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| 01:23.57 | SeRi | dijib: you around? |
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| 01:43.33 | futuresoon | how do i allow anyone with a sip account from linphone.org to call my asterisk servedr |
| 01:44.41 | gusto | no idea |
| 01:45.00 | gusto | the allowing as such is not the problem |
| 01:45.13 | gusto | the question is if ppl can call you |
| 01:45.31 | gusto | like yourdevice@yourasteriskdomain.tld |
| 01:45.57 | futuresoon | they can't just call mydevice@some.ip.address? |
| 01:48.06 | futuresoon | do most sip services allow you to call other sip service addresses? |
| 01:48.33 | futuresoon | i figured there's just some kind whitelists and blacklists or something |
| 02:31.37 | ChannelZ | make a peer with the host=linphone.org and do whatever you want with it |
| 02:33.15 | gusto | that would assume that linphone.org is one IP and that these ppl who are calling come out from that IP |
| 02:33.29 | gusto | allowguest would be more reliable |
| 02:33.31 | ChannelZ | Yup. |
| 02:36.35 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 02:47.09 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:58.39 | SeRi | interesting. pfsense was future on vuc. |
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| 03:14.13 | SeRi | p3nguin: http://www.ubnt.com/edgemax#edge-router |
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| 03:57.23 | SeRi | sort of quite around here |
| 03:58.21 | WIMPy | Start to experimet with multicast RTP :-) |
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| 04:07.01 | SeRi | :) |
| 04:07.10 | SeRi | making my recordings now |
| 04:17.00 | *** join/#asterisk greenwolf (6c2211fd@gateway/web/freenode/ip.108.34.17.253) |
| 04:28.13 | slav3_kitten | SeRi, i really need to get a female british sound pack |
| 04:28.29 | slav3_kitten | and better moh music. god i'm lazy |
| 04:28.35 | SeRi | slav3_kitten: ? |
| 04:28.40 | SeRi | lol |
| 04:29.11 | SeRi | slav3_kitten: I am making my own. |
| 04:29.22 | SeRi | for my auto attendant |
| 04:29.38 | greenwolf | anyone have any great vps hosting services for asterisk systems? |
| 04:29.46 | slav3_kitten | SeRi, ii'm not a chick nor british :{ |
| 04:29.54 | SeRi | rofl |
| 04:30.12 | SeRi | greenwolf: there is a few. some here use linodes |
| 04:31.39 | greenwolf | they see pretty expensive any other ideas? |
| 04:33.03 | SeRi | greenwolf: not sure. I dont use them. |
| 04:33.10 | SeRi | ~vps |
| 04:33.10 | infobot | Virtual Private Server: a virtual machine which is functionally equivalent to a separate physical computer, is dedicated to the individual customer's needs, has the privacy of a separate physical computer, and can be configured to run as a server computer (i.e. to run server software). |
| 04:33.29 | greenwolf | thanks infobot but i need a low-cost service |
| 04:33.37 | SeRi | well looks like infobot only has the def and not a recommendation |
| 04:33.43 | SeRi | lol |
| 04:33.51 | greenwolf | if anyone here has any experience with other vps companies plz let me kno |
| 04:33.53 | greenwolf | lol |
| 04:33.54 | greenwolf | :) |
| 04:41.21 | SeRi | greenwolf: I thought amazon If not mistaken some people had got an asterisk instance running on the free amazon vps |
| 04:42.21 | WIMPy | Free? |
| 04:42.47 | SeRi | WIMPy: amazon has a free micro instance. |
| 04:42.57 | SeRi | I think is free for a year or something like that |
| 04:43.24 | SeRi | AWS Free Usage Tier |
| 04:44.28 | WIMPy | To get people hooked? |
| 04:44.46 | SeRi | probably... |
| 04:45.02 | SeRi | I had major issues with them in the past |
| 04:45.04 | SeRi | never again |
| 04:47.28 | WIMPy | I've got one at Hetzner, but I don;t run Asterisk on it (yet). |
| 04:49.40 | WIMPy | Maybe if I ever get around to continue on the conference system I started 3 weeks ago and didn't touch since then. :-( |
| 04:50.10 | SeRi | WIMPy: you should. |
| 04:50.23 | SeRi | I am giving my confbridge heavy usage with my new job |
| 04:50.35 | SeRi | I have customers dial in when we need a conf |
| 04:50.50 | WIMPy | I know. There's uncountable things I should finish :-( |
| 04:51.08 | SeRi | lol |
| 04:51.18 | SeRi | I just got floowme working |
| 04:51.33 | SeRi | s/floowme/followme/ |
| 04:51.35 | WIMPy | Unfortunately I've got an issue that eats lots of time. I somehow got in to politics. |
| 04:51.51 | SeRi | wtf. Yeap that would do it. |
| 04:51.52 | SeRi | lol |
| 04:52.06 | SeRi | gov politics? or business? |
| 04:53.02 | WIMPy | Party / city council |
| 04:53.26 | WIMPy | And some NGOs. |
| 04:54.16 | slav3_kitten | WARNING[20068][C-000000cb] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input:=1 < any idea what conf file to look in for this error? |
| 04:54.29 | greenwolf | can asterisk run on mac? |
| 04:55.21 | WIMPy | slav3_kitten: If you use an empty variable, the whole expression vanishes. Just like in bash. Use quotes or add garbage. |
| 04:55.33 | WIMPy | greenwolf: Sure |
| 04:56.17 | slav3_kitten | WIMPy, so i forgot to init a variable someplace? |
| 04:56.28 | dijib | it choppy tonight |
| 04:57.09 | WIMPy | slav3_kitten: I don't know what you expect there. I just say that empty variables need to be cared for. |
| 04:57.14 | greenwolf | so if i download the tar.gz asterisk i can untar and compile and actually run it on a mac computer? |
| 04:57.22 | greenwolf | are there any install instructions for using asterisk on mac |
| 04:57.43 | slav3_kitten | WIMPy, i don't know where to start looking config file wise to find the empty var |
| 04:58.19 | WIMPy | greenwolf: You will have unusuall installation paths, but that should be about it. |
| 04:58.23 | slav3_kitten | i am using asterisk db to do persistent caller id blocking |
| 04:59.07 | WIMPy | slav3_kitten: Where you got the error. If you can't recognize anything from the error, you will have to turn up verbose. |
| 04:59.28 | slav3_kitten | *nod* |
| 04:59.35 | greenwolf | umm...can i get a dualboot system and put linux on a mac machine |
| 04:59.41 | greenwolf | maybe that would be much easier |
| 04:59.54 | greenwolf | then just put the asterisk on the linux side of the mac system |
| 05:00.00 | greenwolf | if thats even possible |
| 05:00.20 | slav3_kitten | greenwolf, how many channels do you plan on using with your asterisk? |
| 05:00.40 | WIMPy | You probably won't find support for interface cards under MacOS. But otherweise I wouldn't expect much difference. |
| 05:02.24 | greenwolf | im going to need alot of current call channels |
| 05:02.33 | greenwolf | prbably around 300 to 500 |
| 05:02.46 | greenwolf | where going to be hosting a conference system using asterisk and opensips |
| 05:02.50 | slav3_kitten | i'd just build a dedicated asterisk box instead of dual booting an such |
| 05:04.14 | greenwolf | ok but can i build a dedicated machine on a mac machine |
| 05:04.25 | slav3_kitten | ppc or intel? |
| 05:04.27 | greenwolf | or would i just be better off using a PC with linux then |
| 05:05.05 | slav3_kitten | just throw a small hdd with linux installed in the mac box and put the mac hdd aside |
| 05:05.13 | slav3_kitten | dual boot seems silly imho |
| 05:06.22 | nightrid3r | a mac is expensive compared to a pc so unless you have a spare mac |
| 05:07.03 | greenwolf | ok |
| 05:11.09 | p3nguin | slav3_kitten: You can also pastebin the entire extension that is giving that warning. |
| 05:13.44 | slav3_kitten | p3nguin, that's all i got so far in my log. i set verbose up a bit and i'll keep an eye out for it |
| 05:14.27 | SeRi | p3nguin: whats going on... |
| 05:14.52 | p3nguin | I'm watching a movie before I go off to bed. |
| 05:15.07 | slav3_kitten | what movie? |
| 05:15.21 | p3nguin | Paranormal Activity 3 |
| 05:16.24 | slav3_kitten | removes 3 respect points from p3nguin |
| 05:16.44 | p3nguin | The wife wanted to watch it. |
| 05:16.59 | slav3_kitten | adds 4 respect points to p3nguin |
| 05:17.30 | SeRi | p3nguin: cool. quality time |
| 05:17.35 | WIMPy | So in total you get +1 respect point for obeying the wife? |
| 05:17.44 | slav3_kitten | yep |
| 05:18.02 | slav3_kitten | i like obeying women, i'm odd |
| 05:18.26 | p3nguin | I could have vetoed the movie, but I don't really care too much. |
| 05:18.43 | slav3_kitten | but had you initiated that movie on your own.. |
| 05:19.45 | ChannelZ | This is why I'm not married |
| 05:19.49 | SeRi | my wifes falls to sleep on movies |
| 05:20.05 | unicron | women don't like it when you obey all the time |
| 05:20.05 | SeRi | unless they are cartoons she will sleep for sure |
| 05:20.18 | slav3_kitten | that sucks... |
| 05:21.07 | SeRi | nope. It actuially works for me |
| 05:21.14 | SeRi | I dont have to go see pussy movies |
| 05:21.25 | SeRi | I get to choose action movies when me and her go out |
| 05:21.42 | SeRi | Is awesome |
| 05:21.59 | WIMPy | There's even a genre where you can get both at the same time :-) |
| 05:22.11 | p3nguin | If I would have picked the movie, I probably would have picked the one where I shut off the TV and go to bed. |
| 05:22.27 | SeRi | rofl!!! |
| 05:22.27 | SeRi | hahahahaha |
| 05:22.59 | slav3_kitten | yea but she will fall asleep during aliens and that's a tragedy |
| 05:23.35 | SeRi | tragedy? |
| 05:23.59 | slav3_kitten | yus |
| 05:24.04 | p3nguin | Tragedy is when you cut your finger... |
| 05:24.12 | p3nguin | Comedy is when you fall into an open sewer and die. |
| 05:24.26 | SeRi | lol |
| 05:25.03 | slav3_kitten | and falling asleep during aliens is a tragedy |
| 05:35.38 | SeRi | dfgas-cr48: You around? |
| 05:44.58 | *** join/#asterisk futuresoon (~futuresoo@cpe-74-66-25-137.nyc.res.rr.com) |
| 05:45.28 | futuresoon | is there a simple way i'm missing to let anyone join a conference call without having to make a special user for each of them? |
| 05:46.04 | WIMPy | futuresoon: I don't know of any other way. |
| 05:46.11 | WIMPy | What are you doing? |
| 05:46.33 | futuresoon | well my experience with conference calls that i've been on is that you get a number, and anyone can call in on that number |
| 05:46.56 | futuresoon | now since this is sip i figure there's no straight telephony number |
| 05:46.57 | WIMPy | For example, yes. |
| 05:47.22 | futuresoon | but if someone has an sip account with some random provider, they can't use that account to connect to my server? |
| 05:47.26 | WIMPy | SIP has an URI scheme (or "is"?), yes. |
| 05:47.39 | *** join/#asterisk evilman_home (kvirc@2.92.147.239) |
| 05:47.56 | futuresoon | now i don't have a domain i just have an ip |
| 05:48.04 | WIMPy | Probably not, but then they don't need a provider. SIP is P2P. |
| 05:48.30 | WIMPy | You can use IPs in URIs just as you can use host names. |
| 05:48.45 | WIMPy | And then there's dyndns, off course. |
| 05:49.01 | futuresoon | okay so if my conference is at 12345@123.45.678.910 |
| 05:49.12 | futuresoon | don't most sip clients require they have an account? |
| 05:49.16 | futuresoon | most or even all? |
| 05:49.30 | WIMPy | That depends on the client. |
| 05:49.50 | WIMPy | The issue is more likely they they will try to use an account if they have one. |
| 05:49.59 | WIMPy | ymmv |
| 05:50.23 | futuresoon | ahh okay |
| 05:50.29 | futuresoon | so i don't need an account on some clients |
| 05:50.57 | WIMPy | Generelly no. |
| 05:52.04 | futuresoon | awesome! okay i'm going to get THAT client then |
| 05:52.06 | futuresoon | thank you :-) |
| 05:53.35 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 07:05.46 | SeRi | it will be a looooong night |
| 07:12.04 | WIMPy | What did you break? |
| 07:12.53 | dfgas-cr48 | lol |
| 07:17.09 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:17.55 | dijib | SeRi: where u at? |
| 07:19.20 | *** join/#asterisk bmg505 (~leon@196-209-99-140.dynamic.isadsl.co.za) |
| 07:24.45 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 07:34.56 | SeRi | dfgas-cr48: you around? |
| 07:37.20 | dfgas-cr48 | sup |
| 07:44.32 | SeRi | dfgas-cr48: you still in dd-wrt |
| 07:45.01 | dfgas-cr48 | yah y? |
| 07:45.10 | SeRi | just curious |
| 07:45.15 | SeRi | how is it working? |
| 07:45.50 | dfgas-cr48 | this morning like garbage so i did a reset and it works great now |
| 07:46.00 | SeRi | awesome |
| 07:46.07 | SeRi | voip working ok? |
| 07:47.00 | dfgas-cr48 | havent really used much |
| 07:49.11 | SeRi | dfgas-cr48: you did anything special to get it to work beside port forwarding? |
| 07:50.44 | dfgas-cr48 | didnt even do that |
| 07:51.35 | SeRi | nice |
| 07:51.38 | SeRi | very nice |
| 07:52.30 | dfgas-cr48 | didnt even do that:) |
| 07:52.34 | dfgas-cr48 | opps |
| 07:52.36 | dfgas-cr48 | :) |
| 07:52.40 | SeRi | lol |
| 07:53.19 | dfgas-cr48 | tired |
| 07:57.39 | SeRi | wired |
| 07:57.47 | SeRi | coffee |
| 07:57.50 | SeRi | lots of it |
| 08:06.37 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 08:20.09 | SeRi | yawns... |
| 08:20.18 | SeRi | time for some more coffee |
| 08:29.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 08:41.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 09:25.03 | *** join/#asterisk fritz09 (~Adium@2002:542e:50e1:e472:655f:d34a:d039:8949) |
| 09:39.30 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 09:39.57 | gavimobile | is there a security issue with remote users? |
| 09:40.21 | gavimobile | I thought I read something like that once. |
| 09:40.32 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.174) |
| 09:40.50 | gavimobile | also what is encrypted calls for? |
| 09:48.15 | SeRi | gavimobile: not that i am aware of I am on 1.8.16.0. |
| 09:48.37 | gavimobile | SeRi: I am using 1.8 |
| 09:48.54 | SeRi | encrypted calsl do just that encrypt calls.... |
| 09:49.09 | gavimobile | why would someone want to encrypt thei calls?\ |
| 09:49.10 | SeRi | gavimobile: remote users as in manager.conf? |
| 09:49.19 | SeRi | gavimobile: security. |
| 09:49.32 | gavimobile | SeRi: no, remote users from outside the nat |
| 09:49.43 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:50.12 | gavimobile | what type of security is there to have encrypted calls? people cannot listen in to the conversation? |
| 09:50.31 | SeRi | ok. Than yes no issues here with remote user. |
| 09:50.56 | SeRi | Thats right to prevent people from eves droping |
| 09:51.36 | gavimobile | SeRi: I know this might sound silly but I thought I read that using port 5060 and not using tls (5061) for remote users shows passwords in plain text or something like that, and wifi users can pickup packets |
| 09:52.31 | SeRi | not true. the password is hashed when going accross. Is not in plain txt. |
| 09:52.36 | ChannelZ | It's still hashed |
| 09:52.38 | gavimobile | do you have any idea of what I might have read? is there any issue somewhat related to what I just said right now |
| 09:53.18 | SeRi | gavimobile: no. not sure where you got that from. |
| 09:53.35 | gavimobile | SeRi: ok cool! |
| 09:53.47 | gavimobile | back to the encrypted calls, how can someone tune in? |
| 09:54.01 | SeRi | a compromise hop in your route |
| 09:54.23 | SeRi | they reall and I mean really have to show interest on listening to your calls |
| 09:54.26 | SeRi | is not simple |
| 09:55.05 | SeRi | Usually when somebody is eves droping your sip they are really after you so worrie about something esle... lol |
| 09:55.22 | gavimobile | exactly |
| 09:55.36 | gavimobile | im not worried if people listen to my calls |
| 09:55.40 | gavimobile | they are not private |
| 09:55.55 | gavimobile | I thought tls had something to do with securing my remote users |
| 09:55.57 | SeRi | :) yeap (: |
| 09:56.04 | gavimobile | tls uses port 5061? |
| 09:56.42 | SeRi | gavimobile: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 09:57.38 | gavimobile | SeRi: im not trying to make secure calls |
| 09:57.44 | gavimobile | im trying to secure my server |
| 09:57.53 | SeRi | Than you dont need tls |
| 09:58.00 | SeRi | nor worrie about it |
| 09:58.09 | gavimobile | port 5060 is opened to the public.\ |
| 09:58.30 | gavimobile | im using very long passwords that are 15 characters long with numbers lowercase and uppercase letters |
| 09:58.37 | SeRi | iptables/fail2ban/good passwords/good device names/asterisk hardening |
| 09:58.50 | gavimobile | im using fail2ban to block users after 3 wrong attempts they get blocked for 3 days |
| 09:58.50 | SeRi | gavimobile: ^^ |
| 09:59.08 | SeRi | gavimobile: It looks like you are in the right path. |
| 09:59.28 | gavimobile | im looking to secure even more if possible |
| 09:59.35 | gavimobile | im a little worried the 5060 is opened though |
| 09:59.50 | gavimobile | I see a lot of people are attacking my server |
| 10:00.06 | SeRi | gavimobile: that will all ways happen |
| 10:00.14 | SeRi | no matter what you do. |
| 10:00.23 | SeRi | they will allways try |
| 10:00.44 | SeRi | To prevent that I use trusted source |
| 10:00.45 | gavimobile | SeRi: is there anything else in addition I can do? |
| 10:00.54 | gavimobile | or that you would recommend? |
| 10:00.56 | SeRi | trust only the source |
| 10:01.10 | SeRi | but if you have many peers that close to impossible |
| 10:01.13 | gavimobile | I didn't understand. which source? |
| 10:01.32 | SeRi | the client to your asterisk box |
| 10:01.39 | SeRi | or the itsp to your asterisk |
| 10:02.00 | gavimobile | I trust my itsp |
| 10:02.13 | gavimobile | I don't use thoese chitty ones |
| 10:02.19 | gavimobile | which don't have support numbers |
| 10:02.33 | SeRi | I dont mean that. I mean in iptables |
| 10:02.43 | gavimobile | the source address? |
| 10:02.50 | SeRi | only recive SIP/RTP from a trusted source in iptables |
| 10:02.58 | SeRi | yes |
| 10:03.09 | gavimobile | SeRi: well I have one user that his ip isn't static |
| 10:03.16 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 10:03.19 | gavimobile | and he needs to connect from different locations\ |
| 10:03.32 | SeRi | use dinamic dns if possible |
| 10:03.38 | SeRi | is the way I do it with my clients |
| 10:04.27 | gavimobile | SeRi: so I give this user a dynamic dns name and iptables has a rule to allow this dynamic dns name and he users this name to connect? |
| 10:04.43 | gavimobile | hrm... I cannot think of the logic here |
| 10:05.47 | unicron | i'll just leave this here http://www.qcnetwork.com/vince/doc/divers/udp_over_ssh_tunnel.html |
| 10:06.26 | SeRi | unicron: owl? |
| 10:06.49 | gavimobile | unicron: was that for me |
| 10:07.00 | unicron | perhaps you can have that user make excessively large ssh keys to login securely without a password, then do voip over the tunnel? |
| 10:07.00 | SeRi | gavimobile: I am sure it was. |
| 10:07.47 | unicron | another guide http://zarb.org/~gc/html/udp-in-ssh-tunneling.html |
| 10:07.48 | gavimobile | ssh is open to one local user |
| 10:07.55 | gavimobile | and I connect to that pc using rdp |
| 10:08.06 | gavimobile | than I connect to ssh locally |
| 10:08.54 | unicron | you like to overcomplicate things don't you |
| 10:09.20 | SeRi | unicron: looks that way. |
| 10:09.21 | unicron | if your user is anything like that, just have him install cjdns |
| 10:09.24 | gavimobile | unicron: I don't want my ssh to be attacked |
| 10:09.40 | unicron | it's plenty fast enough for voip, and provides an ipv6 tunnel over encrypted udp packets |
| 10:09.48 | SeRi | gavimobile: Than learn iptables |
| 10:10.04 | gavimobile | SeRi: ip tables is easy to understand |
| 10:10.56 | unicron | it provides many of the advantages of tor hidden services (minus the anonymity) but it does cloak your IP from casual snooping, and it can expose a whole range of ports behind a firewall or nat |
| 10:11.18 | SeRi | than I have nothing else to say. Good luck. :) maybe unicron has better ideas. |
| 10:11.35 | unicron | https://projectmeshnet.org/ |
| 10:11.39 | gavimobile | ok, thanks |
| 10:11.40 | SeRi | unicron: just curious are you part of the project? |
| 10:11.55 | unicron | cjdns? |
| 10:12.09 | SeRi | projectmesh |
| 10:12.38 | unicron | i run cybertron.decepticons.net and cyberia.biomancer.net |
| 10:13.01 | unicron | hyperboria nodes |
| 10:13.03 | SeRi | interesting |
| 10:13.10 | SeRi | no bsd port? |
| 10:13.32 | unicron | it runs in bsd supposedly, i haven't tried it yet |
| 10:13.39 | SeRi | Ill tyry to dl and compile it and see if it runs. |
| 10:13.49 | unicron | i got mad at netbsd and am building dragonfly now |
| 10:13.56 | SeRi | same for cjdns |
| 10:14.30 | SeRi | cool |
| 10:14.40 | unicron | but i'm a little nervous |
| 10:14.43 | unicron | this box has issues |
| 10:14.51 | unicron | and dragonfly's only detecting 1 of the four xeons in it |
| 10:14.55 | unicron | =[ |
| 10:15.09 | unicron | yet they all stand ready at POST |
| 10:15.50 | SeRi | :/ |
| 10:15.57 | unicron | dfly's funky new scheduler is the whole reason i wanted to try it, too |
| 10:16.15 | SeRi | ha. I was reading about that now |
| 10:16.35 | unicron | it can move closer related threads to closer together cores, improving cache latency |
| 10:17.08 | SeRi | very interesting. |
| 10:17.46 | SeRi | got your cert yet? |
| 10:18.27 | unicron | my ssl cert yes, my oscp cert no :P |
| 10:18.35 | unicron | i forget which one i may have been drunkenly bitching about |
| 10:18.48 | unicron | brb |
| 10:38.12 | SeRi | shift is over. time to go to bed. |
| 10:38.20 | SeRi | g/n all |
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| 11:35.37 | Tarang | Hi guys, is there a way to check the packetization (i.e is it 20ms or 30ms) of an active channel? e.g via sip show channels |
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| 11:38.38 | tompaw | Hello. |
| 11:39.23 | ghost75 | is there no gsm plugin for foobar? |
| 11:47.43 | ghost75 | found only winamp |
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| 11:58.06 | greenwolf | question... |
| 11:58.48 | greenwolf | would i achieve good results on a i-5 proccessor with 6GB ram 64-bit OS but it being a laptop? |
| 11:58.59 | greenwolf | good results for a asterisk based phone system that is? |
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| 12:02.25 | ghost75 | yes but why laptop? |
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| 12:03.30 | greenwolf | laptop is going to act like 1u rackmount server |
| 12:03.43 | ghost75 | wtf |
| 12:03.48 | greenwolf | is basically what i have to work with and its alot better machine and equipment than the desktop i have |
| 12:04.01 | ghost75 | a laptop is not designed to run 24x7 |
| 12:04.18 | greenwolf | better system than an old desktop im working with :) i know its now the standard thing to do |
| 12:04.35 | ghost75 | sell it and get other hardware |
| 12:04.35 | greenwolf | so u think i would ruin it then? |
| 12:04.46 | greenwolf | ok just wondering if it was possible |
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| 12:05.03 | greenwolf | but lets say i ran the system part time would it do well for asterisk at those specs? |
| 12:05.23 | ghost75 | i am running asterisk on atom |
| 12:05.42 | WIMPy | Laptops are great servers. Built-in UPS. |
| 12:06.07 | ghost75 | but are prone to overheating and are hard to repair |
| 12:06.12 | greenwolf | yea i thought the new laptops dont get hot because there is no moving parts now |
| 12:06.16 | greenwolf | they dont overheat anymore |
| 12:06.22 | ghost75 | they do |
| 12:06.24 | greenwolf | no moving parts in the harddrive |
| 12:06.43 | ghost75 | cpu and gpu gets hot |
| 12:06.51 | WIMPy | The hot parts are CPU and GPU. |
| 12:07.02 | ghost75 | and no raid |
| 12:07.04 | greenwolf | solid state CPU |
| 12:07.17 | ghost75 | solid what? |
| 12:07.18 | greenwolf | how could a solid state CPU overheat? |
| 12:07.25 | ghost75 | you mean solid state disk |
| 12:07.25 | greenwolf | read about it its called solid state |
| 12:07.27 | WIMPy | Thay have been for several decades :-) |
| 12:07.51 | WIMPy | Solid stte doesn't mean cool. |
| 12:07.57 | greenwolf | wimpy: have you ran an asterisk server off a laptop? |
| 12:08.04 | greenwolf | and had good results? |
| 12:08.18 | WIMPy | But: In a server you don;t need the GPU and depending on what it does the CPU might stay cool as well. |
| 12:08.21 | ghost75 | i have run laptop as linux home server |
| 12:08.56 | greenwolf | excuse me its got 8GB Ram and i-5 hip |
| 12:08.58 | greenwolf | chip* |
| 12:09.03 | greenwolf | i think it should run pretty good |
| 12:09.04 | WIMPy | Nope. I didn't. But old laptops are popular routers. |
| 12:09.18 | ghost75 | how many users should run there |
| 12:09.22 | greenwolf | so then this is a bad idea? |
| 12:09.37 | greenwolf | the laptop will just handle incoming call connections |
| 12:09.43 | greenwolf | no extensions |
| 12:09.57 | greenwolf | a phone system that allows callers to call in and retrieve account info |
| 12:10.03 | greenwolf | on a asterisk based phone system |
| 12:10.28 | ghost75 | i guess this laptop would be overkill from hardware |
| 12:10.49 | WIMPy | Dosn't sound too complicated. |
| 12:10.58 | WIMPy | Yeah. Probaly. |
| 12:11.00 | greenwolf | so you think with that type of hardware running in this laptop would be overkill for what im trying to do? |
| 12:11.33 | greenwolf | and do u think this would run well for what im trying to do |
| 12:11.38 | ghost75 | how many concurrent calls |
| 12:11.42 | WIMPy | Unless you have lots of data to process for a huge amount of calls, yes. |
| 12:12.01 | greenwolf | maybe around 200-400 current calls |
| 12:12.26 | ghost75 | at same time? |
| 12:12.30 | greenwolf | the asterisk system would also be accessing mysql data to retrieve account information for incoming calls |
| 12:12.45 | greenwolf | but again no extensions or outbound calls from the system |
| 12:12.58 | greenwolf | just an incoming call center to access the customers account info via phone lines |
| 12:13.09 | greenwolf | all voip btw |
| 12:13.22 | greenwolf | i think i-5 with 6GB ram should be ok |
| 12:13.25 | ghost75 | and if it breaks what then? |
| 12:13.35 | WIMPy | Well, taht is quite a number, but if you only do a bit of file playback, the should be fine. What your applications needs is another matter, off course. |
| 12:13.37 | ghost75 | can you allow few days of downtime? |
| 12:14.24 | greenwolf | ghost75: why do u say that? |
| 12:14.45 | ghost75 | because this is not designed to run 24x7 |
| 12:14.49 | greenwolf | i have many backup laptops for failover and thinking about using opensips for asterisk distribution to allow that many current calls |
| 12:15.03 | WIMPy | Because a laptop usually takes longer to replace. |
| 12:15.15 | ghost75 | so many calls and cannot afford reliable server hardware? |
| 12:15.35 | greenwolf | its just for a week or so till i get the server hardware here |
| 12:15.52 | greenwolf | so we can have something up and running while were waiting to get proper hardware obiously |
| 12:16.02 | WIMPy | Oh. Well, just do it then. |
| 12:16.04 | greenwolf | plus its only going to run at night |
| 12:16.15 | greenwolf | 8 hours at a time with 8 hours rest |
| 12:16.36 | greenwolf | ok then i thought so...just wanted everyone elses input |
| 12:16.41 | greenwolf | :) |
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| 13:23.25 | EmleyMoor | has a problem with his DAHDI FXO - it's "dropping" caller ID, so preventing me processing calls based on it |
| 13:24.06 | EmleyMoor | It's probably not an unknown problem in the UK - but is there a fix for 1.8.13.1? |
| 13:49.57 | WIMPy | The real fix is to go digital. |
| 14:02.55 | unicron | that's not always possible |
| 14:03.02 | unicron | especially in my case |
| 14:03.19 | unicron | which sucks because i would love to have dtmf out of band, as inband has been giving me nothing but issues |
| 14:07.06 | WIMPy | As long as you don;t have OOB DTMF from peer to peer, which ist rather unlikely still, that might not help much. |
| 14:09.13 | unicron | since i switched to dtmf inband, dialing from a cell phone has been extremely finicky |
| 14:09.32 | unicron | it eats my digits or gets them out of order |
| 14:09.54 | WIMPy | Sounds like you have some real networking issues. |
| 14:10.41 | unicron | i don't think so |
| 14:10.43 | ghost75 | is there a predefined exten called "fax" ? |
| 14:10.52 | WIMPy | yes |
| 14:11.40 | unicron | my termination and origination provider has a point of presence in the same city where the pbx vm is colo'd |
| 14:12.06 | unicron | and it's got a fat pipe regardless |
| 14:12.50 | unicron | and i'm the only user so far |
| 14:12.52 | unicron | :/ |
| 14:13.19 | ghost75 | http://www.freefaxbutton.com <- is this really free? |
| 14:13.28 | WIMPy | If you receive DTMF out of order there is something VERY wrong. |
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| 14:19.47 | ghost75 | damn, only 2 fax per day |
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| 15:02.39 | ghost75 | [Nov 24 16:01:38] WARNING[18864]: app_fax.c:432 transmit_audio: channel 'SIP/arcor_in-00000005' refused to negotiate T.38 |
| 15:02.39 | ghost75 | [Nov 24 16:01:38] WARNING[18864]: dsp.c:1354 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 15:02.51 | ghost75 | this means t.38 doesnt work with g.729 ? |
| 15:03.26 | unicron | try setting dtmfmode=rfc2833 in sip.conf |
| 15:04.05 | unicron | or change your codec to ulaw |
| 15:05.40 | ghost75 | when i allow ulaw/alaw then all phone calls will go with this :< |
| 15:08.47 | unicron | i am too much of a newbie to help with that |
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| 15:11.55 | ghost75 | the sender needs to send in t.38 too? |
| 15:12.47 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:13.07 | p3nguin | Wouldn't that be something if a sender could use any method they want, and you can still receive in T.38? |
| 15:14.26 | ghost75 | from what i understood udptl is used instead g.711 |
| 15:14.42 | p3nguin | <ghost75> is there a predefined exten called "fax" ? <------ No, there is no predefined extension. You have to CREATE your extensions for things to work. |
| 15:15.40 | ghost75 | ok, didnt use it even |
| 15:15.42 | p3nguin | 'fax' is a special extension, just like a, e, i, o, t, T, h, and s. |
| 15:16.26 | p3nguin | You use the fax extension when you use fax detection. The call will "jump" to the fax extension when the tones are detected. |
| 15:16.38 | ghost75 | roger tht |
| 15:17.42 | ghost75 | in sip.conf i have t38pt_udptl = yes and faxdetect = yes |
| 15:18.13 | p3nguin | Your peer must also support t.38 for t.38 to work. |
| 15:18.52 | ghost75 | i think it does: [Nov 24 16:01:38] WARNING[18864]: app_fax.c:432 transmit_audio: channel 'SIP/arcor_in-00000005' refused to negotiate T.38 |
| 15:19.13 | ghost75 | hmm but saying transmit, not receive |
| 15:20.09 | ghost75 | my itsp lists T.38 |
| 15:20.42 | p3nguin | I'm not certain, but I think you need to start the call in ulaw/alaw. |
| 15:20.56 | ghost75 | that would suck |
| 15:21.06 | p3nguin | Test it to see if it works. |
| 15:21.30 | ghost75 | have to wait anyway till tomorrow because free fax only allows 2 faxes |
| 15:23.29 | ghost75 | whats the easiest way to send fax from windows pc ? |
| 15:32.30 | ghost75 | https://reviewboard.asterisk.org/r/310/diff/?expand=1 |
| 15:32.38 | ghost75 | T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. |
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| 15:43.21 | ghost75 | t.38 needs directmedia=yes ? |
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| 15:52.58 | WIMPy | Doh. Switch without power has high packet loss. |
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| 16:51.33 | ghost75 | can i display all settings from sip.conf in cli ? |
| 16:52.09 | [TK]D-Fender | no |
| 16:52.26 | [TK]D-Fender | not in one shot anyway. |
| 16:52.30 | ghost75 | hard to know whats set default |
| 16:52.55 | [TK]D-Fender | "sip show peers", "sip show peer [X}", "sip show registry", "sip show settings" |
| 16:53.14 | [TK]D-Fender | ghost75: Relying on "default" instead of setting it yourself is a very bad idea. |
| 16:53.54 | ghost75 | i am curious about t38pt_rtp and tcp |
| 16:56.38 | ghost75 | i got this and not sure why: app_fax.c:432 transmit_audio: channel 'SIP/arcor_in-00000005' refused to negotiate T.38 |
| 16:57.00 | ghost75 | during receiving a fax |
| 16:57.10 | [TK]D-Fender | And di you put the proper T.38 settings in your peers? |
| 16:57.33 | unicron | does anybody use asterisk 11? |
| 16:58.00 | ghost75 | what means proper peer setting? |
| 16:58.07 | WIMPy | unicron: Set up a limesurvey. |
| 16:58.31 | ghost75 | have only allow=g729 |
| 16:59.20 | ghost75 | and i put dtmfmode=rfc2833 |
| 17:01.11 | ghost75 | and of course t38pt_udptl = yes in global |
| 17:01.55 | ghost75 | general, not global |
| 17:07.48 | [TK]D-Fender | and maybe THEY don't suport T.38 |
| 17:12.02 | ghost75 | they say they do |
| 17:12.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:12.52 | jpsharp | which provider? |
| 17:13.00 | ghost75 | vodafon |
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| 17:43.34 | ghost75 | but spandsp already supports t.38 no? |
| 17:46.38 | ghost75 | i received fax but i think it went over g711: http://pastebin.com/H85cqj3P |
| 17:57.53 | SeRi | ghost75: if you are trying to fax over itsp than your very limited on what you can use |
| 17:58.29 | SeRi | some people would say different things but my luck with spandsp has been horrendous. |
| 17:58.49 | ghost75 | maybe my asterisk is too old |
| 17:59.15 | SeRi | ghost75: you know of a working solution of fax over ip? |
| 17:59.27 | SeRi | If the itsp does not support it than there is nothing you can do |
| 17:59.44 | ghost75 | is the first time i try fax over ip |
| 18:00.17 | SeRi | Is not about asterisk been to old is about the support of a analog service over digital |
| 18:00.35 | ghost75 | i read also that spandsp doesnt support t.38 but not sure if thats old |
| 18:00.43 | SeRi | digium fax has worked very well for me |
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| 18:01.13 | SeRi | all faxes over digium fax ha ve completed 100% |
| 18:01.41 | ghost75 | you mean that free fax from digium which requires licensE? |
| 18:02.34 | SeRi | yes which is free |
| 18:02.45 | ghost75 | but needs to be installed extra |
| 18:02.56 | SeRi | dont be lazy |
| 18:03.06 | ghost75 | i rather go to asterisk 1.8 before |
| 18:03.49 | SeRi | which version you are on? |
| 18:03.55 | ghost75 | 1.6.2 |
| 18:04.33 | SeRi | what a waste of time. Than I will answer your statement... Yes asterisk is to old... In your case |
| 18:04.53 | WIMPy | So would 1.8 be. |
| 18:04.58 | ghost75 | wanted to update anyway but not sure if this will change anything |
| 18:05.57 | SeRi | WIMPy: 1.8 is still LTS |
| 18:06.22 | ghost75 | and for 1.8 i just need to do an apt-get |
| 18:06.28 | WIMPy | That doesn't make it newer. |
| 18:06.41 | SeRi | It makes it supported. |
| 18:07.02 | ghost75 | 1.8 > 1.6 |
| 18:07.07 | SeRi | and compare to 1.6? Yes it is newer |
| 18:07.07 | WIMPy | And there have been changes to T.38. But I don't know when exactely as that's not somethign I care about. |
| 18:07.24 | WIMPy | I'm not a support fetishist. |
| 18:07.25 | SeRi | WIMPy: same here :) |
| 18:07.30 | SeRi | lol |
| 18:07.53 | WIMPy | But I'd check for changes to fax and get the appropriate version. |
| 18:08.03 | SeRi | you like to make your own patches and fixes. nothing wrong with that :) |
| 18:08.07 | SeRi | WIMPy: +1 |
| 18:08.53 | WIMPy | Support only makes sense if it's clear what it's all about, anyway. |
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| 18:10.35 | ghost75 | t.30 means g.711 ? |
| 18:11.47 | WIMPy | Short answer: yes |
| 18:14.13 | ghost75 | and asterisk is also only able to make passthrough when i send a fax ? |
| 18:14.29 | WIMPy | Depends on the version. |
| 18:14.40 | jpsharp | unless you add on the SpanDSP fax package or the Digium Fax for Asterisk package. |
| 18:15.04 | SeRi | I am going to try and get dCAA |
| 18:15.11 | ghost75 | spandsp can relay fax ? |
| 18:15.20 | SeRi | yes |
| 18:15.24 | ghost75 | wow |
| 18:15.49 | SeRi | I didnt have luck with it though. |
| 18:16.32 | jpsharp | I've had pretty decent luck with SpanDSP + Gafachi |
| 18:20.00 | SeRi | gafachi... MhhMhh |
| 18:20.47 | jpsharp | their t38 is a bit hokey :) |
| 18:22.09 | ghost75 | is there any free fax send software for win which supports t.38 |
| 18:24.46 | SeRi | ghost75: again your itsp has to support it. |
| 18:25.19 | ghost75 | yes but looking for tool |
| 18:26.54 | SeRi | there right in your asterisk all ready |
| 18:27.08 | ghost75 | no on windows |
| 18:27.11 | SeRi | and WIMPy gave you a good advice |
| 18:27.26 | SeRi | this is not a windows support chan. |
| 18:27.28 | SeRi | I am out |
| 18:27.31 | WIMPy | Who? Me? |
| 18:27.32 | SeRi | going to take the dcaa |
| 18:27.58 | WIMPy | ended up there once... |
| 18:28.19 | SeRi | WIMPy: how was it? |
| 18:28.39 | WIMPy | easy. You just have to read carefully. |
| 18:28.47 | SeRi | cool. |
| 18:29.09 | SeRi | My job wants me to take it. they finally agree to get me some gethoducation.... |
| 18:29.23 | SeRi | cheap asses |
| 18:29.26 | WIMPy | What's that? |
| 18:29.57 | SeRi | Is when a company does not want to send you for training and send you to this eonline courses that get you now where |
| 18:30.16 | SeRi | s/now where/nowhere/ |
| 18:30.17 | WIMPy | aye |
| 18:32.03 | SeRi | Is the test 11hrs? or am I reading this wrong? |
| 18:32.12 | WIMPy | What? |
| 18:32.30 | WIMPy | Minutes perhaps. |
| 18:32.36 | SeRi | o ok |
| 18:32.38 | SeRi | lol |
| 18:34.01 | WIMPy | I had two wrongs. Unfortunatly you don't find out what you did wrong. |
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| 18:37.46 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 18:39.33 | *** join/#asterisk sambagirl (~sambagirl@unaffiliated/sambagirl) |
| 18:53.23 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:59.59 | slav3_kitten | meowls |
| 19:05.45 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 19:14.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:17.53 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 19:20.24 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:22.47 | sambagirl | if your using a hosted service for asterisk, and they don't have digium or sangoma timing hardware on their servers, how will that affect the broadcasting quality? if your local on your servers you have these products but that isnt necessisarily availble on remote services. |
| 19:22.50 | sambagirl | ? |
| 19:23.02 | p3nguin | .kick sambagirl enough! |
| 19:23.12 | sambagirl | oh go to hell |
| 19:23.16 | *** part/#asterisk sambagirl (~sambagirl@unaffiliated/sambagirl) |
| 19:24.56 | *** join/#asterisk Stilo (~textual@a85-138-2-114.cpe.netcabo.pt) |
| 19:25.54 | ChannelZ | well she told you |
| 19:26.07 | p3nguin | I know. I'm devastated. |
| 19:28.23 | ghost75 | dont like samba? |
| 19:28.31 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.244) |
| 19:28.35 | p3nguin | I don't like people acting like idiots. |
| 19:28.45 | [TK]D-Fender | More a fan of the horizontal mambo ;) |
| 19:28.55 | ghost75 | i dont even understand question |
| 19:29.20 | p3nguin | I wouldn't worry over it. |
| 19:29.25 | WIMPy | Whyu do you assume it was acted? |
| 19:29.26 | [TK]D-Fender | It was vague and possibly not even relevant. |
| 19:29.45 | p3nguin | I'll rephrase. |
| 19:29.58 | p3nguin | I don't like people behaving like idiots. |
| 19:30.25 | p3nguin | I have a bad tendency to erroneously use those terms synonymously. |
| 19:30.40 | p3nguin | I know better, but for some reason it is a habit. |
| 19:33.03 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 19:41.35 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:46.52 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-223-064.pools.arcor-ip.net) |
| 19:56.28 | *** join/#asterisk situssoft (~situssoft@host109-154-134-154.range109-154.btcentralplus.com) |
| 20:03.21 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:12.37 | situssoft | This is my first time on IRC, I'm looking for help with Asterisk. |
| 20:12.50 | p3nguin | ~ask |
| 20:12.50 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:12.52 | p3nguin | ~book |
| 20:12.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:15.33 | situssoft | I am placing calls to SIP devices using AMI, the calls are handled using AGI. I need to create a test system is it possible to have Asterisk respond as if there were a SIP device with a human attached? |
| 20:15.53 | p3nguin | Yes. |
| 20:16.25 | p3nguin | Create an extension (extensions.conf) that answers the call and plays sound files. See Playback() and BackGround(). |
| 20:17.55 | situssoft | Its been a long day, I think I've tried that. It seems that if the extension is not answered by a real SIP device it doesn't work for me. |
| 20:18.32 | situssoft | I'll have another go and see if I can get some more precise questions now that I have cut my teeth in IRC |
| 20:18.43 | situssoft | thanks for repsponding |
| 20:19.01 | p3nguin | If you make an attempt and it doesn't work out for you, bring us some evidence so we can troubleshoot it. |
| 20:19.07 | p3nguin | ~pb |
| 20:19.07 | 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. |
| 20:19.31 | p3nguin | You can pastebin your extension and any verbose/debug info. |
| 20:19.46 | situssoft | ok, thanks. What kind of evidence should I be looking to produce? |
| 20:20.00 | p3nguin | core set verbose 3 |
| 20:20.04 | p3nguin | Then generate the call. |
| 20:20.07 | situssoft | ok, understood. I'll have another go with that in mind |
| 20:20.12 | p3nguin | Something should spew out on the CLI. |
| 20:20.18 | situssoft | Oh it does |
| 20:20.26 | p3nguin | That's what you pastebin. |
| 20:20.49 | p3nguin | Everything pertaining to the call where the issue is, that's what we want to see to help solve a problem. |
| 20:20.55 | situssoft | I have to go offline now, but I will be back. Thanks |
| 20:23.40 | SeRi | WIMPy: you where right |
| 20:23.50 | SeRi | you have to pay attention |
| 20:23.57 | SeRi | they are sneaky with their questions |
| 20:24.13 | p3nguin | Are you DoD compliant, now? |
| 20:24.24 | SeRi | Indeed I am Sr. |
| 20:24.39 | SeRi | ISO5889633200 Comliant as well |
| 20:24.50 | SeRi | s/Comliant/Compliant/ |
| 20:25.02 | *** join/#asterisk mokmeister (~mokmeiste@109.78.107.7) |
| 20:25.03 | SeRi | lunch time |
| 20:26.17 | SeRi | I wonder if I will ever get to know the score |
| 20:26.22 | SeRi | I am interested to know |
| 20:26.58 | SeRi | ha I did. |
| 20:26.59 | SeRi | lol |
| 20:31.41 | jpsharp | You took the dcap test online? |
| 20:32.23 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 20:32.32 | Stilo | forward my home calls to my cell phone with asterisk is possible? |
| 20:32.38 | jpsharp | Yes. |
| 20:32.39 | p3nguin | Yes. |
| 20:32.40 | *** join/#asterisk TimeRider (~steve@02de05c4.bb.sky.com) |
| 20:32.42 | jpsharp | Yes. |
| 20:32.53 | unicron | Yes. |
| 20:32.59 | Stilo | Anyone have a good tutorial to do this? |
| 20:33.05 | p3nguin | ~book |
| 20:33.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:33.07 | Stilo | Is hard to do it? |
| 20:34.01 | p3nguin | No, it is not hard to do. |
| 20:34.18 | p3nguin | That is one of the most basic things. |
| 20:34.20 | ChannelZ | unless you're starting from nothing |
| 20:34.46 | jpsharp | The hardest part is figuring out how to get your home phone line connected to Asterisk. |
| 20:35.03 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 20:36.11 | Stilo | jpsharp you already did it? |
| 20:36.23 | jpsharp | Years ago :) |
| 20:38.09 | p3nguin | If you have a POTS line from the telco, the cheapest method is likely to be the use of an ATA that has an FXO port on it. |
| 20:47.14 | ghost75 | 2 new words in one sentence |
| 20:48.43 | p3nguin | Which two? |
| 20:48.58 | ChannelZ | POTS and FXO I'd guess |
| 20:49.06 | ghost75 | yes |
| 20:49.47 | p3nguin | ~pots |
| 20:49.47 | infobot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 20:49.50 | p3nguin | ~fxo |
| 20:49.50 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 20:50.36 | p3nguin | Hmm. Does that seem like ~pots has FXO and FXS reversed? |
| 20:51.01 | ghost75 | i thought ata is only for analog phones to make them ip |
| 20:51.31 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 20:51.44 | p3nguin | That's essentially what I intended to mean. |
| 20:51.58 | ChannelZ | No, it's right |
| 20:52.17 | ghost75 | so telco = analog phones |
| 20:52.23 | p3nguin | But rather than an ATA with an FXS port (which a phone connects to), I suggest an FXO port (which a phone line connects to). |
| 20:52.37 | p3nguin | The keyword was POTS. |
| 20:53.03 | p3nguin | POTS, Plain Old Telephone Service, is an analog service. |
| 20:53.16 | p3nguin | At least what customers see usually is. |
| 20:53.58 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 20:54.00 | darkdrgn2k | hi all |
| 20:54.16 | darkdrgn2k | whast the main diff between the pap2t and the spa2102 ? |
| 20:54.28 | ghost75 | pap2t is older |
| 20:55.15 | darkdrgn2k | but when it comes to quality of voice and such |
| 20:55.17 | darkdrgn2k | any major diff? |
| 20:55.25 | ghost75 | no |
| 20:55.27 | p3nguin | The PAP2T is just an ATA. |
| 20:55.29 | SeRi | jpsharp: no. I took the dcaa online. I wanted to see where I am as far as asterisk knowladge. |
| 20:55.38 | p3nguin | The SPA-2102 is an Ethernet router and ATA. |
| 20:55.48 | *** join/#asterisk bandroid (~bandroidx@2607:f358:1:fed5:4:0:414:11) |
| 20:55.55 | p3nguin | The ATA portion is the same in both. |
| 20:56.03 | darkdrgn2k | well no caus the 2102 support .38 |
| 20:56.08 | p3nguin | It does? |
| 20:56.18 | darkdrgn2k | yep |
| 20:56.22 | p3nguin | I wasn't aware of T.38 support in that. |
| 20:56.32 | jpsharp | SeRi: Oh, dcaa. Not dcap. |
| 20:57.03 | SeRi | jpsharp: yeap. That I know of dcap is not online |
| 20:57.14 | p3nguin | If that is actually a fact, then the ATA portion isn't exactly the same... but the main difference is the router in the SPA-2102. |
| 20:57.18 | jpsharp | I should do the same thing one of these days. I've been fondling Asterisk for 11 years now, time I made it official. |
| 20:57.51 | p3nguin | DCAA is related to Asterisk? |
| 20:57.52 | Stilo | What type of things can we do with Asterisk without any hardware? |
| 20:58.02 | ghost75 | router is used when phone directly talks to isp ? |
| 20:58.02 | SeRi | jpsharp: I am doing it to expand on my branch. |
| 20:58.46 | darkdrgn2k | http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a35e44.shtml |
| 20:58.48 | jpsharp | p3nguin: dcaa in this case is Digium Certified Asterisk Administrator. Not the defense department stuff. |
| 20:58.56 | [TK]D-Fender | Stilo: It makes a good coaster if you burn it on a CD |
| 20:59.16 | p3nguin | Well wtf. I asked seri if that made him DoD compliant and he said yes. |
| 20:59.23 | SeRi | p3nguin: LOL. |
| 20:59.36 | SeRi | p3nguin: I thought you where joking man |
| 20:59.39 | ghost75 | spa is said to be easier to setup |
| 20:59.40 | SeRi | LOL |
| 20:59.46 | SeRi | p3nguin: sorry! |
| 20:59.54 | p3nguin | Weirdo! |
| 20:59.59 | SeRi | ROFL! |
| 21:00.01 | SeRi | HAHAHAHA |
| 21:00.13 | p3nguin | Bad info in, bad info out. |
| 21:00.53 | p3nguin | Okay, so how much is the test fee for the asterisk thing? |
| 21:01.07 | SeRi | p3nguin: Free if you pass teh first time |
| 21:01.20 | SeRi | than you move on to dcap |
| 21:01.22 | p3nguin | I anticipate being able to pass it with ease, and that would allow me to add it to my collection. |
| 21:01.29 | Stilo | SeRi what can I do with Asterisk without hardware? |
| 21:01.43 | ghost75 | oh lol spa is already outdated |
| 21:01.44 | p3nguin | Nothing! Software needs hardware. |
| 21:01.59 | SeRi | p3nguin: I am sure you will with your eyes closed |
| 21:02.13 | p3nguin | Software requires a shell, shell requires a kernel, kernel requires hardware. |
| 21:02.38 | p3nguin | seri: How do I get started? |
| 21:02.44 | [TK]D-Fender | Kernel leads to fear. Fear leads to hate. Hate leads to the Dark Side..... |
| 21:03.14 | SeRi | p3nguin: go to the digium site and log in |
| 21:03.27 | SeRi | than go to training |
| 21:03.35 | SeRi | Than click on DCAA |
| 21:03.40 | SeRi | and take the test |
| 21:03.50 | p3nguin | Sounds easy enough. I'll probably do it later today or tonight. |
| 21:04.37 | SeRi | it was simple. took me a few min to go through it. I will pass you the same advice WIMPy gave me which for you will not be a problem.... Read carefully. |
| 21:04.53 | SeRi | for me that was a small issue. LOL |
| 21:05.52 | SeRi | p3nguin & [TK]D-Fender that will be my new signature for the rest of my life |
| 21:06.09 | p3nguin | What will be? |
| 21:06.26 | SeRi | Software requires a shell, shell requires a kernel, kernel requires hardware. Kernel leads to fear. Fear leads to hate. Hate leads to the Dark Side. |
| 21:06.43 | SeRi | poetry at best |
| 21:06.47 | SeRi | LOL |
| 21:07.04 | Stilo | i want to do any possible project with asterisk without hardware, i got cloud servers they can be used for something? |
| 21:07.10 | Stilo | and want work directly with mobiles |
| 21:07.14 | *** join/#asterisk spooky7 (~spooky@athedsl-221013.home.otenet.gr) |
| 21:07.18 | SeRi | k out to get whiskey |
| 21:07.19 | p3nguin | Yes, that will work. |
| 21:07.31 | p3nguin | You can install Asterisk on someone else's hardware. |
| 21:07.36 | SeRi | cellebrate my new cert |
| 21:07.40 | spooky7 | hello to all asterisk users |
| 21:07.49 | [TK]D-Fender | Someone else's hardware = hardware |
| 21:07.54 | SeRi | p3nguin: you will join me later on today. |
| 21:08.00 | Stilo | yes p3nguin |
| 21:08.05 | Stilo | but what type of projects can i do |
| 21:08.19 | [TK]D-Fender | Stilo: Whatever you want |
| 21:08.20 | p3nguin | I would say the most common thing is to... |
| 21:08.24 | p3nguin | make phone calls. |
| 21:08.30 | [TK]D-Fender | Stilo: My Asterisk server is a jukebox and makes me coffee |
| 21:08.36 | spooky7 | Can anyone help me with my asterisk installation ? |
| 21:08.41 | [TK]D-Fender | ~ask |
| 21:08.41 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:08.43 | [TK]D-Fender | spooky7: ^^ |
| 21:08.58 | Stilo | [TK]D-Fender |
| 21:09.00 | Stilo | without hardware |
| 21:09.02 | Stilo | it's limited |
| 21:09.06 | Stilo | can't do nothing basically |
| 21:09.20 | p3nguin | (1507.49) <[TK]D-Fender> Someone else's hardware = hardware |
| 21:09.41 | [TK]D-Fender | Stilo: Depends on your definition of "nothing". |
| 21:10.06 | Stilo | That's "nothing" is what I want to know what is possible. |
| 21:10.15 | Stilo | Because I don't know anything possible to do without hardware |
| 21:10.32 | [TK]D-Fender | Stilo: * will do whatever you configure it to do with whatever services & hardware you set it up with |
| 21:10.48 | [TK]D-Fender | Stilo: You are using the word "hardware" is a VERY dangerously vague way. |
| 21:11.12 | [TK]D-Fender | Stilo: This will not lead to the answers you could and should be getting. |
| 21:11.38 | Stilo | i mean i have a cordless phone not telephone/pbx connected to a wall jack and have an iphone |
| 21:11.59 | [TK]D-Fender | Stilo: Your IP phone is hardware.... |
| 21:12.00 | ghost75 | iphone should be sipable |
| 21:12.16 | [TK]D-Fender | Stilo: * is sofware that can speak to to it... seems THAT is capabble of doing thigs together |
| 21:12.17 | ghost75 | sippy phone :) |
| 21:12.35 | [TK]D-Fender | Stilo: Your analog cordless phone ... that would require something else to let it become useful. |
| 21:12.56 | [TK]D-Fender | iPhone could have a VoIP client on it... |
| 21:13.10 | ghost75 | in two and a half man, she calls alan also sippy? |
| 21:13.15 | Stilo | so it's possible to do projects with asterisk and iPhone :))))) |
| 21:13.21 | Stilo | just have to know which projects are possible |
| 21:14.20 | [TK]D-Fender | Stilo: what "projects"? * is a TELEPHONY TOOLKIT. you can do whatever you want with it. |
| 21:14.54 | [TK]D-Fender | Stilo: Dozens? hundreds? Thousands? How many "thinks" can you build with LEGO? It's limited by your imagination and whatever OTHER resources you have available. |
| 21:15.01 | spooky7 | I have one PSTN line and I want my asterisk to distinguish an incoming call if it is a voice call then ring all my internals but if it is a fax call to tranfer the call to iaxmodem . Is tha possible ? |
| 21:15.12 | [TK]D-Fender | Stilo: You are being too vague with "projects" jsut like with "hardware". |
| 21:15.36 | [TK]D-Fender | spooky7: Yes, * has fax detection for PSTN lines on common interfaces you might use |
| 21:16.00 | [TK]D-Fender | spooky7: there is even an Asterisk Standard Extension expressly for this |
| 21:16.28 | spooky7 | Can you give me details ? |
| 21:16.32 | [TK]D-Fender | ~book |
| 21:16.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:16.41 | [TK]D-Fender | ^ |
| 21:17.04 | p3nguin | faxdetect and the fax extension |
| 21:17.04 | [TK]D-Fender | Asterisk Standard Extensions are Dialplan 101 knowledge. Youshould know how they are used.... |
| 21:18.04 | Stilo | [TK]D-Fender: to forward my home calls(cordless phone) to my iPhone, possible? |
| 21:18.49 | [TK]D-Fender | Stil what are "home calls"? |
| 21:18.53 | spooky7 | Sorry but I am new to asterisk |
| 21:20.25 | spooky7 | I have the next dialplan [incoming] |
| 21:20.25 | spooky7 | exten => s,1,Wait(0.5) |
| 21:20.25 | spooky7 | exten => s,n,Answer() |
| 21:20.25 | spooky7 | exten => s,n,Playtones(ring) |
| 21:20.25 | spooky7 | exten => s,n,Waitexten(2,m) |
| 21:20.25 | spooky7 | exten => s,n,Macro(DialHomePBX,250) |
| 21:20.26 | spooky7 | ; If this is a fax, dial fax line |
| 21:20.26 | spooky7 | exten => fax,1,StopPlaytones |
| 21:20.27 | spooky7 | exten => fax,2,Dial(IAX2/iaxmodem) |
| 21:20.27 | spooky7 | exten => fax,3,Hangup |
| 21:20.31 | p3nguin | New to IRC, too. |
| 21:20.32 | spooky7 | But not working |
| 21:20.35 | [TK]D-Fender | spooky7: PASTEBIN <- |
| 21:20.37 | spooky7 | yes |
| 21:20.37 | [TK]D-Fender | ~pb |
| 21:20.37 | 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. |
| 21:20.38 | p3nguin | Do not flood. |
| 21:20.39 | [TK]D-Fender | ^^^ |
| 21:21.10 | [TK]D-Fender | spooky7: You should also be telling us what you're using..... |
| 21:21.30 | Stilo | [TK]D-Fender if someone call to my home telephone and get no answer get a notification on iphone |
| 21:21.50 | spooky7 | Centos 6.3 asterisk 11 x100p fxo iaxmodem hylafax |
| 21:21.56 | [TK]D-Fender | stil You've just added something NEW to this picture. Before you had a cordless PHONE, an iPhone .... and now an ANALOG LINE? |
| 21:22.35 | [TK]D-Fender | spooky7: "faxdetect=both" in your DAHDI config |
| 21:22.42 | spooky7 | yes |
| 21:22.51 | [TK]D-Fender | spooky7: I'd also wait more than 2 s. |
| 21:23.04 | spooky7 | ok |
| 21:23.06 | [TK]D-Fender | spooky7: PB it all up. |
| 21:23.12 | p3nguin | Do not flood. |
| 21:23.28 | spooky7 | all what ? |
| 21:23.42 | [TK]D-Fender | spooky7: DAHDI configs, dialplan, call attempt, all of it |
| 21:23.58 | spooky7 | hmmmmmmm |
| 21:24.10 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 21:24.13 | Stilo | i don't have analog line |
| 21:24.38 | [TK]D-Fender | Stilo: Where is this "call" coming from then? |
| 21:24.49 | p3nguin | He must have satellite phones in his house. |
| 21:25.00 | [TK]D-Fender | The voices just "appear" I guess.... |
| 21:25.37 | [TK]D-Fender | It's shocking how clueless people can be about what they have and what is happening around them. |
| 21:25.40 | *** join/#asterisk Mr_Break_it (~klem@c-75-71-128-197.hsd1.co.comcast.net) |
| 21:26.23 | cusco | hmm little question, I'm trying to set up some small dialplan on my frinds router (ddwrt) - it has asterisk 1.4 and it states it cannot join queue |
| 21:26.24 | Tarang | Can anyone help me get asterisk to use 20ms iLBC frames? |
| 21:26.25 | *** join/#asterisk Stilo (~textual@a85-138-2-114.cpe.netcabo.pt) |
| 21:26.26 | cusco | <PROTECTED> |
| 21:26.27 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 21:26.29 | cusco | [Nov 24 21:26:00] WARNING[18566]: app_queue.c:4077 queue_exec: Unable to join queue 'casa' |
| 21:26.32 | cusco | can't figure why |
| 21:26.40 | [TK]D-Fender | cusco: Stop using | as a delimiter first |
| 21:26.49 | cusco | Iam not |
| 21:26.51 | cusco | Im using , |
| 21:26.54 | cusco | its 1.4 |
| 21:27.02 | cusco | it writes to cli that way |
| 21:27.04 | p3nguin | All those pipes seem to say otherwise. |
| 21:27.10 | [TK]D-Fender | cusco: next check any limits on member availability,, make sure the names right, etc |
| 21:27.20 | cusco | Iset joinwhenempty=yes |
| 21:27.35 | [TK]D-Fender | cusco: cusco I think that was "joinempty". |
| 21:27.46 | [TK]D-Fender | -when |
| 21:27.49 | cusco | sorry, indeed: joinempty = yes |
| 21:27.57 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 21:28.00 | [TK]D-Fender | cusco: check it all, the PB it all |
| 21:29.36 | cusco | ow, Iset joinempty in [queue] not only in [general] |
| 21:29.39 | cusco | sorry totrouble you |
| 21:32.10 | spooky7 | here is my pastebin http://pastebin.com/QZECVv0j |
| 21:32.51 | [TK]D-Fender | faxdetect = both faxdetect = incoming faxdetect = outgoing <- you have THREEE. **BAD** |
| 21:33.35 | [TK]D-Fender | spooky7: that LAST one says NO INBOUND |
| 21:34.23 | spooky7 | trying to see it |
| 21:35.27 | spooky7 | I cannot see what you mean |
| 21:35.36 | [TK]D-Fender | http://pastebin.com/QZECVv0j |
| 21:35.38 | [TK]D-Fender | 33, 34,35 |
| 21:35.43 | [TK]D-Fender | you have THREE lines of this |
| 21:35.47 | [TK]D-Fender | they are OVERRIDING each other |
| 21:36.05 | [TK]D-Fender | that LAST one is what "sticks" meaning you are detecing OUTBOUND only, not INBOUND |
| 21:37.11 | p3nguin | You'll have to spell it out for him to remove the two that are wrong. |
| 21:37.44 | spooky7 | OK I fixed it to faxdetect=both |
| 21:38.11 | spooky7 | I will try again |
| 21:38.18 | spooky7 | just a moment |
| 21:39.12 | ghost75 | both is same as yes |
| 21:39.31 | p3nguin | Oh yeah? |
| 21:41.46 | spooky7 | Yeah!!!!!!!!!! |
| 21:41.53 | spooky7 | That worked |
| 21:42.00 | spooky7 | You are the best |
| 21:42.04 | spooky7 | Thank you |
| 21:42.32 | [TK]D-Fender | spooky7: You're welcome. |
| 21:42.42 | spooky7 | So simple |
| 21:43.10 | spooky7 | Let's go one step beyond |
| 21:44.04 | SeRi | hell no |
| 21:44.16 | spooky7 | Is possible when a call is received the caller id immediately stored in mysql and not only when the call is finished |
| 21:44.18 | spooky7 | ? |
| 21:45.55 | p3nguin | As soon as the call reaches ANY extension that is capable of running the appropriate app/function, you can write it to the database. |
| 21:45.55 | [TK]D-Fender | spooky7: Store it yourself |
| 21:46.07 | [TK]D-Fender | spooky7: This is OUTSIDE of CDR |
| 21:46.15 | [TK]D-Fender | CDR is it's own thing and only gets written at the end |
| 21:46.25 | jpsharp | Ohwow. I logged into my Digium account and there's all kinds of people assocated with it that I've never even talked to, much less "managed". |
| 21:46.44 | p3nguin | [tk]d-fender: its |
| 21:47.16 | [TK]D-Fender | p3nguin: Just making sure you feel validated #grammarnazi ;) |
| 21:47.25 | SeRi | lmao |
| 21:47.40 | spooky7 | So it not possible |
| 21:47.54 | [TK]D-Fender | Occasionally people get nabbed and replaced with near-perfect duplicates. Grammar is a good tell-tale sign ... and the presence of pods ;) |
| 21:48.12 | [TK]D-Fender | spooky7: Not in the CDR DB |
| 21:48.20 | [TK]D-Fender | spooky7: in your own, yes |
| 21:48.33 | spooky7 | how ? |
| 21:48.53 | [TK]D-Fender | spooky7: lookup the DB options in the book.. |
| 21:48.54 | [TK]D-Fender | ~book |
| 21:48.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:49.10 | [TK]D-Fender | func_odbc, MySQL() dialplan app, etc. |
| 21:49.29 | spooky7 | OK I will give it a try |
| 21:49.42 | [TK]D-Fender | Call an AGI and use its (p3nguin) API's |
| 21:49.48 | *** join/#asterisk nicknam1232 (021d232e@gateway/web/freenode/ip.2.29.35.46) |
| 21:58.52 | ghost75 | [TK]D-Fender: T.38 is completely NOT using G.711 right? |
| 22:03.38 | jpsharp | I seem to have passed the dcaa test. Yay me. |
| 22:04.33 | ghost75 | bows down |
| 22:04.56 | ChannelZ | OK so type=peer matches by IP and port. But I guess the exception is for initial registrations, in which case it has to try to use the From: name to match a device name since obviously it doesn't know their IP if their host=dynmic.. yes? |
| 22:05.28 | jpsharp | ghost75: True T.38 calls do not use G711 at all. |
| 22:07.07 | p3nguin | The whole matching IP/port thing is for the call. The REGISTER is always username:secret. |
| 22:08.07 | p3nguin | Once the device registers, you can pretend that the host is now set to the contact address rather than "dynamic". |
| 22:08.14 | p3nguin | Then INVITEs are matched on IP/port. |
| 22:09.19 | ghost75 | do you use type=friend on phones? |
| 22:09.27 | p3nguin | Nope. |
| 22:09.43 | ChannelZ | Yes but 'username' is the device name matched from the From: name, yes? |
| 22:09.46 | p3nguin | I use type=peer for everything. (few exceptions exist) |
| 22:10.01 | ghost75 | friend is peer+user ? |
| 22:10.06 | [TK]D-Fender | yes |
| 22:10.21 | p3nguin | I'd have to look to see if REGISTER has the same From: field. |
| 22:10.59 | ghost75 | this user brings any more security or what |
| 22:11.34 | p3nguin | Type of 'user' only allows for a device to make calls inbound to asterisk, not receive calls from asterisk. |
| 22:14.12 | p3nguin | That's not really more "secure" if the device can still make calls. |
| 22:15.34 | ghost75 | The type=friend is a device type that accepts both incoming and outbound calls, |
| 22:15.34 | ghost75 | ; where Asterisk match on the From: username on incoming calls. |
| 22:16.58 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-187.phlapa.fios.verizon.net) |
| 22:17.07 | p3nguin | Are you still having problems understanding peer vs. user vs. friend? |
| 22:17.39 | ghost75 | friend seems not to be a useful option |
| 22:18.18 | p3nguin | It's useful when you have multiple devices using the same IP address. |
| 22:18.44 | wdoekes | .. or devices that do not register before calling |
| 22:18.57 | p3nguin | If multiple devices have the same IP address, you can't match each device based on the host address, so you have to use something unique, such as the user name. |
| 22:19.30 | p3nguin | Both peer and friend can take calls without registration, so that's not really a good example. |
| 22:19.34 | ChannelZ | although peers match by IP AND port number unless their port numbers aren't consistent.. |
| 22:19.46 | ghost75 | i see |
| 22:20.36 | ghost75 | http://forums.asterisk.org/viewtopic.php?t=78053 |
| 22:20.49 | p3nguin | A common place where you have multiple user names but a single IP address is on a phone where you have to configure multiple line keys with separate accounts. |
| 22:21.08 | p3nguin | For that, you use friend instead of peer. |
| 22:21.43 | p3nguin | If you prefer to match by user name rather than IP address in other cases, that's fine too. friend does that. |
| 22:22.33 | ghost75 | is matching only inbound or both |
| 22:22.46 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 22:23.00 | p3nguin | "inbound" is a relative term. What specifically do you mean? |
| 22:23.08 | p3nguin | A call from the device inbound to asterisk? |
| 22:23.59 | ghost75 | think so |
| 22:24.07 | p3nguin | That would be an INVITE from the phone, which asterisk matches against the peer entries. |
| 22:26.20 | ChannelZ | Put it this way. A 'user' is a device which can send a call TO Asterisk, but that you cannot send a call to FROM Asterisk (IE you can't Dial(SIP/thatuser). A 'peer' and a 'friend' can do both, but they match peers in sip.conf in a different way. |
| 22:26.23 | p3nguin | When asterisk determines that there is a matching entry and that the entry has a secret set, it will challenge the device to authenticate. |
| 22:27.22 | p3nguin | If the entry has type=friend, the authentication will use the name in the From: field and the secret which come from the phone. |
| 22:28.37 | p3nguin | If the type is peer, asterisk will have to check the registry to put the name with the address. |
| 22:28.58 | p3nguin | (or the host entry for static hosts) |
| 22:29.13 | ChannelZ | The explanation in sip.conf is still wrong BTW |
| 22:29.21 | p3nguin | What does it say now? |
| 22:29.44 | ghost75 | see here: http://forums.asterisk.org/viewtopic.php?t=78053 |
| 22:30.02 | ChannelZ | Look at "Naming devices" - I think they mean to say #2 should be type=friend |
| 22:31.44 | ChannelZ | ghost75: that only tells a small part of the story; Yes a 'friend' is considered both a 'user' and a 'peer' but the important part is what a user or a peer actually means. |
| 22:32.17 | p3nguin | That seems right to me. #2 indicates that it matches the address from the From: where #1 says the username in the From: |
| 22:33.40 | p3nguin | I agree that it is quite unclear, though. |
| 22:33.45 | ChannelZ | But #2 says "Asterisk checks the From: addres (sp) and matches the list of devices with a type=peer" but peers don't match by From: - unless they are talking things other than INVITEs, in which case the explanation is at best totally unclear |
| 22:33.57 | ghost75 | if on type user cannot be called, whats that used for |
| 22:34.23 | p3nguin | It's unclear for sure, but I'm thinking it could mean the host portion of the From: as opposed to the user portion. |
| 22:34.31 | ChannelZ | ghost75: I'd say it rarely is, but think of an ITSP which sends you incoming calls from a different server. You might have a 'user' setup for just that. |
| 22:34.45 | p3nguin | It would be better stated the same as #1 but type=friend, as you indicated. |
| 22:35.08 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.151) |
| 22:35.12 | p3nguin | Yay for uncertainty. |
| 22:35.57 | ChannelZ | Unfortunately the book isn't any clearer, as they don't elaborate about any of it it seems. It's always been the weakest and most confusing part of Asterisk IMHO because it's seemingly so poorly documented. |
| 22:36.21 | p3nguin | I do occasionally use type=user for an ITSP that only sends calls in to my asterisk and that I cannot dial out through. |
| 22:36.29 | p3nguin | It's rare, but I have done it. |
| 22:37.08 | p3nguin | peer just seems like the right answer most of the time. |
| 22:39.29 | wdoekes | peer = find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0); |
| 22:39.30 | wdoekes | /* Then find devices based on IP */ |
| 22:39.30 | wdoekes | if (!peer) { peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type); |
| 22:40.05 | wdoekes | should really be only #1 and #3 |
| 22:41.04 | ChannelZ | well but a friend has both a user and a peer entry. I guess an important thing to know is in what order are they searched? |
| 22:41.25 | ChannelZ | Seems like users first (by name) then peers (by IP/port) |
| 22:41.26 | wdoekes | when looking for peers, friends and peers are considered equally important |
| 22:41.29 | p3nguin | Friend should check the name first. |
| 22:41.43 | p3nguin | address second. |
| 22:42.14 | wdoekes | the above code does that.. #1 look by From, #3, look by IP |
| 22:42.59 | ghost75 | so on type=peer the username= is useless |
| 22:43.18 | wdoekes | ghost75: no.. username is not the username you think it is |
| 22:43.20 | ChannelZ | not necessarily, but username= probaby isn't what you think it is |
| 22:43.24 | ChannelZ | heh jinx |
| 22:43.34 | wdoekes | [username] is the only username used for matching incoming calls |
| 22:43.44 | p3nguin | The username field is what you SEND to that device. |
| 22:43.46 | p3nguin | from asterisk. |
| 22:44.11 | p3nguin | Think ITSP requiring your username to make calls. |
| 22:44.23 | ghost75 | like login name |
| 22:45.13 | SeRi | ok back |
| 22:45.14 | p3nguin | It's not a login, though. |
| 22:45.26 | p3nguin | It's just authentication due to the INVITE. |
| 22:45.37 | ghost75 | then fromuser is used together with type=user |
| 22:45.48 | p3nguin | You send the INVITE, they return that you are not authorized, then you send the authentication. |
| 22:48.30 | slav3_kitten | p3nguin, should i use only user for my voip.ms which is inbound only? |
| 22:48.41 | p3nguin | No. |
| 22:49.03 | p3nguin | You don't use voip.ms for termination services? |
| 22:49.21 | slav3_kitten | nope, it's my DID only, i use flowroute for outbound termination |
| 22:49.37 | slav3_kitten | they are a good bit cheaper for the calls to the uk g/f's mobile |
| 22:49.52 | p3nguin | But slightly higher for calls within the US. |
| 22:50.18 | slav3_kitten | fraction of a cent from what i saw |
| 22:50.31 | slav3_kitten | and my outbound volume is... very limited |
| 22:51.17 | p3nguin | Anyway, they don't send a username that you can match on, so type=peer is the best way for that. |
| 22:51.37 | p3nguin | type=peer and host=theirhostnameoraddr |
| 22:51.54 | slav3_kitten | *nods* i got ya now |
| 22:52.09 | p3nguin | All calls that they send to you have a username of the calling phone number. |
| 22:52.29 | slav3_kitten | ah, that'd not work well i don't think |
| 22:52.36 | slav3_kitten | btw iax2 whoot |
| 22:53.09 | p3nguin | Good for high call volume and for NAT traversal. |
| 22:53.44 | slav3_kitten | yea i couldn't get those moronic friends to get sip working right... so they now use iax2 without incident |
| 22:54.41 | SeRi | slav3_kitten: why moronic? wasnent you who was helping them? |
| 22:55.14 | SeRi | s/wasnent/wasent/ |
| 22:55.27 | p3nguin | still wrong |
| 22:55.33 | p3nguin | wasn't <------------ |
| 22:56.00 | SeRi | well I try. LOL |
| 22:56.18 | slav3_kitten | SeRi, yea but i can't drive over an setup their network equipment |
| 22:56.34 | SeRi | engraves wasn't to brain |
| 22:56.36 | p3nguin | teamviewer! |
| 22:57.27 | slav3_kitten | p3nguin, hell no. then they will blame me for every network problem they have |
| 22:57.44 | slav3_kitten | and a number of softphones support iax2 |
| 22:57.59 | SeRi | p3nguin: I wont argue with that. unless is linux they are remoting too. |
| 22:58.28 | SeRi | lol |
| 22:59.02 | slav3_kitten | too bad you can't ssh into p3nguin's brain an fix wasn't |
| 22:59.30 | p3nguin | pf is probably blocking his access. |
| 22:59.46 | slav3_kitten | pf? |
| 23:00.00 | p3nguin | ~pf |
| 23:00.00 | infobot | i heard pf is pink floyd, or a BSD packet filter (not related to Pink Floyd), or http://www.openbsd.org/faq/pf/ (no, that's not related to Pink Floyd either) |
| 23:00.15 | SeRi | pf is awesome |
| 23:01.05 | slav3_kitten | i dunno about tha |
| 23:01.07 | slav3_kitten | t |
| 23:02.15 | SeRi | dont hate on what you dont understand ;) |
| 23:02.48 | *** join/#asterisk greenwolf (42570424@gateway/web/freenode/ip.66.87.4.36) |
| 23:03.30 | SeRi | I use pf and no problem. |
| 23:03.46 | jpsharp | pf ftw |
| 23:03.53 | SeRi | +1 |
| 23:04.28 | slav3_kitten | i run cisco |
| 23:05.22 | unicron | any ideas why i might be able to send dtmf but not voice? can receive voice... |
| 23:05.42 | unicron | it's one way audio but the dtmf is two way |
| 23:06.19 | p3nguin | ~sipnat |
| 23:06.19 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 23:07.00 | unicron | this box is on 1:1 NAT |
| 23:07.46 | p3nguin | I've heard of translation issues with 1:1. |
| 23:08.55 | carrar | I'd guess you not sending DTMF inband |
| 23:09.23 | carrar | :) |
| 23:09.34 | p3nguin | If it was inband, I guess it would also be one-way. |
| 23:09.39 | carrar | yup |
| 23:10.37 | unicron | dtmf is inband |
| 23:10.57 | unicron | i called my friend's phone and he could hear me pressing digits but not my voice |
| 23:11.24 | p3nguin | And then you had him pushing buttons and you heard them? |
| 23:11.38 | unicron | forgot to test that part... one sec |
| 23:12.21 | carrar | ok thats a confusing statement |
| 23:12.53 | unicron | yes, he can press buttons and i can hear them |
| 23:13.15 | p3nguin | So we know DTMF isn't a problem. |
| 23:13.28 | SeRi | unicron: is this over ssh? |
| 23:13.32 | p3nguin | Try another microphone. |
| 23:13.43 | p3nguin | Or try reconfiguring the input for your phone. |
| 23:13.54 | unicron | i'll just have him call me instead |
| 23:14.26 | p3nguin | If he calls you, you get two-way audio? |
| 23:15.26 | unicron | yes, 2 way if he calls me |
| 23:15.30 | p3nguin | That eliminates your mic and input from being the problem. |
| 23:15.57 | unicron | but we were both dialing into the same DID |
| 23:16.07 | unicron | from the PSTN |
| 23:16.10 | p3nguin | I'd be trying to examine the NAT. |
| 23:16.30 | SeRi | seems like a nat issue to me |
| 23:16.52 | SeRi | or one of his crazy complex security setup... |
| 23:17.03 | SeRi | :P |
| 23:17.10 | unicron | the calls aren't gonna be crypted at all unless users have zrtp |
| 23:17.32 | unicron | (or unless the pbx is connected to over cjdns, but that's another story) |
| 23:18.08 | unicron | i just need to protect metadata and forum posts and such |
| 23:18.41 | slav3_kitten | forum posts? |
| 23:19.16 | p3nguin | You've never heard of the voice forums in asterisk? |
| 23:19.16 | slav3_kitten | slips on his "i'm a noob" shirt |
| 23:19.30 | slav3_kitten | nope... |
| 23:19.40 | p3nguin | Most people call them "conferences." |
| 23:20.18 | slav3_kitten | like conference calls? |
| 23:20.39 | p3nguin | I guess that went right over your head. |
| 23:21.04 | slav3_kitten | i'm honestly quiet tired |
| 23:21.19 | slav3_kitten | so yes, it entirely went over my head |
| 23:21.38 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:23.00 | unicron | well p3nguin |
| 23:23.19 | unicron | i can't afford a real ssl cert |
| 23:23.28 | unicron | so i have a forum with a cheap cert on shared hosting |
| 23:23.44 | unicron | and i need people to register there, and end up with sip accounts on a pbx elsewhere |
| 23:23.56 | unicron | with security and encryption at every step |
| 23:24.33 | carrar | So you've removed this system from the internet :) |
| 23:24.37 | p3nguin | I don't understand the relationship between a forum and sip. |
| 23:24.49 | slav3_kitten | that's where i've gotten confused at |
| 23:24.57 | unicron | it's an unholy union which i am creating through a custom API |
| 23:25.38 | slav3_kitten | i know that there are forums for asterisk as well jsut the sip + forums = confusion |
| 23:26.04 | unicron | i could make sip accounts manually when people register, but i'd rather solve a harder problem once than repeat a task |
| 23:26.31 | unicron | carrar: paranoia makes a good servant but a poor master :) |
| 23:26.32 | jpsharp | So you throw everything into a database and set up realtime on Asterisk. |
| 23:27.27 | unicron | i was thinking of just having the pbx run a web server that only listens on localhost, and then the forum sends its api calls through an ssh tunnel |
| 23:27.46 | slav3_kitten | god parents need to fucking give me a call when their computer starts to have an issue. instead i get a call at 5pm on a weekend "my computer is acting up, it's been screwing up for a couple months though so i don't know why it's now not working with anything" |
| 23:28.25 | jpsharp | unicron: That's all kinds of rube goldbergish. |
| 23:31.05 | unicron | well |
| 23:31.38 | slav3_kitten | ohh web forms not fourms? |
| 23:32.32 | unicron | i guess i'll set up the forum db to run on the pbx as well |
| 23:33.53 | p3nguin | You wouldn't rather have a centralized DB and let both your forums and your pbx use it? |
| 23:34.30 | unicron | that would be ideal |
| 23:35.04 | p3nguin | Time to deploy another virtual machine. |
| 23:35.20 | unicron | i'll see if i can arrange it |
| 23:36.39 | unicron | i had another idea involving storing a key in memory after gaining a passphrase over a vsphere console before continuing to boot |
| 23:36.58 | unicron | but if i put up another box, i can just encrypt its drive and be done with it |
| 23:37.55 | unicron | (using mlock() and prctl(PR_SET_DUMPABLE,0)) |
| 23:42.48 | *** join/#asterisk citywok (~kvirc@208.186.203.146) |
| 23:44.51 | citywok | I just moved PBX's, and have the oddest problem. when dialing a meetme conf, it always says invalid conf pin when i type in a pin, even though it exists in meetme.conf, the gui says it parses meetme.conf, and the only pin in meetme.conf is the pin i specified. thoughts? |
| 23:45.37 | p3nguin | Need more info. |
| 23:45.46 | p3nguin | Show the dial plan and the meetme.conf. |
| 23:45.57 | citywok | http://pastebin.com/UrrjsNYd |
| 23:46.02 | citywok | was just getting to that heh |
| 23:46.06 | slav3_kitten | i thought meetme was depreciated for confbridge in the new versions |
| 23:46.14 | citywok | oh, meetme(,ipscM) is the dialplan |
| 23:46.22 | p3nguin | depreciated means it lost monetary value. |
| 23:46.28 | citywok | lol |
| 23:47.27 | slav3_kitten | it means it's diminished in value over time. money is not implied |
| 23:48.18 | citywok | p3nguin: any ideas? it's... freakin odd. i cut out every meetme pin for testing. this is a copy/paste of all the configs from the old PBX. |
| 23:48.37 | citywok | i cranked up debugging but it isn't saying anything |
| 23:48.50 | p3nguin | Nevertheless, you probably meant deprecated. |
| 23:49.41 | slav3_kitten | i entirely did, but that's besides the point |
| 23:51.00 | slav3_kitten | deprecated works much better in that |
| 23:51.30 | slav3_kitten | it's only one letter different too |