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| 02:58.50 | carrar | mooska |
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| 04:09.20 | zauberfisch | I am looking for a working example of a music on hold config please. |
| 04:09.38 | zauberfisch | my setup logs "starting music ..." but no sound is played |
| 04:09.58 | WIMPy | Look at the sample configs. |
| 04:10.08 | zauberfisch | they don't seem to be working |
| 04:10.10 | WIMPy | Maybe it just doesn't liek the files you gave it? |
| 04:10.14 | zauberfisch | I must be doing something wrong then |
| 04:10.29 | zauberfisch | I have setup an asterisk on a raspberry using the pre built image from raspberry-asterisk.org and removed the freepbx web interface thingy and configured it manually with config files in /etc/asterisk |
| 04:10.49 | zauberfisch | everything I want is working great so far expect music on hold |
| 04:10.53 | zauberfisch | my config: http://pastebin.com/py7kc4dR |
| 04:11.15 | WIMPy | Does it tell you anything when you increase debug level? |
| 04:11.30 | zauberfisch | WIMPy: not sure. possible. I converted an mp3 to ulaw and put it into a custom folder |
| 04:11.40 | zauberfisch | WIMPy: but when using Playback() it works |
| 04:12.00 | zauberfisch | so I presumed the file is ok. but perhaps Playback() and music on hold work differently |
| 04:12.38 | zauberfisch | with high verbose level asterisk tells me: "-- Started music on hold, class 'default', on ...." |
| 04:12.42 | WIMPy | It should probably be the same. |
| 04:12.52 | zauberfisch | but doesn't make a sound at all |
| 04:13.01 | WIMPy | Debug is somthing else. |
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| 04:15.05 | zauberfisch | oh, I should also mention, my asterisk is version 11 |
| 04:16.03 | zauberfisch | WIMPy: could you share a link to documentation how I can debug asterisk |
| 04:16.27 | WIMPy | 'core set debug <level>' |
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| 04:17.05 | zauberfisch | oh, damn it. I'll have to do afk for a bit |
| 04:17.10 | zauberfisch | WIMPy: thank you so far |
| 04:17.21 | zauberfisch | I'll be back in a few hours |
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| 04:45.49 | xochilpili | hi all |
| 04:46.10 | xochilpili | i have wrote in #freepbx |
| 04:46.13 | xochilpili | im using sipml5 with freepbx and asterisk, i can make the call, and it rings in the other side, when i answered there's no sound. Does someone have solve this? |
| 04:46.39 | xochilpili | i got some : "SIP/2.0 405 Method Not Allowed" |
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| 08:33.30 | xochilpili | anyone? |
| 08:37.02 | xochilpili | i have some audio issues with asterisk and sipml5 in both sides... |
| 08:37.38 | xochilpili | if i make a call from zoiper to another extension it works, also, calling to sipml5 extension via web, it rings, but no audio |
| 08:38.23 | xochilpili | from sipml5 calling to *43 (freepbx) code, to echo test, no hear anything, in the CLI seems to response; from zoiper to *43 works fine |
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| 09:21.26 | linocisco | how to install asterisk 12 on ubuntu server ! any detail guide like http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/ is appreciated |
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| 09:41.28 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F3F7.dip0.t-ipconnect.de) |
| 09:43.28 | Dpunkt | Can someone help me debug some bridging issue when trying to route an incoming call via a different endpoint? |
| 09:44.31 | Dpunkt | Incoming calls work, outgoing calls work. But when trying to dialout to Provider B when i got a call to provider A i have no Audio |
| 09:44.35 | linocisco | how to install asterisk 12 on ubuntu server ! any detail guide like http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/ is appreciated |
| 09:46.31 | Dpunkt | it seems to be an issue in the incoming half, when changing provider a with C it works, when changing B to A or C the behavior stays the same |
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| 09:49.24 | Dpunkt | the channels got bridged via simple_bridge but i see no "probation passed" |
| 09:58.13 | Dpunkt | when using Dial(,,D(123)) it works |
| 09:58.20 | phix | h |
| 09:58.27 | phix | my dahdi is wack |
| 10:01.05 | Dpunkt | it seems asterisk ignores the "direct_media=no" when using the anonymous pjsip endpoint |
| 10:04.03 | file | have you looked at the RTP to see if traffic is being exchanged in the non-working case? |
| 10:07.56 | Dpunkt | when usig rtp debug on i see nothing when its not working |
| 10:08.13 | file | are you behind NAT? |
| 10:08.17 | Dpunkt | yes |
| 10:08.26 | Dpunkt | but incoming calls work |
| 10:08.38 | file | have you configured your pjsip.conf transport with your external IP address and also forwarded RTP ports? |
| 10:08.39 | Dpunkt | (to phones direct connected to asterisk) |
| 10:09.20 | Dpunkt | ports are forewarded |
| 10:09.25 | phix | close(file) |
| 10:09.36 | phix | or is it file.close() |
| 10:09.36 | Dpunkt | local_net is set, external ip is dynamic |
| 10:09.40 | phix | file: are you OO? |
| 10:09.53 | file | external IP is dynamic, in that you haven't set it? |
| 10:09.59 | Dpunkt | yes |
| 10:10.07 | file | because what's probably happening is that both ITSPs are waiting for you to send media to them |
| 10:10.40 | file | or they are sending it and it isn't reaching you because of the wrong IP address |
| 10:11.17 | Dpunkt | when i set my dialplan to ring a local phone when got a incoming call it works |
| 10:11.33 | file | D(123) works because that causes Asterisk to send DTMF, which then allows the remote server to send you media, which gets forwarded to the other server, which then since it has received media starts sending media |
| 10:11.54 | file | but if both sides are either sending media to the wrong address or waiting until you send media... |
| 10:11.57 | file | then you got nothin' |
| 10:12.49 | Dpunkt | so they "correct" their wrong ip address when they receive media from a different adress? |
| 10:13.13 | file | yes |
| 10:13.29 | file | it's a common way to help with NAT, send it to the IP address+port you receive it from |
| 10:13.34 | Dpunkt | ok, so setting external_ip wound resolve the issue |
| 10:13.40 | file | dunno! maybe |
| 10:13.58 | file | depends on what the underlying problem is - all I can do is guess/offer suggestions as I don't control the equipment at your providers |
| 10:14.30 | Dpunkt | the old sip stack had "external_host" which was pretty handy to me to get the external ip |
| 10:14.59 | file | external_signaling_address and external_media_address accept hostnames |
| 10:15.16 | Dpunkt | are they re-lookuped any time? |
| 10:15.19 | file | no |
| 10:15.27 | file | not currently |
| 10:17.00 | Dpunkt | im trying setting the external_ |
| 10:17.09 | Dpunkt | adresses to see whats happening |
| 10:17.19 | phix | file: for i in file: <3 |
| 10:17.45 | file | and now I think I'll take my dog for a walk |
| 10:18.39 | phix | file: or talk to me ;) |
| 10:18.52 | file | nah |
| 10:18.53 | phix | file: then again, can you take my dog too? |
| 10:19.03 | phix | file: her name is Charli |
| 10:19.05 | file | I can not |
| 10:19.05 | phix | she is awesome |
| 10:19.11 | Dpunkt | thanks for help |
| 10:19.18 | phix | she will try and bite you though, cause she does what she wants |
| 10:19.31 | file | my dog is... independent, but loyal |
| 10:19.45 | file | runs off |
| 10:19.46 | phix | file: sounds just like Charli :) |
| 10:20.13 | phix | she is napping on my bed with her ass in my pillow just like I told her not to |
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| 10:49.30 | file | and now I exist again |
| 10:50.18 | eirirs | file file |
| 10:55.23 | phix | file: so what are you? open(file) or file.open()? |
| 10:55.47 | file | classified |
| 10:56.05 | phix | file: Illegal! |
| 10:56.31 | phix | file: You cannot "classify" simple requests like that |
| 10:56.38 | file | sure can |
| 10:56.49 | phix | You live in NL or US? |
| 10:57.35 | file | neither |
| 10:59.50 | phix | yaya |
| 10:59.56 | phix | I like you more then |
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| 11:38.36 | probonic | Should Asterisk be remotely bridging when the two sides of the bridge have different codecs? |
| 11:39.01 | phix | probonic: if you want to |
| 11:40.06 | probonic | well it is, and I'm not getting any audio as a result. If it is remotely bridging the two sides when they have different codecs, how does that work? |
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| 12:55.35 | newtonr | probonic, probably want to file a bug if you can reproduce it. you'll need to attach a packet capture, debug logs and configuration with it. |
| 12:55.55 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
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| 13:01.15 | Fira | Hey guys :) Looking for some suggestions |
| 13:01.49 | Fira | I have N users in M mixing bridges in an ARI application. Let's say i want to bridge them together /temporarily/ |
| 13:02.01 | Fira | Is there a better option than moving everyone back and forth :/ ? |
| 13:02.49 | Fira | I tried looking into the snooping functions but you can only snoop on channels, not bridges, so that'd involve M+N snoop channels. This'd also mean people would hear themselves... and there's a recently filled bug about snooping killing CPU altogether on the bugtracker |
| 13:03.40 | Fira | Or N*2 snoop channels to have everyone snoop to and from a common bridge... |
| 13:05.41 | Fira | TL;DR: I'm looking for a way to do something similar to Mumble's Channel Linking |
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| 13:06.49 | Fira | What i've been doing so far is save the position of every channel, move them to a common bridge, then move them back -- but this sounds really messy to me |
| 13:10.34 | file | that is the only mechanism |
| 13:10.40 | file | and fundamentally what would end up happening anyway |
| 13:10.43 | Rico | hello |
| 13:10.46 | newtonr | Fira, I don't have an immediate answer to your question, but you may want to cross-post in #asterisk-ari |
| 13:10.56 | newtonr | and I just saw file's response |
| 13:11.27 | Rico | I've got a lot of T38 faxes which arrives incomplete (tiff is not A4 format) but asterisk res_fax_digium module show them as result: 'SUCCESS' (FAX_SUCCESS) |
| 13:11.38 | Rico | anybody here with a similar problem , |
| 13:11.39 | Rico | ? |
| 13:11.54 | Fira | file: newtonr: Hmm, alright, thanks |
| 13:16.33 | coppice | Rico: if they are not A4, what size are they? |
| 13:17.07 | Rico | width is the A4 one, but height is not, as if the fax was cut in middle of transmission |
| 13:17.16 | Rico | sometimes I have 4 cm, sometime half page, ... |
| 13:17.47 | coppice | are you sure that's not how they were sent? most FAX machines are capable of sending short pages |
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| 13:18.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:25.13 | Rico | i'm sure : |
| 13:25.16 | Rico | exemple : |
| 13:25.42 | Rico | http://pastebin.com/MaSP1QGd |
| 13:26.54 | coppice | It says 2 pages. were both pages cut short? |
| 13:27.29 | Rico | mmh |
| 13:27.32 | Rico | first page is cut |
| 13:27.45 | Rico | second is complete |
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| 13:28.48 | Rico | really strange |
| 13:29.45 | coppice | well, its not uncommon for VoIP systems to cause the modem to lose sync mid page, but that should be noted in the final outcome |
| 13:30.11 | coppice | does spandsp perform better than the digium module in this context? |
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| 13:30.25 | Rico | I've never test spandsp |
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| 14:54.10 | rogersja | Iâm having issues with the orientation of received faxes in asterisk, any ideas or experience with this? Iâve tried sending test faxes from different fax machines (brands, models, etc) |
| 14:54.46 | aquaguy | Hello, I'm a noob in the PBX world. I've installed asterisk and I'm trying to make internal calls (calls between two local extensions located in the same network). I've created a Dial Plan with a "internal" Outgoing calling rule matching the pattern _6XXN. I've downloaded Asterisk COnnect from the play store and configured it. I'm trying to check the voicemails calling 0000 from the app but nothing happens at all. It shows c |
| 14:55.11 | rogersja | all faxes are received in landscape orientation, even though the content is not rotated. Such that it just squishes the 8.5â x 11â fax to 11â x 8.5â |
| 14:55.33 | rogersja | Iâve tried different viewers and image converters |
| 14:56.19 | rogersja | this doesnât happen all the time, but does happen a majority of the time. |
| 14:57.03 | [TK]D-Fender | aquaguy: You're typing too much and it cut you off at "It shows c" |
| 14:57.42 | aquaguy | sorry |
| 14:57.43 | aquaguy | It shows call in progress, 0000 Callback and the black phone image wiggles but there is no sound at all. Anyone can help me? |
| 14:58.42 | [TK]D-Fender | We don't support whatever that app is here, but for the * side, show us what it's doing |
| 14:58.51 | [TK]D-Fender | ~pb |
| 14:58.51 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:58.53 | [TK]D-Fender | ^^^ |
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| 15:29.08 | flam_ | is it possible to pass sip headers to an AGI script? |
| 15:30.40 | [TK]D-Fender | Thre is effectively no such thing as "passing" at all |
| 15:30.44 | [TK]D-Fender | AGI = dialplan processing |
| 15:31.16 | [TK]D-Fender | You can read all the same vars , call all the same applications & functions.... |
| 15:31.55 | flam_ | so all headers are readable there? |
| 15:32.19 | [TK]D-Fender | [11:31][TK]D-FenderYou can read all the same vars , call all the same applications & functions.... |
| 15:32.22 | [TK]D-Fender | Says it all |
| 15:32.27 | [TK]D-Fender | AGI = dialplan |
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| 15:34.35 | Dodewapwap | I'm trying to make calls from SIP phones go to an XMPP account, and for that it looked like I should use res_xmpp and chan_motif |
| 15:35.07 | Dodewapwap | But currently it isn't working, and everywhere I look people use these only for Google Voice |
| 15:35.33 | Dodewapwap | So can anyone confirm I'm on the right (or wrong) path ? |
| 15:41.30 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/xmpp.conf.sample?revision=420494&view=markup |
| 15:45.43 | Dodewapwap | Ok, so I guess it'll be ok to continue the way I started |
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| 15:51.34 | monsterco | I am working on a high volume of calls system, how can I turn verbose to 9 for a specific extension? |
| 15:53.08 | [TK]D-Fender | monsterco: You can't |
| 15:53.33 | monsterco | can I use grep and pipe using tail -f with asterisk full log? |
| 15:53.55 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 15:54.08 | [TK]D-Fender | monsterco: It's a text file.... |
| 15:54.30 | monsterco | wish this was a feature of asterisk CLI - very handy |
| 15:58.02 | monsterco | I am trying to send a fax and I am getting the following errors (this is a SPA2111): |
| 15:58.05 | monsterco | [2015-06-18 11:57:16] NOTICE[31171] chan_sip.c: FAX CNG detected but no fax extension |
| 15:58.05 | monsterco | [2015-06-18 11:57:22] WARNING[2530] chan_sip.c: Unsupported SDP media type in offer: image 4812 udptl t38 |
| 15:58.05 | monsterco | [2015-06-18 11:57:22] WARNING[2530] chan_sip.c: Failing due to no acceptable offer found |
| 15:58.19 | monsterco | I looked at ATA and enabled/disabled t.38 and in both cases it fails |
| 15:58.32 | monsterco | it is not a fax machine issue because it works on another voip line just fine |
| 15:58.49 | monsterco | where should I look for the problem? |
| 16:00.50 | [TK]D-Fender | monsterco[2015-06-18 11:57:22] WARNING[2530] chan_sip.c: Unsupported SDP media type in offer: image 4812 udptl t38 <- this is very clear |
| 16:01.04 | [TK]D-Fender | UDP received and not allowed by your config |
| 16:01.30 | [TK]D-Fender | You set another one up properly. You set this one up wrong |
| 16:01.43 | monsterco | hmmmm, so when I call the fax line I hear the fax tone - I guess this is one way audio issue? |
| 16:03.04 | [TK]D-Fender | because? |
| 16:11.31 | *** join/#asterisk monsterco (~monsterco@206-248-138-120.dsl.teksavvy.com) |
| 16:17.56 | Fira | Hmm hey |
| 16:18.32 | Fira | I'd need a way to add/edit/manage/remove users on-the-fly to Asterisk... Is there a better way than writing a config parser/writer and reloading ? |
| 16:18.51 | Fira | I see some sql modules around for logging but nothing quite like it for the SIP config |
| 16:19.40 | [TK]D-Fender | Fira: Realtime <- well documented in the BOOK |
| 16:19.41 | [TK]D-Fender | ~book |
| 16:19.42 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:19.51 | [TK]D-Fender | And I still wouldn't call it "better". |
| 16:20.13 | Fira | Hmm, i'll have a look still, thanks for the info :) |
| 16:22.17 | Fira | [TK]D-Fender: Well, it's not so 'realtime' so i see what you mean, but it's still better than writing your own config parser :S... |
| 16:24.02 | Fira | oh, i was looking at the static variation :) |
| 16:30.16 | monsterco | [TK]D-Fender - I am not sure ; just speculating; I checked and it seems all good. Do you mean this is a codec issue or a NAT and no UDP transfer at all? |
| 16:30.28 | [TK]D-Fender | It's a NEGOTIATION issue. |
| 16:30.44 | [TK]D-Fender | They insist on T.38. You don't support it. Fail. |
| 16:55.54 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-dizkvryshkuedqfo) |
| 16:55.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:03.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:05.00 | *** join/#asterisk monsterco (~monsterco@206-248-138-120.dsl.teksavvy.com) |
| 17:06.12 | monsterco | [TK]D-Fender - hmmm, thanks. I though T.38 is an optional thing. So, does T.38 happen on the ATA or on the Fax Machine? |
| 17:06.37 | [TK]D-Fender | It happens at some conversion point |
| 17:07.12 | monsterco | So, there is ITSP, then FreePBX (Asterisk), then ATA, then Copier Fax machine |
| 17:07.25 | monsterco | the only options for T.38 I have is on the ATA (SPA2111) |
| 17:07.54 | monsterco | which I have played with and didn't make a different. Since canreinvite = no, I thought all voice will go directly to fax machine without any manipulation |
| 17:08.17 | monsterco | so I don't know where T.38 gets invovled |
| 17:08.37 | [TK]D-Fender | the side the call is coming from. |
| 17:16.47 | monsterco | So, if T.38 is a common thing and everyone is asking for it then I should have support for it. Would this be an ITSP issue or simply my end of equipment like ATA and Fax machine? |
| 17:16.56 | monsterco | [TK]D-Fender ^^^ |
| 17:20.05 | monsterco | [TK]D-Fender - I also connected the same fax machine to another voip line and it works just fine so i guess it is supporting T.38... |
| 17:21.56 | Milenco | hey guys |
| 17:22.00 | Milenco | just a random question |
| 17:22.19 | Milenco | does it matter in what order a context gets loaded? |
| 17:22.24 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 17:22.41 | Milenco | when including extensions.d/*.conf i mean |
| 17:23.18 | Milenco | i define the list of extensions in entensions.conf and include extensions.d/*.conf from there. |
| 17:23.33 | Milenco | does it matter which context gets loaded first from that folder? |
| 17:23.46 | Milenco | it shouldn't right? |
| 17:23.56 | [TK]D-Fender | It can. |
| 17:24.03 | [TK]D-Fender | If you are defining constants |
| 17:24.24 | [TK]D-Fender | Also depending on the other contexts that are getting included. |
| 17:24.52 | [TK]D-Fender | If 1 file ends with a context name and you want a specific file to continue to ADD to that same context... then the order clearly matters |
| 17:25.02 | Milenco | if i link to a context from another context, and that context isn't loaded yet, then the config would fail to load? |
| 17:27.09 | Milenco | I currently have it like this: |
| 17:27.10 | Milenco | http://pastebin.com/N6i0ZuZy |
| 17:29.32 | *** part/#asterisk aquaguy (~Arkaitz@100.83-213-53.dynamic.clientes.euskaltel.es) |
| 17:30.08 | Milenco | lets assume 04_iax.conf isnt there and every context has it own file, could i strip the 00_, 01_, etc? |
| 17:31.39 | WIMPy | Not sure if I get that right, but the whole dialplan is loaded before it is used. |
| 17:32.17 | WIMPy | And is that paste literal? The empty contexts that is? |
| 17:32.40 | Milenco | yes, i add to the contexts from within the files |
| 17:32.59 | Milenco | 01_incoming.conf starts with [incoming](+) |
| 17:33.03 | WIMPy | Then do it there and only there. |
| 17:33.40 | Milenco | i've just tested removing some of the prefixes and reloading asterisk on a testserver works fine |
| 17:34.43 | Milenco | it seems the order only matters when adding within the same context, the order doesn't seem to matter for contextes(?) as a whole |
| 17:34.49 | Milenco | is that correct? |
| 17:38.10 | WIMPy | yes |
| 17:38.22 | WIMPy | The order of contexts doesn't matter either way. |
| 17:38.30 | Milenco | contexts, thanks :P |
| 17:38.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:38.48 | Milenco | oke, thanks :) |
| 17:38.53 | WIMPy | But the order within a context also only matters for include and switch statements. |
| 17:39.04 | Milenco | hmm it does..? |
| 17:39.21 | Milenco | is that why dialplan show shows it alphabetically? |
| 17:39.41 | WIMPy | Extensions within a context will be sorted to have a most specific type matching. |
| 17:39.46 | WIMPy | Exactely. |
| 17:40.17 | Milenco | thanks |
| 17:41.08 | Milenco | i've inherited this asterisk setup some time ago and noticed that external parties could potentially make outsides calls, so looking in to fixing that, but that means redoing the contexts |
| 17:43.20 | Milenco | (i'm still thinking it out in my head :P) |
| 17:43.27 | WIMPy | If you always use explicit priorites (i.e. no n) you could have all the lines of a context in random order. Would make it completely unreadable but Asterisk would sort it anyway. |
| 17:44.43 | Milenco | are there any best practices for setting up these contexts to allow for security and stuff, like only let local clients make outside calls |
| 17:45.10 | WIMPy | Use includes. |
| 17:45.23 | Milenco | yeah i think i got a setup in mind |
| 17:45.38 | Milenco | this setup already uses some includes, but in a not so thoughtful way |
| 17:45.49 | WIMPy | Make contexts per allowed functions and include them to contexts assined to the peers. |
| 17:45.54 | Milenco | everything is linked together basically for all parties |
| 17:46.20 | Milenco | be it external voip companies offering us pbx trunks, other sites, etc. |
| 17:46.56 | Milenco | there hasn't been any abuse yet from outside parties and i manage all the internal things for all sites, but i rather restrict and redo the contexts |
| 17:47.28 | Milenco | one question remaning (maybe 2) :P |
| 17:48.27 | Milenco | can i differentiate incoming iax calls based on a default user and everybody else? |
| 17:49.05 | Milenco | i have a external party offering us calls and i dont want them to reach internal clients directly, while keeping this option open for other parties |
| 17:49.27 | WIMPy | Sure. Just the same as with sip. Theres a general context and one per peer. |
| 17:49.27 | Milenco | like, other site locations |
| 17:50.01 | Milenco | all my own managed server are using the same iax credentials for internal traffic |
| 17:50.31 | Milenco | so i would like to give them some extra permissions while not doing so for all the others/default |
| 17:51.24 | WIMPy | That's what the contexts are there for. |
| 17:51.34 | Milenco | hmm my iax.conf is stripped down, let me grab a default one |
| 17:51.44 | Milenco | duhhh |
| 17:51.48 | Milenco | nvm i see it |
| 17:51.50 | Milenco | thanks |
| 17:52.04 | Milenco | didnt figure iax had the same concept of contexts as sip |
| 17:52.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:53.44 | WIMPy | It's the same in most channeltypes. |
| 17:53.46 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 17:54.39 | Milenco | now that you say this i think, of course, you are right, how could i miss this :P |
| 17:54.46 | Milenco | i've configured this for dahdi as well multiple times |
| 17:55.09 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-198-253.dynamic.qsc.de) |
| 17:55.10 | Milenco | im doing asterisk as i go but havent really taken much take into deepening myself with the sbject |
| 17:55.55 | Milenco | hmm, basically i only need 4 contexts i think |
| 17:56.26 | Milenco | internal_calls, external_calls, incoming_trust, incoming_untrust, voiphints |
| 17:56.56 | *** join/#asterisk angryuser (~angryuser@LCaen-656-1-198-99.w193-251.abo.wanadoo.fr) |
| 17:56.56 | Milenco | in the pastebin above i would also need a clicktocall context |
| 17:57.34 | Milenco | of course, i would need other function specific contexts, but in essense i think i could do with that |
| 17:57.43 | Milenco | any thoughts/things i might be missing? |
| 17:59.44 | WIMPy | That's pretty specific top your setup. But the basic idea look right. |
| 18:01.59 | Milenco | ok, thanks again :) |
| 18:02.55 | Milenco | last question is about the order: lets say i have this on line 1: "exten => 1[234]X" and line 2: "exten => 1[456]X" |
| 18:03.11 | Milenco | would the order in which the config is loaded matter when i call 140? |
| 18:03.58 | Milenco | (there might be other situations where there is an even match in exactness) |
| 18:04.29 | WIMPy | No. It's always sorted. |
| 18:05.22 | WIMPy | In this case the first one would match due to them being equally specific, but literally before the other. |
| 18:09.00 | rmudgett | Milenco: See https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 18:09.12 | Milenco | simplifying configs is hard.. :P |
| 18:09.45 | Milenco | thanks rmudgett, reading now |
| 18:10.25 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:11.16 | [red] | Milenco: if you're more comfortable with using a specific programming language, asterisk AGI works very nicely |
| 18:11.22 | [red] | or you can play around with lua |
| 18:11.31 | [red] | i enjoy using lua a lot rather than .conf |
| 18:11.50 | Milenco | yeah, i've been using AMI for some things |
| 18:12.01 | Milenco | should still look into AGI |
| 18:12.29 | kfife | Maybe a dumb question, but what's the benefit of using same => vs. exten =>? |
| 18:12.31 | Milenco | but i'm trying to master the asterisk config style rather then doing it externally |
| 18:12.40 | kfife | besides one character ;-) |
| 18:12.45 | zauberfisch | lol7 |
| 18:12.45 | Milenco | same => saves the hassle of writing it again |
| 18:12.55 | Milenco | also its easier when moving stuff around when working on something |
| 18:12.58 | kfife | hassle of writing exten? |
| 18:13.04 | Milenco | same => n, is what i use mostly |
| 18:13.27 | WIMPy | No, the extension. |
| 18:13.28 | kfife | how's that different than exten => n, |
| 18:13.31 | Milenco | yeah, sometimes you want multiple extensions doing partly the same thing |
| 18:13.39 | [red] | other than a preferred syntax, there's no real difference |
| 18:13.46 | Milenco | no it works the same |
| 18:13.51 | WIMPy | exten needs an extension. same implies the last one. |
| 18:13.54 | Milenco | its just for confenience |
| 18:14.09 | kfife | so it's the savings exactly of one character? |
| 18:14.15 | kfife | or is it |
| 18:14.18 | Milenco | no |
| 18:14.20 | WIMPy | No |
| 18:14.21 | Milenco | normally you'de do: |
| 18:14.23 | kfife | good |
| 18:14.26 | Milenco | exten 1234,1,.... |
| 18:14.29 | Milenco | exten 1234,2,.... |
| 18:14.31 | Milenco | exten 1234,3,.... |
| 18:14.32 | WIMPy | It saves the extension plus two chars. |
| 18:14.37 | Milenco | etc, but with same its like this: |
| 18:14.43 | Milenco | exten 1234,1,.... |
| 18:14.46 | Milenco | same => n,... |
| 18:15.22 | WIMPy | Where using the n priority has nothing to do with same. It can be used the other way round as well. |
| 18:15.26 | Milenco | you could still do same => 2,... if youd like btw |
| 18:15.26 | [red] | yeah, it's less typing than exten => 1NXXNXXXXXX,1,Answer() lol |
| 18:15.36 | Milenco | true WIMPy |
| 18:17.10 | kfife | so for exten => s,1,Noop(Hello) |
| 18:17.51 | kfife | is same => s,n, noop(world) != exten => s,n,noop(world) ? |
| 18:18.04 | kfife | OH! |
| 18:18.05 | Milenco | no, its the same |
| 18:18.05 | kfife | I get it. |
| 18:18.08 | kfife | I get to drop the S |
| 18:18.10 | kfife | 's' |
| 18:18.13 | kfife | as well |
| 18:18.15 | kfife | ? |
| 18:18.16 | [red] | yup |
| 18:18.17 | Milenco | you can use same and n independent from each other |
| 18:18.20 | Milenco | sorry for the confusion |
| 18:18.27 | [red] | s is techically an extension |
| 18:18.29 | WIMPy | Yes, no repeating the extension. |
| 18:18.34 | Milenco | yeah, same => s,n, its the correct syntax |
| 18:18.38 | kfife | Got it. |
| 18:18.39 | [red] | so when using same => drop the 's' |
| 18:18.41 | kfife | Makes sense. |
| 18:18.49 | Milenco | isn't* |
| 18:18.56 | kfife | I knew it had to be something. :-) |
| 18:19.01 | Milenco | man i confuse everything/everybody now |
| 18:19.06 | Milenco | better get a drink |
| 18:19.20 | [red] | 5 o'clock somewhere |
| 18:19.30 | kfife | Yes, get a drink. Water is necessary for good health |
| 18:19.36 | kfife | (tee hee..._ |
| 18:22.05 | *** join/#asterisk airjump (~Thunderbi@p5B0A0596.dip0.t-ipconnect.de) |
| 18:23.16 | kfife | Any opinions on a the best open-source MCU for asterisk and video conferencing? |
| 18:25.54 | Milenco | haha just got a coke (cola) |
| 18:26.13 | Milenco | whats a MCU :P |
| 18:26.43 | Milenco | ohw, an asterisk video conferencing end-point? |
| 18:26.48 | Milenco | no, sorry :P |
| 18:26.52 | suYin | crashed today one of our customer server... because of shift + g.. the "g" was live in line 1 :( |
| 18:31.38 | TazzNZ | kfife, we are using the Polycom CX5500's |
| 18:31.55 | TazzNZ | they are pretty good, but they do have quite a high $$$ |
| 18:42.22 | *** join/#asterisk airjump (~Thunderbi@p5B0A0596.dip0.t-ipconnect.de) |
| 18:46.04 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 18:46.29 | Milenco | yeah those video conferencing endpoints are pretty expensive |
| 18:46.41 | Milenco | we are using lifesize here |
| 18:47.01 | Milenco | didn't like the original product that much, many firewall traversal issues and such |
| 18:47.18 | WIMPy | Get a RPi with camera module and instal a SIP client? |
| 18:47.20 | Milenco | but we've upgraded to lifesize cloud (with the same hardware) and thats working out pretty nice |
| 18:47.35 | Milenco | RPi? |
| 18:47.45 | Milenco | raspberry pi? |
| 18:47.49 | WIMPy | Raspberry Pi |
| 18:48.05 | Milenco | then you're probably not the one needing to educate your users |
| 18:49.05 | Milenco | corporations dont have that much time on there hand designing these things with little hassle and nice featured gui's |
| 18:49.13 | WIMPy | Do they need to know somethign they don't need to know to operate any other phone? |
| 18:49.33 | Milenco | lifesize cloud costs us 6000 euro yearly for 25 users, including conference rooms |
| 18:49.47 | WIMPy | Oh, I always thought tiome doesn't matter. The employees are payd anyways. |
| 18:49.47 | Milenco | you'd think you'd be way easier off installing skype |
| 18:49.53 | Milenco | or doing something else |
| 18:50.08 | Milenco | but these corporate guys want all the nice things and no hassle and such |
| 18:50.32 | WIMPy | Or at least the promise thereof. |
| 18:50.47 | Milenco | haha true, but employees are expensive as well |
| 18:50.53 | Milenco | at least, it people where i work :P |
| 18:51.01 | WIMPy | How many of these no hassle solutions are really hassle free? |
| 18:51.13 | Milenco | none |
| 18:51.21 | Milenco | i believe, or almost none |
| 18:51.44 | WIMPy | That's what I found. |
| 18:51.48 | Milenco | but little hassle is still less expensive then lots of hassle when doing it yourself :P |
| 18:52.20 | WIMPy | Maybe, maybe not. It's a gamble. |
| 18:52.29 | Milenco | in our case its also a thing of employee shortage |
| 18:52.50 | Milenco | we can't let someone work on it for a few months |
| 18:53.09 | Milenco | we've been/are short on people for years :P |
| 19:00.10 | TazzNZ | Milenco, we did they skype thing |
| 19:00.21 | TazzNZ | we offer the corp. a "$200" solution |
| 19:00.31 | TazzNZ | and they took it over the "$10k" solution |
| 19:00.49 | Milenco | how did it work out? |
| 19:00.50 | TazzNZ | turns out - it was shit |
| 19:01.03 | TazzNZ | problem with skype is BW |
| 19:01.08 | TazzNZ | and you can't control it |
| 19:01.24 | TazzNZ | at least with Asterisk I can say - that IP to this IP is high prio. |
| 19:01.34 | Milenco | maybe its better now that they have launched skype for business |
| 19:01.38 | TazzNZ | (we running cross-country VC's) |
| 19:01.47 | TazzNZ | Milenco, sadly, we are going that route |
| 19:02.03 | Milenco | same here :), although i dont know what a VC is :P |
| 19:02.06 | TazzNZ | but they will still be front ended with Asterisk :) |
| 19:02.14 | TazzNZ | Video Conf :) |
| 19:02.29 | Milenco | the lifesizes were already in place when i was there, recently been bought/configured |
| 19:02.41 | Milenco | i always hated them because they were so expensive and didnt even work really well |
| 19:02.52 | TazzNZ | and btw Milenco - good luck with that freaking install of "Skype for Business" - aka Lync 2015 |
| 19:02.58 | Milenco | and thought we could do way better just using skype or something like that |
| 19:03.19 | TazzNZ | I must say though - our problem was more device related |
| 19:03.32 | Milenco | but earlier this year i was tasked with finding a succesor for our current solution and figured i could do way better |
| 19:03.33 | TazzNZ | "cheap" web cam in a big room doesn't work |
| 19:04.15 | Milenco | turns out all the cheap options dont by far meet all the corporate requirements, and the once that do are super expensive (10k+ invest per site conference room) |
| 19:04.22 | TazzNZ | imho, you need good sound/lighting/camera - then worry about the medium to carry it :) |
| 19:04.43 | TazzNZ | Milenco, yip - welcome to my project :D |
| 19:04.47 | Milenco | lifesize got us covered for about 6k per year now without needing to upgrade our devices |
| 19:05.01 | Milenco | so id like to think i did well, since it works pretty oke now |
| 19:05.29 | Milenco | but it was quite a surprise for me that skype and asterisk sip and such were in shape for these corporate things |
| 19:05.36 | Milenco | were not* |
| 19:05.56 | TazzNZ | I disagree with you there :) |
| 19:06.08 | TazzNZ | Asterisk conf. is pretty much up there |
| 19:06.14 | Milenco | Asterisk may have all the technical requirements, indeed |
| 19:06.26 | TazzNZ | it's the wrapper that you are missing |
| 19:06.27 | Milenco | but it doesn't offer good endpoints |
| 19:06.28 | TazzNZ | :) |
| 19:06.31 | Milenco | for conference rooms |
| 19:06.33 | Milenco | indeed |
| 19:06.46 | TazzNZ | uhm, that CX5500 is a SIP phone |
| 19:07.02 | TazzNZ | I got the same quality via Asterisk as I did via Lync |
| 19:07.35 | Milenco | ohw, you mean using third party endpoints with sip support with your own asterisk |
| 19:07.45 | Milenco | yeah that could work out pretty nice :) |
| 19:08.08 | Milenco | our lifesizes dont offer nice video sip conference options unfortunately |
| 19:08.12 | TazzNZ | yeah - other than the Digium phones you only get third party :D |
| 19:08.13 | WIMPy | See where te RPi might fit in? |
| 19:08.23 | Milenco | no i still dont :) |
| 19:08.56 | WIMPy | It could be the video phone for Asterisk. Full HD in and out if you want. |
| 19:09.20 | Milenco | i mispoke when i said asterisk wasnt up for the corporate video conferencing task |
| 19:09.27 | Milenco | i meant asterisk/digium alone |
| 19:09.31 | TazzNZ | the bigger thing that is missing, imho, is the "how do I setup a meeting" - Lync does this *very* well. The just added a button to Outlook and people can make a meeting without thinking |
| 19:09.54 | WIMPy | Well, usually people miss video mixing capabilitioes from Asterisk. |
| 19:09.58 | Milenco | yeah lifesize has the same thing |
| 19:10.05 | Milenco | a plugin for outlook and such |
| 19:11.48 | Milenco | ah well, if i had to do it all over i'd considered involving asterisk more |
| 19:15.04 | WIMPy | Didn't you say you have a lack of man power? How does using Asterisk fit in there? |
| 19:15.45 | Milenco | asterisk has always been the companies choice |
| 19:16.06 | Milenco | and the maintenance isnt that bad, we got complex and extended needs |
| 19:16.28 | Milenco | so a normal pbx solution wouldnt be able to manage everything we do |
| 19:16.34 | Milenco | or would at a very high price |
| 19:16.37 | TazzNZ | Milenco, there is a hybrid world, where you use Lync for the conf. room and Asterisk for the normal telephony |
| 19:16.39 | Milenco | perhaps we could switch to freepbx |
| 19:16.53 | Milenco | yes we use asterisk for our normal telephony and callcenters |
| 19:17.10 | Milenco | and a different video solution |
| 19:17.32 | WIMPy | Ok, so for that task it's ok to pay people insted of vendors, but for video it's the other way round? |
| 19:17.58 | Milenco | no, its a matter of cost |
| 19:18.12 | Milenco | which is also the reason we are still with lifesize |
| 19:18.13 | WIMPy | Looks a little random to me. |
| 19:18.33 | Milenco | i understand why i may seem that way |
| 19:18.44 | Milenco | but our asterisk base config dates back years |
| 19:18.57 | Milenco | we've been rapidly extending for about 3 years now |
| 19:19.28 | Milenco | its cheaper for us to maintain asterisk and spending some hours every here and thing to improving/updating things |
| 19:19.41 | Milenco | then to switch entirely to a new system |
| 19:19.59 | Milenco | a non asterisk based system that is |
| 19:21.11 | *** join/#asterisk tuxd00d (~tuxd00d@71-223-220-39.phnx.qwest.net) |
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| 19:50.46 | iulhk | using asterisk-13.0.0 Error: "res_config_mysql.c:631 update_mysql: MySQL RealTime: Failed to update database: Incorrect integer value: '' for column 'port' at row 1", any idea please ? |
| 19:51.38 | rrittgarn | your column type is wrong would be my guess |
| 19:51.42 | rrittgarn | in your db |
| 19:52.10 | rrittgarn | change to varchar or allow null on that column |
| 19:54.17 | iulhk | <rrittgarn>: null was already allowed , i changed type to varchar and its fine , thank you |
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| 20:06.07 | raj | can asterisk be used as a sip phone like gizmo5? |
| 20:08.16 | iulhk | can anybody help me to figure out, call file by using MessageSend "http://pastebin.com/rtvz0vHn" ? |
| 20:08.22 | newtonr | raj, there are console drivers to let you use the mic and speakers on the host system |
| 20:09.24 | raj | newtonr, so what features did gizmo5 have that asterisk doesn't? |
| 20:11.18 | newtonr | raj, I don't know anything about gizmo5 |
| 20:11.21 | raj | oh |
| 20:12.11 | WIMPy | Asterisk is not a phone, but it can be used as one. |
| 20:12.21 | raj | I see |
| 20:17.52 | [TK]D-Fender | Asterisk would make a shitty phone.... |
| 20:17.56 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-didjcbyujfyegypl) |
| 20:19.52 | raj | [TK]D-Fender, ok thank you, that's kinda what I was looking for |
| 20:19.52 | malcolmd | lack of built-in acoustic echo cancellation would be one problem :D |
| 20:20.07 | raj | [TK]D-Fender, can you suggest something better? |
| 20:20.43 | [TK]D-Fender | raj: Anything. Asterisk is NOT meant to be a "phone" |
| 20:20.46 | [TK]D-Fender | ~softphone |
| 20:20.47 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 20:20.48 | [TK]D-Fender | ^^ |
| 20:20.58 | *** join/#asterisk monsterco (~monsterco@UNVLON5554W-LP130-01-2925010583.dsl.bell.ca) |
| 20:21.51 | WIMPy | He, why do they have to be voip calls? |
| 20:22.30 | WIMPy | Softphones have even existed for modems, haven't they? |
| 20:25.16 | [TK]D-Fender | He was comparing to another softphone, that's why |
| 20:26.03 | WIMPy | doubts that infobot knew about that. |
| 20:26.54 | *** join/#asterisk sjeijk (~SJ@a83-163-238-141.adsl.xs4all.nl) |
| 20:27.43 | [TK]D-Fender | And ... checkout time, BBL |
| 20:27.53 | sjeijk | hi all, iâm making a telephione installation and was wondering if someone could tell me if this idea is possible in asterisk |
| 20:28.11 | sjeijk | the idea is: |
| 20:28.41 | sjeijk | iâm going to send someone a telephone and when it arrives iâm gonna call that number |
| 20:29.03 | sjeijk | when he/she picks the phone up, i have a callcenter sound installation programmed in c++ |
| 20:29.36 | sjeijk | i want to stream that over voip, i have a sip available |
| 20:30.18 | WIMPy | What's a "callcenter sound installation"? |
| 20:30.32 | WIMPy | And what does "i have a sip available" mean? |
| 20:31.07 | sjeijk | is this the asterisk pbx irc? |
| 20:32.08 | sjeijk | with callcenter sound installation i mean, we have a scripted voice menu where people can navigate thru |
| 20:32.10 | *** part/#asterisk mjordan (mjordan@nat/digium/x-oyjnukcyygklqfxx) |
| 20:33.00 | iulhk | can anybody help me to figure out, call file by using MessageSend "http://pastebin.com/rtvz0vHn" ? |
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| 20:59.03 | kfife | Is the Features.conf 'automixmon' function able to insert a tone into the media stream? |
| 20:59.46 | kfife | I know that in 1.x it couldn't, but not sure if that's been changed. |
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| 21:08.52 | Milenco | hmm WIMPy, are you sure you can set a default context for iax? |
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| 21:33.16 | WIMPy | yes |
| 21:38.14 | kfife | what's the best way to insert a courtesy tone into the media stream when automixmon is invoked? |
| 21:38.51 | kfife | I can't find a straight answer at Google University. I think I'm missing a concept. |
| 21:39.47 | kfife | I have automixmon working, |
| 21:40.09 | kfife | Is that the same as Touch Monitor? |
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| 23:46.51 | iulhk | how to read info messages in dialplan, receiving from caller and send to callee? WARNING[28297][C-00000080]: chan_sip.c:21765 handle_request_info: Unable to parse INFO message from 430f4513-9d9f-3675-afcc-ebcf985ed883. Content-Type:manylink/camera-status? |
| 23:51.09 | [TK]D-Fender | You can't |
| 23:53.12 | iulhk | how to add custom sip header, i just want to pass "Content-Type:manylink/camera-status" from caller to callee, or callee to caller ? |
| 23:54.00 | [TK]D-Fender | You can only add to an INVITE you are placing |
| 23:54.01 | iulhk | my client already sending this header, i just want it should pass through asterisk ? |
| 23:54.11 | [TK]D-Fender | "core show applications like sip" |
| 23:54.12 | [TK]D-Fender | ^ |
| 23:55.11 | iulhk | so during the call, is there any posibility ? |
| 23:55.55 | [TK]D-Fender | no |
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| 23:59.04 | iulhk | ok, one more, caller sending invite audio call, at receiver-end after answer the call, video screen opened automatically, how to stop this? |