| 00:02.20 | josefig | [TK]D-Fender: now i'm trying with local channels, but i think i did something wrong: https://pastebin.ubuntu.com/p/3pJYkhn9BV/ the call is placed but when I pick up it doesn't play de audio and it shows error that the context is not found |
| 00:03.28 | sibiria | seems to work, at least for #32Ol |
| 00:03.32 | sibiria | oops |
| 00:03.35 | sibiria | when using a pre-dial handler for Dial() and having it Return() the newly created channel's call-id, GOSUB_RETVAL doesn't return outside of the newly create channel. any easy of solving this? |
| 00:03.42 | [TK]D-Fender | YOU'RE STILL NOT DOING WHAT THOSE INSTRUCTIONS TOLD YOU TO DO. |
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| 00:28.02 | josefig | [TK]D-Fender: Then, I don't understand how to accomplish the instructions :/ |
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| 01:14.04 | snadge | why would someone send an INVITE with no number in it.. im trying to understand this call flow |
| 01:14.35 | snadge | eg.. the request line is INVITE sip:203.149.74.xxx:5071 SIP/2.0 |
| 01:15.06 | snadge | why am i still doing voip stuff |
| 01:19.06 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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 |
| 01:21.18 | snadge | the pcap would probably do a better job of explaining it than i ever could it seems |
| 01:21.33 | [TK]D-Fender | <PROTECTED> |
| 01:21.45 | [TK]D-Fender | then just say "mobile" as if that say how it gets there or what it means |
| 01:22.07 | [TK]D-Fender | Then something about some kind of transfer |
| 01:22.15 | [TK]D-Fender | WORDS. Use them |
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| 01:23.12 | snadge | i know i do support.. i feel like a failure as person.. i have to give the correct concise information |
| 01:24.52 | snadge | ok.. so A party calls external number, a cell phone, call connects is working.. they transfer the call to another external number..it connects.. then we lose audio from them |
| 01:25.11 | [TK]D-Fender | "External number" does mean ANYTHING |
| 01:25.13 | [TK]D-Fender | you should know this |
| 01:25.31 | [TK]D-Fender | Who is THEY? |
| 01:26.17 | snadge | i am provider .. and they are customer i guess.. with SONUS SBCSWeLite 6.1.2v104 Sonus |
| 01:26.24 | [TK]D-Fender | ... |
| 01:26.32 | snadge | it is trunked with us.. and calls are working etc.. its just a particular case that they lose audio when transferring |
| 01:26.35 | [TK]D-Fender | There are 3 parties in this call. |
| 01:28.16 | snadge | i wonder if its something stupid like a problem with the destination number.. i might just ask them to test dialling the number they're transferring to and seeing if they get audio |
| 01:28.52 | [TK]D-Fender | How is a NUMBER a problem? |
| 01:29.51 | [TK]D-Fender | Yeah.... you don't really want help .... |
| 01:29.55 | snadge | oh wow i think i just realised what the problem was.. call transfer is a pbx feature |
| 01:30.03 | snadge | this is a trunk |
| 01:30.04 | [TK]D-Fender | I'll just leave you for the next emotional support tech. |
| 01:30.10 | snadge | what the f$#@ is this customer talking about |
| 01:30.25 | [TK]D-Fender | Please stay in the channel to maintain your call priority |
| 01:31.07 | snadge | theres two problems here.. i fail big time at trying to explain what my problem is |
| 01:31.31 | snadge | and the second problem is.. wtf is this customer even doing.. this is a trunk.. why are they talking about problems transferring calls |
| 01:33.13 | snadge | thats entirely the problem of whatever this sonus crap is and i dont care.. if he can call each number seperately and it works |
| 01:33.31 | snadge | then i dont care why it doesn't work when he transfers the call on his crappy pbx |
| 01:33.35 | snadge | thats his problem |
| 01:34.04 | snadge | he can contact sonus ;) |
| 01:37.18 | snadge | you know what.. i think you have helped me more than you realise [TK]D-Fender |
| 01:37.37 | snadge | im clearly in the wrong job |
| 01:37.52 | snadge | i dont give a fuck about any of this shit |
| 01:39.10 | snadge | i cant wait until i get this php programming job.. and i can go back to doing something that i enjoy, and give a shit about |
| 01:39.16 | snadge | and i can tell this place to shove it |
| 01:47.35 | snadge | thanks for your time.. and i apologise for being too retarded to figure out how to ask for help properly |
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| 01:55.49 | snadge | is anyone other than [TK]D-Fender in here? |
| 01:56.17 | snadge | maybe someone who isn't as much of a fucking asshole? |
| 01:58.09 | [TK]D-Fender | I'm an asshole now am I? |
| 01:58.31 | snadge | well you basically just said to me.. fuck off you're a retard and i have no time for you |
| 01:58.37 | snadge | which i totally understand |
| 01:58.41 | [TK]D-Fender | You come in with zero description, talk about random things like "number" with no service or device involved |
| 01:58.55 | Samot | Well.. |
| 01:58.56 | Samot | First.. |
| 01:58.57 | [TK]D-Fender | and about other people's systems you don't control running software we don't support |
| 01:59.02 | Samot | Sonus is an SBC not a PBX |
| 01:59.03 | [TK]D-Fender | And WE owe you something? |
| 01:59.16 | Samot | Also, it's enterprise. It's a thing a lot of the big boys use. |
| 01:59.30 | [TK]D-Fender | Don't come in here and cry |
| 01:59.36 | Samot | So to call it a "crappy PBX" means there no understanding of what is being used |
| 01:59.41 | snadge | yeah i've kind of gathered that.. and did the best i can, responded to the customer and suggested they engage with sonus support |
| 02:00.05 | snadge | they can call either number and get audio via this SBC |
| 02:00.15 | snadge | but if they call both numbers and try to join them together, then they lose audio |
| 02:00.29 | [TK]D-Fender | Wahtever. They gear. We can't help |
| 02:00.37 | snadge | so my limited understanding of the situation tells me.. thats probably something to do with the sonus, and therefore nothing any of us has to deal with |
| 02:00.38 | [TK]D-Fender | their* |
| 02:00.42 | snadge | exactly |
| 02:00.45 | Samot | Well |
| 02:00.56 | Samot | You haven't described an actual clear issue |
| 02:01.02 | Samot | So it's hard to say |
| 02:01.10 | Samot | Nor have I seen any debugs or logs of anything... |
| 02:01.22 | Samot | Could be them, could be you. |
| 02:01.24 | snadge | all i have to go on is a pcap.. and my sip knowledge is extremely limited |
| 02:01.51 | Samot | Isn't there another tech you can ask there? |
| 02:01.56 | Samot | You can't be the only tech. |
| 02:02.10 | snadge | the place im currently working.. im a senior voice engineer |
| 02:02.17 | Samot | What? |
| 02:02.17 | snadge | if i dont know.. nobody else will |
| 02:02.23 | Samot | How is that possible? |
| 02:02.30 | Samot | You didn't even know what a Sonus SBC was... |
| 02:02.48 | snadge | im only supposed to know about our side of it.. so the Sonus SBC is just a customer |
| 02:02.55 | Samot | OK |
| 02:02.58 | snadge | i dont technically need to be able to support that |
| 02:03.07 | Samot | That's not my point. |
| 02:03.21 | Samot | To be a "Senior Voice Engineer" is to know what a Sonus SBC is. |
| 02:03.27 | Samot | Because it's in that realm. |
| 02:03.42 | Samot | Of what Senior Voice Engineers would use/recommend. |
| 02:05.20 | snadge | ive been doing voice for about 5 years.. in my last role, we had an asterisk based cluster.. it was built and designed by someone else |
| 02:05.41 | snadge | all i had to do was support it.. so my knowledge is fairly limited really, admittedly so |
| 02:05.53 | Samot | Asterisk doesn't "cluster" |
| 02:05.53 | Samot | oK |
| 02:05.58 | Samot | 8:14:02 PM <snadge> why would someone send an INVITE with no number in it.. im trying to understand this call flow |
| 02:05.58 | Samot | 8:14:34 PM <snadge> eg.. the request line is INVITE sip:203.149.74.xxx:5071 SIP/2.0 |
| 02:06.08 | Samot | ^^ The answer is "They have something misconfigured" |
| 02:06.22 | Samot | Because an INVITE without a user portion is incomplete |
| 02:06.41 | snadge | ok thats why i opened with that leading question.. and your answer is along the lines of what i was looking for |
| 02:07.06 | snadge | then tkd got me flustered by asking me questions i didn't really know how to respond to.. combined with my inability to describe the problem |
| 02:07.37 | Samot | Please do not describe yourself as a "Senior Voice Engineer" when asking for help. |
| 02:07.44 | [TK]D-Fender | No, you followed that "no number" statment... with 4 more unrelated things |
| 02:07.53 | [TK]D-Fender | YOU added unrelated shit into the picture |
| 02:07.55 | Samot | It was a mess. |
| 02:07.56 | snadge | i know thats what i mean by flustered |
| 02:08.11 | snadge | i completely acknowledge i was entirely at fault.. and feel bad about it.. for what its worth |
| 02:08.17 | Samot | It's incoherent and doesn't describe the issue in any way understandable. |
| 02:08.46 | snadge | yes.. then whilst i realised that.. for whatever reason i became agitated and frustrated and it just got worse |
| 02:09.17 | [TK]D-Fender | Nothing to see, nothing to do... |
| 02:10.04 | [TK]D-Fender | And off to more entertaining things |
| 02:10.11 | snadge | thanks both of you anyway.. and again i apologise |
| 02:10.30 | snadge | and feel embarassed |
| 02:11.06 | Samot | And we still don't know what the actual issue is. |
| 02:14.30 | snadge | ok just overview.. customer has sonus sbc trunked with us.. when they place call to two different numbers, that works |
| 02:14.57 | snadge | if they make a call to one of those numbers, then attempt to transfer the call to the other number.. they lose audio in both directions |
| 02:15.16 | snadge | around the time that happens.. we get that odd looking INVITE thats missing a user portion |
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| 02:16.14 | snadge | for now .. i have managed to put it back on the customer and say, it must be some kind of issue with call bridging that is related to the sonus.. that we have customers with asterisk, and 3cx server etc and they can transfer calls just fine |
| 02:17.51 | snadge | so he is now engaging with a sonus technician.. and im doing another pcap with a different test.. probably to verify the same result |
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| 05:27.39 | malby | hello, I've finally reached "peak telemarketer" and I don't think I can handle another call from some one phoning from what appears to be an Australian number, but is actually overseas, and I'm wondering if it is possible to do something like this:- |
| 05:27.55 | malby | 1) Keep our voip line without any hardware changes (we have a Gigaset A510 handset).. or as minimal hardware changes as possible (other than asterix hardware) |
| 05:28.05 | malby | 2) Whitelist a set of known phone numbers that will result in the phone ringing. |
| 05:28.13 | malby | 3) Anything not matching one of the known numbers gets a automated response that says "If you are human, please press 1 to get through", where if they press 1, our phone rings |
| 05:28.16 | malby | and thats it |
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| 11:45.38 | andi | Hi, is there a way to lookup phone numbers from a caller in a mysql table and if it's not in there, insert this number? I'd like to create a nice way to manage a phonebook. With a small webfrontend I'd like to insert names to numbers that don't have a name, yet. With the correct xml representation of the data I will be able to add this phonebook to my snom phones via a webserver. |
| 11:49.02 | sibiria | do it with an AGI application |
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| 11:50.34 | sibiria | scripting that in the dial plan, or even using Lua, will be messier than just doing it with PHP/PDO or Perl/DBI |
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| 11:55.39 | andi | So I found that there is a mysql command in asterisk: https://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
| 11:56.13 | andi | Do you mean with an AGI script to call an external maybe php script to insert the data? Explained here: https://www.voip-info.org/wiki/view/Asterisk+AGI |
| 11:56.29 | sibiria | yes |
| 11:58.33 | andi | Now I'm hanging at the point where to step into the dialplan. What is the first part that is executed? I think I need to add a part in extensions_custom.conf, but where? |
| 11:59.55 | sibiria | not sure i understand the question |
| 12:01.45 | Samot | extensions_custom.conf == FreePBX |
| 12:02.18 | andi | So to execute the AGI script I need to execute an AGI(mytestscript.php, some, parameter, for, the, script) within my dialplan. Changes to freepbx normally go to extensions_custom.conf. So I need to find the point where the very beginning of an external call starts to be able to implement that script there. |
| 12:02.30 | Samot | Uhm. |
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| 12:02.34 | andi | Samot: Yes, exactly, maybe I could have said that earlier. |
| 12:02.38 | sibiria | at some point, defined in asterisk, your incoming call will begin in a pre-set context |
| 12:02.39 | Samot | FreePBX already has a GUI interface for this |
| 12:02.41 | sibiria | usually "default" |
| 12:02.46 | Samot | It's called Asterisk Phonebook |
| 12:03.12 | Samot | It won't generate the XML, you'll still have to do that |
| 12:03.36 | Samot | But there is already a method into which to lookup numbers and have them associated with a contact. |
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| 12:09.11 | andi | For any reasone the asterisk phonebook is greyed out in my system. So I need to check why that's the case. Thank you for the information, I'll check that. |
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| 12:34.33 | sibiria | does 'maxcalls' in asterisk.conf regard incoming, outgoing, or both together? |
| 12:34.48 | sibiria | i cannot find a detailed description |
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| 12:36.11 | sibiria | the latter, i presume |
| 12:38.00 | Samot | ;maxcalls = 10; Maximum amount of calls allowed. |
| 12:38.12 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/asterisk.conf.sample |
| 12:41.17 | sibiria | i have a feeling it was meant for total calls, but in practice (in 18.13.2) it seems to regard only incoming |
| 12:42.00 | sibiria | unless the default really is 0 (as in unlimited), not 10 |
| 12:42.17 | Samot | It's the maximum amount of calls Asterisk will accept. |
| 12:42.18 | sibiria | the setting has been left commented out, and our systems regularly have 50-100 simultaneous outgoing calls |
| 12:42.27 | Samot | It's up to you to control how it makes calls |
| 12:42.31 | Samot | And tracks them. |
| 12:42.39 | Samot | That's what GROUP() is for. |
| 12:43.28 | Samot | Because the default is no limit |
| 12:43.33 | Samot | Unless you set maxcalls |
| 12:44.01 | sibiria | hence documentation lacking. default is 0, not 10 as might be implied |
| 12:47.31 | Samot | Thats why its commented out |
| 12:47.42 | Samot | Thats a sample config |
| 12:47.55 | Samot | The uncommented atuff is what you need |
| 12:48.08 | Samot | Not to mention it is under a options section |
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| 13:29.46 | polysics | hey! is asterisk-dev access approval manual? one of my colleagues has signed up a while ago |
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| 13:32.20 | Samot | I just joined with no issue |
| 13:32.35 | file | I don't believe it is, but if you send a large message then it'll go into queue to get released |
| 13:32.59 | Samot | I think your nick needs to be registered. |
| 13:33.27 | file | (if you are referring to mailing list) |
| 13:34.06 | Samot | Oh, yeah that maybe. |
| 13:34.23 | Samot | #asterisk-dev channel mode is +cnrt |
| 13:34.36 | Samot | So it requires a registered/identified nick to enter the channel |
| 13:34.51 | file | yes, also required for here |
| 13:35.33 | polysics | yes, the mailing list - I'll tell him to check the spam folder then |
| 13:36.01 | polysics | man, delving into SDP is a nightmare :D |
| 13:36.05 | file | I meant it'll go into the mailing list queue to get released by an admin |
| 13:36.12 | file | that's WebRTC |
| 13:36.18 | file | if you are going to use it, then that's life |
| 13:36.29 | polysics | yes, it is indeed |
| 13:36.39 | polysics | WebRTC also doesn't actually work |
| 13:36.50 | polysics | we can make it pretend it does |
| 13:38.40 | Samot | Meh. |
| 13:38.50 | Samot | WebRTC will take off like video calls did. |
| 13:41.24 | polysics | well, it does have advantages like zero installation etx |
| 13:43.04 | Samot | Yes but the disadvantages... |
| 13:43.08 | Samot | Like no standards. |
| 13:43.15 | Samot | Each browser has it's own method of web sockets |
| 13:43.21 | Samot | How it handles WebRTC. |
| 13:43.37 | Samot | You're back to the 1990's with "Best used in X browser" land |
| 13:45.40 | Samot | When I was at the LEC we spent a shit tonne of $$$$ and man hours on video calls. |
| 13:45.54 | Samot | It sold like..... |
| 13:45.58 | Samot | Nothing. |
| 13:46.26 | Samot | Not a single nibble ever. |
| 13:46.31 | Samot | 1) Doesn't work on the PSTN |
| 13:46.40 | Samot | 2) Requires a phone the supports video with a camera |
| 13:47.29 | Samot | The only real video call request was for something like GoToMeeting. |
| 13:47.31 | Samot | That was it. |
| 13:48.24 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:48.52 | Samot | Turned out that cutting a deal with the guys at GoToMeeting and bundling that as a third-party addon feature was the better option. |
| 13:48.56 | Samot | And sold more. |
| 13:50.18 | Samot | Something I need to do again because it turns out, in the market area I am in that's exactly how the LECs/ISPs do it. |
| 13:50.37 | Samot | They offer GoToMeeting as the video/web conference solution for their services. |
| 13:53.11 | Samot | WebRTC is going to be the same thing. |
| 13:53.39 | Samot | A niche service that will be requested by a small, small percentage over the overall market base. |
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| 13:57.03 | sibiria | as someone who's done quite some developing of websockets stuff both server-side and browser-side, i cannot really say there's any merit to "no standard; each browser has its own methods" |
| 13:57.32 | polysics | in our experience if you can mandate specific conditions it can be pretty slick |
| 13:57.32 | sibiria | it's all quite uniform across the major browsers, and has been for years |
| 13:57.42 | polysics | ie. you "HAVE to usse Chrome" |
| 13:58.29 | sibiria | i don't see the need for that. i've never had to jump through hoops or make exceptions for this or that browser |
| 13:58.45 | sibiria | websockets isn't webgl, or css3 etc. |
| 13:59.06 | sibiria | it's one of the rare cases that don't suffer from this popular problem |
| 13:59.14 | Samot | WebRTC is a good local use solution. |
| 13:59.32 | Samot | There is no video support on the PSTN |
| 13:59.47 | Samot | The average user really doesn't want to be on a video call when calling a company |
| 14:00.08 | sibiria | obviously. and 64 kbit/s doesn't go far even with good implementations of modern codecss |
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| 14:00.10 | Samot | Nor do they want to take extra steps to load a browser and go to a special web page to make a call |
| 14:00.16 | sekil | there was...back in ISDN/H.323 time |
| 14:00.29 | polysics | audio calls are useful too :) |
| 14:00.48 | Samot | I've been working at ITSP/LEC's for over 15 years doing this |
| 14:00.54 | sibiria | i think video calls as such is an overrated feature of the past's sci-fi predictions |
| 14:01.08 | Samot | The amount of video or WebRTC requests for service is pretty much 0 |
| 14:01.09 | sibiria | web meetings is one thing, btu i don't really believe people want video calls for day to day communications |
| 14:01.14 | sibiria | they certainly aren't asking for it |
| 14:01.17 | sibiria | it's just a hassle |
| 14:01.37 | Samot | 8:48:52 AM <Samot> Turned out that cutting a deal with the guys at GoToMeeting and bundling that as a third-party addon feature was the better option. |
| 14:01.38 | Samot | ^^^^ |
| 14:01.47 | Samot | That's what LECs and ITSPs do now.. |
| 14:01.48 | sekil | many companies still want video...Polycom sold bunch of video stuff |
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| 14:01.52 | Samot | They just bundle GoToMeeting |
| 14:01.55 | Samot | Because it's easier |
| 14:02.01 | Samot | And it's a low feature request... |
| 14:02.04 | sibiria | as said, web meetings... this is where video calls have a chance to survive |
| 14:02.16 | Samot | Right |
| 14:02.25 | Samot | But LECs/ITSPs are really building it out |
| 14:02.30 | sibiria | best of luck to them |
| 14:02.32 | Samot | They are just going with the leading company |
| 14:02.55 | Samot | Because it's not, I repeat not, worth the investment for most of them. |
| 14:03.06 | Samot | It's not a big sales leader. |
| 14:03.07 | Samot | At all |
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| 14:04.06 | polysics | going back to my original topic: in normal conditions, there should be no SDP interop problems on the SFU between Chrome and FF, correct? |
| 14:04.13 | polysics | if we have a problem it's on "our" end |
| 14:04.34 | Samot | That's going to be up to the browser. |
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| 14:04.43 | Samot | They each will have their own SDP implementation. |
| 14:04.57 | Samot | How close they follow whatever standards are set for WebRTC is on them. |
| 14:05.05 | Samot | You could make this work like a champ today |
| 14:05.10 | file | the SDP generated in the latest release candidate is accepted and works with Firefox last I tried |
| 14:05.14 | Samot | And next month FF could push an update that breaks it |
| 14:05.28 | Samot | Or Chrome. |
| 14:05.37 | Samot | These are things that have already happened. |
| 14:06.11 | Samot | Chrome and FF have pushed changes over the years that have made people have to redo their WebRTC deployment. |
| 14:08.16 | file | I'll also add that looking from the Asterisk side is limited as I said on the mailing list, the browser really is a black box |
| 14:12.01 | polysics | so the first step is installing the latest RC and re-testing? |
| 14:12.07 | polysics | can do that :) |
| 14:12.27 | file | there was a fix done in it for Firefox |
| 14:12.31 | file | or recently, lemme look |
| 14:12.48 | file | https://gerrit.asterisk.org/#/c/7883/ |
| 14:12.52 | file | in 15.3.0-rc1 |
| 14:13.27 | file | I'll also add: it took getting internal Firefox debug logging to figure that out |
| 14:13.39 | file | and a week of time |
| 14:14.22 | sibiria | is #asterisk-dev usually active? say i want to discuss a possible bug |
| 14:14.28 | sibiria | or quirky behavior |
| 14:14.39 | file | depends on the time of day and what people are doing |
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| 14:38.36 | polysics | oh wow, thank you @file |
| 14:39.48 | file | hmm? |
| 14:40.06 | polysics | for that fix :) |
| 14:40.10 | file | ah |
| 14:40.29 | file | yes, always try the latest thing out (be it release or candidate) before investing time in trying to figure WebRTC out |
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| 15:57.44 | Steeldutch | Hello |
| 15:57.56 | Steeldutch | Can someone help me with a issue in asterisk |
| 15:59.47 | pawiecki | Steeldutch: ask your question, don't ask to ask. |
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| 16:00.10 | Steeldutch | My issue is that i have a voicemail running |
| 16:00.48 | Steeldutch | when i call a intern extension the voicemail is working perfect with codec ulaw when i call the voicemail with a extern number |
| 16:00.53 | Steeldutch | i get error's |
| 16:01.15 | Steeldutch | app_playback.c:494 playback_exec: Playback failed on SIP/ |
| 16:02.01 | Steeldutch | i have not this file goedemiddag.slin i have only the .ulaw , .wav, .gsm and i want that the system always use ulaw |
| 16:02.06 | Steeldutch | how can i fix this issue |
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| 16:04.53 | [TK]D-Fender | "extern number" doesn't mean anything |
| 16:05.13 | [TK]D-Fender | You need to look at what codec the call is actually using and what modules you have loaded |
| 16:05.49 | Steeldutch | Format Name Extensions |
| 16:05.49 | Steeldutch | ------ ---- ---------- |
| 16:05.49 | Steeldutch | slin16 wav16 wav16 |
| 16:05.51 | Steeldutch | slin wav wav |
| 16:05.53 | Steeldutch | adpcm vox vox |
| 16:05.55 | Steeldutch | g729 g729 g729 |
| 16:05.57 | Steeldutch | gsm gsm gsm |
| 16:05.59 | Steeldutch | ilbc iLBC ilbc |
| 16:06.01 | Steeldutch | g726 g726-16 g726-16 |
| 16:06.03 | Steeldutch | g726 g726-24 g726-24 |
| 16:06.05 | Steeldutch | g726 g726-32 g726-32 |
| 16:06.07 | Steeldutch | g726 g726-40 g726-40 |
| 16:06.09 | Steeldutch | h263 h263 h263 |
| 16:06.11 | Steeldutch | gsm wav49 WAV|wav49 |
| 16:06.13 | Steeldutch | g723 g723sf g723|g723sf |
| 16:06.15 | Steeldutch | g722 g722 g722 |
| 16:06.17 | Steeldutch | ulaw au au |
| 16:06.19 | Steeldutch | alaw alaw alaw|al|alw |
| 16:06.21 | Steeldutch | ulaw pcm pcm|ulaw|ul|mu|ulw |
| 16:06.23 | Steeldutch | h264 h264 h264 |
| 16:06.25 | Steeldutch | 18 file formats registered. |
| 16:06.27 | Steeldutch | ah that was not a good idea sorry |
| 16:06.34 | Steeldutch | This are my codecs when i ask asterisk with core show file formats |
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| 16:10.04 | igcewieling | Steeldutch you will NEVER get help if you do that again. Use a pastebin |
| 16:10.41 | Steeldutch | I say sorry it is first time i use irc |
| 16:12.32 | [TK]D-Fender | CODECS, not just "formats" |
| 16:12.55 | [TK]D-Fender | ""format" is required to read a file, but CODEC is required to transcode to/from it |
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| 16:15.06 | [TK]D-Fender | so format != codec |
| 16:17.29 | Steeldutch | can i change the order of the formats that need to load first |
| 16:18.14 | igcewieling | Steeldutch: no. If you could it wouldn't help. It is the codecs you care about not the formats. |
| 16:19.32 | [TK]D-Fender | load order means nothing |
| 16:19.43 | [TK]D-Fender | * will always play the most efficient version it can |
| 16:20.19 | Steeldutch | ok i have also try to unload the slin module then the voicemail tape is working only the recording is then .wav and that is then broke |
| 16:20.23 | [TK]D-Fender | You haven't shown us the full call witht he full error, proof of what versions exist there, etc |
| 16:20.58 | Steeldutch | Asterisk 13.17.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. |
| 16:21.00 | igcewieling | Steeldutch: stop. the solution is simple. disable the codecs you don't use with a deny= line in sip.conf. |
| 16:21.50 | igcewieling | and go read the asterisk book. |
| 16:24.37 | Steeldutch | i will do that i have put a deny= line in sip.conf |
| 16:24.44 | Steeldutch | i get this error |
| 16:24.49 | Steeldutch | pastebin |
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| 22:19.10 | josefig | is this correct ? exten => s,n,ExecIf($[${DIALSTATUS} = "ANSWER"]? Playback/${AUDIOFILE}) |
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| 22:41.53 | rmudgett | josefig: You need quotes around the result of ${DIALSTATUS}: "${DIALSTATUS}" to compare the result with "ANSWER" |
| 22:44.30 | rmudgett | The application to exec also needs to be expressed as: Playback(${AUDIOFILE}) |
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| 22:51.33 | josefig | yes, i found the issue now it works but now I have different issue, it seems when it generates the call and I pickup the call it generates 2 calls, but I'm not sure why, https://pastebin.ubuntu.com/p/bbMQM4x5zb/ |
| 22:51.59 | josefig | it generates a bridge but I'm not sure why |
| 22:57.07 | [TK]D-Fender | -- Attempting call on Local/s@local-channel-files for s@local-channel-files:1 (Retry 1) |
| 22:57.09 | [TK]D-Fender | You should be sure |
| 22:57.26 | [TK]D-Fender | because you very clearly are sending both legs to the SAME PLACE |
| 22:57.39 | [TK]D-Fender | Local/s@local-channel-files |
| 22:57.45 | [TK]D-Fender | for s@local-channel-files:1 |
| 22:57.54 | [TK]D-Fender | When the FIRST place answer .. you are calling the SAME THING |
| 22:59.22 | josefig | [TK]D-Fender: I see, what I want to do is when call answer it does a Playback with my ${AUDIOFILE} |
| 22:59.48 | [TK]D-Fender | Well you told it to do the same thing twice |
| 22:59.54 | [TK]D-Fender | Stop doing that |
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| 23:00.22 | josefig | How would be the correct way ? |
| 23:01.10 | [TK]D-Fender | Call out... then send it to the dialplan where you have your playback |
| 23:01.25 | [TK]D-Fender | You can't send it to the SAME dialplan ... without it calling out AGAIN. |
| 23:01.30 | [TK]D-Fender | So don't |
| 23:01.37 | [TK]D-Fender | make som OTHER place for it to go |
| 23:01.45 | josefig | I gotcha, let me try |
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| 23:02.48 | [TK]D-Fender | Channel: WHERE YOU CALL |
| 23:03.06 | [TK]D-Fender | Context: Exten: Priority: WHERE YOU GO |
| 23:03.13 | [TK]D-Fender | this was never a hard thing |
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| 23:15.14 | josefig | [TK]D-Fender: worked now, I appreciate it, problem of concepts amigo :P |
| 23:15.18 | josefig | Thanks |
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