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| 09:13.26 | litn | hello. I'd like to be able to get see how many unheard voicemails an extension has, via either sql or by ami or I guess by any means |
| 09:13.31 | litn | how do I go about doing it? |
| 09:19.27 | litn | nvm, I got it |
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| 10:16.50 | Maliuta_ | did you try doing an 'ls | wc -l' on the directory? :) |
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| 14:52.32 | Forbidd3n | Samot: you around? |
| 14:55.07 | Forbidd3n | I am now getting around to implement what you suggested on Friday for sip headers. I currently pass this with Twilio SDK $dial->sip('sip:x.x.x.x?SESSID=TestingSessionId'); and then on my custom extension I have the following -- exten => 1000,1,SET(SESSID=${SIP_HEADER(SESSID)}) and exten => 1000,2,AGI(capture.php,${SESSID})) and when I look at the asterisk log the SESSID isn't being set -- Set("SIP/sip.twilio.com-00 |
| 14:55.07 | Forbidd3n | 000015", "SESSID=") in new stack |
| 14:56.13 | Samot | Show an actuall debug |
| 14:56.34 | Samot | Prove the header exists in the SIP message |
| 14:57.52 | Forbidd3n | Do you have a pastebin you prefer to use? |
| 14:58.13 | Samot | Just pastebin.com or whatever |
| 14:58.40 | Forbidd3n | Did you want to see the asterisk full log of the initial call coming in |
| 14:58.41 | Forbidd3n | ? |
| 14:59.13 | [TK]D-Fender | * CLI w/ SIP debug enabled |
| 14:59.24 | [TK]D-Fender | never "log files" |
| 14:59.37 | Forbidd3n | ok one sec |
| 15:03.47 | Forbidd3n | Ok I looked at the CLI log and using the Twilio SDK concept to pass custom headers I do not see my header variable name in the CLI debug. -- https://www.twilio.com/docs/api/twiml/sip |
| 15:07.21 | Forbidd3n | Samot: https://pastebin.com/BEQAK8Q6 |
| 15:07.44 | Samot | X-SESSID: sdfdfadfdsfdsfdgdgd |
| 15:07.57 | Samot | Please read all of Twilio documents, they use X headers. |
| 15:08.17 | Samot | So they prepend X- to your header name. |
| 15:08.39 | Forbidd3n | Samot: Sorry I didn't see that in the documentation |
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| 15:09.49 | Forbidd3n | Wouldn't I see X-SESSID in the log headers? |
| 15:10.01 | Samot | No. |
| 15:10.19 | Samot | Custom SIP headers are not looked at by Asterisk unless you tell it to look for them. |
| 15:10.40 | Samot | SET(SESSID=${SIP_HEADER(SESSID)}) |
| 15:10.53 | Samot | There is no SIP header named SESSID |
| 15:10.56 | Samot | It's X-SESSID |
| 15:11.00 | Forbidd3n | I understand |
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| 15:12.56 | Forbidd3n | Samot: Any good books you can recommend on learning Asterisk? |
| 15:13.56 | Samot | ~book |
| 15:13.56 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:14.01 | Forbidd3n | Thanks |
| 15:14.06 | Samot | That's the closest you're going to get. |
| 15:14.13 | Samot | There's nothing current. |
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| 16:45.30 | Samot | Yo, [TK]D-Fender. You ever use Local channels to add a caller to a queue? |
| 16:46.02 | [TK]D-Fender | to do the adding? Or adding a Local channel as a member? |
| 16:46.56 | Samot | More like instead of taking SIP/caller-channel and do a Queue() for that.. |
| 16:47.17 | [TK]D-Fender | wording is still a bit off... |
| 16:47.21 | Samot | I want to call waiting to be Local/exten@context... |
| 16:47.42 | [TK]D-Fender | "to call waiting"? |
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| 16:48.01 | Samot | I'm trying to see if I can push a "placeholder" caller in the queue.. |
| 16:48.03 | [TK]D-Fender | your phrasing is pretty jumbled right now... |
| 16:48.15 | [TK]D-Fender | You can have any channel join a queue |
| 16:48.21 | Samot | Right |
| 16:48.24 | [TK]D-Fender | Local is no different |
| 16:48.34 | [TK]D-Fender | you can use that as a call-back placeholder, or whatever you want |
| 16:48.36 | Samot | I'm wondering if you've done it to get an idea of what happens.. |
| 16:48.37 | [TK]D-Fender | dialplan is dialplan |
| 16:48.52 | [TK]D-Fender | it'll do exactly what you should expect |
| 16:48.56 | Samot | So it won't dial Local/exten@context until it dials the agent? |
| 16:49.28 | [TK]D-Fender | if you have a local channel as a member it'll go through that for as much time as it is permitted. If added as a "caller" to Queueu() it will wait just eh same as any other channel |
| 16:49.32 | Samot | And yes, I should have just call "callback placeholder" |
| 16:49.59 | [TK]D-Fender | You're referring to a local channel as a MEMBER then, not as a CALLER |
| 16:50.04 | Samot | No. |
| 16:50.11 | Samot | I want a caller to breakout |
| 16:50.16 | Samot | Select an option to hold their place.. |
| 16:50.22 | Samot | Enter their phone number.. |
| 16:50.37 | [TK]D-Fender | oops. |
| 16:50.48 | [TK]D-Fender | Yeah I somehow reversed that. |
| 16:50.49 | Samot | I can dump a call file but that would be time based.. |
| 16:51.16 | Samot | And when the call leaves it changes the positions.. |
| 16:51.34 | [TK]D-Fender | Not sure if there is an "exit position" for the caller when Queue uses an exit context |
| 16:51.41 | ack-nack | Am I right that there's no way to create a bona-fide dialplan function? i.e. gosub & the GOSUB_RETVAL function is as close as I'll get? |
| 16:51.53 | Samot | So if that dump out at position 5 I want to throw a local channel in position 5 to hold and then call back |
| 16:51.54 | [TK]D-Fender | if you don't have that then you'll need a tracker to determine that and use the position option to place your call-back in priority. |
| 16:51.55 | igcewieling | Have you considered using a ConfBridge joined to the queue, then you can put callers into or out of the bridge without Queue knowing anything. |
| 16:52.03 | Samot | Yes. |
| 16:52.04 | Samot | There is |
| 16:52.19 | Samot | I can get the callers position when they breakout |
| 16:52.25 | Samot | That's not a problem. |
| 16:52.33 | Samot | I can get all the queue details and the position when they breakout |
| 16:52.39 | Samot | It's trying to figure out how to hold them. |
| 16:52.46 | [TK]D-Fender | Then you're in the realm of jsut setting that position and originating the call-back |
| 16:53.26 | [TK]D-Fender | not sure I follow the "hold" I was getting the impression you wanted to the caller to be able to exit and hangup and get called back |
| 16:53.32 | [TK]D-Fender | What did I miss here? |
| 16:53.32 | Samot | Yes. |
| 16:53.42 | Samot | OK, I'm in position 5 |
| 16:53.48 | Samot | There are 4 other callers in front of me.. |
| 16:53.53 | Samot | Who knows how many after me.. |
| 16:53.56 | Samot | I breakout |
| 16:54.02 | Samot | I enter my callback number. |
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| 16:54.09 | Samot | The moment I break out, I'm not longer position 5 |
| 16:54.16 | Samot | The person in 6 is now 5 |
| 16:54.54 | [TK]D-Fender | You then Originate a Local-channel passing that number & position. You have that channel answer and loop a "Press X to proceed to dial the client". You then hang up on the caller. |
| 16:55.16 | Samot | OK.. |
| 16:55.24 | [TK]D-Fender | The queue goes around and around. Then an agent takes that call. They hear the looping message and press the ACK option to proceed with the dialout |
| 16:55.30 | Samot | Right |
| 16:55.35 | [TK]D-Fender | You need to do an ACK for this concept |
| 16:55.40 | Samot | OK, that's what I needed to know.. |
| 16:55.45 | [TK]D-Fender | because that Dialplan is active while in queue |
| 16:55.50 | Samot | The queue member ask to confirm taking the call. |
| 16:55.52 | [TK]D-Fender | yup |
| 16:55.54 | Samot | Right. |
| 16:56.00 | Samot | That's what I wanted to confirm.. |
| 16:56.19 | Samot | the second leg is in the queue holding... |
| 16:59.32 | Samot | When the caller leaves the queue, hangup or breakout.... |
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| 17:00.38 | Samot | ${QUEUE_PRIO}, ${QUEUE_MAX_PENALTY}, ${QUEUE_MIN_PENALTY}, ${QUEUESTATUS}, ${QUEUEPOSITION} are dumped as dialplan variables |
| 17:01.26 | Samot | 0 = completed call for QUEUEPOSITION |
| 17:01.44 | Forbidd3n | Samot: the other day you stated that when I do a sip call through Twilio SDK the call is now on the PBX server. I would expect the call to stay open until I hang up the phone or call the Hangup method for Asterisk. It doesn't. It does run through the extension items, but then the call is dropped on Twilio's end. I tried a simple example in the book you recommend. In the CLI it does trigger the answer, playback and |
| 17:01.45 | Forbidd3n | the hangup, but it doesn't play the hello world audio. |
| 17:01.50 | Forbidd3n | exten => 2000,1,Answer() |
| 17:01.50 | Forbidd3n | exten => 2000,n,Playback(hello-world) |
| 17:01.51 | Forbidd3n | exten => 2000,n,Hangup() |
| 17:02.01 | Forbidd3n | that is what I have for my extension setup. |
| 17:02.42 | Samot | When you say it doesn't playback the audio... |
| 17:02.47 | Samot | How are you determining that? |
| 17:02.56 | Samot | You mean you don't hear it on the call? |
| 17:02.56 | Forbidd3n | I don't hear it on the phone. |
| 17:02.58 | Samot | OK |
| 17:03.05 | Forbidd3n | It seems to play it in the CLI |
| 17:03.10 | Forbidd3n | or says it did I mean |
| 17:03.14 | [TK]D-Fender | Go prove audio is being sent to the right palce |
| 17:03.19 | Samot | So if Asterisk says it's playing, it's playing. |
| 17:03.27 | [TK]D-Fender | And always show us the call. That dialplan doesn't prove anything |
| 17:03.29 | Samot | Yup, sounds like an SDP issue. |
| 17:03.40 | Samot | So again, full SIP debug so we can see the SDP stuff |
| 17:03.52 | Samot | And don't "clean" it |
| 17:03.57 | [TK]D-Fender | "Here's a brochure for my car. Now tell my why it crashed into that wall" |
| 17:04.02 | Forbidd3n | [TK]D-Fender: what do you mean 'being sent to the right place' ? |
| 17:04.03 | Samot | You mean scrub needed data. |
| 17:04.03 | [TK]D-Fender | me* |
| 17:04.17 | Samot | RTP requires an IP:PORT |
| 17:04.25 | [TK]D-Fender | <Forbidd3n> [TK]D-Fender: what do you mean 'being sent to the right place' ? <- where do we see what IP ADDRESS it's thinking it should send the audio to? |
| 17:04.25 | Samot | If that is wrong, no audio |
| 17:04.41 | [TK]D-Fender | Forbidd3n, Look at the CALL. |
| 17:04.50 | Samot | This is basic troubleshooting |
| 17:05.46 | Forbidd3n | I will post the CLI |
| 17:05.57 | [TK]D-Fender | with SIP debug enabled |
| 17:05.59 | [TK]D-Fender | ^^^^ |
| 17:06.06 | Forbidd3n | yeah |
| 17:06.08 | ack-nack | found the issue in the car brochure: * Self driving |
| 17:08.02 | Forbidd3n | https://pastebin.com/wq1r0ARs |
| 17:09.02 | Samot | Well no kidding. |
| 17:09.25 | Forbidd3n | ? |
| 17:09.34 | Samot | SIP/2.0 403 Forbidden |
| 17:09.43 | Forbidd3n | I just seen that as well. |
| 17:09.56 | Samot | No matching peer for '+19162439854' from '54.172.60.0:5060' |
| 17:10.07 | Samot | SIP/2.0 603 Declined |
| 17:11.04 | Samot | [2018-03-19 13:05:44] WARNING[23445][C-00000036]: chan_sip.c:17293 check_auth: username mismatch, have <Twilio Truck>, digest has <1000> |
| 17:11.04 | Samot | [2018-03-19 13:05:44] NOTICE[23445][C-00000036]: chan_sip.c:26361 handle_request_invite: Failed to authenticate device "+19162439854" <sip:+19162439854@sip.twilio.com>;tag=08181152_6772d868_772be938-8a2e-41a4-a1c6-b23737132a38 |
| 17:11.14 | Samot | So you have your Twilio trunks authing?! |
| 17:11.17 | Samot | For incoming?! |
| 17:11.21 | Forbidd3n | Yes. |
| 17:11.25 | Samot | And then you don't have one of their IPs.. |
| 17:11.30 | Samot | Well then make it work right |
| 17:11.33 | [TK]D-Fender | Found peer 'Twilio Truck' for '+19162439854' from 54.172.60.1:5060 |
| 17:11.38 | Samot | Because it's broken. |
| 17:11.47 | [TK]D-Fender | they are getting challenged by * |
| 17:12.06 | Samot | You should probably confirm your Twilio peers are actually working right. |
| 17:12.26 | Forbidd3n | So my inbound route isn't accepting twilio's request basically |
| 17:12.31 | Samot | Why are you authing the inbound?! |
| 17:12.42 | Samot | Are you not expecting them? |
| 17:12.57 | Samot | No one auths their providers incoming calls |
| 17:13.25 | [TK]D-Fender | <Forbidd3n> So my inbound route isn't accepting twilio's request basically <- what do you mean "inbound route"? |
| 17:13.26 | Forbidd3n | Samot: ok, this is new to me. I saw it in their docs somewhere so I provided it. |
| 17:13.37 | Samot | What is new to you? |
| 17:18.18 | Forbidd3n | Ok I removed the authentication and now just getting the 603 declined error |
| 17:18.31 | Samot | Because you don't have a peer for that IP |
| 17:18.38 | Samot | Twilio sends calls from like 5 or so IPs |
| 17:18.44 | Samot | You have to have a peer for each IP |
| 17:18.47 | Samot | It's in their documents. |
| 17:20.34 | Forbidd3n | New to Asterisk, IVR and FreePBX |
| 17:20.58 | Forbidd3n | What do you mean by peer? |
| 17:22.38 | [TK]D-Fender | That SIP thing you made <- |
| 17:22.51 | [TK]D-Fender | The entry that defines the other side |
| 17:23.52 | Forbidd3n | [TK]D-Fender: the other side, meaning something I have to setup to allow in FreePBX? |
| 17:24.12 | [TK]D-Fender | the thing you define your connection to your provider with |
| 17:27.12 | Forbidd3n | sorry for my confusion... :/ |
| 17:27.41 | [TK]D-Fender | Also... you are using FreePBX and its GUI terms are not appropriate here |
| 17:27.55 | [TK]D-Fender | your TRUNK defines how to auth incoming and outgoing calls |
| 17:28.27 | Forbidd3n | ok |
| 17:28.51 | [TK]D-Fender | Call comes in and your PEER you defined in your trunk initially matches the caller ... and then your definition is set to challenge them. they do not accept being challenged ad don't respond with auth and your system tells them to go away |
| 17:29.30 | [TK]D-Fender | "Inbound route" has nothing to do with this. That tells your system where to send the call based on the # theya re requesting. However they are not accepted as a sender period |
| 17:29.50 | [TK]D-Fender | "I don't know & trust you so I don't care what you want" |
| 17:31.15 | Forbidd3n | TK - I have peer details setup that authenticate the call coming in from Twilio the way they specify in their documentation |
| 17:31.32 | Forbidd3n | under peer details I have user, type, secret and host |
| 17:39.37 | [TK]D-Fender | Their documentation may be wrong. |
| 17:39.40 | [TK]D-Fender | LINK IT |
| 17:40.02 | [TK]D-Fender | And there is always more you should be specifying at the very lest, also including CODEC's, etc |
| 17:40.42 | Forbidd3n | https://www.twilio.com/resources/images/docs/UsingFreePBXwithTwilioElasticSIPTrunking-06152015.pdf |
| 17:40.51 | Forbidd3n | Page 6 |
| 17:42.58 | [TK]D-Fender | Yup.... just wrong. |
| 17:43.25 | [TK]D-Fender | Provider guides are often wrong if not simply outdated |
| 17:43.40 | [TK]D-Fender | "insecure=port,invite" |
| 17:43.49 | [TK]D-Fender | "disallow=all" |
| 17:43.54 | [TK]D-Fender | "allow=YOURCODEC" |
| 17:44.09 | [TK]D-Fender | fill that in with what you should be using with them between alaw or ulaw |
| 17:44.45 | Samot | Guys. |
| 17:44.53 | Samot | Page 6 is for OUTBOUND |
| 17:45.16 | Forbidd3n | correct |
| 17:45.25 | Samot | Please go to page 8 |
| 17:45.36 | Forbidd3n | so it authenticates with Twilio with the peer settings |
| 17:45.38 | Samot | Configuring Your Twilio Origination (inbound) Trunks in FreePBX <-- |
| 17:46.09 | [TK]D-Fender | page 8 is ALSO fucked |
| 17:46.15 | Samot | Twilio sends calls from 3-4 IP addresses, depending on your region. |
| 17:46.15 | Samot | We are going to create a separate peer for each IP address, something similar to this: |
| 17:46.28 | [TK]D-Fender | using a "user context" .. with "type=peer" = BULLSHIT |
| 17:46.40 | *** join/#asterisk pchero (~pchero@x1-6-d8-d7-75-5e-99-79.cpe.webspeed.dk) |
| 17:46.50 | [TK]D-Fender | And doesnt' fix the auth issue |
| 17:47.08 | [TK]D-Fender | Add the lines I gave you and be prepared to enter trunk information for their other peers |
| 17:49.06 | Forbidd3n | Samot: so do I just repeat what is in USER DETAILS for each IP Address? |
| 17:49.14 | [TK]D-Fender | yes |
| 17:49.57 | [TK]D-Fender | Also there is an assumed inherited NAT setting |
| 17:50.02 | [TK]D-Fender | they should also be "nat=no" |
| 17:50.11 | [TK]D-Fender | So we've gone from 3 lines each to 7 just to start. |
| 17:50.14 | [TK]D-Fender | And..... |
| 17:50.27 | [TK]D-Fender | you haven't prevented re-invites while could also screw audio of bridged calls |
| 17:50.33 | [TK]D-Fender | "directmedia=no" |
| 17:50.37 | [TK]D-Fender | so now 8 lines. |
| 17:50.40 | [TK]D-Fender | and they show 3. |
| 17:50.43 | [TK]D-Fender | Shitty guide |
| 17:52.43 | Forbidd3n | [TK]D-Fender: do I need to setup a new trunk for each incoming peer or just put them all in the one incoming peer trunk? |
| 17:53.18 | Forbidd3n | I am guessing a new trunk for each one |
| 17:54.23 | *** join/#asterisk pchero (~pchero@x1-6-d8-d7-75-5e-99-79.cpe.webspeed.dk) |
| 17:56.01 | [TK]D-Fender | 1 each |
| 17:57.38 | Forbidd3n | so then I need to setup a new route for each one of these. It seems a lot of duplicate stuff |
| 17:59.52 | Forbidd3n | [TK]D-Fender: ^ |
| 18:01.33 | *** join/#asterisk pchero (~pchero@x1-6-d8-d7-75-5e-99-79.cpe.webspeed.dk) |
| 18:02.57 | Forbidd3n | nvm I can select multiple trunks per outbound route |
| 18:04.06 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 18:08.57 | *** join/#asterisk pchero (~pchero@x1-6-d8-d7-75-5e-99-79.cpe.webspeed.dk) |
| 18:13.20 | [TK]D-Fender | no |
| 18:13.32 | [TK]D-Fender | you use 1 trunk for outbound period |
| 18:15.57 | *** join/#asterisk sekil (~sekil@cable-89-216-232-238.dynamic.sbb.rs) |
| 18:16.40 | Forbidd3n | [TK]D-Fender: I thought I had to setup multiple trunks for each Twilio IP address |
| 18:16.58 | [TK]D-Fender | to match their inbound |
| 18:17.05 | [TK]D-Fender | not outbound |
| 18:17.56 | Forbidd3n | ok so one inbound route for each trunk setup |
| 18:18.02 | [TK]D-Fender | no |
| 18:18.38 | *** join/#asterisk craigify (~craigify@ip184-186-18-123.no.no.cox.net) |
| 18:19.16 | Forbidd3n | so on page 9 of that pdf do I put all the hosts in the user details with a break between each one? |
| 18:19.44 | [TK]D-Fender | NO |
| 18:20.14 | Forbidd3n | Ok, I currently have 4 trunks, one for each Twilio IP |
| 18:23.39 | [TK]D-Fender | 1 trunk per pace that is SENDING YOU CALLS |
| 18:23.51 | [TK]D-Fender | INBOUND ROUTE is to match what NUMBEr they are sending |
| 18:23.59 | [TK]D-Fender | you pay for ONE phone number right? |
| 18:24.03 | [TK]D-Fender | that should mean ONE inbound toure |
| 18:24.06 | [TK]D-Fender | route* |
| 18:26.46 | Forbidd3n | ok, I have the inbound route setup. I have the destination set to hit the inbound trunk, but I can only select one trunk |
| 18:28.16 | Forbidd3n | [TK]D-Fender: I originally had my inbound connecting to a custom extension |
| 18:28.36 | Forbidd3n | now I have the destination set to the trunk |
| 18:29.28 | [TK]D-Fender | that makes no sense |
| 18:45.26 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il) |
| 18:49.11 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 19:00.51 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 19:03.29 | *** join/#asterisk paulgrmn (~paulgrmn@198-0-107-153-static.hfc.comcastbusiness.net) |
| 19:29.35 | Forbidd3n | [TK]D-Fender: do you know why I would be getting this error? ERROR[60874]: res_pjsip.c:3019 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' |
| 19:29.57 | Forbidd3n | I am using Chan_SIP not PJSIP |
| 19:30.27 | [TK]D-Fender | that is clearly hitting the chan_pjsip port |
| 19:30.39 | [TK]D-Fender | "I am using" = meaningless. |
| 19:30.46 | [TK]D-Fender | it's hitting the port it is hitting |
| 19:31.33 | [TK]D-Fender | And you seem to be moving the target |
| 19:33.04 | Forbidd3n | [TK]D-Fender: I went through that PDF 100% and now I am getting this error. Not trying to be a moving target. |
| 19:33.43 | [TK]D-Fender | nothing is telling me WHAT is trigger that message |
| 19:33.53 | [TK]D-Fender | Including anything saying it has anything to do with your provider |
| 19:34.02 | [TK]D-Fender | Because I'm not seeing DEBUG to match |
| 19:34.18 | [TK]D-Fender | Have we solved the last problem we spent time on yet? |
| 19:35.35 | Forbidd3n | The last problem was getting it to play the playback. I setup the peers as they were in the PDF I found. This is where I am atm |
| 19:35.41 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 19:36.04 | Forbidd3n | This is in my CLI - [2018-03-19 15:34:16] WARNING[62108][C-00000001]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 54.172.60.3" |
| 19:36.20 | [TK]D-Fender | ... |
| 19:36.20 | Forbidd3n | I have an incoming trunk that allows this peer |
| 19:36.33 | [TK]D-Fender | PASTEBIN THE WHOLE COMM |
| 19:36.58 | [TK]D-Fender | You are wasting our time with single-line C&P's like that |
| 19:38.18 | Forbidd3n | https://pastebin.com/63yc7BY9 |
| 19:38.48 | [TK]D-Fender | No matching peer for '+19162439854' from '54.172.60.3:5060' |
| 19:38.59 | [TK]D-Fender | Whatever peer you thought you created ... doesn't match them |
| 19:39.43 | Forbidd3n | do I need to add the port number to each host peer I have setup? |
| 19:40.40 | Forbidd3n | I have that peer setup for incoming -- |
| 19:40.45 | Forbidd3n | type=peer |
| 19:40.45 | Forbidd3n | host=54.172.60.3 |
| 19:40.45 | Forbidd3n | context=from-trunk |
| 19:42.07 | Forbidd3n | [TK]D-Fender: ^ |
| 19:42.08 | [TK]D-Fender | "sip show peers" <- |
| 19:42.17 | [TK]D-Fender | And I told you to add a LOT MORE TO THAT |
| 19:44.46 | Forbidd3n | [TK]D-Fender: I have four trunks setup TwilioIn1-4 with IP address 54.172.60.0-3 -- It is only picking up the first one |
| 19:44.47 | Forbidd3n | Twilioln1 54.172.60.0 No No 5060 Unmonitored |
| 19:44.57 | [TK]D-Fender | Because you screwed up the others |
| 19:45.05 | [TK]D-Fender | NOTHING belongs in the "incoming" sections. |
| 19:45.10 | [TK]D-Fender | PEER ONLY <- |
| 19:45.17 | Forbidd3n | one sec |
| 19:45.20 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 19:45.25 | [TK]D-Fender | go fix all of them ad you'd better include all of those things I told yuo to add |
| 19:48.19 | Forbidd3n | Ok I have those fixed now |
| 19:48.26 | Forbidd3n | sip show peers shows them all |
| 19:48.30 | Forbidd3n | Here is my recent CLI |
| 19:48.31 | Forbidd3n | https://pastebin.com/DBfHsA0P |
| 19:48.34 | Forbidd3n | [TK]D-Fender: |
| 19:49.18 | [TK]D-Fender | Looking for s in from-trunk (domain 74.85.131.124) |
| 19:49.38 | [TK]D-Fender | and now it's matching your trunk and looking for "s" |
| 19:50.01 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:1ae:fea1:4845:577a) |
| 19:50.02 | [TK]D-Fender | because your registration didn't specify a number to dial |
| 19:50.14 | [TK]D-Fender | user:pass@host/YOURDIDNUMBERHERE |
| 19:51.03 | *** join/#asterisk miralin (~Thunderbi@194.8.128.64) |
| 19:51.10 | [TK]D-Fender | "core set verbose 10" <- also you should always have this enabledwhen collecting debug |
| 19:52.58 | Forbidd3n | so would the number to dial be the extension number I have setup? |
| 19:53.19 | [TK]D-Fender | no, it should the number you PAY them for |
| 19:53.30 | [TK]D-Fender | YOURDIDNUMBERHERE <- what part of this was unclear |
| 19:54.26 | Samot | JFC. |
| 19:54.28 | Samot | Stop. |
| 19:54.29 | Forbidd3n | [TK]D-Fender: Just trying to understand, sorry for my ignorance. I have the DID Number set in my inbound routes |
| 19:54.46 | [TK]D-Fender | Well they aren't ASKING for that number |
| 19:54.46 | Samot | I've said this already, Twilio does NOT DO INBOUND AUTH |
| 19:54.50 | [TK]D-Fender | so fix your trunk registration |
| 19:55.00 | Samot | The Inbound "trunks" are not "Outbound" trunks. |
| 19:55.00 | [TK]D-Fender | Samot, We just got that resolved apparently |
| 19:55.06 | Samot | No |
| 19:55.10 | Forbidd3n | Samot: I don't have auth set for inbound |
| 19:55.12 | Samot | There's no f'ing REGISTER string |
| 19:55.27 | Samot | What are we looking at an inbound call? |
| 19:55.33 | [TK]D-Fender | if he isn't registering it should be part of the URI they are told to dial to him |
| 19:55.35 | Samot | s/What/Why/ |
| 19:56.00 | [TK]D-Fender | because that's what we're trying to deal with |
| 19:56.00 | Samot | Twilio does straight up SIP URI calling... |
| 19:56.05 | Samot | OK |
| 19:56.06 | [TK]D-Fender | So go fix the URI |
| 19:56.10 | Samot | so there's no REGISTER string. |
| 19:56.27 | Samot | It's sip:NUMBER@IP:PORT |
| 19:56.30 | [TK]D-Fender | and disregard my previous comment about REGISTER in your trunk if you are providing them a fixed IP |
| 19:57.00 | Forbidd3n | Samot: this is the sip call I am making with Twilio through Twiml code - $dial->sip('sip:74.85.131.124:5160?SESSID=TRSASGSADGDSDESX'); |
| 19:57.21 | Samot | There's no user |
| 19:57.25 | Samot | JFC. |
| 19:57.30 | Samot | ARe you new to SIP as well? |
| 19:58.17 | Forbidd3n | Samot: Yes, working on learning it with simple testing and the documentation online isn't that well documented imo |
| 19:58.31 | Samot | Well |
| 19:58.39 | Samot | Why are you trying to write SIP applications? |
| 19:58.55 | Samot | Because Twilio is a SIP development provider. |
| 19:59.14 | Samot | It's meant more for those wanting to use SIP applications, APIs, etc. |
| 19:59.28 | Samot | They kinda figure you're not new to the whole SIP game. |
| 19:59.49 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:00.50 | Forbidd3n | I have complete IVR written using Twilio, the only issue is that Twilio isn't PCI compliant. So I spun up a new server, installed FreePBX and Asterisk so when the user gets to a point in the call to ask for PCI data, I send the call to the PBX server, collect it, store it, then go back to the call. When I need to process the transaction I will send API call to retrieve the data the user inputed with DTMF tones and |
| 20:00.50 | Forbidd3n | process the transaction. |
| 20:01.02 | Samot | INVITE sip:74.85.131.124:5160 SIP/2.0 |
| 20:01.08 | Samot | ^^ That's your INVITE |
| 20:01.11 | Samot | Completely wrong. |
| 20:01.15 | [TK]D-Fender | no target # in there |
| 20:01.20 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:01.20 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:01.34 | Samot | $dial->sip('sip:NUMBER@74.85.131.124:5160?SESSID=TRSASGSADGDSDESX'); |
| 20:01.37 | Samot | ^^^ THAT |
| 20:01.40 | Samot | FFS. |
| 20:02.06 | Forbidd3n | Would number be the extension on the PBX box I want it to call when it connectS? |
| 20:02.17 | Samot | facepalms |
| 20:02.20 | Samot | YES |
| 20:02.43 | Forbidd3n | Samot: I just want to confirm what I am understanding is all. I don't just want answers. :) |
| 20:03.00 | Samot | This is basic SIP 101 |
| 20:03.20 | Samot | You're writing an application without understanding the basics of what you are writing the application for. |
| 20:03.31 | [TK]D-Fender | <Forbidd3n> Would number be the extension on the PBX box I want it to call when it connectS? <- YOUR FUCKING PHONE NUMBER YOU ARE PAYING THEM FOR |
| 20:03.44 | Samot | No. |
| 20:03.46 | Samot | Actually. |
| 20:03.48 | [TK]D-Fender | yOU WANT TO KNOW IT IS them CALLING YOU BECAUSE YOU HAVE A CALL FROM THE NUMBER YOU ARE paying them for |
| 20:03.50 | Samot | He's using the SDK |
| 20:03.55 | Samot | He's originating a call basically. |
| 20:03.56 | Forbidd3n | Samot: correct |
| 20:04.07 | Samot | It can be destination he wants |
| 20:04.24 | Samot | 1) The guide he referred to is for using their Elastic SIP trunks |
| 20:04.34 | Samot | 2) This is a Programmable SIP Trunk using the SDK |
| 20:04.40 | [TK]D-Fender | Why is his app going through an ITSP not related to a DID service? |
| 20:04.40 | Samot | So, not really the same thing. |
| 20:04.42 | igcewieling | eats more popcorn |
| 20:04.55 | Samot | Because TWILIO is where the all lives. |
| 20:05.09 | Samot | He's only using Asterisk for the part of the call that needs PCI compliance. |
| 20:05.26 | Samot | So unless the call needs PCI compliance, Asterisk is never used. |
| 20:05.30 | Samot | It stays at Twilio. |
| 20:05.34 | Samot | Twilio is not a standard ISP |
| 20:05.38 | Samot | Twilio is not a standard ITSP |
| 20:05.46 | Samot | They are an applications platform. |
| 20:05.59 | [TK]D-Fender | Whateve. Pick a number. Make an inbound route to match that # |
| 20:06.18 | [TK]D-Fender | Don't care if it's 12345\ |
| 20:06.27 | Samot | Right |
| 20:06.35 | Samot | He could do the dial string and sent it to 100@IP:port |
| 20:07.11 | [TK]D-Fender | just make it unique so you don't go and stupidly confuse with an Extension or anything else on your system |
| 20:09.33 | Forbidd3n | Samot: so the 100 would be an extension I would have setup to run AGI scripts to collect the DTMF, correct? |
| 20:09.47 | Samot | Sure. |
| 20:09.54 | Samot | I'm just using it as an example. |
| 20:10.14 | Forbidd3n | Ok, let me put in the custom extension I have setup and see what the CLI says then |
| 20:10.48 | [TK]D-Fender | NO. |
| 20:10.53 | [TK]D-Fender | do not use an EXTENSION NUMEBR |
| 20:11.12 | [TK]D-Fender | pick something FUCKING UNIQUE. You're confused enough as it is. |
| 20:11.41 | Forbidd3n | Ok |
| 20:12.05 | [TK]D-Fender | [TK]D-Fender> just make it unique so you don't go and stupidly confuse with an Extension or anything else on your system |
| 20:14.35 | Forbidd3n | Current CLI - https://pastebin.com/QqZ9Sf5h |
| 20:14.42 | Forbidd3n | Samot, [TK]D-Fender |
| 20:15.24 | [TK]D-Fender | <[TK]D-Fender> "core set verbose 10" <- also you should always have this enabledwhen collecting debug |
| 20:15.27 | [TK]D-Fender | .... |
| 20:15.31 | [TK]D-Fender | FOLLOW DIRECTIONS |
| 20:16.01 | Forbidd3n | [TK]D-Fender: I have it set to 10 |
| 20:16.26 | [TK]D-Fender | that is not * CLI at verbose 10 |
| 20:16.28 | Forbidd3n | not sure how it got turned back off |
| 20:16.46 | Forbidd3n | probably by me applying config in FreePBX - I will repost |
| 20:16.48 | Forbidd3n | sorry |
| 20:18.03 | Forbidd3n | https://pastebin.com/Jy8nKVNv |
| 20:19.03 | Forbidd3n | so the number before the @ has to be the DID then and not just any number - Executing [100@from-trunk:2] NoOp("SIP/Twilioln4-00000007", "Received an unknown call with DID set to 100") in new stack |
| 20:20.02 | [TK]D-Fender | Clearly don't have an inbound route to match that # |
| 20:20.44 | Forbidd3n | I have an inbound route that matches the DID number I have with Twilio |
| 20:20.57 | [TK]D-Fender | [100 <----------------------- THIS FUCKING NUMBER |
| 20:21.01 | [TK]D-Fender | the one they are SENDING |
| 20:21.44 | Forbidd3n | Ok, so you were correct, the number before the @ has to be the inbound DID number not the 100/any number that I thought Samot said |
| 20:24.27 | Samot | I said it was an example. |
| 20:24.38 | Samot | You still need to be able to accept what you send on the other side. |
| 20:24.40 | Samot | Again, SIP 101 |
| 20:25.02 | Forbidd3n | Samot: ok |
| 20:25.54 | Forbidd3n | I have set the request like this now -- $dial->sip('sip:+18044098564@74.85.131.124:5160?SESSID=TRSASGSADGDSDESX'); |
| 20:25.57 | *** join/#asterisk kharwell (~kharwell@user-24-214-15-130.knology.net) |
| 20:25.57 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 20:26.22 | Forbidd3n | Current CLI - https://pastebin.com/pf4tKkMy |
| 20:27.13 | *** join/#asterisk elguero (~miguel323@74-95-21-41-Connecticut.hfc.comcastbusiness.net) |
| 20:28.00 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 20:28.01 | [TK]D-Fender | -- Executing [+18044098564@from-trunk:16] Goto("SIP/Twilioln4-00000008", "twilio-custom,1000,1") in new stack |
| 20:28.07 | [TK]D-Fender | <PROTECTED> |
| 20:28.15 | [TK]D-Fender | It's a MIRACLE |
| 20:28.43 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:1ae:fea1:4845:577a) |
| 20:28.45 | [TK]D-Fender | So you called a PHP script as an AGI |
| 20:28.53 | [TK]D-Fender | Which did NOT answer the call |
| 20:29.48 | [TK]D-Fender | given how massively you've managed to get everything wrong up until this point I'm scared to think of how bad that AGI might look |
| 20:29.52 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 20:33.20 | [TK]D-Fender | packs up to head home |
| 20:34.25 | Forbidd3n | Samot: Now that I have the connection, can you tell me why it isn't playing the audio back? |
| 20:34.26 | Forbidd3n | https://pastebin.com/BqzG7SPv |
| 20:35.55 | Forbidd3n | sorry wrong pastebin - this one |
| 20:35.56 | Forbidd3n | https://pastebin.com/a8nYeQpe |
| 20:36.18 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 20:37.50 | Samot | I have no clue. |
| 20:37.59 | Samot | You're initiating a NEW call from Twilio.. |
| 20:38.08 | Forbidd3n | I mean it plays it in the logs, but I don't hear it |
| 20:38.10 | Samot | I don't know if you need to bridge the caller and this call togther.. |
| 20:38.11 | Samot | Right |
| 20:38.21 | Samot | I don't know if the two channels at Twilio are bridged together.. |
| 20:38.32 | Samot | You're going to need to look at Twilio.. |
| 20:38.35 | Samot | I got nothing on this. |
| 20:38.39 | Forbidd3n | Ok, thanks |
| 20:39.02 | Forbidd3n | At least I have a successful connection now I can dwell into what I have to have done |
| 20:39.05 | Forbidd3n | Thanks you again! |
| 21:13.11 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:18.41 | Forbidd3n | [TK]D-Fender: Now that I have the connection, do you know why the ulaw file plays but I can't hear it on the phone? |
| 21:18.42 | Forbidd3n | https://pastebin.com/a8nYeQpe |
| 21:24.10 | [TK]D-Fender | Unless it's a local firewall issue it's the other side |
| 21:28.36 | Forbidd3n | on the firewall the ports 10000-20000 range should be open for media, is this correct? |
| 21:30.06 | [TK]D-Fender | yes |
| 21:49.50 | Forbidd3n | [TK]D-Fender: can you point me in the right direction of what I should be looking for to collect DTMF tones from user, when they hit the pound key, go to the next item which will store the DTMF in a variable and pass to AGI script |
| 21:50.04 | [TK]D-Fender | Go read the book |
| 21:50.19 | [TK]D-Fender | You've started this having done none of the research and you have no idea what you're doing |
| 21:50.44 | [TK]D-Fender | We aren't here to write your entire setup for you |
| 21:50.54 | Forbidd3n | [TK]D-Fender: I am asking where I should start looking, not asking you how |
| 21:51.01 | [TK]D-Fender | `book |
| 21:51.02 | [TK]D-Fender | ^ |
| 21:51.05 | [TK]D-Fender | go learn the dialplan |
| 21:51.11 | [TK]D-Fender | If you intend to use AGI, it's also in the book |
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| 21:59.55 | Samot | OK, I already explained this |
| 22:00.05 | Samot | I will try again. |
| 22:00.52 | Samot | Call comes into Twilio. Twilio answers the call, does all the stuff you want to do on there. When it's time for the PCI compliance part you are initiating a NEW call. |
| 22:01.25 | Samot | There is no proof that Call A (caller to Twilio) is connecting to Call B (Twilio to Asterisk) |
| 22:03.52 | Samot | Are you actually interacting with the call once it's connected or are you just playing that recording back on answer? |
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