| 00:16.49 | ChannelZ | I just couldn't remember |
| 00:17.49 | ChannelZ | Though 'config' could stand to be renamed to something a bit more logical |
| 00:27.12 | igcewieling | I always hated the "config" name |
| 00:30.30 | [TK]D-Fender | "make init" <- much clearer |
| 00:35.24 | igcewieling | *nod* |
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| 00:43.42 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 04:49.46 | dijib | SeRi: |
| 04:56.43 | *** join/#asterisk aaotech (~aaotech@75.95.13.174) |
| 04:58.35 | aaotech | Hey asterisk guys! How can I create a variable with a variable? TRUNK=Gtalk/google/${EXTEN}@voice.google.com doesn't work. |
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| 05:08.47 | ectospasm | aaotech: that should work, are you using it in a Set() application call? |
| 05:09.03 | aaotech | trying to set it as a global :p |
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| 05:21.42 | al_nz1 | evening all. Asterisk newbie here. Looking to install it at home |
| 05:22.05 | al_nz1 | evening all. Asterisk newbie here. Looking to install it at home) is an option? |
| 05:26.50 | Weezey | sure |
| 05:27.06 | al_nz1 | Hi Weezey |
| 05:27.45 | al_nz1 | I am reading docs and watching videos now. I want to connect to my 3rd party internet based SIP provider....but this looks like a lot of learning! |
| 05:28.14 | al_nz1 | (talking just one line for hoe phone use) |
| 05:28.20 | Weezey | yup yup, lots to learn |
| 05:28.40 | Weezey | my hoes have their own DIDs |
| 05:28.52 | al_nz1 | hoes? lol |
| 05:31.10 | al_nz1 | any suggestions for a pre brewed installation? the problem it seems is too much information at the moment |
| 05:31.24 | al_nz1 | finding the right info is hard! |
| 05:31.51 | ectospasm | aaotech: yeah, ${EXTEN} doesn't have a referent in a global declaration... |
| 05:32.32 | ectospasm | ~tb |
| 05:32.32 | infobot | well, tb is TeraByte! |
| 05:32.35 | ectospasm | er |
| 05:32.39 | ectospasm | ~thebook |
| 05:32.39 | 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 |
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| 07:50.43 | [TK]D-Fender | aaotech: "core show function EVAL" <- |
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| 10:27.11 | danfromuk | Has anyone set up asterisk with openvpn? Do you simply give the asterisk box a private ip address for the openvpn users to connect to? |
| 10:27.35 | WIMPy | yes and yes |
| 10:30.14 | danfromuk | Ok, thanks. I'll give it a go. |
| 10:30.24 | danfromuk | Trying to get round some router and isp issues. |
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| 12:53.59 | vlad_starkov | Question: Could anyone recommend SIP client with TLS+SRTP support for OS X? |
| 12:56.43 | nullxff | Hello, I have been "hacked" (Asterisk 1.6.2.9) I have the full logs. Somebody managed to call International Premium Numbers (Sierra Leone). Offering $50 to anybody who can tell how it was done exactly and how to prevent it |
| 12:58.07 | brabo | nullxff: 1.6 is vuln afaik, someone can get a remote shell and fiddle with stuff |
| 12:58.19 | brabo | http://downloads.asterisk.org/pub/security/AST-2012-014.html |
| 12:58.34 | brabo | oh i see 1.6 isnt in there |
| 12:58.38 | brabo | so it may be something else |
| 12:59.12 | nullxff | It is 1.6X |
| 12:59.15 | nullxff | YEs |
| 12:59.47 | brabo | not really good with asterisk, just here to see the chatter hehe |
| 13:00.02 | brabo | atm i'm trying to exploit the vuln version 11.0.0 :p |
| 13:00.22 | nullxff | It is just a small hobby box, but I want to know How |
| 13:00.42 | brabo | if it was a vuln 1.8 10 or 11 i could tell you how |
| 13:01.00 | brabo | broad lines anyway, i still need to find 1 thing in order to do a remote exploit |
| 13:02.14 | nullxff | I think maybe it is a misconfigured dial plan or something |
| 13:02.24 | brabo | but since 1.6 isnt included in the security advisory of this particular exploit, it most likely will be different |
| 13:02.43 | brabo | nullxff: gimme sec, lemme look up if your version contains a vuln |
| 13:02.47 | nullxff | But it hard to tell from the logs, at least for me |
| 13:03.16 | nullxff | It contains some |
| 13:03.22 | nullxff | http://www.cvedetails.com/version/125574/Asterisk-Open-Source-1.6.2.9.html |
| 13:03.44 | brabo | http://www.cvedetails.com/cve/CVE-2012-2414/ |
| 13:03.46 | brabo | rce |
| 13:03.58 | brabo | that would do to get in, and mess with everything |
| 13:04.20 | nullxff | I think that the mannager is disabled |
| 13:04.26 | brabo | but it's only for remote authed users |
| 13:04.37 | brabo | mkay, then i have no idea ;) |
| 13:04.54 | brabo | but as i said, i'm totally not an asterisk wizz, unlike i guess a lot of other ppl here hehe |
| 13:05.06 | brabo | i just like to play with vulnerablities :) |
| 13:05.14 | nullxff | K |
| 13:06.45 | brabo | so sorry, i cant really be of any use ;) |
| 13:07.00 | nullxff | Anyway, if somebody can help me for $50 contact me : info at 0xff org |
| 13:07.51 | brabo | nullxff: best stick around for a bit |
| 13:07.59 | nullxff | K |
| 13:08.06 | brabo | this s irc, someone will see your question and reply sooner or later ;) |
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| 13:43.29 | elmargol | Hi got a SPA3102 can someone please point me to a nice howto forward incoming calls to asterisk? |
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| 14:11.56 | elmargol | is there a gui or something to config asterisk? |
| 14:13.54 | [TK]D-Fender | elmargol: Several, some free, most commercial. Most supported free version is FreePBX |
| 14:13.56 | [TK]D-Fender | ~freepbx |
| 14:13.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:13.58 | [TK]D-Fender | ^ |
| 14:14.10 | elmargol | I have asterisk installed on debian |
| 14:14.14 | elmargol | are there packages for this? |
| 14:14.32 | [TK]D-Fender | elmargol: www.freepbx.org |
| 14:15.03 | [TK]D-Fender | elmargol: Source tarball currently |
| 14:16.40 | elmargol | Or maybe a decent documentation? |
| 14:17.58 | [TK]D-Fender | elmargol: For Asterisk itself? |
| 14:18.00 | [TK]D-Fender | ~book |
| 14:18.00 | 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 |
| 14:18.26 | [TK]D-Fender | elmargol: then http://wiki.asterisk.org |
| 14:19.03 | elmargol | I have configured my SPA3102. It should now forward PSTN calls to my asterisk box |
| 14:19.10 | [TK]D-Fender | elmargol: And 3rd (often outdated on some syntax, but some good bits there) ... |
| 14:19.12 | [TK]D-Fender | ~wikis |
| 14:19.12 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:19.24 | elmargol | The issue I have is that I get wrong password at the asterisk console |
| 14:19.34 | [TK]D-Fender | elmargol: Plenty of guides you can Google up specifically for that |
| 14:19.42 | [TK]D-Fender | ~pb |
| 14:19.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:19.50 | [TK]D-Fender | elmargol: Your friend ^^^ |
| 14:20.15 | [TK]D-Fender | elmargol: * CLI, "core set verbose 10" , "sip set debug on" |
| 14:26.29 | elmargol | now it has status Unmonitored |
| 14:28.21 | [TK]D-Fender | elmargol: means you didn't set "qualify=yes" for the peer. This is not necessarily bad. |
| 14:28.31 | [TK]D-Fender | elmargol: Where is the SPA relative to your server? |
| 14:28.57 | elmargol | I don't get the question |
| 14:29.06 | elmargol | asterisk is running inside a vm |
| 14:29.17 | elmargol | the spa is on my desk :) |
| 14:30.03 | [TK]D-Fender | Local LAN then |
| 14:30.19 | [TK]D-Fender | make sure your VM is running its LAN in BRIDGED mode and not NAT |
| 14:30.56 | [TK]D-Fender | So go to * CLI as I told you, and pastebin up the complete debug of the failed call and show your sip.conf configs for it |
| 14:31.03 | [TK]D-Fender | mask only the secret |
| 14:31.12 | elmargol | I think it works now |
| 14:31.20 | elmargol | there are two peers showing up |
| 14:31.41 | elmargol | sip show peers |
| 14:32.06 | [TK]D-Fender | It now shows you the IP? |
| 14:32.13 | elmargol | yes |
| 14:32.32 | [TK]D-Fender | And yyou did not set it as host=IP in your peer but rather host=dynamic? |
| 14:33.52 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:34.26 | elmargol | yes |
| 14:34.43 | elmargol | can I call a peer from the CLI? |
| 14:34.47 | [TK]D-Fender | elmargol: Good sign then. Go actually use it and see what you get |
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| 14:45.35 | elmargol | [TK]D-Fender, is Spandsp the right solution to receive a fax from SIP? |
| 14:45.47 | elmargol | I'd like to email the received fax |
| 14:53.30 | [TK]D-Fender | Yup |
| 14:53.59 | elmargol | how about iaxmodem and hylafax? |
| 14:54.08 | elmargol | Most guides I can find use iaxmodem |
| 14:54.29 | [TK]D-Fender | elmargol: You should be using Asterisk 8 or higher, and using a provider that supports T.38 |
| 14:54.45 | elmargol | I don't use a provifrt |
| 14:54.54 | elmargol | I have a dedicated fax line |
| 14:55.04 | [TK]D-Fender | Hylafax is for more than just simple reception. That makes for a full-blown fax solution |
| 14:55.26 | [TK]D-Fender | Well you said "from SIP". You mean from your SPA? |
| 14:55.34 | [TK]D-Fender | (the PSTN side) |
| 14:55.35 | elmargol | yes |
| 14:56.01 | elmargol | PSTN -> SPA3102 -> SIP -> Asterisk -> software -> Email is the solution I try to implement |
| 14:56.19 | [TK]D-Fender | elmargol: Make sure to enable T.38 on the SPA & * side |
| 14:56.46 | [TK]D-Fender | Just to receive, SpanDSP is fine. |
| 14:57.16 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:00.54 | elmargol | FAX Enable T38 = Yes |
| 15:02.08 | [TK]D-Fender | there are a few sip.conf parameters for *'s end |
| 15:09.21 | elmargol | to me this all looks very scripted |
| 15:09.29 | elmargol | Most howtos I find use some shellscripts |
| 15:09.42 | [TK]D-Fender | For what? |
| 15:09.54 | elmargol | https://issues.asterisk.org/jira/secure/attachment/36635/Configuring%20Asterisk%20for%20Receiving%20FAX%20to%20Email.txt |
| 15:10.07 | elmargol | receiving a fax and sending it over email |
| 15:10.27 | [TK]D-Fender | That is a single shell script... |
| 15:10.39 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:10.46 | [TK]D-Fender | and you COULD just use a 1-line call to "sendmail" ... but this makes it LOOK nice, etc |
| 15:10.52 | elmargol | yes i'm wondering why such a thing is now built in... |
| 15:11.03 | elmargol | now/not |
| 15:11.26 | [TK]D-Fender | elmargol: because that assumes e-mail |
| 15:11.37 | [TK]D-Fender | The point is that YOU have control over what you do with it.. |
| 15:11.43 | [TK]D-Fender | and this action is Ttiny |
| 15:11.49 | [TK]D-Fender | and nothing to complain about |
| 15:12.12 | [TK]D-Fender | You can use * for whatever you want. I use mine as a jukebox and coffee timer |
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| 15:54.11 | elmargol | exten = fax,1,Gosub(faxreceive,s,1) |
| 15:54.15 | *** part/#asterisk stork (stork@unaffiliated/stork) |
| 15:54.28 | elmargol | does this call faxreceive if extension fax is ringing? |
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| 16:00.16 | [TK]D-Fender | elmargol: this is based on fax detection. |
| 16:00.29 | elmargol | yes |
| 16:00.36 | [TK]D-Fender | elmargol: If this line dedicated to faxing? |
| 16:00.39 | elmargol | I used the example from the link |
| 16:00.49 | [TK]D-Fender | is* |
| 16:00.53 | elmargol | yes is |
| 16:00.57 | elmargol | i have this line in [general] |
| 16:00.58 | [TK]D-Fender | then you hav no need for that. |
| 16:01.07 | [TK]D-Fender | just go direct to ReceiveFax |
| 16:01.16 | [TK]D-Fender | in the exten your SPA calls |
| 16:02.14 | elmargol | do i use a number or a name as exten ? |
| 16:03.16 | [TK]D-Fender | Depends on what you told the SPA to use |
| 16:03.40 | [TK]D-Fender | I might hope whatever guide you may have found suggested using the phone # of the line you connected it to as the number it should pass on |
| 16:04.40 | elmargol | the SPA has diaplan (xx.) |
| 16:04.54 | elmargol | This is how the extension is generated? |
| 16:05.05 | elmargol | sorry I'm totally new to SIP |
| 16:06.11 | *** part/#asterisk tigermatt (~tigermatt@host81-157-115-19.range81-157.btcentralplus.com) |
| 16:06.15 | *** join/#asterisk tigermatt (~tigermatt@host81-157-115-19.range81-157.btcentralplus.com) |
| 16:07.01 | [TK]D-Fender | elmargol: When the call comes in on the PSTN port it has to be told to send the call to *. You'll normally have a dialplan line on that port like (S0:1234567890), etc ot say to "send this to number 1234567890 on the registered server), or some such |
| 16:07.20 | [TK]D-Fender | elmargol: It's more than just SIP, it's more specific knowledge of how the SPA works... |
| 16:07.36 | elmargol | Yes I have to start from 0 :( |
| 16:08.26 | elmargol | Thank you for your help. I take a break and continue tomorrow :) |
| 16:08.35 | [TK]D-Fender | elmargol: Ok, keep at it... |
| 16:08.51 | tigermatt | Am trying to set up call queues using both 'ringall' and dynamic penalty changes using queuerules.conf. The usecase is 4 agents, 2 should be dialled first for 10 seconds, then add in the other 2 after a further 10 seconds. I was trying to avoid using Local channels and wanted to do it using the built-in queue penalties if possible. Added penaltychange => 10,+5 to queuerules.conf and defined this in the queue definitio |
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| 18:32.00 | vlad_starkov | Question: Anyone has successful experience installing A2billing with PostgreSQL support? |
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