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| 00:20.45 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.1 (2019/7/11) 16.4.1 (2019/7/11), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 07:32.04 | stevenm | anyone familiar with using a OneAccess ONE100 box as an ISDN2 gateway? had this thing hanging around, completely unused - for years :S |
| 07:32.25 | stevenm | the UI on it is sloow and seems to hate new browsers too - thinking of just putting it in the bin |
| 07:33.54 | stevenm | the oneaccess website doesn't seem to let you get firmware or manuals any more either |
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| 08:44.16 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.1 (2019/7/11) 16.4.1 (2019/7/11), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 11:10.45 | FuriousGeorge | hey all |
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| 11:14.16 | FuriousGeorge | i'm trying to write a loop, either using a callfile or nested while, to call all my extensions in order and playback a file |
| 11:14.44 | FuriousGeorge | at first i tried with a nested loop in the dialplan, but i was having trouble continuing after an answer |
| 11:15.20 | FuriousGeorge | then i started to write a script to generate call.files, but started to doubt that it was a better approach before finishing |
| 11:16.28 | FuriousGeorge | actually, maybe that's the way to go, and then use the result of the playback command to tell if there was success or not |
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| 12:37.20 | FuriousGeorge | i'm stuck here. i have fxs channel banks with analog phones set to autoanswer. i'm trying to devise a way to test if they are working as expected |
| 12:37.41 | FuriousGeorge | if they autoanswer, then they work, whereas if they ring for a few seconds, they don't |
| 12:37.57 | FuriousGeorge | every test i devise gives me some false positives or negatives depending on the situation |
| 12:38.06 | FuriousGeorge | here's what i have now: |
| 12:39.44 | FuriousGeorge | https://pastebin.com/B9YfcjwE |
| 12:40.13 | FuriousGeorge | i've tried it a couple of other ways too. in this case, even if the sip channel im using as a standing extension does not answer, playback returns success for whatever reason |
| 12:40.32 | FuriousGeorge | jumping based on dialstatus is impossible |
| 12:40.50 | FuriousGeorge | when using a call file, that is |
| 12:41.30 | FuriousGeorge | and if i just write a loop in the dialplan and call it from another extension, hoping to get it to cycle through, then when an extension answers the loop stops |
| 12:56.46 | Samot | How does the phone autoanswer? |
| 12:56.56 | Samot | It's an analog phone. |
| 12:57.08 | Samot | FuriousGeorge: ^^^^ |
| 12:57.31 | FuriousGeorge | Samot: it has some circuitry that autoanswers |
| 12:57.46 | FuriousGeorge | i don't think it can be made to ring normally, actuallt |
| 12:57.46 | Samot | And does it autoanswer on an normal call? |
| 12:58.03 | FuriousGeorge | Samot: on any call, im pretty sure |
| 12:58.11 | Samot | That's not what I asked. |
| 12:58.14 | Samot | I asked if it does. |
| 12:58.24 | Samot | "Pretty sure" means you haven't got a firm answer. |
| 12:58.38 | FuriousGeorge | what i mean is that there may be SOME way to make it ring, but i don't think so |
| 12:58.43 | FuriousGeorge | by default it auto-answers |
| 12:58.50 | Samot | That wasn't my question. Again. |
| 12:59.00 | Samot | Does it autoanswer when you call it normally? |
| 12:59.03 | Samot | Without a call file? |
| 12:59.16 | FuriousGeorge | yes, in every case that i've found it autoanswers, by default |
| 12:59.39 | Samot | And what should happen when it auto answers? |
| 13:00.20 | FuriousGeorge | i need to distinguish between when the continuity on the line is good, and it autoanswers, or if the continuity is not good, in which the ATA just rings the circuit |
| 13:00.44 | FuriousGeorge | i can do this easily enough with a point to point call, jumping based on dialstatus |
| 13:00.59 | Samot | So what should happen? |
| 13:01.04 | FuriousGeorge | but i need to call 200+ extensions, and don't want to sit there and do it, so i'm trying various ways to write a loop |
| 13:01.18 | Samot | What should happen when it auto answers? |
| 13:01.23 | Samot | Is there a playback? |
| 13:01.29 | Samot | Is there a live call? |
| 13:01.31 | FuriousGeorge | yes, i'm using playback |
| 13:01.31 | Samot | What? |
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| 13:01.54 | FuriousGeorge | currently it playsback tt-weasels |
| 13:02.01 | Samot | And then phone is not auto answering when you use a call file? |
| 13:03.11 | FuriousGeorge | no, it still autoanswers. i can't jump on dialstatus, though, nor can I jump based on PLAYBACKSTATUS. the former gives a false negative (answered calls go to NOANSWER, and the latter gives a false positive (the call times out, but playbackstatus is success) |
| 13:03.26 | FuriousGeorge | that's when i use a callfile |
| 13:03.49 | FuriousGeorge | again, if i just manually called an extension, then i'm sure it would jump like i'm accustomed to it jumping |
| 13:04.02 | Samot | But you haven't tested. |
| 13:04.04 | FuriousGeorge | but i'm trying to do this either with an external script or a dialplan loop |
| 13:04.12 | Samot | Because you are again giving "pretty sure" type answers. |
| 13:04.23 | FuriousGeorge | no, i've tested many different ways |
| 13:04.34 | Samot | So it does jump properly then? |
| 13:04.46 | FuriousGeorge | i gave a pretty sure answer to if it always autoanswers because i didn't want to speak in absolutes |
| 13:05.04 | FuriousGeorge | i was not thinking that you mean callfile vs "regular" in which case, absolutely yes it autoanswers |
| 13:05.33 | Samot | Yes. That is what I was asking. When not using a call file the call does what it is supposed to do all the way through? |
| 13:05.54 | Samot | Auto answers, playsback, gives the proper responses and does your jump... |
| 13:06.32 | FuriousGeorge | there is no difference in the behavior of the phone when using a call file or not |
| 13:06.47 | Samot | And the rest of it? |
| 13:06.58 | Samot | The playback is proper? |
| 13:07.08 | Samot | You get the proper dialstatus or playback status? |
| 13:07.16 | Samot | It does the jump based on said status? |
| 13:07.28 | FuriousGeorge | playback is fine. right now i have sip extensions that im using to test an i'm manually answering. my problem with tracking the dialstatus remains. |
| 13:07.37 | FuriousGeorge | so it has nothing to do with the extension and how it answers |
| 13:07.44 | Samot | Sigh. |
| 13:08.04 | Samot | So the issue is, no matter if it's a call file or not the dialstatus/playback status is not gotten or followed? |
| 13:08.05 | FuriousGeorge | no matter what the endpoint is, if i use a callfile, i cannot jump on dialstatus or playbackstatus and get useful information |
| 13:08.12 | FuriousGeorge | with the example that i pasted above |
| 13:08.20 | Samot | That's probably because it was answered. |
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| 13:08.35 | Samot | Which would require the g option in the Dial() to continue on. |
| 13:09.10 | FuriousGeorge | g would continue if the called party hangup, but, if im not mistaken, when playback stops the calling party is the one hanging up |
| 13:09.11 | FuriousGeorge | not sure |
| 13:09.15 | file | you're also using a call file incorrectly... |
| 13:09.41 | file | "Dial this channel, once answered send it to this context and extension" |
| 13:09.42 | Samot | I was going to get to that once I determined it worked as expected without the call file... |
| 13:12.44 | FuriousGeorge | file Samot should i be using application dial, then? |
| 13:13.01 | FuriousGeorge | and pointing it at a local channel that playsback some file? |
| 13:13.34 | FuriousGeorge | would it be easier to do this without a callfile? i had a nested while set up before, which worked great to output noops correctly |
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| 13:14.53 | file | you can use a call file, but you have to think about what happens when the outgoing channel is answered |
| 13:15.02 | file | for example - you can send it to the Wait application in the call file |
| 13:15.06 | file | which will do nothing except wait. |
| 13:16.34 | FuriousGeorge | file the problem is that sending it to the dial application, which would certainly be the intuitive thing to do, does not give me anything useable in $dialstatus. so i need to think of some other way |
| 13:16.46 | FuriousGeorge | just like playback doesn't give me anything useful for playbackstatus |
| 13:16.50 | FuriousGeorge | this is hardly intuitive |
| 13:17.01 | file | fix the problem above first |
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| 13:17.19 | FuriousGeorge | "Dial this channel, once answered send it to this context and extension" |
| 13:17.30 | file | because, as it is, you would end up potentially dialing the thing twice, and having the dialplan execute twice at the same time |
| 13:17.34 | FuriousGeorge | im guessing "this channel" should be a local channel |
| 13:17.45 | file | which will just confuse and make no sense |
| 13:17.58 | file | yes |
| 13:23.40 | sibiria | i think you're still understanding that you are dialing _twice_ |
| 13:23.52 | sibiria | the fundamental function of a call file is that it will always begin with asterisk dialing |
| 13:24.04 | sibiria | once connected, it jumps into the extension you specified |
| 13:24.35 | sibiria | it's not a means of _only_ jumping into a dial plan |
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| 13:24.48 | sibiria | still not understanding* |
| 13:29.26 | FuriousGeorge | i have to specify a channel and an extension, so i assume those are supposed to represent the two times i am dialing |
| 13:30.02 | FuriousGeorge | i can take this example right out of the docs |
| 13:30.03 | FuriousGeorge | https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels |
| 13:30.49 | Samot | Your call file is going to call loop,s,1 |
| 13:31.12 | Samot | That is the leg 1 channel. That's is going to be answered and then it's going to dial SIP/201. That is leg 2. |
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| 13:31.48 | FuriousGeorge | thought i tried it that way already, but let me try again |
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| 13:32.48 | Samot | Did you execute a call file? |
| 13:32.52 | Samot | Then that is what happens. |
| 13:35.49 | FuriousGeorge | Samot: you mean the call file i'm going to make, or the call file i already made? |
| 13:36.34 | Samot | I mean call files in general. |
| 13:36.37 | Samot | That's how they work. |
| 13:37.21 | Samot | They are going to call the context,extension,priority or application with arguments to execute. |
| 13:37.45 | Samot | So in your case its calling Local/201@loop/ |
| 13:38.16 | Samot | So when it hits loop,s,1 it's going to consider that answered and execute what is in loops,s,1. |
| 13:38.29 | Samot | Again, in your case you have a Dial(SIP/201) so that is what it is going to do. |
| 13:48.47 | FuriousGeorge | this is probably useable: https://pastebin.com/ZtpZRUgg |
| 13:49.02 | FuriousGeorge | because i get a correct response on no-answer, which is my most likely fail case |
| 13:49.12 | FuriousGeorge | still, it would be nice to also get a correct response on answer |
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| 14:03.08 | Samot | https://www.irccloud.com/pastebin/ZzGPqAGS/ |
| 14:03.19 | Samot | FuriousGeorge: Try that. |
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| 14:03.43 | Samot | OOps. |
| 14:03.45 | Samot | One sec. |
| 14:04.05 | Samot | same => n,NoOp(${DIALSTATUS}) <--- Missed the closing ) on that line, #7 |
| 14:04.27 | Samot | Fixed. |
| 14:05.50 | sibiria | FuriousGeorge: you will still be dialing twice |
| 14:06.15 | sibiria | and to be practically correct, what you are doing is making a call-within-a-call - a call transfer of sorts |
| 14:06.16 | Samot | Try the example I gave. |
| 14:06.50 | sibiria | calling Dial() in a dial plan originating from a call file is the most basic type of call transfer. this is what you are doing |
| 14:07.16 | sibiria | i.e. bridging your initial call with a new one |
| 14:07.38 | Samot | It's not a transfer. |
| 14:07.48 | Samot | Leg 1 is local, it can stay on the machine. |
| 14:08.13 | Samot | Leg 2 is the Dial(SIP/202) that is going to call the destination and bridge the local channel legs. |
| 14:08.14 | sibiria | if you absolutely must dial with some option, e.g. 'g', then you should originate your call in another way, not through a call file |
| 14:08.16 | Samot | Like a normal call. |
| 14:08.38 | Samot | sibiria: His call file was derped to begin with. |
| 14:08.42 | Samot | That was pointed out. |
| 14:08.51 | Samot | My example fixes that derpness. |
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| 14:21.53 | FuriousGeorge | sibiria: g does nothing to help, so i don't think i need it |
| 14:23.04 | FuriousGeorge | samot: just noticed that. thanks |
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| 15:01.39 | FuriousGeorge | Samot: that might work, except that ${DEST} is the number I'm testing, and that will be autopopulated by the script generating the files |
| 15:02.00 | Samot | OK. |
| 15:02.06 | Samot | It's an EXAMPLE. |
| 15:02.10 | Samot | Does it work? |
| 15:02.16 | Samot | If yes, modify to your needs. |
| 15:02.32 | Samot | In no, then why didn't work? |
| 15:02.39 | Samot | s/In/If/ |
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| 15:04.29 | FuriousGeorge | Samot: i did modify it such that the dialplan looked like this: |
| 15:04.30 | FuriousGeorge | [loop]exten => _XXX,1,NoOp(Dialing Destination)same => n,Dial(SIP/201) |
| 15:04.34 | FuriousGeorge | oops |
| 15:04.47 | FuriousGeorge | [loop]exten => _XXX,1,NoOp(Dialing Destination)same => n,Dial(SIP/${DEST}) |
| 15:04.47 | Samot | OK. |
| 15:04.50 | Samot | Did it work? |
| 15:04.53 | FuriousGeorge | like that ^^^ the second one |
| 15:04.56 | Samot | Wait.. |
| 15:05.05 | FuriousGeorge | that's why i set ${DEST} in the callfile |
| 15:05.15 | FuriousGeorge | but i see what you did there |
| 15:05.20 | FuriousGeorge | you set it to _XXX |
| 15:05.35 | Samot | Because you had Local/201@loop |
| 15:05.39 | Samot | 201 is the EXTEN |
| 15:06.10 | Samot | You need to make it Local/s@loop if you don't want to specific an extension to match against. |
| 15:06.31 | FuriousGeorge | i see it now. i take out my ${DEST} |
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| 15:07.28 | FuriousGeorge | and then i have my script modify the 201 part, because it will always hit _XXX in loop |
| 15:08.11 | Samot | Does the example work? |
| 15:08.17 | FuriousGeorge | it will in a second |
| 15:11.59 | FuriousGeorge | Executing [101@loop:1] NoOp("Local/101@loop-00000022;2", "Dialing Destination") in new stack |
| 15:12.05 | FuriousGeorge | i think i let it ring too long |
| 15:12.06 | FuriousGeorge | sec |
| 15:20.02 | FuriousGeorge | Samot: https://pastebin.com/ajchwdwH |
| 15:20.47 | FuriousGeorge | like one of the other ways, we can deduce the difference between an answer and no-answer, just not the way we intend to |
| 15:20.56 | file | Samot misspelt weasels in that |
| 15:21.13 | FuriousGeorge | misspelled? |
| 15:21.16 | FuriousGeorge | ;) |
| 15:21.30 | Samot | I did. |
| 15:22.02 | Samot | However, the results are what were expected. |
| 15:22.07 | Samot | The playback happened. |
| 15:22.43 | amferreira | Hello. I'm looking for recommendations for CDR software, mainly for reporting. any ideas/suggestions? |
| 15:22.57 | Samot | No. |
| 15:23.06 | Samot | CDR reporting is generally a business logic. |
| 15:23.20 | Samot | There are a couple out there but no guarantee they give you what you want. |
| 15:24.09 | FuriousGeorge | the second case does give you the NOANSWER dialstatus, but not as a result of the NOOP. instead it comes from chanlocal in the related console output |
| 15:24.09 | FuriousGeorge | <PROTECTED> |
| 15:24.36 | FuriousGeorge | however, on the success, you get a PLAYBACKSTATUS of SUCCESS |
| 15:24.52 | FuriousGeorge | so this is by far the best result, in that it's pretty clear what's happening |
| 15:28.55 | amferreira | Samot, sure but it's supposed to integrate with Asterisk, so.. What's wrong with the question? |
| 15:30.57 | Samot | There's nothing wrong with it. |
| 15:31.06 | Samot | 11:23:20 AM <Samot> There are a couple out there but no guarantee they give you what you want. |
| 15:31.20 | amferreira | Ok, thanks |
| 15:31.31 | Samot | In other words, they will do reporting on Asterisk CDRs but not provide you with the data sets you might be looking for. |
| 15:31.37 | Samot | Or the reports you would want. |
| 15:31.43 | Samot | It's all basic. |
| 15:33.40 | amferreira | I don't need billing. Just reporting... X made a call to Y, with Z seconds. Mainly call logging. |
| 15:34.19 | amferreira | And a nice way to display this. |
| 15:34.43 | Samot | AsterNIC CDRs. |
| 15:35.05 | Samot | It will give you basic reporting on what calls where made and by where. |
| 15:35.16 | amferreira | I'll give it a shot, thanks |
| 15:35.28 | sibiria | if that's all you need, the native CDR framework will do it just fine |
| 15:35.47 | Samot | Except for that lack of a GUI part |
| 15:35.55 | Samot | Or the code being done. |
| 15:36.31 | sibiria | even the default cdr_csv does exactly that, with more detail |
| 15:36.47 | amferreira | I need the GUI part ;) |
| 15:37.05 | sibiria | read the CSV in libre/office ;) |
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| 15:54.33 | cusco | Hello... when a extension performs a Dial(SIP/something); and that channel answers, how can I know which end ended the call, the caller or the callee? |
| 15:56.36 | Samot | There really isn't a way. |
| 15:57.45 | cusco | even if there is a gracefull hangup? a sip .. something .. byew .. ? |
| 15:57.46 | [TK]D-Fender | "g" dial option. |
| 15:57.58 | cusco | the g, makes it go on with the dialplan, correct? |
| 15:58.03 | [TK]D-Fender | if it continues having answered you'll know. |
| 16:01.48 | cusco | got it, thanks |
| 16:02.03 | Samot | cusco: You can't get the BYE details. |
| 16:02.16 | Samot | You can just see that the log shows a channel being hung up. |
| 16:02.29 | Samot | There's is nothing to guarantee the hangups are in sequential order. |
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| 16:09.23 | sibiria | does pjsip have an equivalent to chan_sip's "useragent" channel variable? |
| 16:09.35 | sibiria | or does it simply not offer anything to read the remote's user-agent |
| 16:09.50 | cusco | Samot: I understand... shouldn't that be something to address in asterisk? |
| 16:10.11 | Samot | No. |
| 16:10.21 | Samot | It's a B2BUA not a SBC/Switch/Proxy |
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| 22:44.02 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.1 (2019/7/11) 16.4.1 (2019/7/11), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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