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| 04:31.41 | al_nz1 | anyone here able to try to login to my server please? I need to test fail2ban |
| 04:31.46 | al_nz1 | SIP login that is |
| 04:32.25 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:32.33 | Penguin | Send me your IP addr with /notice |
| 04:36.17 | ChannelZ | Send me $1 |
| 04:36.45 | Penguin | Do you accept bitcoin? |
| 04:38.56 | ChannelZ | no |
| 04:39.24 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 04:39.30 | ChannelZ | Grr. Linux traffic control is starting to work my last good nerve. |
| 04:39.55 | Penguin | What's the problem? |
| 04:40.17 | nextime | Hello all. As asterisk-gui is now deprecated and i don't like freepbx, is there any non-so-intrusive/fat gui out there suggested to use with asterisk 11 ? |
| 04:40.50 | ChannelZ | Is 'nano' considered a GUI? |
| 04:41.04 | Penguin | vim might be. |
| 04:41.08 | nextime | ChannelZ : no, it isn't. My personal gui is VIM |
| 04:41.25 | nextime | ChannelZ : but as i like and prefer to use config files, the same doesn't apply to my customers |
| 04:41.32 | Penguin | That is the preferred method of asterisk configuration. |
| 04:41.37 | ChannelZ | well we all make bad choices once in awhile. |
| 04:42.02 | nextime | Penguin : i agree, and *my* asterisk installs are all without gui. |
| 04:42.27 | ChannelZ | Doesn't digium still make something? |
| 04:42.37 | Penguin | If you're going to let people who have no business configuring a PBX configure a PBX, FreePBX is probably the right answer. |
| 04:42.41 | nextime | but if a customer ask me to have a gui and it doesn't like to edit files, i have to make him happy with a gui. |
| 04:43.13 | nextime | Penguin : no, not really, they just need to add/remove users, and freepbx is too fat for my religion |
| 04:43.32 | Penguin | The PBX doesn't need to know about the users. |
| 04:43.41 | Penguin | It only needs to know about the phones and the numbers. |
| 04:43.43 | ChannelZ | And my traffic control quandry is that I can't get anything workable. I've got a basic grasp of hfsc but when I start putting things into practice I get results I don't expect or understand. |
| 04:43.45 | nextime | i was using asterisk-gui some years ago for this very same issue, just some js and html, no mysql, no php, no apache/ngnix |
| 04:43.50 | nextime | but now it is deprecated |
| 04:44.41 | nextime | Penguin : ok, they need to add/remove phones/extensions, it's the same for them |
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| 04:52.40 | al_nz1 | you get the ip pengiun? |
| 04:52.51 | Penguin | Yes. |
| 04:53.01 | Penguin | One moment. |
| 04:53.07 | al_nz1 | k |
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| 04:59.47 | Penguin | Is that enough? |
| 05:00.09 | Penguin | You're still sending me 200 OK after all the bad stuff I did. |
| 05:00.42 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:01.05 | ChannelZ | It's the Obama administration of SIP! |
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| 05:07.27 | Penguin | al_nz1: You certainly are not very attentive. |
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| 05:14.04 | ChannelZ | Let's hack something! |
| 05:14.23 | Penguin | You might as well start with al_nz1's asterisk, since fail2ban doesn |
| 05:14.27 | Penguin | 't stop that sort of thing. |
| 05:16.59 | Penguin | Let me clarify... |
| 05:17.10 | Penguin | HIS fail2ban doesn't stop that sort of thing. |
| 05:17.22 | Penguin | fail2ban is capable when configured to do so. |
| 05:18.41 | ChannelZ | I guess it's failing 2 ban. HA! |
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| 06:23.56 | al_nz1 | sorry Penguin - had to change a shitty nappy |
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| 06:24.13 | al_nz1 | let me look at some logs |
| 06:27.53 | al_nz1 | still there Penguin |
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| 06:34.11 | ChannelZ | it must have been on the walls |
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| 07:49.37 | r00f | i've heard that asterisk guys will be present at GITEX 2014 |
| 07:49.45 | r00f | looking forward to meet |
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| 08:26.55 | Stefan27 | 'asterisk guys'? |
| 08:27.07 | eirirs | <*> Hi. |
| 08:27.09 | eirirs | see, he said hi |
| 08:27.24 | Stefan27 | hi |
| 08:27.32 | al_nz1 | are there girls here too? |
| 08:27.42 | eirirs | I thought * was unisex |
| 08:28.29 | Stefan27 | girls dont like telephony systems as much as guys i think |
| 08:28.38 | Stefan27 | in general |
| 08:29.03 | Stefan27 | im sure there's some |
| 08:32.17 | eirirs | but why are you asking? |
| 08:32.36 | eirirs | I don't see how it's related to the asterisk business |
| 08:35.15 | al_nz1 | they just like to use phones! |
| 08:35.34 | al_nz1 | eirirs: becausew he said hi asterisk guys |
| 08:35.39 | al_nz1 | what about the girls |
| 08:35.43 | eirirs | true |
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| 08:38.30 | Stefan27 | it's probably okay to pollute this irc chat with social stuff at this hour anyways, most developers are asleep |
| 08:38.52 | Stefan27 | what is GITEX 2014 anyways |
| 08:39.35 | al_nz1 | Stefan27: you a developer? |
| 08:39.47 | Stefan27 | not for asterisk |
| 08:40.21 | Stefan27 | but digium is american right |
| 08:43.33 | al_nz1 | ur in the US? |
| 08:43.41 | Stefan27 | I'm in Sweden |
| 08:43.56 | Stefan27 | you? |
| 08:43.58 | al_nz1 | NZ |
| 08:44.13 | Stefan27 | cool |
| 08:44.15 | Stefan27 | 12 hours off :) |
| 08:44.39 | Stefan27 | earth-polar-opposites |
| 08:44.42 | al_nz1 | defo |
| 08:44.53 | Stefan27 | If a meteor hits the earth, at least one of us will survive the initial impact |
| 08:46.50 | snadge | such a comforting thought |
| 08:56.51 | al_nz1 | anyone able to try to hack my IP? |
| 08:56.55 | al_nz1 | if I give it to you |
| 08:56.59 | al_nz1 | particularly on SIP |
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| 10:48.41 | MadHatter42 | has anyone implemented trunk rotation |
| 10:48.56 | MadHatter42 | lets say you have 5 providers and want to distribute calls evenly |
| 10:49.02 | MadHatter42 | for outbound |
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| 11:30.07 | wdoekes | ~poll |
| 11:30.07 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 11:30.11 | wdoekes | ~polls |
| 11:30.12 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 11:30.34 | wdoekes | oh, he left |
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| 12:05.21 | *** join/#asterisk gain (~gain@2001:1418:100:858b::ff21) |
| 12:05.25 | gain | hi all |
| 12:06.04 | gain | I'm using asterisk 1.8.10 and I want to know if I can start a call from Asterisk CLI for testing purposes... |
| 12:06.13 | gain | just tried dial command, but not works |
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| 12:12.01 | sekil | gain: originate |
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| 12:48.02 | pzn | I need to test a sip equipment. however I don't want to bother a real person at the other side of the phone call. is there any "easy" way to make a "fake call"? (simulate a call-answer)? a simple Playback(file) will simulate the same sip commands that answer a regular phone call? |
| 12:48.12 | gain | sekil: thanks |
| 12:53.52 | [TK]D-Fender | pzn: AMI/CLI Originate, Call-files. |
| 12:58.35 | pzn | in extension.conf ${EXTEN} has the destination extension. which variable can I get the origin extension, or the origin sip name/id (or anything related to origin)? |
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| 12:59.06 | pzn | [TK]D-Fender, great! AMI originate solves it. tks |
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| 13:02.17 | mjt | hello. I'm trying out asterisk, started with a zoiper sip client on my mobile phone, using a wifi network (the acces point is the same machine as asterisk server). Which codecs are recommended? I've a "hardware" voip client (a soho voip router installed by my ISP) which demonstrates quite good sound quality, but this my mobile phone client sound is rather bad. |
| 13:02.22 | mjt | or is it something else? |
| 13:03.51 | pzn | mjt: my samsung note 2, GT-N7100, with its native voip client, via wifi; has a good sound quality (not perfect, but good for conversation), however the voice "delay" is bad, about 0.5-1second |
| 13:04.57 | *** join/#asterisk theron_ (~theron@199.201.64.132) |
| 13:05.08 | mjt | pzn: my phone is also note2 :) |
| 13:07.59 | mjt | well yes, voice delay is also quite large. But I also have "creaky" sound ("gurgling") |
| 13:18.23 | pzn | mjt, my sound is good, no noises |
| 13:18.52 | pzn | mjt, but not as good ad my "wired phone" an old linksys pap2 |
| 13:21.06 | *** join/#asterisk babak_ (413144be@gateway/web/cgi-irc/kiwiirc.com/ip.65.49.68.190) |
| 13:28.24 | mjt | pzn: that's something similar to what I have from my POTS company (they use ONT (optical) terminal but the idea is the same, they run a voip gateway inside) |
| 13:34.57 | mjt | so really, what's the recommended codec to use? Is it not the default? |
| 13:35.46 | pzn | mjt, I've been using ulaw or alaw |
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| 13:40.19 | [TK]D-Fender | mjt: There is no such thing as "default" codec for SIP |
| 13:40.33 | [TK]D-Fender | mjt: It is your job to define everything in your system |
| 13:43.03 | mjt | well yes, I start from a default configuration |
| 13:43.15 | mjt | which comes with debian, at least ;) |
| 13:43.39 | mjt | but anyay, that's a recommended codec(s)? |
| 13:44.16 | *** join/#asterisk zerous (~naveens@88.201.43.78) |
| 13:46.17 | pzn | mjt, you need to know what is needed. low bandwidth or high bandwidth? that should guide you to choose a codec. in local network, I always stay with ulaw or alaw (64bkps bandwidth) |
| 13:46.51 | mjt | I wonder what codecs these "hardware voip gateways" use... |
| 13:47.01 | mjt | 64kbps is perfectly fine |
| 13:48.09 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 13:49.46 | *** join/#asterisk bklang (~bklang@coquette.alkaloid.net) |
| 13:50.39 | bklang | Hello all. I see that Asterisk 13 has VP8 passthrough support. Does that mean it can be used in ConfBridge with active speaker? Can Asterisk simply forward the VP8 frames, or does it need "real" support for that? |
| 13:57.03 | pzn | I have to discuss about using videoconference at my work, however I never needed that. In fact I've been using skype and others at my phone as app. is there any "protocol" supported by asterisk for video conferences? |
| 13:59.39 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bejcjmvkeypxwkuu) |
| 13:59.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:00.23 | bklang | pzn: SIP supports video conferences, but maybe I misunderstand your question |
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| 14:02.02 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:02.49 | [TK]D-Fender | SIP supports video CALLS. "Conferencing is a media processing concept. |
| 14:03.05 | gain | hi guys... I have connected ekiga with my asterisk and if I try to call with it I have no response, the problem is that I have nothing in logs at all... never seen somthing like this... sip show peers shows my ekiga is OK... what else can I check? O.O |
| 14:03.43 | [TK]D-Fender | [09:50]bklangHello all. I see that Asterisk 13 has VP8 passthrough support. Does that mean it can be used in ConfBridge with active speaker? Can Asterisk simply forward the VP8 frames, or does it need "real" support for that? <- clearly won't work. |
| 14:03.53 | [TK]D-Fender | bklang: Passthough != transcode |
| 14:04.57 | mjordan | well, we don't use the video stream in any format to determine speaker |
| 14:05.22 | mjordan | VP8 is just the video; it isn't the audio stream. Since the video media and the audio media are tightly coupled to a channel, we just do speaker detection by looking at the audio stream |
| 14:05.39 | [TK]D-Fender | mjordan: I think that Italians might take offense at that ;) |
| 14:05.51 | mjordan | possibly! |
| 14:06.23 | [TK]D-Fender | mjordan: So ... would an Italian quadriplegic be considered "mute |
| 14:06.25 | [TK]D-Fender | "? |
| 14:06.42 | mjordan | is unsure |
| 14:07.27 | bklang | mjordan: so does that mean VP8 can be used with ConfBridge and active speaker? |
| 14:07.44 | mjordan | It should work |
| 14:07.50 | mjordan | I haven't tested it :-) |
| 14:07.53 | bklang | ok :) |
| 14:08.13 | mjordan | But active speaker only cares about the audio. If we can transcode the audio stream into SLIN (which is usually a "yes"), then we can do active speaker |
| 14:08.30 | bklang | right, that's exactly what I want |
| 14:08.34 | [TK]D-Fender | Actually... if * littlay just frame swaps for follow the speaker, I suppose the video code should just pass through. |
| 14:08.47 | mjordan | we should, when we switch speakers, queue up a frame to the new speaker that asks for a picture update from the source. With VP8, that should hit the RTCP layer and ask for an update |
| 14:08.49 | bklang | actually, I don't even care about active speaker in this case, I'd be happy with a ConfBridge and a named channel as the video source |
| 14:08.57 | bklang | mostly I wanted to make sure that COnfBridge would forward VP8 |
| 14:09.03 | [TK]D-Fender | literally* |
| 14:09.05 | mjordan | bklang: which is also supported, as you can lock the speaker to a channel |
| 14:09.09 | mjordan | bklang: yup, should "just work" |
| 14:09.30 | mjordan | but I personally haven't tested VP8. I know Lorenzo, who wrote that patch, did, and things haven't really changed with it |
| 14:10.24 | mjordan | bklang: if you are testing video out in 13, then I would (for right now) limit yourself to a single video codec. There's a few bugs that we're aware of that we're working out right now where things get a bit confused if you list every video codec under the sun |
| 14:10.41 | bklang | ok, thanks for advice, I'll try that now |
| 14:11.50 | bklang | Are these warnings indications that the media is being entirely rejected? SRTP unprotect failed with: authentication failure 110 |
| 14:12.10 | bklang | I'm getting audio, but not video, and those warnings only appear occasionally |
| 14:16.06 | jeev | has anyone interconnected with ipsec to verizon? |
| 14:18.16 | mjordan | bklang: hm. If it is occassional, I'd suspect that the RTCP packets aren't getting unprotected. But that shouldn't affect your RTP stream. |
| 14:18.24 | mjordan | which phone are you using? |
| 14:18.27 | mjordan | chrome? |
| 14:18.38 | mjordan | (phone, ha) |
| 14:19.40 | bklang | mjordan: ha :) Chrome on one end and Bria 4 on the other |
| 14:19.50 | bklang | also using FF in place of Chrome |
| 14:19.56 | bklang | let me try a Chrome to FF call... |
| 14:20.30 | mjordan | are you using DTLS for the media? |
| 14:21.45 | bklang | as far as I know, yes. encrpytion and avpf are enabled |
| 14:21.51 | mjordan | heh |
| 14:22.01 | mjordan | using chan_sip? |
| 14:22.05 | bklang | also, I'm using chan_sip, because ...yeah. |
| 14:22.24 | mjordan | encryption refers to SDES-SRTP |
| 14:22.31 | bklang | ah |
| 14:22.48 | mjordan | if you take look at the sip.conf.sample, there's a slew of DTLS options you have to set up |
| 14:22.55 | bklang | ok |
| 14:23.06 | bklang | well I guess that answers my other question, chan_pjsip isn't required |
| 14:23.10 | bklang | even if strongly encouraged |
| 14:23.11 | mjordan | nope |
| 14:23.39 | mjordan | you can run 13 without any of the chan_pjsip work. You will need pjproject now, however, for WebRTC, as we use that for ICE/STUN/TURN |
| 14:24.07 | mjordan | pjproject was completely removed after 11, and is no longer embedded - so if you need ICE/STUN/TURN, you have to install the shared object libraries for that |
| 14:24.18 | bklang | ok I *do* have dlts enabled |
| 14:24.25 | bklang | do I need to remove encryption=yes? |
| 14:24.43 | mjordan | I would on your WebRTC peers |
| 14:24.46 | bklang | I have pjproject compiled (version 2.3) and I have Asterisk 13 built against it |
| 14:24.49 | bklang | pl |
| 14:24.50 | bklang | err ok |
| 14:25.19 | mjordan | er |
| 14:25.30 | mjordan | I should recant |
| 14:25.39 | mjordan | make it look like this: |
| 14:25.39 | mjordan | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 14:25.39 | mjordan | :-D |
| 14:26.27 | voip | i found on my setup when isntalling bigbluebutton and using bria for video conferencing along with portgopro on a PC, it was all about the video codecs |
| 14:26.48 | bklang | mjordan: thanks :) |
| 14:27.14 | voip | once the video codec was the same across the board, and i had disabled the rest of them, html5 worked seamlessly and video was streaming |
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| 14:28.55 | bklang | interesting, that example has encryption=yes |
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| 14:58.50 | ovoshlook | Hello. I use AMI originate and get issue with name of CallerID field. Name of my device something like company.device-deviceid. When call originates asterisk delete "-" and ".". After this all I have callerId as "companydevicedeviceid". How I need to escape this elements at code? |
| 15:01.36 | WIMPy | Are you talking baout numbers or names? And what does AMI have to do with it? |
| 15:05.21 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
| 15:06.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:08.50 | gain | tried channel originate SIP/mysip/<phone_number> extension <my_context_name> but not works... and can't find valid examples on how to use it... can anyone help me? I need it for testing purposes |
| 15:12.13 | [TK]D-Fender | gain: Show us exactly what you have done, the dialplan you're supposedly sending it to, and the CLI output. |
| 15:12.16 | [TK]D-Fender | ~pb |
| 15:12.16 | 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. |
| 15:12.18 | [TK]D-Fender | ^^^ |
| 15:16.32 | newtonr | gain, if you search the wiki, you'll find an Example I wrote: https://wiki.asterisk.org/wiki/display/AST/Creating+and+Manipulating+Channels+from+the+CLI |
| 15:17.14 | newtonr | It isn't super detailed, but it is a simple command. |
| 15:17.15 | ovoshlook | WIMPy: I about numbers and names. AMI originate a call (or I not anderstand question) |
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| 15:18.46 | ovoshlook | gain: Try ... extension my_ext@my_context |
| 15:19.31 | gain | newtonr: here is my test http://pastebin.com/3eQcSsMK |
| 15:21.07 | gain | ops |
| 15:21.54 | gain | extensions.conf http://pastebin.com/j7bCsq6v |
| 15:23.02 | WIMPy | ovoshlook: So we share a problem: I don't understand your question, either. What's the link between caller IDs and AMI. And I'm still unsure what part of the caller ID it's about. |
| 15:29.43 | ovoshlook | I have AMI originate action generator: It generates AMI action where one of fields os CallerId, that look likes CallerId: company.device-deviceid <company.device-deviceid>. So AMI deletes all "-" and "." when action going to asterisk (for exammple at CALLERID(num) i see companydevicedeviceid) |
| 15:37.00 | *** join/#asterisk zerous (~naveens@88.201.43.78) |
| 15:37.26 | zerous | could any one tell me the command for checking whether a sip client is connected to the asterisk server.? |
| 15:37.35 | zerous | i tried sip show peers |
| 15:37.49 | zerous | but it says no such command. please help ? |
| 15:38.01 | mjordan | if you don't have that command, chan_sip isn't loaded. |
| 15:38.11 | zerous | oh.. |
| 15:38.23 | zerous | i have already configured it. |
| 15:38.23 | mjordan | what does 'module show like chan_sip' display? |
| 15:38.37 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 15:38.37 | zerous | it says modules not configured when asterisk starts. |
| 15:38.52 | mjordan | Do you have a modules.conf? |
| 15:38.53 | tenspeed705{AFK} | how reliable is DNS SRV records for failover/load balance? |
| 15:38.59 | zerous | mjordan: no. |
| 15:39.11 | zerous | so i guess i should have it.right ? |
| 15:39.14 | mjordan | zerous: you may want to read the basics of setting up Asterisk |
| 15:39.16 | mjordan | ~thebook |
| 15:39.16 | 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 |
| 15:39.39 | mjordan | if you're missing that, you're probably missing some other things too :-) |
| 15:39.50 | zerous | mjordan: could you give me a link to it. I guess i am in between and skipping certain things |
| 15:40.11 | zerous | infobot: thank you.. i will look into it. |
| 15:40.11 | infobot | bitte, zerous |
| 15:42.45 | zerous | mjordan: i can see that i have modules.conf in /etc/asterisk/ |
| 15:42.52 | newtonr | gain, try looking at 'logger show channels' to see what logger channels you have going to the console |
| 15:43.18 | [TK]D-Fender | gain: faxsrv2*CLI> channel originate SIP/centralino extension 3200787746@fax <- this is wrong |
| 15:43.18 | newtonr | gain, then look at logger.conf to configure that, and be sure to 'logger reload' or restart Asterisk to make changes take effect |
| 15:43.33 | [TK]D-Fender | gain: You are starting by dialing your peer.. but NOT# passing the a NUMEBR to dial. |
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| 15:45.47 | zerous | mjordan: i can see modules.conf in /etc/asterisk/ but there is no modules folder |
| 15:48.24 | mjordan | how did you install Asterisk? |
| 15:48.37 | mjordan | and where are you looking for your modules? |
| 15:48.45 | zerous | mjordan: by downloading the tarball and installing it. |
| 15:48.52 | zerous | mjordan: in /etc/asterisk/ |
| 15:48.58 | mjordan | that's not where they would be |
| 15:49.06 | zerous | oh.. |
| 15:49.19 | mjordan | default location for Asterisk's modules is /usr/lib/asterisk/modules |
| 15:49.28 | newtonr | zerous, https://wiki.asterisk.org/wiki/display/AST/Directory+and+File+Structure |
| 15:50.46 | [TK]D-Fender | zerous: To config what your installation is doing "core show settings" from * CLI |
| 15:50.53 | [TK]D-Fender | confirm* |
| 15:51.40 | zerous | [TK]D-Fender: yes i see that. |
| 15:52.06 | zerous | is there a way i can understand whether my sip client is registered with the asterisk server. |
| 15:53.45 | zerous | mjordan: i can see chan_sip.so in the modules folder. |
| 15:54.59 | newtonr | zerous, did you every verify that chan_sip is even loaded? |
| 15:55.04 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:55.31 | newtonr | zerohalo, run "module show like chan_sip.so" |
| 15:55.59 | newtonr | err, zerous |
| 15:56.01 | zerohalo | :D |
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| 15:56.32 | zerous | newtonr: when i tried that, it says 0 modules loaded. |
| 15:58.38 | zerous | and i read autoload=yes in modules.conf |
| 15:59.14 | newtonr | zerous, okay, so you likely have a typo in sip.conf preventing it from loading, but it could something else. You can look through debug and verbose messages on Asterisk load to figure it out. Or a simple way to troubleshoot could be to try using a blank sip.conf and see if chan_sip loads with that. |
| 16:00.58 | newtonr | if it loads with a blank sip.conf, but not with your configuration, then you have a typo in your config preventing it from loading correctly. |
| 16:08.55 | zerous | newtonr: its working now. :) thanks a lot. |
| 16:09.24 | zerous | i mean when i make the call from the sip client, it is answered by the machine. but the asterisk cli doesn't show the call coming. |
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| 16:18.44 | newtonr | zerous, the missing console messages probably stem from your logger.conf configuration. I mentioned to you how to check that out.. |
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| 16:38.29 | file | falls over |
| 16:38.59 | eirirs | ...into a file drawer. |
| 16:47.17 | zerous | newtonr: i checked logger.conf.. is there something else that i need to check. i am not getting the console messages ? |
| 16:55.28 | ovoshlook | I have AMI originate action generator: It generates AMI action where one of fields os CallerId, that look likes CallerId: company.device-deviceid <company.device-deviceid>. So AMI deletes all "-" and "." when action comes to asterisk (for exammple at CALLERID(num) i see companydevicedeviceid). How to escape this characters? |
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| 17:33.49 | newtonr | zerous, so you have verbose,notice,error,warning all going to the console? you restarted asterisk and now "logger show channels" shows all of those on the console? If so, then make sure verbose is at 3 or so, and you should definitely see output when the dialplan is being executed. |
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| 20:24.22 | newtonr | [TK]D-Fender, added an article that can help for people with the typical missing commands issue. https://wiki.asterisk.org/wiki/display/AST/Troubleshooting+Asterisk+Module+Loading |
| 20:24.59 | newtonr | It isn't crazy detailed. |
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| 21:09.29 | bibz2 | hey theer |
| 21:09.30 | bibz2 | there* |
| 21:09.46 | bibz2 | is it possible to unmask a "anonymous" or unknown callerid? |
| 21:09.59 | bibz2 | seems like 2008 it was, but does this still work? |
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| 21:16.30 | WIMPy | Unmask? |
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| 21:28.49 | wdoekes | https://i.imgur.com/eK2BbIC.jpg |
| 21:29.16 | bibz2 | seeing the number of a anonymous call |
| 21:29.36 | bibz2 | or displaying/forwarding it.. so it doesn't display "unknown/anonymous" on my phone.. |
| 21:30.04 | bibz2 | the article from 2008 states something about a privacy setting in the header, but that isn't existent in my pbx |
| 21:30.36 | wdoekes | did you mention which telephony protocol you were using? |
| 21:41.36 | bibz2 | oh, i'm sorry. I'm using SIP |
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| 21:51.49 | ChannelZ | You can't know what you're not told |
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| 22:03.55 | greenwolf | hi |
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| 22:21.40 | ChannelZ | oh hai |
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| 23:42.30 | MasterChen | boxybrown: hey dude |
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