| 00:13.35 | Slade | damn, on hold with intuit, they're a terrible company, but they've actually got good jazzy hold music |
| 00:13.39 | Slade | long time since i've heard good hold music |
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| 00:17.48 | *** 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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| 00:30.33 | Samot | Well it's good to see your money going for that at least. |
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| 00:38.30 | Slade | its my tax money thats going away because of them thats more annoying |
| 00:39.19 | Slade | but yes. it must have set them back a few dollars for hold music |
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| 01:05.33 | Reinhilde | heh |
| 01:05.54 | Reinhilde | why do psych wards in wisconsin have callerids that end in 7 0s? |
| 01:06.01 | Reinhilde | is that a form of blocked number? |
| 01:09.32 | Samot | Yes. |
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| 02:58.35 | LiuYan | 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 |
| 02:59.42 | LiuYan | (I use files mode in music on hold, and sort=random) |
| 03:04.20 | Samot | Try with a higher verbosity. |
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| 10:20.45 | _ganapathi_ | hi, am using asterisk 16 with WebRTC(sipxml5) . here rtpstart and rtpend port defined as 10000-20000. But asterisk server port is using in those range only. how do i define to use the range on client system as well for rtp |
| 10:21.02 | _ganapathi_ | any configuration need to change for that |
| 10:21.25 | file | no, there is no configuration |
| 10:21.33 | file | the path that is used is negotiated through ICE |
| 10:22.14 | _ganapathi_ | not possible to restrict client port then ? |
| 10:22.21 | _ganapathi_ | or through ICE it's possible. |
| 10:22.35 | file | the client controls all of that, or if NAT is involved then the NAT implementation |
| 10:23.07 | _ganapathi_ | there is no NAT. |
| 10:23.11 | file | each side is independent and can use what it wishes |
| 10:23.57 | _ganapathi_ | i understand. But due to some security issue i need to restrict port ranges on client system as well |
| 10:24.02 | _ganapathi_ | to defined range. |
| 10:24.12 | _ganapathi_ | client & server both are in local LAN only. |
| 10:24.15 | file | that's not anything Asterisk can do |
| 10:24.27 | file | you'll have to figure out how to make the browser do as you wish |
| 10:24.38 | _ganapathi_ | Oooh |
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| 11:03.18 | _ganapathi_ | through stun it's possible to restrict ? |
| 11:03.39 | file | STUN does not inherently restrict it, no |
| 11:03.59 | file | STUN allows discovery of the visible IP address/port and is used by ICE underneath to do the negotiation |
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| 20:04.05 | megaTherion | How can I do outbound calls to numbers with a leading '+' as with the international format? |
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| 20:17.24 | megaTherion | I've found adding '+' to the extern pattern like _+X. works but then I need to duplicate those rules for numbers with leading + and those without... kinda annoying - any better way? |
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| 20:18.15 | file | use a Goto to strip it out and jump to the non +? |
| 20:19.16 | megaTherion | heh Im quite new to asterisk... I've to figure out how this all works - I've found google results which talk about Goto stuff and Noop but IM not sure I completly understand the mechanic of it |
| 20:19.26 | megaTherion | I dont want to blindly copy everything I find because its for my phone XD |
| 20:19.47 | file | it moves to a new extension/context/priority |
| 20:20.03 | megaTherion | hmmm |
| 20:20.13 | megaTherion | contexts are always defined with brackets like '[xxx]'? |
| 20:20.17 | file | Goto(default,${EXTEN:1},1) would jump to the first priority in the default context with the first character of the current extension stripped out |
| 20:20.20 | file | yes. |
| 20:20.34 | file | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-1.1 |
| 20:21.01 | megaTherion | Another thing I dont understand the first field with the placeholders... how is it working really? I mean I've seen things like _XXXXXXX .. which seem to be a specific amount of digits, but _X works for all numbers? |
| 20:22.58 | file | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6 |
| 20:23.00 | megaTherion | file: yeah thanks, I'll read that stuff |
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| 20:24.33 | megaTherion | file: thanks for the response, I've tried asking a couple of task but nobody responded :| In fact I did a lot of googling but there are also many hits which include deprecated options and things you shouldnt do this way any longer... it was confusing me. ;) |
| 20:30.10 | [TK]D-Fender | <megaTherion> Another thing I dont understand the first field with the placeholders... how is it working really? I mean I've seen things like _XXXXXXX .. which seem to be a specific amount of digits, but _X works for all numbers? <- it doesn't |
| 20:33.56 | [TK]D-Fender | Also I don't believe anything in dialplan matching has changed since I started using it... |
| 20:37.04 | megaTherion | Well that comment wasnt related to the dialplan only but also to sip.conf, however it seems that priority 'n' and keywords such as 'same' are added improvements - as I just reading |
| 20:37.14 | megaTherion | and of course I have 0 clue since when you use asterisk ;) |
| 20:37.53 | megaTherion | I'll continue reading so that I better understand what Im doing here ;) |
| 20:38.56 | [TK]D-Fender | "n" and "same" as priorities have been around for about a decade |
| 20:39.52 | [TK]D-Fender | sip.conf had some changes back then, and is pretty much an evolutionary dead end now with pjsip taking over. |
| 20:40.58 | megaTherion | pjsip is what? |
| 20:41.11 | [TK]D-Fender | a replacement channel driver for the SIP protocol |
| 20:41.32 | [TK]D-Fender | chan_sip (sip.conf) is the old one that * started with) |
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| 20:47.02 | megaTherion | another thing is, can I somehow display/filter/alter the incoming numbers I see on an VoIP app? |
| 20:50.53 | [TK]D-Fender | depends what you mean by that |
| 20:51.11 | [TK]D-Fender | Btypically an app will show what it is given |
| 20:51.33 | [TK]D-Fender | So you set what you want it to see within the spec before you call it. |
| 20:51.42 | megaTherion | well right now I seem to receive numbers in the form of +49XXXXXXXX@mybox - that doesnt really match out with the address book entries |
| 20:52.00 | megaTherion | "within the spec", what is *the* spec? :) |
| 20:58.44 | igcewieling | megaTherion: from my dialplan, https://pastebin.com/uNCxgM5c |
| 20:58.49 | [TK]D-Fender | SIP spec for formatting CID, etc |
| 20:59.12 | igcewieling | it is for USA numbers, but you should understand the basic idea. |
| 20:59.23 | [TK]D-Fender | When the call comes in you can change the callerID however you like |
| 21:00.35 | igcewieling | megaTherion: this one is correct: https://pastebin.com/mJ9jGRNJ |
| 21:01.16 | igcewieling | if you are referring to the dialed number, then listen to me. if you are referring to CallerID, then listen to [TK]D-Fender, |
| 21:01.49 | [TK]D-Fender | as for the dialed number ... they send you what they send you. You have to be able to match it to begin processing. |
| 21:09.35 | megaTherion | ok sorry, I'll read up - I was busy |
| 21:10.28 | megaTherion | I refer to the CallerID, not the number I've dialed |
| 21:10.30 | megaTherion | so incoming calls |
| 21:10.40 | [TK]D-Fender | So go change it |
| 21:30.23 | [TK]D-Fender | heads home |
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