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| 12:54.29 | admin0 | hi .. which asterisk version is good for use with a2billing ? |
| 12:55.29 | WIMPy | You will have to ask the a2billing guys. |
| 12:56.38 | admin0 | hmm.. true |
| 12:56.38 | admin0 | ok |
| 12:57.05 | admin0 | is there a way for codec preference selection for sip users ? |
| 12:57.16 | admin0 | or it is something that asterisk will automatically determine ? |
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| 14:42.20 | admin0 | i did apt-get install asterisk and immediately i started to get this: [Oct 20 16:39:17] NOTICE[19389]: chan_sip.c:22622 handle_request_invite: Call from '' (87.98.245.45:5070) to extension '900972597103443' rejected because |
| 14:42.21 | admin0 | extension not found in context 'default'. .. |
| 14:42.35 | admin0 | does the default install of asterisk broadcast/register itself somewhere ? |
| 14:43.26 | [TK]D-Fender | no |
| 14:43.41 | [TK]D-Fender | And there is no such thing as "default install" |
| 14:44.58 | admin0 | default as in apt-get install asterisk |
| 14:45.08 | [TK]D-Fender | the "sample configs" don't, if that's more specifically what you're referring to. Mind you you should never use any of the channel driver or dialplan config samples even as a starting pointt, but rather build your own from scratch and use them as documentation and reference only |
| 14:45.32 | [TK]D-Fender | admin0: debian packaging is also not made by Digium so I can't say what else they might have done |
| 14:46.32 | admin0 | ok |
| 14:46.43 | [TK]D-Fender | admin0: You are almost certainly just getting caught by a SIP mass-scanner trying to find vulnerable systems to defraud |
| 14:47.06 | admin0 | ok |
| 14:47.08 | admin0 | looks like it |
| 14:47.36 | [TK]D-Fender | Go firewall them out |
| 14:47.59 | [TK]D-Fender | And better secure your sip configs |
| 14:48.44 | [TK]D-Fender | "allowguest=no" unless you specifically want to allow anonymous callers to reach you. If so do make sure they point to a context that only allows the specific things you want them to be able to do |
| 14:49.16 | admin0 | before starting up asterisk, i had renamed the default sip/iax/extensions .conf and used blanks in their places |
| 14:50.37 | [TK]D-Fender | then you should know that there is no "default" |
| 14:50.45 | [TK]D-Fender | because you cleared everything out |
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| 16:40.01 | WIMPy | Wow. A PRI card for 3.61 isn't bad. |
| 16:42.48 | ChannelZ | 3 dolla? |
| 16:43.09 | WIMPy | No, Euros. |
| 16:44.41 | vlad_starkov | Question: Is it possible to substitute channel variables in [applicationmap] (features.conf) like this: `some_feature => *,peer/caller,Dial(Local/${SPECIAL_EXT}@test_context)` ? |
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| 17:25.48 | slav3_kitten | dialplan design. should my international calling extension matches all go to a single dial command or should i have it done on an individual basis for each country |
| 17:30.14 | WIMPy | Why do you have multiple extensions? |
| 17:35.31 | lnb | phone has multiple lines, line 2 is mapped to line 1. line 1 is in queue. if 1st caller is answered by line 1, will second caller go to line2 or will queue see line 1 as occupied? |
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| 17:37.11 | vlad_starkov | Question: what is the best way to monitor and kill Confbridge's zombie peers? |
| 17:42.24 | lnb | ask another way. Need queue to ring line 2 if line 1 is busy. does that mean map line 2 -> line 1 or need to make new extensions for line 2 on various phones? |
| 17:45.43 | slav3_kitten | WIMPy, extensions was the wrong word |
| 17:46.37 | slav3_kitten | i have multiple masks for different countries because i don't want a universal international mask as it's my home phone an i don't want people to be able to inadvertantly call countries that will run up my itsp bill |
| 17:47.11 | WIMPy | Sounds like "extension" was the right word. |
| 17:47.54 | slav3_kitten | well it's more of a dial mask |
| 17:48.07 | slav3_kitten | extension i think of as more the phones in the house |
| 17:48.33 | WIMPy | Not in Asterisk terms. |
| 17:50.01 | vlad_starkov | Question: Is it possible to execute CLI command from dialplan? |
| 17:50.13 | ChannelZ | extensions are things you dial. Devices are the phones on your desk. |
| 17:50.15 | WIMPy | Not directly. |
| 17:51.10 | vlad_starkov | WIMPy: You mean `system(/usr/sbin/asterisk -rx 'my_command here')` ? |
| 17:51.19 | WIMPy | yes |
| 17:51.25 | vlad_starkov | WIMPy: ok |
| 17:53.24 | vlad_starkov | WIMPy: So if I need to make possible conference admin to kick all conference participants, I need do this: `system(/usr/sbin/asterisk -rx 'confbridge kick conf_name_here all')` or kinda? |
| 17:54.19 | WIMPy | For example. |
| 17:55.07 | vlad_starkov | WIMPy: It looks like `confbridge kick conf_name_here all` is not applicable. It there a way to kick all the conference participants at once? |
| 17:57.11 | WIMPy | Wasn't that added recently? |
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| 17:57.52 | WIMPy | You can play with marked users, I guess, or parse the participants list. |
| 17:58.40 | vlad_starkov | WIMPy: I think the later is only option currently for 11.5.1 |
| 17:58.42 | rhamnett | Hi wondering if someone can help. I want to know if there is a simple way to detect the audible tone of the fax handshake of a call upon answering and hang up without having to use any third party modules. I am using alaw/ulaw codecs |
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| 18:02.24 | WIMPy | rhamnett: dahdi should be able to do it. |
| 18:03.13 | ChannelZ | faxdetect in sip.conf as well. Then just make your fax extensions hangup. |
| 18:03.37 | rhamnett | WIMPy, I am pure SIP I have read the documentation at http://www.voip-info.org/wiki/view/Asterisk+fax but I can't seem to get asterisk to recognise the fax after enabling faxdetect = yes |
| 18:03.58 | rhamnett | ChannelZ, appreciate the pointer, that's what I was trying to do but can't get anywhere with it |
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| 18:04.22 | rhamnett | i set faxdetect=yes and then made a extent => fax,1,hangup() |
| 18:04.23 | ChannelZ | Your dialplan has to be such that asterisk has some time to listen |
| 18:04.46 | rhamnett | ChannelZ, perhaps that is what I will try next, do I have to answer the channel and wait for a few sec? |
| 18:04.48 | ChannelZ | Like, you have to answer the channel, and then play back a welcome message or something.. or just wait |
| 18:04.52 | vlad_starkov | Question: is it possible to hangup the channel if it does not answer for 60 seconds? (For example if there is Internet connection issues on callee side) |
| 18:05.11 | rhamnett | ChannelZ, can it listen whilst it is playing a welcome prompt without doing anything special? |
| 18:05.13 | ChannelZ | vlad_starkov: use a timeout on Dial |
| 18:05.19 | ChannelZ | rhamnett: yes |
| 18:05.32 | rhamnett | ok awesome ill try it |
| 18:06.40 | vlad_starkov | ChannelZ: you mean to use main timeout in Dial app? It will hangup channel in set number of seconds after the channel has been answered. It doesn't take into account whether this channel is alive or zombie. |
| 18:07.14 | vlad_starkov | ChannelZ: I need something more intelligent |
| 18:07.19 | WIMPy | rtptimeout, session-timers |
| 18:07.35 | vlad_starkov | WIMPy: sip.conf? |
| 18:07.38 | ChannelZ | you said "does not answer" |
| 18:07.45 | WIMPy | yes |
| 18:08.37 | vlad_starkov | ChannelZ: oops, sorry my mistake. Saying "does not answer" I meant it does not response anymore, like broken connection. |
| 18:08.59 | vlad_starkov | WIMPy: I'll check |
| 18:12.13 | rhamnett | ChannelZ, still not getting anything with a wait of 3 seconds, is that too unreasonably short |
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| 18:13.19 | ChannelZ | depends on the machine. I think the average machine sends the CNG tone every 3 or 4 seconds |
| 18:13.32 | vlad_starkov | WIMPy: so, here we go: "rtptimeout=60; rtpholdtimeout=60, rtpkeepalive=60" I think this should be enough |
| 18:13.57 | WIMPy | That allows for long interruptions. |
| 18:14.01 | vlad_starkov | WIMPy: session-timers is a bit more tricky thing |
| 18:14.48 | WIMPy | If you're not using directmedia, rtptimeout is the first thing. |
| 18:14.55 | vlad_starkov | WIMPy: "rtptimeout=15; rtpholdtimeout=15, rtpkeepalive=60" |
| 18:15.02 | WIMPy | But it requires rtp, obviousely. |
| 18:15.38 | WIMPy | Not sure about the hold thing, but I guess the keepalive must be smaller than the timeout. |
| 18:16.12 | WIMPy | And rtptimeout has a off by one thing. Setting it to 15 will kill the call after the 16th second without rtp. |
| 18:16.30 | vlad_starkov | "directmedia" – is a term which describes the schema when Asterisk stays in audio-path all the time? |
| 18:16.48 | WIMPy | No, when it doesn't. |
| 18:17.30 | vlad_starkov | WIMPy: Then I don't use directmedia |
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| 18:18.17 | vlad_starkov | WIMPy: In this case, probably the "session-timers" is what I need? |
| 18:18.38 | WIMPy | No, if you have rtp, rtptimeout is fine. |
| 18:19.13 | vlad_starkov | WIMPy: I just need the Asterisk to ping the end-point from time to time to make sure it alive. Otherwise Asterisk should hangup that channel. |
| 18:19.31 | WIMPy | That's the session-timers. |
| 18:20.09 | vlad_starkov | WIMPy: I'm just thinking, is there VoIP clients that stops sending RTP when user turn off it's mic? |
| 18:20.14 | WIMPy | rtptimeout kills a call if no rtp is received on that channel. |
| 18:20.32 | WIMPy | Don't think so, but definitely possible. |
| 18:20.51 | vlad_starkov | WIMPy: Yepp. So then I should try session-timers? |
| 18:21.03 | WIMPy | They would hopefully send some keep-alives then. |
| 18:21.14 | vlad_starkov | WIMPy: `hopefully` |
| 18:21.31 | WIMPy | With sip, everything is just hope. |
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| 18:21.50 | WIMPy | Sometimes you're lucky, sometimes you aren't. |
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| 18:31.21 | vlad_starkov | WIMPy: Do all the phones support session-timers? (SIP extension, RFC 4028) |
| 18:31.48 | WIMPy | Off course not. |
| 18:32.52 | vlad_starkov | WIMPy: then it's not reliable way |
| 18:33.36 | WIMPy | If you want something reliable, don't use sip. |
| 18:34.01 | rhamnett | ChannelZ, got it working :D had not waited enough |
| 18:34.10 | rhamnett | die fax spammers!! |
| 18:38.34 | ChannelZ | :) |
| 18:38.51 | rhamnett | One of my customers was getting 700 fax calls a day |
| 18:38.56 | rhamnett | to toll free |
| 18:38.58 | rhamnett | US |
| 18:39.02 | ChannelZ | lovely |
| 18:39.16 | rhamnett | telco are useless and will not trace them |
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| 18:48.12 | doug | seems my old extensions.conf used ${CHANNEL(PEERNAME)} to get the current channel |
| 18:48.19 | doug | what's that now (with 1.8+)? |
| 18:52.17 | WIMPy | peername is still peername, but it's not the channels name, but surprisingly the peers name. |
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| 20:56.19 | luka | is trying to configure cisco phone 7961 with asterisk |
| 20:56.52 | luka | but cisco phone is not sending "Authorization:" sip header and thus failing to connect to asterisk |
| 20:56.59 | luka | anybody know how to fix that? |
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| 20:58.33 | nafg_ | Hello |
| 20:59.35 | nafg_ | I have a question about the asterisk REST api |
| 20:59.54 | ChannelZ | ϖ |
| 20:59.55 | nafg_ | First, is it released yet, or when is it scheduled? |
| 21:00.12 | ChannelZ | Well it's set for asterisk 12 which is still being developed. |
| 21:00.25 | ChannelZ | Their previous timetable is shot, not sure if there is any new ETA. |
| 21:00.40 | nafg_ | Hi ChannelZ. My question is this: it says the events API will use web sockets, which is not HTTP |
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| 21:01.05 | nafg_ | my question is, couldn't they use HTML5 server side events (EventSource), or similar (like CouchDB)? |
| 21:01.12 | nafg_ | Would make interop a lot easier |
| 21:02.17 | nafg_ | ChannelZ: is this the right place to discuss it? or is there a better way to communicate with the developers? |
| 21:02.53 | nafg_ | why should I have to use a websockets client lib when I can use plain HTTP. Btw websockets is specifically needed for 2-way communication |
| 21:03.10 | nafg_ | The events rest command would appear to be one way |
| 21:03.59 | ChannelZ | #asterisk-dev perhaps. I doubt they are inclined to switch gears at this point |
| 21:04.06 | lnb | anyone know how to handle 2 lines on sip phone for queue purposes? |
| 21:04.38 | lnb | user wants queue to go to either line if not busy on both |
| 21:04.50 | ChannelZ | The two lines are generally two peers |
| 21:05.00 | ChannelZ | Make them both members of the queue |
| 21:05.05 | lnb | does that mean they have to be individual extension numbers? |
| 21:05.22 | lnb | or can i just map line 2 to line 1 |
| 21:05.34 | ChannelZ | Extensions are arbitrary things |
| 21:05.54 | ChannelZ | Two peers |
| 21:06.03 | lnb | ChannelZ: i don't follow you by 'two peers' |
| 21:06.11 | ChannelZ | I don't know what you mean by 'map line 2 to line 1' |
| 21:06.23 | ChannelZ | That sounds like call waiting with a flashing light |
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| 21:06.54 | lnb | if line 1 is used, then line 2 lights up and rings |
| 21:07.02 | lnb | in 'normal' situation |
| 21:07.32 | lnb | in that case line 2 is mapped to line 1 |
| 21:07.45 | lnb | only one sip account is programmed in. |
| 21:08.18 | lnb | but i am not sure how a queue looks at it if line 1 is busy.. does it look to see there is a 2nd line available on the phone or not |
| 21:08.35 | ChannelZ | It doesn't know anything about "lines" |
| 21:08.56 | lnb | channels then |
| 21:12.12 | ChannelZ | You can look at 'ringinuse' |
| 21:12.57 | lnb | ChannelZ: have you ever configured queue? |
| 21:13.14 | ChannelZ | AFAIK otherwise queues just try to send calls to the members, they either accept them or not. |
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| 21:14.02 | ChannelZ | There used to be call counters but they got removed. |
| 21:14.17 | ChannelZ | And no, I haven't used queues in awhile |
| 21:14.26 | lnb | ok |
| 21:15.09 | lnb | well what i am basically asking is, with a multi line sip phone, how to get the queue to hit all lines not busy |
| 21:15.58 | lnb | i know if i create new extension and configure line x on sip phone and then add to queue it should work. |
| 21:16.12 | ChannelZ | As I said, most "multiline" phones you use separate peers for each "line" |
| 21:16.21 | lnb | i would like to avoid that if possible by just mapping line 2, 3 to line 2 |
| 21:16.24 | ChannelZ | Otherwise all you have is call waiting. Which is fine.. |
| 21:16.26 | lnb | line 1 i meant |
| 21:17.03 | lnb | separate peers = extension numbers? |
| 21:17.18 | ChannelZ | Not necessarily, extensions and peers are unrelated. |
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| 21:18.16 | ChannelZ | In a way you have more of a device-specific question |
| 21:19.02 | ChannelZ | Which is, if someone/something (asterisk in this case) tries to send more than 1 call to the device, what does it do? |
| 21:19.36 | ChannelZ | Asterisk isn't necessarily actively aware of these "multiple lines" as you've described |
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| 21:20.38 | lnb | then I think that means create a new extension for line 2 and line 3... |
| 21:20.59 | lnb | i think it defeats the purpose of the queue but that's what the client wants |
| 21:22.21 | ChannelZ | I think some people can't get over the analog/traditional concept of "lines." |
| 21:22.31 | ChannelZ | Like I have these 4 buttons on my phone, I must have 4 lines! |
| 21:23.23 | lnb | yes that is exactly the thinking |
| 21:23.44 | lnb | but you can have 4 lines |
| 21:24.02 | lnb | you could have 4 businesses each number going to one of those 4 |
| 21:24.11 | ChannelZ | It depends on whether they want to treat them as discreet things. |
| 21:24.36 | ChannelZ | If they want to go old-school and treat them like lines and yell at each other through the building, "Hey Joe, pick up line 3" that's a whole other mess |
| 21:24.37 | lnb | usually for clients i just map line 2/3/4/5/6... to line 1 |
| 21:25.01 | lnb | thats camping |
| 21:25.04 | lnb | thats different |
| 21:25.42 | ChannelZ | If they just want to accept multiple calls, that might just be a setting in the device. |
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