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| 00:38.52 | wonderworld | hey, are there any downsides in using_chan_sip with webRTC? i remember reading that webRTC isn't properly supported by chan_sip, but this was years ago. should I use chan_pjsip or is chan_sip ok? |
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| 00:47.28 | [TK]D-Fender | Chan_sip isn't getting new effort |
| 00:48.24 | [TK]D-Fender | Sounds like you're asking a question you already know the answer to but don't seem to want to take action on |
| 00:52.38 | wonderworld | pretty good analysis i guess |
| 00:52.58 | wonderworld | still couldn't start loving chan_pjsip :( |
| 01:11.55 | Samot | Then you're kinda screwed. It's the primary SIP driver for Asterisk. |
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| 13:22.07 | DannyA | does anyone have experience with Fanvil phones? I bought a bunch of them and Kobaz and I have spent an entire day trying to get them working with Asterisk to no avail. Can't get audio to come through on either end. |
| 13:23.10 | *** join/#asterisk velix (~velix@unaffiliated/velix) |
| 13:23.47 | velix | Funny, I've logged a whole night of getting 400s from Asterisk, but I didn't get the errors from PhonerLite (a sip client for Windows). I'll now have to look, what Asterisk does different: https://file.io/VWbMGY |
| 13:23.54 | velix | Perhaps it's the Username ;) |
| 13:23.58 | velix | The User Agent, sorry. |
| 13:24.21 | velix | Let me parse these timestamps, perhaps it's hourly or so |
| 13:26.29 | velix | Interesting: https://bpaste.net/raw/O6IA |
| 13:27.45 | velix | There's no real pattern: 90, 30, 90, 90, 30, 90 [minutes] |
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| 13:28.04 | velix | Perhaps Asterisk is alife and wants to tell me something. Can anyone read morse? |
| 13:28.35 | [TK]D-Fender | Well there's nothing to read in those... |
| 13:35.41 | velix | [TK]D-Fender: Yeah, the logger files are somewhere else. I'll find them soon (when homeoffice is over). |
| 13:36.09 | velix | I just had a pause and wanted to look into logs :D |
| 13:36.25 | velix | Can I make the history dump its contents? |
| 13:49.29 | *** join/#asterisk deavmi (~quassel@165.0.49.28) |
| 13:53.41 | *** join/#asterisk _ganapathi_ (~Ganapathi@2409:4072:709:9a53:6c4f:58d6:b33:c7a5) |
| 13:53.47 | _ganapathi_ | https://paste.linux.community/view/64eb13b3 |
| 13:54.40 | _ganapathi_ | this is my dialplan. am using confrbridge and defined 1,2 for customer,custinfo extension. |
| 13:55.17 | _ganapathi_ | but sometime some extra DTMF captured then it's going to i then hangup. |
| 13:55.32 | _ganapathi_ | for example if i press 11 instead of 1 as DTMF |
| 13:55.39 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 13:56.17 | _ganapathi_ | is there any chances to check conference is alive or not then use goto to create conference again. |
| 13:56.48 | _ganapathi_ | where am trying to use BRIDGEPEER to check but it's doesn't look like working.. |
| 13:58.20 | _ganapathi_ | if 1 pressed then asterisk putting back to conference by native functionality. but if we pressed other than 1 or 2 DTMF or multiple continuous dtmf then it's hangup the call. |
| 13:58.49 | _ganapathi_ | so how do validate conference is available or not by using channel variables. |
| 13:59.20 | _ganapathi_ | using asterisk 17.2.0 & 16.4.0 |
| 14:09.26 | *** join/#asterisk cryptic (~cryptic@142.196.139.17) |
| 14:10.05 | Samot | Are you supposed to have more DTMF? |
| 14:10.16 | Samot | Should 11 or 13 or 5 or 6 get you somewhere? |
| 14:15.45 | velix | That's my conference setup. Works fine, we had more than 20 conferences on it this week: https://bpaste.net/HGSA |
| 14:16.14 | velix | [TK]D-Fender: Sorry for line 15, but it only works for me with MATH. |
| 14:18.51 | [TK]D-Fender | velix, Show me your other attempt... |
| 14:18.59 | velix | [TK]D-Fender: I used the one you posted here. |
| 14:19.04 | velix | 1 sec |
| 14:19.12 | velix | (some more, my logs have been archived) |
| 14:19.39 | velix | Set(Attempts=$[{Attempts}+1]) |
| 14:20.07 | [TK]D-Fender | that should work. |
| 14:20.16 | velix | Yeah, locally yes. Practically no ;( |
| 14:20.22 | velix | Attempts kept being 1 |
| 14:20.33 | velix | logically yes* |
| 14:20.43 | [TK]D-Fender | I'd like to see a full call to validate that... |
| 14:20.59 | velix | Okay, let's do this if professor goes home :-) |
| 14:21.03 | velix | 2-3 hours |
| 14:21.09 | velix | Perhaps he wants to make a conference. |
| 14:21.10 | igcewieling | I thought $[] always evaluated to 1 or 0 |
| 14:23.36 | igcewieling | perhaps if you tried Set(Attempts={Attempts}+1) |
| 14:23.50 | igcewieling | er... perhaps if you tried Set(Attempts=${Attempts}+1) |
| 14:24.06 | velix | So $[TK]D-Fender evalutes to 0, too? ;) |
| 14:24.20 | velix | igcewieling: ok, I'll try as soon as possible. |
| 14:24.32 | igcewieling | only if it is already not set |
| 14:25.35 | [TK]D-Fender | no |
| 14:25.39 | [TK]D-Fender | $[] is for expressions |
| 14:26.02 | [TK]D-Fender | and you can't just leave that off otherwise you're get literal characters |
| 14:27.35 | igcewieling | looks like tk is right. https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 14:28.29 | velix | Okay, I'll give it a try again soon and paste the output. |
| 14:28.57 | igcewieling | I've been spending too much time using AEL |
| 14:29.01 | velix | By the way: the pins can be entered via website and soon via phone, too. But phone was too complicated (to set dates etc.) |
| 14:29.57 | velix | The quality of Google Wavenet is really good. Never thought that. |
| 14:30.17 | velix | I wonder, why nobody did a voice sampleset ... Have to ask google for permissions. |
| 14:30.43 | velix | Perhaps they don't allow to share the created waveforms. |
| 14:36.40 | velix | Can't I get rid of the "exten => s,", too ? |
| 14:40.44 | Samot | exten => s,n,GotoIf($["${INPUT}" = ""]?begin) < That's going to be an issue if you're trying to track attempts. |
| 14:46.13 | _ganapathi_ | Samot: no. I don't want more than one digits. I want to keep the conference or don't want to hangup call |
| 14:46.38 | Samot | Well them sending two digits isn't going to be accepted. |
| 14:46.46 | Samot | So you need to deal with that. |
| 14:47.08 | velix | Samot: Why? If he is silent for 10 seconds, he starts from scratch. |
| 14:47.33 | _ganapathi_ | first digit accepted and processed by confbridge defined action. But next digits also passing and looking for matching extension. |
| 14:47.33 | *** join/#asterisk retentiveboy (~retentive@c-73-43-121-243.hsd1.ga.comcast.net) |
| 14:47.35 | Samot | velix: So I make two bad PIN attempts, I don't enter anything on the 3rd attempt...I'm now back at 1 |
| 14:47.56 | Samot | Therefore, tracking the attempts to 4 to limit them is going to be skewed. |
| 14:48.12 | velix | Samot: Actually, that's for those guys, who don't find the keyboard on a call from iphone :) |
| 14:48.26 | velix | I've seen some guys, who weren't able to switch in 10 seconds |
| 14:48.30 | Samot | Doesn't matter. |
| 14:48.36 | Samot | It puts them back to 1. |
| 14:48.36 | velix | So they shouldn't be punished by attempts+1 |
| 14:48.44 | Samot | OK but it RESETS them. |
| 14:49.01 | Samot | It doesn't just leave the attempts at what they were. |
| 14:49.06 | Samot | It makes the attempt var 1 again. |
| 14:49.08 | velix | True, I'll fix it. |
| 14:49.11 | velix | Yeah yeah, got ya. |
| 14:49.26 | velix | when they're at attempt 3, they could do silence and are back to attempt 1. |
| 14:49.31 | velix | easy to fix. |
| 14:49.36 | Samot | Or any attempt. |
| 14:49.43 | velix | of course |
| 14:49.43 | Samot | Any attempt that doesn't have digits sent. |
| 14:50.16 | velix | Those DID numbers are really great. |
| 14:50.35 | Samot | What DID numbers? |
| 14:50.49 | velix | aren't they called DID? ;) 1 sec |
| 14:51.02 | velix | I can extent my base number with any number and catch it by extension. |
| 14:51.20 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 14:51.20 | velix | yeah, DID |
| 14:51.22 | Samot | What base number? |
| 14:51.24 | DanFromUK | Still struggling with calls dropping. It only occurs when a single PJSIP channel sent to a PHP AGI script which executes a record_file command. After about 4 minutes, randomly, the SIP call drops. Nothing obvious showing in the asterisk verbose logs. Carrier says that the calls are being ended by the caller (IE not asterisk), but it's callers from all over the country and calls that arent going to this AGI script |
| 14:51.24 | DanFromUK | aren't being affected. So I can only assume it's something on my side that's causing the caller to disconnect. |
| 14:51.50 | velix | Samot: I've got +49234xxxxxxx and can add anything after that. |
| 14:52.09 | velix | +49234xxxxxxx1, +49234xxxxxxx11, +49234xxxxxxx213 |
| 14:52.09 | Samot | Only when it's in the PBX. |
| 14:52.47 | velix | Actually, Asterisk seems them. That's how I identify the conference rooms |
| 14:53.15 | Samot | OK. |
| 14:53.15 | velix | Set(ROOM=${CALLERID(dnid):-2}) |
| 14:53.31 | velix | 21: 2 = department 2, conference room 1 |
| 14:53.41 | velix | 32: department 3, conference room 2 |
| 14:53.53 | Samot | Why do you need to append those to the DIDs? |
| 14:54.02 | velix | I've got a single base number only. |
| 14:54.09 | Samot | OK and? |
| 14:54.17 | igcewieling | German telecom allows appending random digits to TNs and they get passed to the destination |
| 14:54.17 | velix | And so each department can have 1-9 rooms with 1-9 different pins. |
| 14:54.18 | Samot | You have to do something to append those digits. |
| 14:54.25 | Samot | OK. |
| 14:54.39 | velix | igcewieling: Oh, I thought it's an SIP feature? |
| 14:54.50 | Samot | velix: It has no bearing on what you want to do. |
| 14:54.56 | velix | Ah. |
| 14:55.05 | velix | But I'm damn flexible now. |
| 14:55.31 | Samot | Perhaps. |
| 14:55.33 | velix | f.e. administrator of a room could call +49234xxxxxxx1199 for room "11" |
| 14:55.40 | velix | and get a different pin |
| 14:56.04 | velix | but limited to 15 numbers after +49234 |
| 14:56.14 | velix | I think even more work on some providers. |
| 14:56.35 | *** part/#asterisk _ganapathi_ (~Ganapathi@2409:4072:709:9a53:6c4f:58d6:b33:c7a5) |
| 14:57.07 | velix | Even with a conference, Asterisk has damn little usage. And the quality is great. |
| 14:57.29 | Samot | But again, as others have point out, you're pretty standard. |
| 14:58.02 | velix | Yeah, but it solves a lot of problems we had in the past. |
| 14:58.16 | velix | We're a research department connected to the university. Their support is aweful. |
| 14:58.35 | velix | They took 1.5 years to tell us, a conference isn't possible. |
| 14:58.48 | igcewieling | hugs his PRIs |
| 14:59.00 | Samot | PRI's have nothing to do with this. |
| 15:11.58 | Cyrillax | DanFromUK: The carrier can't tell the difference between your PBX hanging up a call and your customer hanging up a call. Basically they're saying, 'it's not us, it's you' |
| 15:15.57 | DanFromUK | DanFromUK: They can tell and have in the past. They are saying that the caller (not us) is disconnecting. But I think that they caller is disconnecting because they are hearing something that makes them think it's stopped working. |
| 15:18.16 | Samot | That's not possible. |
| 15:18.22 | Samot | Asterisk is a B2BUA. |
| 15:18.33 | Samot | The caller's leg is never something that touches the carrier. |
| 15:18.57 | Samot | Only the leg between Asterisk and the Carrier. |
| 15:19.08 | Samot | Because a user can hang up a call and that call doesn't have to stop because, PBX. |
| 15:20.19 | Samot | Also nothing states there has to be an actual person making or answering a call from the carrier. |
| 15:20.25 | Samot | The PBX could be the only thing involved. |
| 15:23.02 | Samot | DanFromUK: Have you looked at the logs? Run live traces of calls? Is this a common problem that can be looked at easily (as it you can replicate it) |
| 15:25.35 | DanFromUK | The logs show nothing unusual. No difference if they hang up intentionally or get disconnected. I'm going to try to replicate it myself, but haven't managed to get it to fail for me. It could be an issue with a specific network operator, but it only appears to be happening for this specific case of recording via a PHP AGI script, which is odd. |
| 15:27.21 | Samot | well then make a call that records that call without going through the AGI |
| 15:27.57 | Samot | Does that call have the issue? If yes, then it's more about the recording than the AGI script. If no, pointing more to the AGI script. |
| 15:28.23 | Samot | Or push a call through the AGI script but don't activate recording, does it happen then? |
| 15:28.37 | Samot | Either one of those can help trace down the issue. |
| 15:32.31 | DanFromUK | Thanks. I'll give those a try. |
| 15:37.29 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 15:37.58 | MLC | Any having trouble with the hold time announced in a queue being way off? I can tell just by looking that the hold time is > 5 minutes, but when a caller enters the queue it is telling them "30 seconds". (Apologies if this was answered yesterday after I left.) |
| 15:54.57 | Cyrillax | DanFromUK: Also might be useful if you post the VERBOSE logs for us. We might see something you missed. |
| 15:56.12 | velix | Sorry for asking again: can I make pjsip history to output all its content? I've activated the logger, but cannot find the logs. |
| 15:57.39 | igcewieling | no. |
| 15:57.42 | *** join/#asterisk ganbold (~ganbold@202.21.108.8) |
| 15:57.55 | igcewieling | turning on the logger is what you have. If you need more, use wireshark or tcpdump./ |
| 15:58.06 | velix | igcewieling: Yeah, but where do the logfiles go? :D |
| 15:58.11 | velix | they're not in /var/log |
| 15:58.19 | igcewieling | to the console and to whatever you specify in logger.conf |
| 15:58.30 | velix | looks it up |
| 15:58.51 | velix | uhm, do I need a special pjsip entry? |
| 15:59.25 | igcewieling | my logger.conf https://pastebin.com/5YaaJpAG |
| 15:59.36 | igcewieling | no, you don't need anything special for pjsip. |
| 16:00.32 | igcewieling | pjsip set logger on is all you should need to see lots of SIP spewage |
| 16:00.47 | Samot | There is nothing in Asterisk that captures the SIP transactions. |
| 16:01.02 | Samot | In the way of building a call ladder or PCAP |
| 16:01.09 | velix | Samot: I don't want this ;) |
| 16:01.18 | Samot | So what are you trying to log? |
| 16:01.18 | velix | Mine looks similar. hmm, logger _is_ on. Interesting. okay, but not problem. |
| 16:01.27 | Samot | Logger is for the SIP messages. |
| 16:01.34 | Samot | Shows the packets. |
| 16:01.41 | igcewieling | output of sip show logger: https://pastebin.com/wVQPMQih |
| 16:01.42 | Samot | Leaving it on will fill up the log files. |
| 16:02.21 | velix | I wanted to see, what Asterisk does every 60/90/30 random minutes to make the University's PBX send an error 400. |
| 16:02.32 | velix | I can see this manually in history. That's fine. |
| 16:03.40 | Samot | And what type of requests are they giving 400's to? |
| 16:05.26 | velix | The normal registration, 1 sec |
| 16:05.35 | velix | https://file.io/VWbMGY |
| 16:05.51 | velix | Timestamps in UTC: https://bpaste.net/raw/O6IA |
| 16:05.52 | Samot | I get a 404 error. |
| 16:06.08 | Samot | The first link doesn't work. |
| 16:06.13 | velix | oh, the file is down already. 1 sec |
| 16:06.16 | Samot | So a list of timestamps is useless. |
| 16:06.42 | velix | https://file.io/zTQW5B |
| 16:06.48 | velix | Samot: You'll see why. |
| 16:06.57 | Samot | I see another 404 error. |
| 16:07.04 | velix | interesting, then the service is broken |
| 16:07.29 | Samot | Why don't you use bpaste like the other? |
| 16:07.32 | Samot | Since that works. |
| 16:07.54 | velix | because it onl allows 250k |
| 16:07.57 | velix | got limited |
| 16:08.04 | velix | like all other services these days |
| 16:08.40 | velix | This one works: https://transfer.sh/9mhyh/logfile.txt |
| 16:08.56 | velix | I'm getting 401, sending auth, getting 200. That's fine. |
| 16:09.04 | velix | But suddenly (at the timestamps posted), I'm getting 400. |
| 16:09.21 | velix | But only with Asterisk. I've run a Windows SIP client in parallel without 400. |
| 16:09.44 | velix | I'm gonna collect all the logfiles now to analyse what was being sent. |
| 16:10.52 | Samot | That tells me nothing. |
| 16:11.27 | velix | I know. that's what I just said. I need to get the logs. |
| 16:11.37 | Samot | They still sent a 401 |
| 16:11.42 | Samot | You still registered. |
| 16:11.49 | velix | No, when the 400 comes, I'm not. |
| 16:11.55 | velix | Calls cannot got out then. |
| 16:12.18 | velix | pjsip allows to re-register pretty fast. chansip didn't allow this. |
| 16:12.37 | velix | But I just wonder, why it's 400 at all. And as you can see in the timestamp, there seems to be a pattern. |
| 16:12.48 | velix | each 30, 60 or 90 minutes |
| 16:12.51 | *** join/#asterisk thansen1 (~thansen@192.74.130.86) |
| 16:12.52 | Samot | That's something you have to ask them. |
| 16:13.00 | Samot | Clearly this place isn't great. |
| 16:13.07 | velix | -> But only with Asterisk. I've run a Windows SIP client in parallel without 400. |
| 16:13.09 | Samot | So they are going to be problematic. |
| 16:13.20 | Samot | Where do you run it from? |
| 16:13.26 | Samot | Not the DC where the Asterisk box is. |
| 16:13.40 | Samot | So then it would point to something at the DC. |
| 16:14.02 | velix | The SIP client runs from my home's computer. |
| 16:15.02 | *** join/#asterisk thansen (~thansen@192.74.130.86) |
| 16:15.40 | velix | Interesting, the last errors today were about 1 hour apart. |
| 16:15.51 | velix | 13:32, 14:31, 15:30, 16:00 |
| 16:15.56 | *** join/#asterisk ruben23 (~rub_e@49.149.110.51) |
| 16:16.07 | Samot | Have you asked them? |
| 16:16.16 | Samot | Why they would send back 400 errors? |
| 16:16.19 | velix | University said: never appeared to any of the phones. |
| 16:16.29 | velix | As of RFC, 400 means: don't try it again. |
| 16:17.57 | Samot | 400 is a bad request |
| 16:18.12 | velix | Yeah, but it comes aout of nowhere. |
| 16:18.14 | Samot | It's a reply saying the request was bad for some reason, malformed/wrong syntax. |
| 16:18.23 | Samot | No, it comes from them. |
| 16:18.33 | Samot | So they would have to know why they do that. |
| 16:18.47 | velix | Samot: But why is it on the PBX side only, not on the SIP client? |
| 16:19.01 | Samot | Well |
| 16:19.04 | Samot | They aren't the same. |
| 16:19.05 | velix | The log is clean, only 401 and 200. Sure THEY can answer it only :-) |
| 16:19.17 | Samot | They are behind different networks. |
| 16:19.27 | Samot | A malformed/bad syntax could be just that. |
| 16:19.28 | *** part/#asterisk igcewieling (~ewieling@199.27.202.69) |
| 16:19.35 | Samot | Something malformed or messed with the packet. |
| 16:20.13 | velix | But every 30/60/90 minutes? that's odd |
| 16:20.47 | velix | It's allowed to run PBX on this provider, others are doing it, too. I've asked before. |
| 16:22.36 | velix | I think, I'll change the "User-Agent" for a test. Maybe innovaphone IP3011 has it blacklisted :D |
| 16:24.45 | Samot | Well you need to found out from them what triggers a 400 response. |
| 16:24.57 | Samot | Make sure it is the standard response for malformed packets. |
| 16:25.14 | Samot | Then you're going to need to have them work with you to SEE the packet as it comes to THEM. |
| 16:25.30 | Samot | After it's left Asterisk, gone through the network and firewall/router, etc. |
| 16:26.13 | Samot | Because they could be throwing a 400 response at anything . |
| 16:26.26 | Samot | Depending on how they are setup. |
| 16:26.51 | velix | I can also run asterisk from my network to check, if it's similar here. |
| 16:34.53 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 16:35.50 | velix | I've just tested MicroSIP, which uses pjsip on Windows. It comes with native ICE support, which works :D |
| 16:47.17 | velix | and stun also works with ICE. Hmm, is this a feature of pjsip or of the application? |
| 16:49.33 | Samot | The device doesn't care if you're using PJSIP |
| 16:49.38 | Samot | It would clearly be a device thing. |
| 16:50.11 | velix | Nah, I'm just playing with the application and it seems like ICE works from there, while it doesn't from Asterisk. |
| 16:50.18 | velix | On asterisk, I'm getting "Unauthorized". |
| 16:52.56 | velix | yep, works: a=candidate:Pc0a8b217 1 UDP 1862270975 ... |
| 17:06.53 | velix | Samot: If you're still interested, here is the latest "400": http://dpaste.com/110E72S (line 400) with the successful registrations before and after. I cannot detect anything here. So I've got to ask the PBX owner and the DC. |
| 17:08.49 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 17:10.43 | Samot | It's you or perhaps them. |
| 17:14.15 | Samot | The only thing I see here when that happens is you send a REGISTER before the Expire is up. |
| 17:15.01 | Samot | The last successful register had an Expire of 60 seconds, the register that gets the 400 happens in 55 seconds from the last one roughly. |
| 17:15.10 | Samot | They may not like that? |
| 17:15.20 | Samot | Other then that I don't see anything wrong with it. |
| 17:15.48 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 17:38.20 | *** join/#asterisk cryptic (~cryptic@142.196.139.17) |
| 17:51.14 | *** join/#asterisk cryptic (~kmm@142.196.139.17) |
| 18:02.11 | velix | Samot: Sorry, I didn't want to leave the conversation. My bouncer got unreachable again. Another DC has routing problems. |
| 18:02.34 | velix | Samot: Oh, that sounds interesting. |
| 18:03.12 | velix | But I also had problems when it was set to 120 and 300. |
| 18:03.28 | velix | Can't I send the SIP load via curl, too? |
| 18:03.44 | velix | Or telnet, to test if that's the reason? |
| 18:06.10 | velix | or using fopen() of PHP |
| 18:06.34 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
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| 18:18.57 | *** part/#asterisk ruben23 (~rub_e@49.149.110.51) |
| 18:22.39 | velix | Samot: While being offline, the pjsip SIP-phone on Windows was connected to their PBX. Up to now, no 400, only 401. |
| 18:27.36 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 18:27.36 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:17.02 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:27.56 | velix | What is so special about the "gerrit" version? |
| 20:36.41 | *** join/#asterisk igcewieling (~ewieling@199.27.202.69) |
| 20:55.14 | Kobaz | velix: it's in development |
| 20:55.17 | Kobaz | the new-new |
| 20:56.32 | velix | Kobaz: Oh okay. |
| 21:12.23 | *** join/#asterisk DannyA (~DannyA@cpe-74-64-125-9.nyc.res.rr.com) |
| 21:12.52 | DannyA | hey all. just in broad conceptual terms, if i have two people talking in a channel, and i want to bring a 3rd person into the mix. ARI -> originate -> ? -> bridge ID |
| 21:13.19 | DannyA | i can get the id of the bridge that the two callers are on, and i know how to originate, but how do i take that new channel i originated, and add them to the existing bridge |
| 21:13.25 | DannyA | is that an AMI, ARI, or Dialplan thing |
| 21:13.49 | Kobaz | i believe you can add people into a simple bridge dynamically |
| 21:14.00 | Kobaz | traditionally before the new bridging, you would have to redirect everyone into a meetme |
| 21:14.08 | Kobaz | confbridge these days |
| 21:17.22 | Kobaz | Doesn't look like you can do it with AMI |
| 21:17.51 | Kobaz | https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Bridges+REST+API#Asterisk16BridgesRESTAPI-addChannel |
| 21:18.07 | Kobaz | you wouldn't need originate necessarily |
| 21:19.04 | DannyA | the 3rd person isn't in the system yet, i need our dispatcher to click a button, have it dial out to someone (a translator), and bridge them into an existing call |
| 21:19.19 | Kobaz | ah |
| 21:19.26 | Kobaz | Okay, yeah so... Originate the new call |
| 21:19.38 | DannyA | so that part i have, how do i get that new channel onto an existing bridge |
| 21:19.38 | Kobaz | Leg A (Channel) is the PJSIP endpoint |
| 21:19.48 | Kobaz | and then leg B is a local channel that kicks off ARI |
| 21:20.01 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 21:20.05 | Kobaz | so launch your node app and then to bridges..addChannel |
| 21:20.40 | Samot | Wait. |
| 21:20.48 | Samot | Dial() creates a two-party bridge. |
| 21:21.04 | Kobaz | Yes |
| 21:21.05 | Samot | Not a multiparty |
| 21:21.39 | Kobaz | mm |
| 21:21.43 | Kobaz | haven't played with that much |
| 21:21.49 | Kobaz | okay, so you might be back to the traditional approach |
| 21:21.53 | Samot | Confbridge/MeetMe = Multi |
| 21:21.56 | Kobaz | Oh, you can ChanSpy then into the bridge |
| 21:22.02 | Samot | Dial() = Core/Native / 2 channel |
| 21:22.09 | Kobaz | right, yeah 2 channel... makes sense |
| 21:22.47 | Samot | You can use incall features codes to do a 3 way call |
| 21:23.02 | Samot | Which would created an adhoc multiparty |
| 21:25.37 | DannyA | right, but the 3 way calling feature means they need to dial out to the 3rd party. i want the 3rd party to be dialed by AMI/ARI, and then patched in |
| 21:25.43 | Kobaz | right |
| 21:25.49 | Kobaz | Originate to a ChanSPy |
| 21:25.51 | DannyA | the dispatchers are too busy to lookup and then dial a phone number for the 3rd party |
| 21:26.01 | DannyA | ChanSpy with whiser i guess? |
| 21:26.10 | Kobaz | yeah whisper to both ends |
| 21:26.12 | DannyA | that creates a fake but functional 3 way call? |
| 21:26.18 | Kobaz | right |
| 21:26.28 | Kobaz | without having to rip everyone out into a meetme |
| 21:30.13 | DannyA | but chanspy is done with a group, right? |
| 21:30.17 | DannyA | not a specific channel id? |
| 21:30.36 | DannyA | so when u originate, the parameters u have to send are context, extension, and arguments |
| 21:30.50 | DannyA | so i can send to an extension which is ChanSpy, and the argument will the existing bridge ID |
| 21:30.58 | DannyA | can i pass that bridge ID into chanspy? |
| 21:31.23 | Kobaz | you can do a specific channel |
| 21:31.36 | Kobaz | You'll need to use the encoding stuff we have for the other originates |
| 21:31.41 | Kobaz | to pass in the channel you want to connect in |
| 21:33.39 | Samot | Question... |
| 21:33.50 | Samot | You said this was to bring on a translator.. |
| 21:34.05 | Samot | Is there a reason the call can't be routed by language before hand? |
| 21:34.54 | DannyA | they don't know till they pick up and talk to the caller |
| 21:35.13 | Samot | OK. This translator in-house? |
| 21:35.29 | DannyA | no, one of our volunteers, many different ones, many different languages |
| 21:35.44 | Samot | OK so why don't you let them select what they need? |
| 21:35.49 | DannyA | they need to find someone on the radio first who's available, then click their unit number on screen, and click a button to connect them |
| 21:36.06 | Samot | So how do they know what they need? |
| 21:36.19 | DannyA | what do u mean |
| 21:36.51 | Samot | Well I answer the call, they do not speak English. How do I know what language they are speaking? |
| 21:37.00 | Samot | How do I know which of the many translators to get? |
| 21:37.06 | DannyA | usually they can get that far :-) |
| 21:37.13 | Samot | OK. |
| 21:37.18 | Samot | So this is an emergency service. |
| 21:37.27 | Samot | That adds time to the response by doing this. |
| 21:37.29 | Samot | Got it. |
| 21:37.45 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 21:37.49 | DannyA | i mean, do u think your local 911 operator speaks 100 different languages? everyone struggles with this... |
| 21:39.27 | Kobaz | use 'the cloud' to figure out the spoken language and route automagically |
| 21:39.40 | Samot | Well I'm from Detroit. |
| 21:39.41 | Kobaz | I'm sure someone at google has some super secret project to handle that :P |
| 21:39.52 | Samot | We have a high Latin and Arabic community |
| 21:40.02 | Samot | Our local PSAPs are equipped for that. |
| 21:45.20 | Samot | You are correct, there is a language gap even PSAPs struggle with but as I said in my area because of the population makeup they have Spanish/Arabic speaking people on staff during shifts. |
| 21:46.39 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 21:52.04 | Samot | DannyA: Why can't the translator call a hotline that prompts them for the agent they want to connect which that then using ChanSpy/ExtenSpy to spy/barge in on the call to talk to all the parties? |
| 21:52.17 | Samot | Since they are already on the radio. |
| 21:53.54 | *** join/#asterisk coldfront (~coldfront@64.225.16.199) |
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| 22:07.37 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 22:08.51 | velix | Interesting. The official Debian build of Asterisk comes with an `init.d` and a `systemd` service. Both have different starting parameters... And both differ from the one in the Asterisk source file ;) |
| 22:09.03 | velix | (its contrib directory) |
| 22:16.10 | Samot | Yes because its built via packets instead of source. |
| 22:16.22 | Samot | s/packets/packages/ |
| 22:17.13 | joepublic | good thing we had infobot or we'd all have been very confused. |
| 22:18.36 | Samot | Or I would have just corrected it another way |
| 22:19.29 | joepublic | sorry, mild sarcasm doesn't carry well. Apologies. |
| 22:22.02 | Samot | Right back at ya. |
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| 22:55.34 | velix | Samot: Actually, I've got 4 different versions of startup files now :D |
| 22:55.43 | velix | Debian's systemd uses -f |
| 22:56.01 | velix | all others allow forks. |
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