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| 02:39.56 | ih8wndz | what changes in pjsip tls effect versions > 13.22? I had to comment out my tls transport after compiling 13.23 & 15.6 |
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| 10:00.37 | GeneralSpongebob | Hi, I'm still having trouble with features.conf. I can't get this to work: INpauseMonitor => #1,peer,GoSub,pauseMonitor |
| 10:01.14 | GeneralSpongebob | When I press '1' or '#1' I can hear a beep but that sub should output a message using NoOp. I have verbosity to 6 |
| 10:04.04 | GeneralSpongebob | The SIP extension and Grandstream GXP 1620 phone are both set to use RFC2833 for DTMF and I'm calling an internal extension that just plays hold music. It seems like features.conf doesn't do anything |
| 10:15.16 | GeneralSpongebob | Is there any way I can test to prove DTMF is working? |
| 10:16.24 | sibiria | have you tried just adding the appropriate extension and doing a verbose ther? |
| 10:16.25 | sibiria | there* |
| 10:16.45 | sibiria | it seems likelier that your "sub" or feature is the problem |
| 10:17.47 | GeneralSpongebob | After dialing the extension I can see the verbose output from it's NoOp but nothing from the sub. I tried changing the feature line to PlayBack,vm-saved but that didn't do anything either |
| 10:19.20 | sibiria | so dtmf works then |
| 10:19.38 | sibiria | else you wouldn't have jumped off to the desired extension in your dial plan |
| 10:21.08 | GeneralSpongebob | no, the extension I dialled matched one in extensions.conf which is what I'm calling to test with. The extension is configured to play the hold music. The DTMF are meant to be picked up by the features configuration but I can't find any indication that it's working which is why I would like to find a way to prove the tones are getting recognised |
| 10:22.32 | sibiria | just make a simple dial plan that offers the extension you want and throws a verbose from it |
| 10:22.56 | sibiria | make an outbound call using that context, press digit on phone, watch verbose from log |
| 10:28.35 | sibiria | also, make sure you've enabled dtmf on one of your configured log outputs |
| 10:28.58 | GeneralSpongebob | how do I make sure it's enabled for log output? |
| 10:29.11 | sibiria | logger.conf |
| 10:29.21 | sibiria | go to the bottom of that file |
| 10:29.42 | sibiria | i suspect you already have an item 'full' configured there, with dtmf as option |
| 10:29.49 | sibiria | if not, just throw dtmf into one of them |
| 10:29.54 | sibiria | and restart |
| 10:30.20 | sibiria | if you already have dtmf enabled for one of the log items, just take a look in that log file |
| 10:32.33 | GeneralSpongebob | DTMF is definitely being recognised so I guess something's wrong with either my features.conf or extensions.conf |
| 10:32.42 | sibiria | yes, as suspected |
| 10:36.01 | GeneralSpongebob | https://pastebin.com/d9FvNGNL Can you see where I've gone wrong? |
| 10:40.01 | sibiria | sorry no, i've never actually used macro features so i don't know their syntax |
| 10:40.57 | sibiria | but i would dig through the full log (or the console while making the call) to see what asterisk has to say |
| 10:41.04 | sibiria | and i'd up the verbose to 9 just to be sure i get everything |
| 10:41.12 | sibiria | verbosity* |
| 10:41.24 | sibiria | and debug, of course |
| 10:42.12 | GeneralSpongebob | Thanks, will try |
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| 11:14.29 | roswell | GeneralSpongebob, there's no point of using GoSub in features, as the execution stack won't know at which point to Return |
| 11:26.49 | GeneralSpongebob | Which module loads features.conf? I don't think the file is being read |
| 11:28.55 | file | dynamic features only apply when Dial() is used. |
| 11:30.32 | GeneralSpongebob | so I just need to dial instead of Gosub/Macro ? |
| 11:31.27 | file | I don't know how to answer that... dynamic features are only listened for and executed when channels are in Dial() |
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| 14:07.56 | GeneralSpongebob | @file Let's say I want to pause the call recording when either side presses number 1 on the phone. How is that accomplished? I'm going in circles through documentation that's out of date |
| 14:08.23 | file | I don't have anything for that at my fingertips. |
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| 14:15.53 | Samot | Show the pastebin again |
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| 14:44.33 | GeneralSpongebob | Samot, https://pastebin.com/xGeCSJD6 I'm now just trying to get it to play something. DTMF is confirmed to be detected by using the logger to the console but the feature is not responding |
| 14:45.21 | file | there is no Dial() in there, thus it won't work regardless |
| 14:46.02 | GeneralSpongebob | right, no one has said what that means though |
| 14:46.07 | [TK]D-Fender | it isn't a magical backgound thing that works all the time. |
| 14:46.26 | GeneralSpongebob | [TK]D-Fender great, so could I have an example or an explanation? |
| 14:46.45 | file | I don't know if I can state it any other way... dialplan steps execute dialplan applications, the dialplan application Dial has to be executed in order for dynamic features to work, and only while that application is active will it work |
| 14:46.48 | GeneralSpongebob | I thought setting DYNAMIC_FEATURES allowed the channel to use those features |
| 14:47.03 | Samot | Right |
| 14:47.09 | Samot | Dial() is what makes a channel |
| 14:47.10 | file | only while the channel has dialed something else and is conversing with them |
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| 14:47.13 | [TK]D-Fender | ; Note that the DTMF features listed below only work when two channels have answered and are bridged together. |
| 14:47.14 | [TK]D-Fender | ; They can not be used while the remote party is ringing or in progress. If you require this feature you can use |
| 14:47.14 | [TK]D-Fender | ; chan_local in combination with Answer to accomplish it. |
| 14:47.17 | Samot | Well for the most part |
| 14:47.30 | GeneralSpongebob | So backing up a sec, are you saying that the active call is not a channel? |
| 14:49.22 | file | there is a single channel executing dialplan in the above |
| 14:49.31 | file | it has not dialed or been bridged to another channel |
| 14:51.09 | file | you may need to do reading to learn some of the fundamentals, as it only gets more complicated |
| 14:51.16 | GeneralSpongebob | then I'm lost. I have dialled the extension from the SIP phone and that's where I've ended up. Is there any working example you can link me to for Asterisk 13 or higher? My goal is to be able to pause and resume call recording by pressing a number on the phone |
| 14:52.38 | file | I don't have anything handy as I previously mentioned |
| 14:52.54 | file | someone else may but without a solid foundation you are likely to run into further problems |
| 14:53.06 | GeneralSpongebob | I've gotten this far |
| 14:53.22 | GeneralSpongebob | I understand what you're saying but "run along little newb" isn't that helpful to me |
| 14:53.54 | [TK]D-Fender | Show us a proper dialplan and call. |
| 14:54.15 | GeneralSpongebob | You mean my entire config, why? |
| 14:54.25 | [TK]D-Fender | the CALL. |
| 14:54.52 | GeneralSpongebob | [TK]D-Fender When dealing with someone who doesn't understand you, it's better to explain rather than shout |
| 14:54.57 | [TK]D-Fender | that last bit of dialplan doesn't listen to features. |
| 14:55.21 | [TK]D-Fender | so show us you're setting it properly and then actually dialing out to where taht call once bridged should listen |
| 14:55.34 | [TK]D-Fender | Also go confirm that your dynamic features are in fact loaded |
| 14:56.21 | GeneralSpongebob | what 'it' are you referring to? "features show" tells me that pause and resume are there as 1 and 3 and both say "no default" if that's relevant? |
| 14:56.44 | igcewieling | GeneralSpongebob: what file did you edit to setup those features? |
| 14:56.49 | GeneralSpongebob | features.conf |
| 14:57.06 | igcewieling | take that file and copy it to a pastebin -- THAT shows us what you did. |
| 14:57.50 | igcewieling | now, ssh into the box, run asterisk -rvvv and then make a call and copy the output to....a pasttbin -- THAT shows the call. |
| 14:58.08 | igcewieling | until you can do those things easily don't expect people to be able to help much. |
| 14:59.14 | GeneralSpongebob | https://pastebin.com/1FbA869Y from grep '^[^;]' features.conf |
| 15:00.30 | GeneralSpongebob | igcewieling: I'm getting tripped up over people assuming I'm an Asterisk expert when I'm clearly not, that's why I'm asking for help. I am a moderately capable linux user and have managed to get pretty far by myseful but this features stuff seems to be different version to version and I've gotten very lost in it |
| 15:00.31 | igcewieling | extensions.conf might be needed too, or at least the parts you are dealing with. |
| 15:01.53 | GeneralSpongebob | The phone I'm calling form exists within the same context as that extension so I'm not sure what other part of extensions.conf would apply?? |
| 15:02.03 | [TK]D-Fender | show us the call |
| 15:02.05 | igcewieling | then paste that part. |
| 15:02.13 | igcewieling | *nod* we are still waiting for the additional stuff. |
| 15:03.25 | GeneralSpongebob | igcewieling that extension is in the paste https://pastebin.com/xGeCSJD6 the only thing not inculded there is the other extensions and the context name. |
| 15:03.49 | igcewieling | until we see the cli output we won't know what part of the dialplan is relevant |
| 15:03.49 | [TK]D-Fender | that hold music stuff is NOT a Dial() |
| 15:04.24 | igcewieling | *nod* I don't see a Dial there either. |
| 15:04.45 | [TK]D-Fender | This only works when 2 channels are bridged in a Dial() |
| 15:05.46 | GeneralSpongebob | https://pastebin.com/JAz2CFN6 |
| 15:05.56 | [TK]D-Fender | NO |
| 15:06.07 | [TK]D-Fender | This is when 2 CHANNELS are talking |
| 15:06.22 | GeneralSpongebob | OK |
| 15:06.24 | [TK]D-Fender | It will NOT listen in the background of other audio you're hearing or during MusicOnHold() |
| 15:06.26 | GeneralSpongebob | WHY ARE WE TALKING IN CAPS |
| 15:06.36 | [TK]D-Fender | Because we've told you this before repeatedly |
| 15:06.53 | igcewieling | GeneralSpongebob: phone A <---- leg1 ---> Asterisk <--leg2---> phone B |
| 15:07.02 | igcewieling | your call has only one leg, that is why it falls over. |
| 15:07.04 | GeneralSpongebob | no, you've made half-assed condescending comments. I appreciate that it's frustrating for you but dont' you fucking think other people need to learn stuff too? |
| 15:07.10 | GeneralSpongebob | thanks igcewieling |
| 15:07.18 | [TK]D-Fender | <file> I don't know if I can state it any other way... dialplan steps execute dialplan applications, the dialplan application Dial has to be executed in order for dynamic features to work, and only while that application is active will it work |
| 15:07.24 | [TK]D-Fender | <file> there is no Dial() in there, thus it won't work regardless |
| 15:07.30 | [TK]D-Fender | <Samot> Dial() is what makes a channel |
| 15:07.41 | igcewieling | GeneralSpongebob: basically your entire test is not valid and can never be valid. |
| 15:08.07 | [TK]D-Fender | [TK]D-Fender> ; Note that the DTMF features listed below only work when two channels have answered and are bridged together. |
| 15:08.14 | igcewieling | Don't feel too bad, back in 2003 I did something similar with call parking. |
| 15:08.15 | [TK]D-Fender | <file> only while the channel has dialed something else and is conversing with them |
| 15:08.19 | GeneralSpongebob | so if I understand now, I can test this by calling a colleagues extension or calling in from a cellphone? |
| 15:08.31 | GeneralSpongebob | *I can only test this... I meant |
| 15:08.51 | [TK]D-Fender | "cellphone" isn't a requirement |
| 15:08.58 | igcewieling | GeneralSpongebob: do either of those two options include a Dial line in your dialplan? If no, then it isn't a valid test |
| 15:09.13 | [TK]D-Fender | Call has to arrive at *. that channel's dialplan has to Dial() something else and get answered and bridged |
| 15:09.36 | [TK]D-Fender | That is the point when dynamic features become relevant |
| 15:09.37 | igcewieling | you know there are plenty of example in the samples directory right? |
| 15:09.42 | GeneralSpongebob | Yes, they do. I'll give that a try, thanks igcewieling for attempting to help. |
| 15:10.09 | GeneralSpongebob | I've gone in circles looking at examples. I was following one to start with then found it applied to a very old version. The documentation is not the easiest thing to navigate |
| 15:10.39 | GeneralSpongebob | Mostly I've probably gone wrong with "testing" it in a way that'll never work without realising |
| 15:12.24 | [TK]D-Fender | That's where we were at. |
| 15:13.05 | [TK]D-Fender | so go apply this to dialplan that will call something else via Dial() |
| 15:14.13 | GeneralSpongebob | igcewieling: Tried with my colleague's phone and it works. All this confusion over something so simple :) Thanks again for your help |
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| 15:20.46 | [TK]D-Fender | \o/ |
| 15:23.55 | igcewieling | GeneralSpongebob: read the Asterisk book, this is basic n00b stuff. |
| 15:24.12 | igcewieling | Now I'm off to beat some with a password. |
| 15:28.24 | GeneralSpongebob | There's a book? |
| 15:28.30 | GeneralSpongebob | :P I'm kidding |
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| 16:23.08 | lwlvl | I have a twin-line with two routers. both sip-servers have the same address, because it is the same sip-server at my provider. but I can only register max. 3 numbers per router/line. so I have to tell 3 out of six numbers, to register on the other router. (every number has it's own account...this is Telefonica...so don't wonder). How can I achieve that 3 accounts take the second router? |
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| 16:32.51 | igcewieling | what is a twin-line? |
| 16:33.25 | igcewieling | you need more than one 1 ip address. |
| 16:33.33 | lwlvl | igcewieling, the line was previously an isdn-line... |
| 16:33.42 | igcewieling | that doesn't help at all. |
| 16:33.47 | lwlvl | igcewieling, you can simply see it as two phone lines... |
| 16:33.56 | lwlvl | with two routers |
| 16:34.03 | lwlvl | and two external IPs |
| 16:34.05 | igcewieling | two phone lines plugged into what? |
| 16:34.35 | igcewieling | Is this an Asterisk question? |
| 16:34.36 | lwlvl | 2 dsl-lines |
| 16:34.43 | lwlvl | yes, it is. |
| 16:35.13 | lwlvl | igcewieling, the lines are up and running...everythings fine with them. |
| 16:35.14 | igcewieling | so what do you need help with in Asterisk? |
| 16:35.44 | igcewieling | if you have 2 endpoints on 2 addresses, then setup 2 peers in Asterisk |
| 16:35.44 | lwlvl | I have 6 numbers / 6 sip-accounts from my provider in common |
| 16:36.11 | lwlvl | every number is being registered with her own account |
| 16:36.13 | [TK]D-Fender | this is a multi-home routing question |
| 16:36.36 | igcewieling | lvlinux: registered from your router or registered from Asterisk? |
| 16:36.46 | [TK]D-Fender | From * |
| 16:36.56 | lwlvl | but one line is only allowed to register 3 of them. when I try to register a fourth one, one of the others disconnects. |
| 16:37.02 | [TK]D-Fender | he wants half to use 1 router (and it's connection), and half to the other |
| 16:37.04 | lwlvl | igcewieling, from asterisk |
| 16:37.16 | lwlvl | [TK]D-Fender, exactly. |
| 16:37.52 | igcewieling | sounds like you are doing something which won't work. That is, Asterisk having more than 1 IP address on the same subnet -- that won't work. |
| 16:37.55 | lwlvl | it's not important which numbers use which line...but it mustn't exceed 3 accounts per line |
| 16:37.56 | [TK]D-Fender | is the provider at the same IP? |
| 16:38.06 | [TK]D-Fender | for both peers? |
| 16:38.20 | igcewieling | lvlinux: stop using the term line -- it means nothing in SIP. |
| 16:38.35 | lwlvl | [TK]D-Fender, you mean, whether the sip-server has the same IP on every connection? -yes |
| 16:39.18 | lwlvl | [TK]D-Fender, but every router has it's own public ip... |
| 16:39.33 | igcewieling | FYI, the "correct" solution is to stop sending money to a company which does not provide the service you need. |
| 16:39.33 | [TK]D-Fender | then the only way I can think of is to have each router offer a completely separate subnet, use chan_sip bound to one, and pjsip to the other |
| 16:39.39 | lwlvl | if I can even use another SIP-server, it would be easy.... |
| 16:40.50 | file | igcewieling: it seems to be a trend with this provider |
| 16:41.10 | igcewieling | *nod* A hostile company. |
| 16:41.19 | file | they've come up across here and discourse enough that I remember them |
| 16:41.34 | file | which is generally a bad sign >_> |
| 16:41.40 | lwlvl | file, you mean O2/Telefonica? |
| 16:41.44 | file | yes. |
| 16:42.09 | igcewieling | *nod* Verizon and Level 3 might be real jerks, but at least they don't do something that stupid. |
| 16:42.40 | lwlvl | file, I also had my problems in the first days....it's even unnecessary to give out one sip account per number.... |
| 16:42.43 | igcewieling | No other providers have numbers? |
| 16:43.06 | igcewieling | Telefonica - Spain, correct? |
| 16:43.12 | lwlvl | it wasn't my decision to go to this provider.... |
| 16:43.21 | lwlvl | but I have to live with that shit... |
| 16:43.27 | lwlvl | Telefonica Germany |
| 16:44.52 | igcewieling | Ah, Ok. you are in an impossible position with impossible requirements. best of luck with that. |
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| 16:45.20 | lwlvl | igcewieling, no, not impossible.... |
| 16:45.22 | file | choices, consequences |
| 16:46.24 | lwlvl | igcewieling, I want to give my customer the advice to transfer the numbers to maybe sipgate....away from telefonica....but I'm really afraid of the process (by thinking of telefonica) |
| 16:47.06 | lwlvl | maybe they're disconnecting the numbers "by accident" and block the transfer for weeks.... |
| 16:47.14 | igcewieling | There are others. For example, the provider I use for testing and personal stuff, vitelity.net has Spanish numbers, I assume they have German numbers. |
| 16:47.46 | lwlvl | I don't want to take the responsibility for 2 weeks no reachability.... |
| 16:48.27 | lwlvl | igcewieling, I have no problem with SipGate....I use them for years....I'm more concerned about telefonica in this process... |
| 16:48.29 | igcewieling | lwlvl: changing providers is disruptive. It is as simple as that. You need to ask yourself is switching providers is more disruptive or if you spending a lot of time trying to figure out a workable solution is more disruptive. |
| 16:48.50 | lwlvl | igcewieling, I think you're right... |
| 16:48.57 | igcewieling | In any case, I doubt I'll be able to help. |
| 16:49.43 | lwlvl | I thought about finding a routing-solution....but the sip-server is the same on both connections...that makes it hard.... |
| 16:50.03 | file | if you have to fight the service to make it work how you need, then one has to question long term usage |
| 16:50.27 | lwlvl | file, yes |
| 16:50.46 | lwlvl | as I said...it wasn't my idea to switch to them.... |
| 16:50.58 | lwlvl | I saved their ass more than one time.... |
| 16:51.37 | lwlvl | they simply agreed to a voip-connection and had a Siemens HiPath 3500 TK-device :D :D :D |
| 16:52.08 | lwlvl | at the day where the sun got up for telefonica, the telefone was dead...also the internet-connection.... |
| 16:52.52 | lwlvl | I connected the TK to the fake-S0 of the voip-routers to make it "work" again.... |
| 16:53.18 | lwlvl | and then I had to reset the complete HiPath....that was "fun"... |
| 16:53.58 | lwlvl | now I installed and configured an asterisk-server with several IP-Phones....and now..."that"... |
| 17:35.05 | GeneralSpongebob | lwlvl: Could you use two Asterisk servers, one per public IP that you have, and have them route between themselves within your network? It's not great but if you can only have three SIP connections per IP then maybe that would at least get you working for now until you have a better environment? |
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| 17:48.28 | lwlvl | GeneralSpongebob, yes, I also thought about the same...but I think it'll better to move the numbers away from O2 |
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