IRC log for #asterisk on 20190903

00:00.48echelonyes, but the issue is that cisco phone configurations aren't as straight forward as entering in the sip uri and credentials
00:01.22echelonyou have to get it to pull the xml config file from a tftp, web server or whatever
00:03.26SamotOther cicso models have been made to work with Asterisk.
00:03.53SamotHow they get provisioned is a Cisco thing so no matter what model, you have the same issue.
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00:19.43*** 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: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), 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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01:40.52LiuYanHi, when using m(moh-class) option in Dial application, how can I know which exact file played to the caller? I tried asterisk -rgcvvv, still can't see logs about that
01:45.29SamotWell that would depend on how you setup the music on hold.
01:45.59SamotIf you have it ordered, then the first file is played and then the rest in order. Random, well it's random. And if you stream it, then that is based on the stream.
01:46.24LiuYanSamot: I use 'files' mode in music on hold, and sort is set to 'random'
01:46.38SamotThen it will pick a random file and start from there.
01:46.57SamotThe logs just tell you what MOH class is in being played.
01:47.07SamotNot the files being played it because there may not be files.
01:47.41LiuYanSamot: sometimes my workmate will ask me about which music i'd place there, but i don't know which exact music he/she is talking about (because of sort=random)
01:48.06SamotSo they want to know the file being used?
01:49.04LiuYanSamot: They may just want to know the music name, which I need to know which file was played to them
01:49.37LiuYans/which//
01:50.05SamotSo like describing a song they heard on the radio.
01:50.16Samotthey'll need to give you some clues and you'll need to match that up with your list.
01:50.48LiuYanSamot: Yes, something like that.
01:51.03SamotThis is no different then, for example, you playing a playlist on random and later your friend going "I really like that one song from earlier. What was it?"
01:51.20SamotYou still need more to find that song in your playlist.
01:52.32LiuYanSamot: True, and also if they 'singing' the rhythm, I will know too, but that would be a little awkward to me :)
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03:26.43echeloncan asterisk support cisco video ip phones' video conferencing? is it a standard protocol?
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03:36.43[TK]D-Fender* doesn't do video mixing
03:37.09[TK]D-Fenderif you are bridging 2 channels them it can pas the video on if the codecs match
03:37.28[TK]D-Fenderif you are using * conferencing it will "follow the speaker" and flick back and forth
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06:10.35devdvdWhat's the "preferred" method now for having asterisk generate text to speech?  Static strings.  The last time I did this, I used festival.  Just wondering if there's something better now.
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13:26.44iulhkhi can anyone guide regarding opensips pua "https://paste.linux.community/view/b26ed41d"?
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14:13.43DanFromUKHi, I've been asked by a client to set up a calling card solution quickly. I am aware A2Billing is no longer supported. Does anyone know of any other premade (and low cost) solutions?
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14:23.52SibertHi everyone
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14:24.38SibertI'm wondering if it's possible to execute a macro when a Dial cmd (https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial) times out, kinda the opposite of the M option for the dial command
14:24.58SibertCould anyone point me in the right direction as how to achieve this?
14:25.08[TK]D-FenderThat'd be the next step in your dialplan....
14:26.05SibertOkay, but isn't that also the step after the call is completed?
14:26.51[TK]D-Fenderno
14:27.17[TK]D-Fenderunoless told otehrwise if a call actually bridges it will terminate the channel when that bridged call ends
14:28.40SibertSo that would mean if I have for example following dialplan:
14:28.42Sibertexten => 51,n,Dial(SIP/51,30)exten => 51,1,Log(NOTICE,some logging)
14:28.54Sibertthen the log would not happen if the call is completed?
14:28.59[TK]D-Fendercorrect
14:29.20Siberthm okay thanks
14:30.23igcewielingSilicium: use hangup handlers.
14:36.32SibertNice to see an active IRC btw :)  mostly when I try to get some support for software over IRC there's no response
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14:37.37filethe community site (https://community.asterisk.org/) is generally more active, but this continues to exist
14:38.24igcewielingall the really smart people hang out on IRC.
14:39.06SibertI tried slack when I needed some help with Firebase, where there were about 9000 people, took about 24h to get an answer
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