IRC log for #asterisk on 20191205

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04:58.21jeffspeffwhere can i find the table structure for realtime followme and followme_numbers?
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05:27.21igcewielingNot bad, my VM which does FTP provisioning hit 903 connections when I rebooted a "all" of our hosted phones.
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12:37.09fores331Hello, experiencing some hiccups in our asterisk/freepbx installation. Anybody here to help?
12:44.23sibiriahave you tried putting it over your shoulder and patting its back?
12:45.18fores331even offered it some tea but still
12:45.21sibiriaif you describe the problem someone who's familiar with freepbx might see and know what's up
12:49.57fores331one provider that worked well suddenly got broken and do not allow any outcoming calls anymore but incoming still work or I think, so I am trying to switch to another provider and to most of them I can connect but still can't call. There is one the calls are going through but the voice is interrupted for couple seconds every 2 seconds or so.
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13:12.20jsmithfores331: Have you looked at the logs?  Taken a packet capture of the SIP traffic?
13:13.16fores331I tried but don't understand much there
13:14.50jsmithThe first thing I would do would be to try to understand why the provider is rejecting the outbound calls (assuming that they *are* making it to the provider, and that the provider is rejecting them)
13:15.12jsmithYou should be able to look for the outbound SIP INVITE to the provider, and look for its response.
13:16.30jsmithThis may be of some help, depending on your familiarity with Wireshark: https://wiki.wireshark.org/VoIP_calls
13:19.02fores331They do it to provider as in their web interface I can see rejection. That provider I am trying to leave and another provider works but with this hiccups. If I connect directly to the trunk with softphone there are no hiccups. So I thought might be some simple settings in freepbx or asterisk sip settings.
13:23.03fores331Never seen wireshark but might try to play with it. Otherwise I would be happy if somebody had a look at the server and fixed it for a reasonable fee.
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15:02.58overyanderHi, I'm setting up pjsip for my provider (Flowroute) that uses IP based authentication. I followed the instructions under "A SIP trunk to your service provider, including outbound registration" at https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples but left out the registration and auth parts. I am able to receive calls from Flowroute but outbound calls fail with https://paste.centos.org/view/c3842b89  what am I missing?
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15:04.51MLCCould not create dialog to invalid URI 'SECRETaccountkey*155512375555'.
15:05.03MLCcheck the specification of the server in your pjsip.conf
15:05.25overyanderspecification of the server?
15:05.44MLCin the aor
15:05.48MLCcontact=
15:05.51filehttps://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
15:06.08fileif you are dialing an extension, it's a different format
15:07.08SamotIt needs to be PJSIP/code*${EXTEN}@FLOWROUTE
15:07.50SamotOr PJSIP/FLOWROUTE/sip:code*${EXTEN}@domain:port
15:08.02SamotI'd go with the first.
15:08.42SamotPJSIP/FLOWROUTE/SECRETaccountkey*155512375555 <-- That is a CHAN_SIP format.
15:09.23davlefouHi, somes one know what is memberdelay in queue option?
15:10.32MLCdavlefou: look in the sample queues.conf file
15:10.33overyanderthank you file and Samot that solved it.
15:12.17davlefouMLC good ideas, thk!
15:13.43MLCAnyone familiar with DPMA? I've added alert= to the phone configuration section, both in the template and directly on the phone itself, but it doesn't show up in "digium_phones show phone xxx".
15:24.28MLCI got it functional, but the alert= still does not show from the command line "digium_phones show phone xxx". Where do I submit a DPMA bug?
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15:57.42fileDigium phones related things would go through Sangoma Technical Support
15:57.51MLCok
16:03.34overyanderI'm having the same issue when trying to dial a local phone extension exten=_XXXX,n,Dial(PJSIP/${EXTEN},30,)
16:03.48overyanderhttps://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels says that should work
16:04.02overyanderand I verified that there is a contact for the extension aor
16:06.19[TK]D-FenderDon't tell us what you verified, SHOW US
16:06.35[TK]D-FenderAssurances mean nothing
16:09.00overyander[TK]D-Fender, what do you need to see to help diagnose the issue?
16:11.32[TK]D-FenderAll of it obviously
16:11.47[TK]D-FenderWe have nothing
16:12.30[TK]D-FenderWe have no proof what you're dialing, the state, that it even made an attempt, that your dialplan is even executing what you say,  or the kind of failure.
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16:23.06overyander[TK]D-Fender, how about this? https://gist.github.com/jeffclay/071284690822aac4646925d850b375cb
16:24.04filethat's the dialplan, not the actual console output and what is happening or the "pjsip show endpoints" and "pjsip show contacts" output
16:24.10[TK]D-FenderDo you should a brochure for your car to a mechanic and then ask what's wrong or do you bring the actual car?
16:24.41overyanderdid you guys not see the error and other outputs in that link?
16:25.11fileit did not embed in IRCCloud, but you also need to show "pjsip show endpoint"
16:25.17fileyou dial an endpoint, which then references an AOR
16:25.33fileoh and your contact is unavailable
16:25.44[TK]D-FenderContact:  1402/sip:1402@192.168.31.27                  2e2bc3d7f9 Unavail
16:25.56[TK]D-FenderSorry ... nobody home
16:27.34overyanderi've added the endpoint info for 1402  please refresh the page.
16:27.48overyanderwhat am i missing that is causing my contact to be unavailable?
16:28.07fileunknown, that means that Asterisk sent an OPTIONS SIP request to the given contact and received no response
16:28.56fileis it an inbound registration, or did you explicitly configure it?
16:29.33overyanderinbound reg
16:29.34fileis it a local device, or is it remote?
16:29.46overyanderremote to the server
16:29.54[TK]D-FenderLook at that IP.
16:30.02fileif it's behind NAT then you need to enable the settings for if a phone is behind NAT
16:30.09[TK]D-Fenderthat is not a "local" IP
16:30.10filerewrite_contact being one
16:42.00overyanderthank you. rewrite contacts did the trick
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20:44.03overyandercan an endpoint have multiple auths?
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20:47.47igcewielingI doubt any phones support multiple passwords for the same account.
20:48.55igcewielingDo you have one which does?
20:58.03seanbright"can an endpoint have multiple auths" yes
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20:58.32seanbrightso sayeth the code
20:59.15overyanderI was thinking more along the lines of having multiple auths for an endpoint to set different usernames AND passwords for an auth ID that is shared by an endpoint. i set my edpoints to be the extension numbers and want to have multiple devices use the same endpoint and aor but have unique authentication.
20:59.23overyanderthanks seanbright
20:59.27seanbrightyw
21:01.35overyanderseanbright, is the auth value in the endpoint config supposed to be comma delimited?
21:02.40seanbrightyes
21:03.06seanbrightlooking at the code, it appears that if you specify 'auth' multiple times it will overwrite previous ones, rather than appending
21:03.11seanbrightso comma delimited is the way to go
21:03.24overyanderthank you again
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21:30.25SamotOh. Dear. God.
21:30.54Samotdrmessano: I have to deal with, get ready for it, a RED PHONE BOX
21:32.40SamotSorry a RedBOX!
21:33.00jpsharpOdd issue:  I have a channel and I set variable __XX=foo.  Then I dial out, but in the dial I use a pre-dial handler and change that variable to "notfoo" via an AGI script.  I can confirm that the variable is set before the pre-dial returns, but when the channel hangs up, I do a dumpchan and I see the variable twice...once with "foo" and once with "notfoo".
21:33.08jpsharpI'm very confused.
21:38.18SamotMan two PRIs that connect to this RedBox that then does T1 signalling to an Adtran TA750 with 16 FXO and FXS ports. Then the RedBox has some special TDM card that is in some ancient FreePBX server, like early 2.x release.
21:38.24jsmithjpsharp: That sounds like a bug.
21:39.02jpsharpjsmith: Yeah, I'm digging through the bug tracker it see if it has been reported.
21:39.45SamotWell it still has Zaptel so yeah, really freaking old.
21:49.20*** join/#asterisk macaruchi (~macaruchi@179.53.195.112)
21:49.27macaruchiHi!
21:49.57macaruchiI wanna know if you know any GSM gateway that can work with Asterisk without any issue
21:50.18macaruchiI need a GSM gateaway with at least 16 gsm SIM
21:50.23macaruchiAny recomendation ?
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22:12.12*** join/#asterisk miralin1 (~Thunderbi@178.34.151.68)
22:19.13*** join/#asterisk hfb (~hfb@47.139.16.213)
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23:27.22Sladethink i'll try flowroute for my apps sms verification
23:28.02*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
23:35.41Sladehmm. i guess problem with that is you're seriously rate limited without a shortcode
23:36.56MLCTwilio does not rate limit on long codes, but their prices are higher.
23:40.18Slade.004 vs .0075
23:40.44SamotTwilio has shortcodes.
23:41.11SladeSamot, think i just wanted to use a long code on flowroute, but its rate limited
23:41.35MLCShort codes are so $$$$$ !
23:43.11Sladetwilio charges .09 if you use their auth service. i figure i can build my own for less than that ;P
23:43.35MLCyeah, there's not much to it
23:44.34SamotWhat is flworoute's rate limit?
23:44.38Sladetrying to find a lib/package out there that does it precanned, so i'm not inventing an already existing wheel.. flowroutes official thing is kinda terribad tho.. uses sqlite db  https://github.com/flowroute/sms-verification
23:46.20Slade1 per second mms,  5 per second sms
23:47.33SamotAnd do you think you're be sending 5 SMS per second for your app verification?
23:49.53Reinhildedo you think you'll be using MMS?
23:50.23Sladeit's not unreasonable to think that 5 people could try to use it at the same time.  but no way it'd be sustained 5per second.   but they require you to handle all the throttling and queuing on your side, which makes it a much bigger challenge.
23:50.45SamotBut would they all do it at the same time?
23:50.54SamotWould all 5 be at 01:00:01
23:51.06SamotOr will they be 01:00:01, 01:00:03
23:51.08SamotEtc.
23:51.44Sladegiven the nature of the app. I think 40-50 people could try within a 1 minute timeframe. giving it a statistical possibility
23:52.22SamotThat's over 1 second per user then.
23:52.31SamotThe rate limit is per second, not per minute.
23:52.32Sladeassuming they all space it out, yes
23:52.51SamotDo you have this in place now?
23:52.58MLCA lot depends on how Flowroute measures.  If they take an average over in a few seconds you'd be fine
23:52.59Sladeno. i'm implementing it now
23:53.11SladeMLC, it doesnt. it drops ones over the 5 per second
23:53.11SamotAnd how many users will be using this app to start with?
23:53.20MLCtacky
23:53.34SamotIt's no different then Calls Per Second.
23:53.38SamotVoice has CPS.
23:54.02MLCpretty simple to drop the requests into a database or file and have a background process to read them from there and send to FlowRoute
23:54.20MLCbut the additional development cost might not be worth it vs something like Twilio
23:54.42Sladeit's not a large number of users, it's just a small number of users that will all be signing up together at the same time.. anyhow. i guess i can take the time to deal with the rejections
23:55.03SamotYou think you'll have 5 requests being sent in a single second?
23:55.06SladeMLC, or if i use a precanned system that already does this. i dont have to worry about it ;0
23:55.19SamotThat's going to be the rarity.
23:55.35MLCtrue, but precanned takes some much of the fun out of it!
23:55.40Sladeindeed. it'll happen, but not very often. maybe its just ok to deal with it
23:55.46SladeMLC, i dont like inventing my own security :P

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