call is anonymous. |
| 23:54.46 | velix | Okay, I need to set a Callerid then. |
| 23:57.48 | velix | WaitTime: 3 |
| 23:57.48 | velix | MaxRetries: 5 |
| 23:57.48 | velix | RetryTime: 3 |
| 23:57.50 | velix | ROFL :DDDD |
| 23:57.59 | velix | Phonebombing |
| 23:59.32 | *** join/#asterisk joepublic (~joepublic@fsf/member/joepublic) |