| 00:23.07 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:23.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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| 11:47.34 | Bast4 | Anyone here that can provide SIP trunk in norway? |
| 11:50.13 | sibiria | Sinch offers in Sweden. latency will be good enough |
| 11:54.13 | Bast4 | sibiria: thanks. I want a norwegian number to show up though when I call, unfortunately this wont work without a norwegian provider :\ |
| 11:54.47 | sibiria | of course it will work. it's up to the trunk provider to either allow or enforce this or that caller id |
| 11:54.57 | sibiria | sinch doesn't limit this for norway |
| 11:55.20 | sibiria | set any caller id you want, dial |
| 12:00.12 | Bast4 | Yeah I have no doubt that custom caller ID works, but the country code however does not change |
| 12:00.32 | Bast4 | The country extension still shows up :\ |
| 12:00.53 | Bast4 | At least thats what happened when i used voip.ms |
| 12:01.07 | sibiria | why would that be a problem, though? |
| 12:01.27 | Bast4 | People will not pick up my calls when a foreign number is calling |
| 12:01.34 | Bast4 | I want my own personal number to be displayed |
| 12:03.13 | sibiria | i beg to differ on that (not pick up when "foreign" number) |
| 12:03.27 | sibiria | surely they also know their own country's prefix |
| 12:04.41 | sibiria | also, ios and android will match domestic number formats in the phonebook to intl. format |
| 12:05.21 | Samot | Yes but not everything is running iOS or Android. |
| 12:05.23 | sibiria | so if you have 41234567 in your phonebook, caller id +4741234567 will match |
| 12:05.30 | sibiria | no but most of the world is |
| 12:05.43 | Samot | By most you mean individuals. |
| 12:05.49 | Samot | Not businesses. |
| 12:05.53 | sibiria | correct |
| 12:06.02 | Samot | Or those with non-mobile services. |
| 12:06.19 | sibiria | the last brave few of that bastion |
| 12:07.35 | Samot | There's also this whole thing where providers can ignore the callerid being sent to them |
| 12:07.44 | Samot | And do their own lookup. |
| 12:08.33 | Samot | It's up to the destination carrier to accept the CallerID being sent or to look it up themselves based on the DID. |
| 12:09.35 | sibiria | it's less of a problem in the west, really |
| 12:10.03 | sibiria | in europe f.e. it's only a problem in one or two countries |
| 12:10.05 | sibiria | turkey is one of those |
| 12:10.11 | Samot | "In the west"? |
| 12:10.16 | Samot | You mean North America? |
| 12:10.23 | sibiria | as in all of europe and to my knowledge north american |
| 12:10.25 | sibiria | america* |
| 12:10.45 | Samot | OK, well I work with carriers in North America. |
| 12:10.49 | Samot | It still is a thing. |
| 12:10.55 | sibiria | so far i've only experienced this to be a domestic level problem with turkey and isreal |
| 12:10.58 | Samot | Depending on the carrier. |
| 12:11.00 | sibiria | israel* |
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| 12:18.43 | ih8wndz | having a problem with pjsip, can make calls, but inbound calls result in an immediate "Called PJSIP/MBPM21" |
| 12:19.18 | ih8wndz | I've been using pjsip for quite a while now, this is really got me |
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| 12:33.30 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:34.18 | [TK]D-Fender | ih8wndz, inbound can't be throwin only that amount of information to CLI. Inbound should show * calling OUT. |
| 12:34.22 | [TK]D-Fender | Show us the actual call debug |
| 12:34.25 | [TK]D-Fender | ~pb |
| 12:34.25 | infobot | i guess pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:38.09 | ih8wndz | [TK]D-Fender: I know, I'm trying to figure this out, it's got me over a barrel |
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| 13:23.59 | ih8wndz | [TK]D-Fender: I know how to: pjsip set logger on, how do I spit that out to /var/log/asterisk/messages |
| 13:24.17 | [TK]D-Fender | * CLI only |
| 13:24.21 | [TK]D-Fender | forget "logs" |
| 13:24.27 | [TK]D-Fender | verbose 10 |
| 13:24.30 | [TK]D-Fender | and SIP debug |
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| 13:32.40 | ih8wndz | [TK]D-Fender: https://pastebin.ca/4019089 |
| 13:35.15 | [TK]D-Fender | So the call comes in, starts processing, you hit a queue that looks like you didn't answer the call first, it dials out to its members and then the calling side aborts |
| 13:35.54 | [TK]D-Fender | Sorry, forget the abort part |
| 13:35.57 | [TK]D-Fender | not the same call |
| 13:37.58 | [TK]D-Fender | I don't seem to see the end of the call in there, nor do I see it getting answered |
| 13:38.49 | ih8wndz | it doesn't ring the device, it aborts immediately, rings though due to que and then voice mail picks up |
| 13:39.15 | [TK]D-Fender | CANCEL sip:12284356149@sip.sharedvps.com SIP/2.0 |
| 13:39.18 | [TK]D-Fender | Sorry, there it is |
| 13:39.26 | ih8wndz | should only be one call |
| 13:39.37 | [TK]D-Fender | ANSWER the call before sending to Queue |
| 13:39.49 | ih8wndz | orly, ok |
| 13:40.04 | [TK]D-Fender | yes because to Flowroute you aren't answering and they seem to have timed out |
| 13:40.09 | [TK]D-Fender | They don't ring you indefinately |
| 13:40.26 | [TK]D-Fender | If you want to be able to maintain the call you have answer it so they don't time out |
| 13:41.10 | ih8wndz | does the same got for Dial()? |
| 13:41.49 | ih8wndz | s/got/go |
| 13:42.33 | [TK]D-Fender | those don't answer the calling side implicitly |
| 13:42.58 | [TK]D-Fender | You have to do it yourself first otherwise you run that risk |
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| 13:48.02 | ih8wndz | [TK]D-Fender: https://pastebin.ca/4019090 |
| 13:48.26 | ih8wndz | included Answer here, no queue on this number |
| 13:49.59 | ih8wndz | same server |
| 13:52.29 | [TK]D-Fender | use something that actually establishes audio |
| 13:52.42 | [TK]D-Fender | playback(silence/1) could do |
| 13:52.49 | [TK]D-Fender | Answer, Playback, then Dial |
| 13:53.08 | ih8wndz | ok |
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| 13:57.19 | ih8wndz | https://pastebin.ca/4019093 |
| 13:59.37 | [TK]D-Fender | I see your peer is matching as Flowroute, but what is this: INVITE sip:16016200852@sip.sharedvps.com SIP/2.0 |
| 14:00.09 | [TK]D-Fender | that is starting to look like you're calling from an intermediate server |
| 14:02.39 | ih8wndz | no, seperate server, my server |
| 14:03.35 | ih8wndz | server a -> flowroute -> ptsn -> flowroute -> server b |
| 14:04.05 | ih8wndz | pstn |
| 14:05.27 | ih8wndz | I can make the call via my cell phone if you want..? |
| 14:11.26 | igcewieling | You would be eliminating at least 50% of troubleshooting by not involving a 2nd server. |
| 14:11.39 | [TK]D-Fender | I'm wondering why I see "vps" if that's actually Flowroute's system directly calling the server whose debug we're looking at |
| 14:11.46 | [TK]D-Fender | I'm not sure why they are concelling |
| 14:11.52 | [TK]D-Fender | how fast does that happen? |
| 14:12.58 | ih8wndz | immediately |
| 14:13.39 | [TK]D-Fender | Don't see a reason here just yet. If that i fact Flowroute directly calling you there I'd contact them to have them debug it in realtime with you |
| 14:24.26 | ih8wndz | call via my cell: https://pastebin.ca/4019099 |
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| 14:29.03 | [TK]D-Fender | <--- SIP read from UDP:216.115.69.144:5060 ---> |
| 14:29.03 | [TK]D-Fender | BYE sip:16016200852@50.116.26.59:5060 SIP/2.0 |
| 14:29.07 | [TK]D-Fender | Still looks kinda instant |
| 14:29.33 | [TK]D-Fender | Try playing back some actual fixed recordings and see if you can keep the call alive in dialplan for a little bit before trying to dial out. |
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| 14:48.13 | ih8wndz | called flowroute, couldn't help. everything looks good on their end, "signal is going to the right place" |
| 14:48.36 | [TK]D-Fender | What do tc&order=pricethey have nothing to say for the instant looking cancel? |
| 14:48.40 | [TK]D-Fender | oops |
| 14:48.48 | [TK]D-Fender | They have nothing to say for the instant looking cancel? |
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| 14:58.07 | ih8wndz | https://pastebin.ca/4019102 |
| 14:59.28 | [TK]D-Fender | Still a really fast cancel.... |
| 14:59.42 | ih8wndz | as the call is making it to my server, and beinging answered by my asterisk. the issue is at the device |
| 15:00.07 | [TK]D-Fender | it's cancelled by the calling end |
| 15:00.14 | [TK]D-Fender | not your phones |
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| 15:02.39 | ih8wndz | i added ss-noservice, which plays. so the "called" does "answer" & "playback". |
| 15:03.03 | ih8wndz | it will even queue, and then go to voicemail, |
| 15:03.27 | ih8wndz | in regards to one of my other DID's |
| 15:04.20 | ih8wndz | but weither it's Dial OR Queue, same issue... Get an immediate "Dialed [extension]" |
| 15:16.28 | [TK]D-Fender | Yeah I really don't see a reason based on what's in the debug |
| 15:16.47 | ih8wndz | frack |
| 15:16.50 | [TK]D-Fender | You are 3 releases behind so I might try upgrading "just because" on grounds that you have issues right now |
| 15:17.39 | ih8wndz | I was on 13.20 |
| 15:18.21 | ih8wndz | I've tried 13.17.2 13.18.0 13.19.2 13.20, all same issue |
| 15:19.38 | ih8wndz | I just reverted to 13.18 from 13.20 on my other servers |
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| 15:21.41 | twanny796 | index.php of atslog is all giberish |
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| 15:55.48 | igcewieling | is starting to wonder if criminalizing the password 1234 might be good thing. |
| 16:00.35 | [TK]D-Fender | That's the stupidest combination I've ever heard in my life! That's the kinda thing an idiot would have on his luggage! |
| 16:02.50 | igcewieling | It seems to me about %75 of users try using that password. |
| 16:04.09 | igcewieling | hmmm... 1) random voicemail boxes with password 1234 2) extortion 3) profit! |
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| 16:53.40 | Kobaz | sooooo |
| 16:54.04 | Kobaz | Anyone recommend something like this that actually has some good reviews and isn't from a noname company? https://www.amazon.com/Anysun-Gateway-bridges-Encryption-Asterisk/dp/B01MA6V00M/ref=sr_1_9?s=electronics&ie=UTF8&qid=1524847966&sr=1-9&keywords=sim+gateway |
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| 17:01.46 | [TK]D-Fender | https://www.beronet.com/products/voip-gateways/gsm-gateway/ |
| 17:01.49 | [TK]D-Fender | https://www.voipsupply.com/voip-gateways/gsm-gateways |
| 17:02.01 | [TK]D-Fender | I would lay odds these are probably acceptable |
| 17:02.03 | Kobaz | you'se used them? |
| 17:02.05 | Kobaz | the bero ? |
| 17:02.32 | [TK]D-Fender | Not personally. I've known both companies for being on the market quite a long time and their names thrown around here obviously |
| 17:02.50 | [TK]D-Fender | You should be able to find reviews for their products to help validate |
| 17:03.45 | [TK]D-Fender | GOIP is one I CONSTANTLY hear only non-English speaking people come in here screaming for "HAELP??" with |
| 17:03.52 | Kobaz | haha |
| 17:04.30 | [TK]D-Fender | GOIP + E6L = Don't even bother answering. |
| 17:04.39 | igcewieling | Sangoma has multi port GSM Voice cards. I don't know if Digium has anything similar. |
| 17:05.24 | Kobaz | yeah goip sounds crap |
| 17:05.31 | file | we don't do GSM |
| 17:05.33 | [TK]D-Fender | I don't see any dedicated gateways from them though. |
| 17:05.33 | Kobaz | Review: The device work good, but if you leave it for a more than 3-4 days then somethings happen and need to be restarted. There is a function for restart every day at specific time, but this does not solve the problem. Need physical restart. |
| 17:05.43 | igcewieling | This is the one I'm thinking of: http://www.se-mena.com/sangoma-w400-gsm-card-en I've not used them as customers have not wanted to spend the extra money. |
| 17:05.52 | Kobaz | i need something external |
| 17:05.57 | Kobaz | no on prem pbx |
| 17:06.03 | Kobaz | so ethernet/sip |
| 17:06.09 | Kobaz | for local survivability |
| 17:06.17 | igcewieling | Oh. Good luck with that. |
| 17:06.26 | Kobaz | there was a company at last years astricon that had some good stuff |
| 17:06.30 | Kobaz | i gotta find their paperwork |
| 17:09.24 | Kobaz | https://www.raspberrypi.org/blog/raspberry-pi-gsm-gateway/ |
| 17:09.34 | Kobaz | heh, i would rather an actual product, but that looks nifty |
| 17:14.51 | [TK]D-Fender | Yeah I had the impression you wanted something integrated, 1-piece, no fuss |
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| 22:51.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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