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| 00:30.29 | moe` | hey kids |
| 00:31.36 | moe` | <PROTECTED> |
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| 00:35.11 | kline | i was at the local hackerspace yesterday and mentioned i was starting with asterisk, and about 15 seconds later someone gave me Asterisk for Dummies and the OReilly Asterisk book, im a happy bunny there |
| 00:35.23 | kline | i was on the edge of just installing freeswitch |
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| 00:51.49 | lankanmon | hi guys |
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| 02:27.19 | snadge | noob question about call flow, and re-invite coming up |
| 02:28.08 | snadge | im starting to suspect a call flow whereby if we send a call to an IP address.. the call may being sent back and diverted elsewhere.. in other words.. im seeing calls going out a trunk thats supposed to only be inbound.. ie.. theres no dialplan which sends these particular calls out that way |
| 02:28.50 | snadge | so ive done a small amount of googling.. canreinvite=no is apparently an old setting, and probably doesn't do what i'm seeming to think it does |
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| 09:25.10 | *** join/#asterisk jamesc (~kvirc@mail2.inetplc.com) |
| 09:25.33 | jamesc | How can I see that I have a certain function? |
| 09:31.59 | Zogot | jamesc: core show functions like X |
| 09:32.49 | jamesc | Zogot: I can't see round and Im on 1.6 which apparently should have it.Do I need to add menu options at build time? |
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| 09:33.37 | Zogot | jamesc: module load func_roun [press tab] |
| 09:34.12 | jamesc | Zogot: its not there :( |
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| 09:35.56 | Zogot | is there a round function |
| 09:36.02 | Zogot | i cant find any docs about it jamesc |
| 09:36.21 | Zogot | oh i see it |
| 09:36.30 | jamesc | Zogot: Set(Units=${ROUND(${MATH(${SecProd}/60)}):1:1}) |
| 09:36.33 | Zogot | try module load func_math |
| 09:36.36 | Zogot | [tab] |
| 09:37.08 | jamesc | Zogot: says Module 'func_math.so' already exists |
| 09:38.25 | jamesc | I grep the source and this is what I see http://paste.fedoraproject.org/336091/75162761/ it looks there |
| 09:42.27 | Zogot | jamesc: and if you just try to use it? |
| 09:43.15 | Zogot | jamesc: exten=>100,1,NoOp(ROUND(1.6))? |
| 09:44.44 | Zogot | jamesc: ah, its only to be used in conditions |
| 09:44.47 | Zogot | i get this |
| 09:45.15 | Zogot | jamesc: exten => 878,1,NoOp($[ROUND(1.6)=2]) with the result of NoOp("SIP/clearvoxtest_cxXoJF7emr-00000040", "1") |
| 09:47.07 | jamesc | ERROR[28477]: pbx.c:2879 ast_func_read: Function ROUND not registered |
| 09:47.29 | Zogot | jamesc: you cant use these functions: https://wiki.asterisk.org/wiki/display/AST/Expr2+Built-in+Functions anywhere except inside conditionals |
| 09:47.35 | Zogot | if i understand it correctly |
| 09:51.02 | jamesc | It works inside $[] weird. Thanks for you help |
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| 12:32.49 | theitguy | hello, i just installed asteriksNOW latest stable distrubition but ARI module is not loaded. How can i make it loaded? |
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| 13:36.26 | pyro25 | morning everyone :P |
| 13:38.00 | pyro25 | I started working a few days ago to work on getting asterisk to do webchat between wss and sip |
| 13:38.10 | pyro25 | is that something people typically easily do? |
| 13:39.53 | *** join/#asterisk Martin` (martin@shell.ipv6.octocore.net) |
| 13:40.06 | Martin` | Hello world! |
| 13:54.07 | [TK]D-Fender | Depends on your definittion of "webchat" |
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| 14:05.17 | pyro25 | hah, right, I meant video calls |
| 14:05.32 | pyro25 | my main issue I really don't get is how to get any video to go through |
| 14:05.54 | pyro25 | I got videosupport enabled in sip, I made sure I got all the codecs loaded |
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| 14:06.21 | pyro25 | then, in my extension, I do a |
| 14:06.23 | pyro25 | same => n,Playback(/srv/media/H263_sample) |
| 14:06.54 | pyro25 | with an extension .h263 |
| 14:07.12 | pyro25 | and no matter what I do, I get Unable to open /srv/media/H263_sample (format (gsm|h263p|h264)): No such file or directory |
| 14:07.14 | [TK]D-Fender | let's see the call |
| 14:07.28 | pyro25 | do I need to transcode the file in some special way? |
| 14:07.39 | [TK]D-Fender | how did you record thatt? |
| 14:08.00 | pyro25 | I went straight on h263 website and picked an official sample =P |
| 14:08.05 | pyro25 | confirmed as well with avprobe |
| 14:08.29 | [TK]D-Fender | do you have an audio file to go with it? |
| 14:09.09 | theitguy | do i need to buy asteriskNOW licence for using asteriks ARI? |
| 14:09.59 | [TK]D-Fender | There is a FREEPBX commercial module. Not "AsteriskNOW" |
| 14:10.08 | [TK]D-Fender | AsteriskNOW just includes FreeePBX. |
| 14:10.15 | [TK]D-Fender | And this is not supported here. |
| 14:10.16 | [TK]D-Fender | ~freepbx |
| 14:10.16 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:10.18 | [TK]D-Fender | ^^^ |
| 14:10.36 | pyro25 | hah! |
| 14:10.38 | pyro25 | [lavf] stream 0: video (h263), -vid 0 [lavf] stream 1: audio (amrnb), -aid 0, -alang und |
| 14:10.57 | Martin` | how can I find out why a grandstream phone (gxp2020) does not accept is provisioning configfile? |
| 14:11.41 | Martin` | I'm sure he downloads the file, but it does not use it |
| 14:11.42 | theitguy | my problem is ARI module is not loaded. and cannot connect ARI. do i need to go #freepbx now? |
| 14:11.57 | pyro25 | am I killing asterisk with amrnb audio? let's see |
| 14:12.05 | [TK]D-Fender | theitguy: yes |
| 14:12.13 | theitguy | thanks. |
| 14:12.42 | mub | Martin`: Those phones have logs. You should set up a fluentD server and send all of your phone logs there! |
| 14:13.02 | mub | I know that doesn't help you, I just wanted to share my 2cents |
| 14:13.19 | Synthase_ | Or a quick rsyslog daemon, FluentD is overkill to see why one phone is being difficult |
| 14:13.32 | Martin` | fluentD is syslog server? |
| 14:13.38 | mub | We send everything there |
| 14:14.28 | Martin` | it supports a lot, lets try it :) |
| 14:14.58 | Synthase_ | An element of ELK stack, replace the L with F. I reiterate, far more than necessary to check one device, normal syslog has you covered. |
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| 14:36.20 | Martin` | mub: ERROR Write /asterisk/phoneprov/cfg000b821a4f97 File size exceeds maximum |
| 14:36.25 | Martin` | what is de max ? |
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| 14:49.20 | mub | Martin`: That doesn't make sense... It works on other gxp2020's, so the configuration works. |
| 14:49.28 | mub | Maybe factory reset the phone? |
| 14:49.33 | mub | I'm not sure that will help though |
| 14:49.34 | Martin` | already did |
| 14:49.52 | Martin` | config works on gxp2130, did not try an other gxp2020 yet |
| 14:50.08 | Martin` | but now I only have a config file with only 1 sip account in it, |
| 14:50.11 | Martin` | still to big |
| 14:50.48 | Martin` | (395 bytes) |
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| 15:02.31 | Martin` | can't find any solution, lets try an other phone |
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| 15:08.15 | Martin` | other phone has same problem |
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| 15:30.27 | Martin` | I guess grandstream broke the provisioning support, because it is end of life, users need to buy new phones if they want to use it |
| 15:32.01 | [TK]D-Fender | What provisioning support? |
| 15:32.23 | [TK]D-Fender | How were your provisioning files actually made? |
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| 16:59.07 | ruied | hello. I'm searching for an acs server to deploy phone configurations. Is there any acs server for linux? What I saw was discontinued projects like openacs... |
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| 17:22.18 | flujan | hello guys, I found this post: http://www.voip-info.org/wiki/view/Asterisk+sip+silencedetecthangup is this option still available on asterisk 11? |
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| 19:46.18 | Kevin` | is it possible to use webrtc without nat? |
| 19:47.32 | Kevin` | i've been fighting with buried error messages for days, current issue is "174843712[7f630924a460]: [main|PeerConnectionImpl] PeerConnectionImpl.cpp:1760: SetRemoteDescription: pc = bc8bee5db848a69d, error = Invalid description, no ice-ufrag attribute |
| 19:47.36 | Kevin` | " |
| 19:47.49 | Kevin` | other people have similar issues but I don't see ice-pwd mentioned as unavailable |
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| 20:05.49 | Kevin` | also ICE(PC:1457553887416961 (id=105 url=http://x-dev/jsSIP/example.html)): Packet received from IP4:10.0.1.35:11888/UDP which doesn't match any known peer |
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| 21:37.46 | jeffspeff | when using make menuselect.makeopts do i need to specify each individual option or only the options i want that aren't normally enabled by default when viewing in make menuselect ? |
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| 22:16.17 | lankanmon | Guys, what is the cheapes hardware that I can get to setup a system with two lines? |
| 22:16.33 | WIMPy | What "lines"? |
| 22:17.17 | lankanmon | I want to make an asteristk system with two physical phone connections |
| 22:17.24 | lankanmon | for personal use |
| 22:17.39 | lankanmon | I am a student and do not have much money |
| 22:17.45 | WIMPy | Now it's phones. You are a little unspecific. |
| 22:18.21 | lankanmon | I am looking to setup a small system at my house (with two phone lines). I want to be able to call the number, and for it to ask me for a pin number (to verify that it is me), then ask me for an outgoing number. Once I dial the outgoing number, it will connect me though to that number using the second line. It should also take all incoming calls to the outgoing number and automatically forward them to my phone. |
| 22:18.21 | lankanmon | This is just an idea, and I want to know what I would need to make this a reality. I am looking for hardware suggestions and software that can handle these specific requirements. -- I would prefer open source options if possible. |
| 22:18.29 | WIMPy | What exactely do you want to physically connect? |
| 22:19.10 | lankanmon | Initially one outbound phone line (to carrier) and the other to a phone |
| 22:19.14 | WIMPy | So you want to connect a POTS line and a POTS phone? |
| 22:19.25 | lankanmon | the system will also connect to a voip phone system |
| 22:19.42 | lankanmon | yes |
| 22:20.13 | lankanmon | the voip connection will hopefully occur online, so no need for a physical line and a phone modem |
| 22:20.21 | WIMPy | You could use one of thos ATAs that also support a line and absolutely ANY kind of computer. |
| 22:21.04 | lankanmon | do they allow for the customization that I am expecting? |
| 22:21.20 | lankanmon | (ask for password and direct calls through |
| 22:21.25 | lankanmon | ) |
| 22:21.45 | WIMPy | Don't know, but you can tell it to send all calls to your computer first. |
| 22:22.26 | lankanmon | I need it to send the incoming calls to the voip service |
| 22:22.49 | lankanmon | and if calls come from the voip service, then I need it to ask what number to dial and to dial it |
| 22:23.09 | lankanmon | (like a phone card service, but for one line) |
| 22:31.16 | lankanmon | Any Ideas? |
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