| 00:00.11 | mjordan | putting more business logic into Asterisk rarely makes anyone happy. |
| 00:00.13 | jmordica | Proprietary sip extensions? |
| 00:00.32 | mjordan | There's a *very* closed issue in the issue tracker for them. |
| 00:02.05 | mjordan | regardless, it's far easier to build these kinds of features treating Asterisk as a media application server, than it is to write all of this in C. |
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| 00:05.53 | [TK]D-Fender | :( |
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| 00:30.41 | Valduare | hi guys |
| 00:30.57 | Valduare | how tough would it be to setup asterisk to handle a phone verification system |
| 00:31.18 | Valduare | ie new user registration - asterisk calls them by number provided and asks to enter a verification code |
| 00:32.12 | ChannelZ | I guess the main question is "triggered by what", but in general. probably not very |
| 00:33.39 | Valduare | site is php based |
| 00:35.32 | ChannelZ | Well ARI would be your friend if you're on a new enough Asterisk. But there's many ways to skin this cat.. AMI, a call file that does some dialplan/AGI perhaps |
| 00:37.02 | Valduare | i dont have asterisk yet very new to this idea |
| 00:37.05 | Valduare | what are these abreviations |
| 00:41.30 | ChannelZ | ARI is the Asterisk REST Interface where you can tell it to do just about anything with HTTP GET and PUTs etc. |
| 00:42.07 | ChannelZ | AGI is a means to launch a script from the dialplan and let it control things from there |
| 00:42.18 | [TK]D-Fender | um..... Don't jump the shark here |
| 00:42.21 | ChannelZ | In either case you can write in just about any language you're comfortable with |
| 00:43.00 | [TK]D-Fender | Everything you probably need is basis func_odbc <- |
| 00:43.05 | [TK]D-Fender | basic* |
| 00:43.42 | [TK]D-Fender | Valduare, So yes, * can interact with DB's on a basic leve right in the dialplan. Then there are other outside scripting possibilities |
| 00:45.01 | Valduare | ok |
| 00:45.19 | Valduare | can I do this through a google voice number atm to learn on |
| 00:45.30 | [TK]D-Fender | Gorget GV. |
| 00:45.41 | [TK]D-Fender | They're about to finilize the teardown of XMPP support |
| 00:45.45 | [TK]D-Fender | Forget* |
| 00:45.56 | [TK]D-Fender | You don't need any kind of number to learn * |
| 00:46.16 | Valduare | thought they already stopped xmpp support long time ago but it was still working |
| 00:46.16 | [TK]D-Fender | Get a softphone and a machine to run * on and you can learn how to do it all from there |
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| 00:57.59 | Valduare | it looks like asterisk runns on pretty small hardware requirements heh |
| 00:58.52 | [TK]D-Fender | Everything depends on what you're doing. |
| 01:00.03 | Valduare | aye |
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| 01:52.54 | phix | Valduare: yup |
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| 07:16.36 | juanmapalad | !ping |
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| 08:23.47 | ChannelZ | pong |
| 08:24.13 | phix | ping! |
| 08:25.13 | phix | hai ChannelZ! I am still having issues with incoming calls |
| 08:27.06 | ChannelZ | Per yesterday- did you get a message "Peer 'xxxx' is now UNREACHABLE!" on the console? (you are running with some verbose yes?) |
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| 08:28.21 | phix | ChannelZ: nope, I didn't get anything logged |
| 08:28.48 | phix | Asterisk thinks everything is fine |
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| 08:29.09 | ChannelZ | Says what? |
| 08:29.28 | phix | It doesn't error out and sip show registry shows it is valid |
| 08:29.31 | ChannelZ | What does 'sip show peers' show? |
| 08:29.42 | phix | I will look at that |
| 08:32.01 | juanmapalad | hi |
| 08:32.13 | phix | oh hai their juanmapalad! |
| 08:32.37 | juanmapalad | what resources/material can you suggest for a beginner like me to learn about asterisk? |
| 08:33.00 | ChannelZ | dejavu |
| 08:33.03 | ChannelZ | ~book |
| 08:33.03 | 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 |
| 08:33.09 | ChannelZ | ~primer |
| 08:33.09 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
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| 08:53.00 | ChannelZ | phix: ...so? |
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| 09:06.41 | ChannelZ | shrugs and goes to bed |
| 09:08.09 | phix | ChannelZ: OK (28 ms) |
| 09:08.28 | phix | and I cannot dial the number, there is no ring tone, after about 10 secs I get a line busy |
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| 13:41.41 | rexwin_ | hi where is issues with vonage supported? |
| 13:43.22 | [TK]D-Fender | vonage.com |
| 13:46.42 | rexwin_ | well, is there an irc channel which can support broadly |
| 13:47.06 | [TK]D-Fender | No. |
| 13:47.12 | [TK]D-Fender | Vonage a consumer product |
| 13:47.16 | [TK]D-Fender | They do sit in chat rooms |
| 13:47.28 | [TK]D-Fender | don't* |
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| 14:56.54 | Onyx47 | hello, does anyone know if there is a way to force Asterisk into T.30 mode while sending / receiving fax? I keep getting T.38 negotiation timing out (expected, service provider does not support T.38). It was fine when I had an ATA connected to a fax machine, but I'm trying to push it through IAX modem now. |
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| 15:30.57 | imihaylov | Hi! Can someone point me how to add calling *30 from extension gives the opportunity to add callerid to the blacklist? I have Asterisk 1.8. |
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| 15:34.54 | ChannelZ | That's something you have to build |
| 15:40.27 | ChannelZ | Do you want them to have to enter the number to add? Or do they have to do *30 while on the call they received they want to block? |
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| 16:32.00 | imihaylov | They will enter the number after dialing *30 |
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| 17:42.11 | AbuAbu | should I see h264 in core show translation? |
| 17:44.40 | mjordan | no. Asterisk doesn't transcode video. |
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| 17:58.35 | Katty | OR DOES IT. |
| 17:58.54 | WIMPy | What is OR? |
| 17:59.05 | Katty | a conjunction |
| 18:01.08 | Katty | WIMPy: https://www.youtube.com/watch?v=RPoBE-E8VOc |
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| 18:55.20 | jab416171 | I'm having a problem adding my google voice account to asterisk. I keep getting the error "Too many created bindings per day." |
| 18:57.50 | jab416171 | https://gist.githubusercontent.com/jab416171/66b521654ece93968ec7/raw/7626c884ad1b95ae40993d769299f05d6187be01/gistfile1.txt |
| 19:06.38 | mjordan | that's coming back from Google. |
| 19:07.50 | jab416171 | xmpp show connections just says 'Retrieving roster' |
| 19:08.18 | mjordan | Google rejected your request. |
| 19:08.52 | mjordan | Most likely, you've violated the limits set upon the account you are using. |
| 19:09.00 | jab416171 | can I disable xmpp? I'd rather not delete and readd the account if I don't have to |
| 19:09.29 | mjordan | XMPP is how Asterisk interoperates with Google Talk/Voice, so no. |
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| 19:09.41 | jab416171 | then can I turn off google talk? |
| 19:09.51 | mjordan | the fact that this works at all still is actually rather shocking, given that Google announced aeons ago that they were moving away from XMPP. |
| 19:09.54 | Penguin | Delete and re-add? Google does not allow that. |
| 19:09.56 | jab416171 | I guess I could just turn off asterisk |
| 19:10.02 | jab416171 | Penguin, delete and re-add from asterisk |
| 19:10.17 | Penguin | It doesn't work that way. |
| 19:10.30 | Penguin | Just unload the module temporarily. |
| 19:10.49 | jab416171 | is that modules.conf? |
| 19:11.02 | jab416171 | which module? |
| 19:11.25 | Penguin | module show like xmpp |
| 19:11.56 | Penguin | Looks to me like it's res_xmpp.so |
| 19:12.05 | Penguin | module unload res_xmpp.so |
| 19:12.35 | jab416171 | and then module load res_xmpp.so? |
| 19:12.45 | Penguin | When you're ready to use it again, yes. |
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| 19:12.59 | jab416171 | if I restart asterisk, it will reload the module, right? |
| 19:13.02 | Penguin | Or it'll also load up the next time asterisk starts. |
| 19:13.15 | gusto | hi Penguin |
| 19:13.21 | Penguin | If you don't want it to load ever, set it to noload in modules.conf |
| 19:13.24 | gusto | Penguin, how are your mirrors? |
| 19:13.50 | jab416171 | even if it says noload in modules.conf, can I still load it from the command line? |
| 19:13.57 | gusto | yes |
| 19:14.13 | gusto | modules.conf only defines what is being loaded automatically on bootup |
| 19:14.17 | Penguin | Yes. The noload setting is for autoloading of modules. |
| 19:14.19 | jab416171 | got it |
| 19:14.25 | gusto | yes |
| 19:14.29 | gusto | I do it other way round |
| 19:14.43 | gusto | I do disable autoloading of modules and then explicitly LOAD the ones I need |
| 19:14.47 | Penguin | autoload yes, noload specific modules |
| 19:16.55 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:707e:bbd9:4194:bf54) |
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| 19:49.35 | minghags | Hello! Im in need of little help if anyone can help :) Im getting this notice: NOTICE[3053]: chan_sip.c:22109 handle_response: Failed to authenticate on BYE to '<sip:0403XX5X3@bX.sXXXXXl.net;user=phone;CPC=Ordinary>;tag=SDdu8i501-snl_004195XX18_NSN_CLIENT' |
| 19:49.59 | minghags | Im using two trunks and its sends back the wrong username |
| 19:50.13 | minghags | is there anything I can do about it? |
| 19:50.34 | minghags | this is the whole "article" http://forums.digium.com/viewtopic.php?f=1&t=92502&sid=8ec271b2db8591f14a48948f0aa04126 |
| 19:50.40 | minghags | thanks in advance |
| 19:51.16 | [TK]D-Fender | minghags: pastebin your sip config for those masking only the secret |
| 19:51.47 | minghags | ok 1 sec |
| 19:54.46 | minghags | http://pastebin.com/EE4L84uh |
| 19:54.52 | minghags | this is trunk config |
| 19:55.32 | [TK]D-Fender | That is going to fail |
| 19:55.43 | [TK]D-Fender | type=peer will match By IP/HOST |
| 19:55.51 | [TK]D-Fender | And the first one will ALWAYS get hit for inbound |
| 19:56.11 | [TK]D-Fender | It does not take the username into account in order to determine which to use |
| 19:56.22 | [TK]D-Fender | It then picks the first every time and then runs into an auth failure |
| 19:56.30 | [TK]D-Fender | You need to be using type=friend |
| 19:56.39 | minghags | on both? |
| 19:56.57 | [TK]D-Fender | yes |
| 19:57.24 | minghags | ok will give it a shot |
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| 20:01.14 | minghags | its the same thing |
| 20:01.15 | minghags | http://pastebin.com/XMFiDUD1 |
| 20:03.53 | [TK]D-Fender | Get rid of the "insecure" as well then |
| 20:04.01 | [TK]D-Fender | And I'll need to see the changes |
| 20:06.44 | minghags | if i get rid of insecure tag the call doesnt get to pbx :/ |
| 20:07.10 | minghags | im connecting to trunk through ipsec if this is anything |
| 20:08.04 | minghags | http://pastebin.com/DvzHXLyc |
| 20:08.09 | minghags | this is config now |
| 20:10.13 | [TK]D-Fender | Chang username to "defaultuser" |
| 20:10.32 | [TK]D-Fender | canreinvite=yes <- that should be "directmedia=no" |
| 20:10.50 | [TK]D-Fender | canredirect=yes <- I don't recall this beign valid at all |
| 20:11.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:11.10 | [TK]D-Fender | nat=yes <- Also virtually no provider ever should be NAT'd |
| 20:14.00 | gusto | lol |
| 20:14.01 | gusto | what? |
| 20:15.10 | gusto | canredirect is an option when you have a PBX for example in between so that it can redirect to a phone for example, when it is a phone it SHOULD NOT redirect |
| 20:15.18 | *** join/#asterisk areski (~areski@80.174.128.2.dyn.user.ono.com) |
| 20:15.31 | gusto | or was it reinvite? |
| 20:15.35 | gusto | whatever |
| 20:16.28 | minghags | if i dont specify insecure call doesnt get through to PBX |
| 20:16.29 | minghags | http://pastebin.com/21tunYRz |
| 20:16.35 | *** join/#asterisk hecatae (~Philip@host-89-240-2-225.static.as13285.net) |
| 20:16.37 | gusto | looks to me like reinvite is when i forward a call and redirect is when I am already on a call and want to redirect the RTP stream to another device |
| 20:16.39 | minghags | tcpdump show something |
| 20:16.44 | [TK]D-Fender | put the insecure back then... |
| 20:16.56 | [TK]D-Fender | And use * SIP debug, never external |
| 20:17.15 | gusto | insecure is quite normal for sip providers, they will not authenticate to you |
| 20:18.23 | gusto | insecure=invite ;-) |
| 20:20.33 | minghags | this is the debug of call http://pastebin.com/R5gvnYUB and new conf http://pastebin.com/7XZnRSq2 |
| 20:20.47 | minghags | any more ideas? and thanks guys byw |
| 20:20.50 | minghags | btw* |
| 20:20.57 | gusto | I have insecure=invite everywhere, because to me not even phones are authentificating themselves, I am autentificating them only by IP address and port |
| 20:21.34 | gusto | this simobil is your sip provider? |
| 20:21.40 | minghags | yes |
| 20:21.56 | gusto | you are from slovenia |
| 20:21.58 | gusto | ha? |
| 20:22.01 | minghags | yop :) |
| 20:22.09 | gusto | I am from Slovakia |
| 20:22.10 | gusto | lol |
| 20:22.21 | gusto | well, let's start |
| 20:22.33 | gusto | simobil is a peer -- not your friend! |
| 20:22.40 | gusto | they do not have an account to you |
| 20:23.04 | gusto | insecure=invite -- but not =port!! because their port will NOT change! |
| 20:23.36 | gusto | you are in Slovenia = Europe - you do NOT use ulaw ... only alaw |
| 20:24.00 | minghags | oh we are working with slovakia company Inoteska :) |
| 20:24.11 | minghags | thanks for helping ;) |
| 20:24.11 | gusto | dtmfmode = info , who said that the peer supports it? I would go with the RFC default |
| 20:25.13 | gusto | why do you have 4 configurations for the same host? |
| 20:25.20 | gusto | one configuration is enough |
| 20:25.32 | gusto | and where do you register to your simobil? |
| 20:25.34 | gusto | ha? |
| 20:25.49 | gusto | how does simobil know that you are online? I am not seeing the register => part |
| 20:25.54 | *** join/#asterisk jamicque (~jamicque@89-71-40-164.dynamic.chello.pl) |
| 20:26.13 | minghags | im connecting through ipsec |
| 20:26.28 | gusto | ??? |
| 20:27.27 | gusto | aha, you have 4 different numbers? |
| 20:27.29 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 20:27.38 | minghags | two trunks |
| 20:27.48 | minghags | but i setup user and peer |
| 20:27.59 | gusto | however |
| 20:28.09 | minghags | for each trunk |
| 20:28.18 | gusto | what is "trunk"? |
| 20:28.38 | minghags | sip trunk |
| 20:28.57 | gusto | there is no such thing as sip trunk, that's only available on IAX2 |
| 20:29.27 | gusto | and why do you need the ipsec btw? because you are doing it over 3G or sth like that? |
| 20:29.27 | jamicque | Hi, lately I'm working with launching asterisk with SIPML5 via websocket. I have configured asterisk as in manual in wiki.asterisk.org. However, I see that I don't get any audio. Nor on Chrome or Firefox and Safari. I have verified if libuuid is compiled into rtp engine, ane it is... In rtp debug I see that there is only one way audio (via ICE). There is no audio from browser. Any ideas wat can be wrong? Does anyone has wo |
| 20:29.27 | jamicque | rking interface with sipml5 and asterisk? |
| 20:30.06 | minghags | ipsec because its there requirement :/ |
| 20:30.30 | gusto | maybe so that providers do not filter it out |
| 20:30.32 | gusto | however |
| 20:30.34 | *** join/#asterisk jamicque (~jamicque@89-71-40-164.dynamic.chello.pl) |
| 20:30.41 | gusto | i do not think that ipsec is enough |
| 20:30.48 | gusto | you have to still make register => to them |
| 20:30.55 | jamicque | I'm back - I've got disconnected |
| 20:31.20 | minghags | Simobil-peer2/3868330183 80.95.239.149 5060 OK (36 ms) |
| 20:31.24 | minghags | simobil-peer/3868330113 80.95.239.149 5060 OK (36 ms) |
| 20:31.24 | gusto | ipsec is on the level of operating system, there is no way asterisk can know what it was about |
| 20:31.30 | gusto | yes, there are your peers |
| 20:31.44 | gusto | but give me sip show registrations |
| 20:31.49 | gusto | or whatever it is calles |
| 20:31.50 | gusto | called |
| 20:32.06 | gusto | sip show registry |
| 20:32.08 | gusto | that one |
| 20:32.11 | minghags | lix*CLI> sip show registry |
| 20:32.11 | minghags | Host dnsmgr Username Refresh State Reg.Time |
| 20:32.14 | minghags | bt.simobil.net:5060 N 3868330183 45 Registered Mon, 16 Feb 2015 21:31:30 |
| 20:32.17 | minghags | bt.simobil.net:5060 N 3868330113 45 Registered Mon, 16 Feb 2015 21:31:30 |
| 20:32.23 | minghags | sorry for pasting here :/ |
| 20:32.24 | gusto | aha |
| 20:32.25 | minghags | mistake |
| 20:32.28 | gusto | now we have something |
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| 20:33.35 | gusto | no problem |
| 20:33.41 | Penguin | <gusto> insecure is quite normal for sip providers, they will not authenticate to you <----- Asterisk won't even expect them to authenticate to you unless you have secret defined (which you should not have defined). |
| 20:34.18 | Penguin | If you must authenticate to them but they will never auth to you, use remotesecret not secret. This is probably your mistake. |
| 20:34.26 | gusto | Penguin, I have that feeling that I have lust to go to McDonald's |
| 20:34.48 | Penguin | I have no idea what that means. |
| 20:35.13 | gusto | yes, I forgot about that ... of course REMOTESECRET is *always* good idea |
| 20:35.27 | gusto | Penguin, that means that I am hungry |
| 20:35.33 | gusto | Penguin, you are from the USA, right? |
| 20:35.47 | Penguin | I ate lunch an hour ago, so I'm not hungry. |
| 20:35.51 | Penguin | Yes I am in the USA. |
| 20:35.53 | gusto | Penguin, we have now a teacher from the USA |
| 20:36.02 | minghags | [2015-02-16 21:35:44] NOTICE[14439]: chan_sip.c:22109 handle_response: Failed to authenticate on BYE to '<sip:040344513@bt.simobil.net;user=phone;CPC=Ordinary>;tag=SD9m56601-snl_0041976156_NSN_CLIENT' |
| 20:36.07 | gusto | yes, but here it's 9 30 PM |
| 20:36.08 | minghags | same :/ |
| 20:36.38 | gusto | that is interesting |
| 20:37.28 | gusto | why is fromuser different from defaultuser? |
| 20:37.42 | gusto | try to set fromuser to the same as defaultuser on all peers |
| 20:37.48 | Penguin | Because they do totally different things! |
| 20:38.03 | gusto | I kow |
| 20:38.05 | gusto | I know |
| 20:38.29 | gusto | but he seems to have more numbers to one provider |
| 20:38.29 | minghags | they require it like this fromuser=phonenumbe |
| 20:38.45 | minghags | two block of numbers |
| 20:38.50 | minghags | blocks* |
| 20:39.42 | Penguin | How could you possibly define two blocks of numbers in a fromuser setting? |
| 20:40.15 | gusto | last time I seen these proxy auth parts, to telefonica voip |
| 20:40.23 | gusto | I solved it over auth= setting |
| 20:40.35 | gusto | maybe it is not a SIP server you are authenticating to |
| 20:41.04 | gusto | so, now I go get something to eat |
| 20:41.05 | gusto | bye |
| 20:42.15 | minghags | Penguin: 38683301130-9 and 38683301130-9 on two different accounts |
| 20:43.05 | minghags | sorry the second one was meant: 38683301830-9 |
| 20:45.20 | jamicque | Hi does anyone have got sipml5 working with asterisk? |
| 20:48.20 | [TK]D-Fender | defaultuser = outbound auth user name. fromuser = changes the "From:" header user |
| 20:49.34 | minghags | http://stackoverflow.com/questions/28544684/asterisk-freepbx-call-doesnt-disconnects-after-hangup |
| 20:49.56 | minghags | i got this answer if its true im in deep crap :P |
| 20:50.18 | [sID] | has anyone here A2Billing? based on Asterisk? |
| 20:53.44 | *** join/#asterisk linuxfool (~james@DHCP-149-228.resnet.ua.edu) |
| 21:05.04 | linuxfool | For those of you who have fielded an Asterisk system, what's the largest number of extensions you had to work with? |
| 21:06.19 | cyford | yes yes [sID] |
| 21:06.54 | cyford | linuxfool i worked with about 500 |
| 21:09.01 | rrittgarn1 | across all my systems i total around 2000 extensions... not all have physical handsets, but that wasnt exactly the question |
| 21:09.14 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 21:10.47 | [sID] | cyford: you ? |
| 21:11.18 | [sID] | While we talk? |
| 21:12.05 | cyford | yes, i managed a 6 schools and locations across us |
| 21:12.40 | Penguin | -= 734 extensions (2843 priorities) in 189 contexts. =- |
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| 21:51.40 | bbryant | hey, how do I setup an asterisk environment that doesn't have dahdi hardware to use meetme? |
| 21:51.50 | bbryant | I've read there's fallbacks written into the driver now |
| 21:52.15 | WIMPy | No driver. Asterisk. |
| 21:52.27 | WIMPy | And it's not new any more. |
| 21:52.36 | bbryant | I haven't used it in a few years |
| 21:56.47 | bbryant | nvm, I figured it out |
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