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00:27.47 | newtonr | have a good weekend all |
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03:36.18 | Joel_re | hey, so I have this context within which I have an extensions defined |
03:36.31 | Joel_re | dialplan show 200@context |
03:36.39 | Joel_re | lists the extension correctly |
03:36.53 | Joel_re | but when I dial in, its just stuck |
03:36.58 | Joel_re | and then hangs up |
03:37.08 | Joel_re | Im using sip, and the phone is registered |
03:37.22 | Joel_re | is there anymore debugging output I can get from asterisk? |
03:38.13 | Penguin | core set verbose 3 |
03:38.16 | Penguin | sip set debug on |
03:38.25 | Penguin | Go make the call. Pastebin the output when done. |
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05:20.33 | Joel_re | hey, http://pastebin.com/GZshXAzk |
05:20.51 | Joel_re | I have the sip account configured and phone registered |
05:21.05 | Joel_re | the calls should drop into the 'joel' content |
05:21.08 | Joel_re | context |
05:21.13 | Joel_re | but nothing happens :\ |
05:21.36 | Joel_re | I dont get any errors in asterisk, the sip phone just disconnects |
05:21.51 | Joel_re | can't figure what Im doing wrong |
05:22.16 | Penguin | (2138.12) <Penguin> core set verbose 3 |
05:22.16 | Penguin | (2138.15) <Penguin> sip set debug on |
05:22.18 | Penguin | (2138.25) <Penguin> Go make the call. Pastebin the output when done. |
05:22.21 | Penguin | Did you do that? |
05:25.11 | Joel_re | oh, I think I missed that |
05:28.51 | Joel_re | hrm I wonder if ekiga is buggy |
05:29.03 | Joel_re | I dont see any sip output after registration |
05:29.06 | Joel_re | when I dial |
05:34.21 | Joel_re | hrm, what could it be |
05:34.44 | Joel_re | I dont see any sip debug output once the phone is registered |
05:35.52 | Joel_re | http://pastebin.com/er9m8H01 |
05:35.54 | Joel_re | is all I have |
05:41.12 | Penguin | You put ekiga and asterisk on the same computer? |
05:41.56 | Penguin | Registration has nothing to do with making a call from the phone to asterisk. |
05:42.29 | Joel_re | Penguin: yes |
05:42.36 | Penguin | Don't. |
05:42.38 | Joel_re | Im testing on my laptop |
05:42.39 | Joel_re | hrm |
05:42.41 | Joel_re | why |
05:42.57 | Joel_re | ok, well I'll get lxc or something running |
05:43.26 | Penguin | SIP devices are both clients and servers. The phone operates on the same port as asterisk, and since you put them on the same computer, they won't work together. |
05:43.56 | Penguin | Unless you change the port in the phone... which is much easier to do in phones that are not ekiga. |
05:47.31 | Joel_re | oh ok |
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09:10.49 | Joel_re | hey, anyone familiar with adhearsion around? |
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09:30.40 | Joel_re | Im trying to have a dialplan in which multiple users dial in and dtmf keys are accepted without a timeout |
09:30.43 | Joel_re | is that possible |
09:31.04 | Joel_re | for a game, so input keys are accepted until they hangup |
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13:20.37 | fornax | Hi, maybe someone can help me. I have a dahdi setup with a beronet quadbri card und had to switch my broken main board. Now everything works fine but the sound quality is very bad in one way. I can understand the other person without problems, but he / she only hears choppy voice |
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13:48.39 | file | la la la |
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14:36.28 | Martin` | my asterisk is broken, "sip show peers" no longer a command |
14:37.14 | Martin` | hmm was still starting? |
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14:43.15 | Martin` | hmm after reboot everything is working fine again. stupid vps provider :P |
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14:58.50 | ghost75 | a non working command nothing has to do with isp |
15:01.40 | ghost75 | the rtcp channel variables seem to be completely f00ked in 1.8 |
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15:07.21 | Martin` | ghost75: it was al responding slow. sip channel took some minutes before it was loaded |
15:08.07 | ghost75 | a non working command nothing has to do with isp |
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16:13.00 | Penguin | ghost75: Actually it does. If chan_sip isn't loaded, sip show peers won't work. |
16:13.42 | WIMPy | fully loaded |
16:13.53 | ghost75 | why it wouldnt be loaded |
16:14.08 | WIMPy | DNS trouble |
16:14.27 | ghost75 | modules are always loaded |
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16:14.54 | Penguin | Whatever their system is doing could easily prevent modules from loading quickly. |
16:15.58 | ghost75 | but this has nothing to do with isp when a command doesnt work |
16:16.11 | Penguin | But I just explained why it does. |
16:16.20 | WIMPy | Loading modules can take very long if something goes wrong, like e.g. DNS. |
16:16.36 | Penguin | If a module is not loaded, the command does not work. If their crap is broken, modules don't load or can take a long time to load. |
16:17.12 | ghost75 | why is * doing thing |
16:17.16 | ghost75 | this |
16:17.33 | Penguin | The modules provide the commands. |
16:18.05 | ghost75 | makes no sense, the modules are stored locally, why it should take longer to load a module when the isp cannot be reached |
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16:27.45 | ghost75 | why i cant scan my sip port from outside with port forwarding enabled on router |
16:28.27 | Penguin | You're listening on UDP but scanning TCP. |
16:28.35 | Penguin | That's the most common reason. |
16:29.06 | ghost75 | hmmm |
16:30.10 | ghost75 | i remember i run both tcp and udp in lan, thats why i saw it |
16:30.26 | ghost75 | cant find any android scanner with udp |
16:31.55 | Penguin | Why do you need to scan it? You know the service is available IF you configured asterisk and IF you forwarded the port. |
16:32.13 | Penguin | Scanning is used to discover things. |
16:32.13 | ghost75 | because i want to :) |
16:34.39 | ghost75 | is not always neccessary to forward port |
16:35.18 | Penguin | If you don't forward the port, things on the outside won't get in to it. |
16:35.39 | Penguin | That's the beauty of NAT. |
16:35.46 | ghost75 | calls from isp still get in |
16:46.23 | sjs205 | hello all, is there a command that I can use to print a variable that used in my dialplan? |
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16:46.42 | WIMPy | Verbose |
16:47.18 | sjs205 | Cheers WIMPy |
16:58.04 | ghost75 | somebody uses the rtcp values stored in channel variable? |
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17:27.50 | khannz | Hello, community. May I have your attention on my very basic issue? I got Arch linux box with asterisk from AUR and everything seems fine, but I can't start it from non-root user. I've changed asterisk.conf to activate runuser and rungroup and pointed it to user & group asterisk, which was created during asterisk install. Now when I'm trying to run simple 'asterisk -cvvv' under my non-root user I got permission denied |
17:27.50 | khannz | <PROTECTED> |
17:30.36 | [TK]D-Fender | probably |
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17:30.41 | [TK]D-Fender | we'd have to see |
17:30.43 | ghost75 | ur user is in group asterisk? |
17:31.11 | [TK]D-Fender | go check all of *'s files |
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17:33.20 | igustin | permissions on directories? |
17:34.49 | [TK]D-Fender | and the files |
17:38.11 | war9407 | [TK]D-Fender: is there a best way to filter silence out of voice mails before asterisk processes them? (I get a lot of spammers who call and then hang up (or stay on and say nothing) -- how best to deal with this? |
17:38.47 | war9407 | minsecs = N doesn't seem to help for the silence issue |
17:39.00 | [TK]D-Fender | war9407: When is "before asterisk processes them"? |
17:39.26 | war9407 | [TK]D-Fender: trying to avoid false positives on voice mail (silence, or when they hang up and don't say anything) |
17:39.59 | [TK]D-Fender | can only find out after the file is done. |
17:40.08 | [TK]D-Fender | And then clean up after it |
17:40.21 | ghost75 | blacklist them |
17:40.27 | war9407 | I do :) |
17:41.03 | war9407 | [TK]D-Fender: ok, so aside from minsecs not too much else can be done settings wise |
17:41.26 | ghost75 | i use a website to check reputation of caller before it gets to ring |
17:42.15 | war9407 | that works too |
17:43.41 | ghost75 | then they get into my honeypot and can talk with * |
17:44.22 | ghost75 | or forward them to lenny |
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18:00.23 | ghost75 | why netstat doesnt show the udp connection to my isp |
18:00.34 | file | udp is connectionless. |
18:01.30 | ghost75 | an iptables rule with -m state --state NEW will not work then? |
18:02.09 | WIMPy | It will work if you have nf_conntrack_sip loaded. |
18:02.13 | [TK]D-Fender | not for UDP it won't |
18:02.32 | WIMPy | Err, no. Yu only need that for RTP. |
18:02.44 | WIMPy | It will always work. |
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18:25.13 | ghost75 | i now created input and output rules for 5060 udp to my isp |
18:27.13 | sjs205 | I keep getting "handle_request_invite: Sending fake auth rejection for device 1002<sip:1002@X.X.X.X>" withc is good, but the IP address is the server addres... is this anything to worry about? |
18:50.21 | ChannelZ | Well possibly |
18:51.05 | WIMPy | No, just a bad message. |
18:51.11 | ChannelZ | That's someone trying to register to you, probably someone you don't want |
18:51.39 | WIMPy | Yes. Someone always will. |
18:54.50 | [TK]D-Fender | That is just showing who they are trying to auth as which is perfectly normal |
18:55.11 | [TK]D-Fender | fake rejecting is just a basic auth challenge |
19:00.51 | sjs205 | Yeah, I guessed that it was someone pretending to be the server... |
19:01.09 | [TK]D-Fender | no. |
19:01.12 | sjs205 | I've locked it right down since my episode of a week ago... |
19:01.20 | sjs205 | [TK]D-Fender, no? |
19:01.52 | [TK]D-Fender | Do you have a device named 1002 in your sip.conf? |
19:11.18 | ghost75 | episode? |
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19:14.37 | ChannelZ | You know, the one where Ross gets his VoIP system hacked and Rachel dumps him over it |
19:16.33 | ghost75 | and then he gets a huge bill |
19:17.12 | sjs205 | [TK]D-Fender, no device named 1002, but i do have a extension 101, which is another extensdion they keep trying... |
19:17.33 | [TK]D-Fender | then that might be a reason to suspect them. |
19:17.50 | sjs205 | ChannelZ, ghost75, episode whereby my server was hacked and caused the company to loose phones for 2 days... oh, and £60... |
19:18.09 | [TK]D-Fender | Because that is a normal challenge to regular devices registering legitimately to avoid discovery |
19:18.22 | [TK]D-Fender | Do not jsut take seeing that message alone as it being a "hacker" |
19:18.23 | ghost75 | 60 not too bad, i read some had couple of 1000 |
19:19.18 | sjs205 | ghost75, yeah, I probably wouldn't have a job anymore if it was more... although in my defence, I'n no PBX engineer... |
19:19.35 | ghost75 | yes if device has wrong password set, you should also see such messages |
19:19.52 | sjs205 | [TK]D-Fender, good advice, but I still don't know why device 1002 is trying... |
19:20.21 | [TK]D-Fender | sjs205: You should look at the actual SIP DEBUG |
19:20.39 | WIMPy | Because they haven't found the right password, yet. |
19:20.44 | sjs205 | [TK]D-Fender, good shout... I've switch it on... watch this space... |
19:21.03 | [TK]D-Fender | sjs205: Not looking at debug an asking if it's a hack attempt makes no sense. |
19:21.53 | sjs205 | [TK]D-Fender, fair one... apologies, I just assumed it was still someone trying to get in... watching and waiting now.... |
19:22.13 | [TK]D-Fender | sjs205: "assume" is a mistake. |
19:22.27 | sjs205 | Ha... well it defo originates from an unknown IP... |
19:22.45 | ghost75 | i just did http://pastebin.com/MQa0Qv0i |
19:28.38 | ghost75 | with old cisco phones, the sip user/pass can be spyed easily |
19:28.50 | sjs205 | [TK]D-Fender, here is the pastebin if you are interested... |
19:28.55 | sjs205 | http://pastebin.com/sTCbSvYB |
19:29.31 | [TK]D-Fender | So go firewall them out |
19:30.06 | sjs205 | [TK]D-Fender, should my fail2ban be doing this? |
19:30.23 | ghost75 | it should |
19:30.42 | ghost75 | but obviously it doesnt |
19:31.45 | sjs205 | ghost75, the funny thing is that fail2ban seems to have set up a rule banning the server's IP address... I guess that is related to this... |
19:32.22 | ghost75 | 209.222.16.18 is you? |
19:33.48 | sjs205 | ghost75, nope... |
19:33.54 | ghost75 | never used fail2ban but doesnt it have any whitelists |
19:34.18 | sjs205 | that is the one that it seems to be mascarading as my server... unl;ess i've got it wrondg... |
19:34.41 | ghost75 | is a scammer |
19:34.56 | ghost75 | but what i read, fail2ban should keep it out * with iptable rule |
19:34.58 | sjs205 | indeed... the server is based in america somewhere... bastards! |
19:35.17 | ghost75 | so you shouldnt get so much messages |
19:35.30 | sjs205 | yeah, that is what i thought, I've got the fail2ban installed just as it says on voip-info |
19:36.33 | [TK]D-Fender | sjs205that is the one that it seems to be mascarading as my server... unl;ess i've got it wrondg... <- you've got it wrong. |
19:36.50 | sjs205 | [TK]D-Fender, how so? |
19:37.08 | [TK]D-Fender | sjs205: they are identifying as a user ON your server. Not AS your server |
19:37.28 | sjs205 | oh, so fail2ban really should be banning them... |
19:37.34 | [TK]D-Fender | Where they are coming from has nothing to do with who they identify as |
19:37.45 | sjs205 | funny how the only thing it seems to have banned is my servers ip address,,, |
19:37.53 | sjs205 | sure [TK]D-Fender |
19:38.54 | ghost75 | do you use voip accounts from isp ? |
19:39.18 | sjs205 | ghost75, yes, one account for outgoing and 1 sip trunk... |
19:39.34 | ghost75 | are all phones inside lan? |
19:40.26 | sjs205 | ghost75, there aren;'t really any phones, other than the ones that I have set up on my home network,. the server is running in our offices... no phones there at all... |
19:41.05 | ghost75 | if you know all ip addresses allowed to connect you could just drop everything else with iptbales |
19:42.10 | sjs205 | ghost75, this is a good point... since I'll only ever be dialing out with this server |
19:42.25 | sjs205 | and all calls will go through the trunk... |
19:42.49 | ghost75 | but its neccessary that ip from isp doesnt change |
19:44.02 | sjs205 | ghost75, yeah, this i'm not sure about, I've just registered two ips in my sip.conf, and this always works... but probably a good idea to check with gradweelll |
19:44.36 | ghost75 | fail2ban should work as well i guess |
19:45.10 | sjs205 | ghost75, yeah, it should be :s |
19:45.15 | sjs205 | I'll check it out... |
19:49.07 | sjs205 | cheers guys |
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21:03.44 | ghost75 | i finally managed to fake the sip header for outbound calls |
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21:34.52 | m0sphere | i'm trying to change the default voice for phpagi's text2wav by editing the voices.scm file for festival, but i can't seem to get it figured out, does anyone have experience with this? |
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21:38.37 | danfromuk | Hi, I'm setting CALLERID(clid), and then doing ForkCDR. It appears that after forkcdr, the CALLERID(clid) is reset to the original details. Any idea how I can resolve this? |
21:38.45 | danfromuk | Is CALLERID(clid) readonly? |
21:44.28 | file | clid is not a valid option to the CALLERID dialplan function |
21:44.36 | WIMPy | clid is not a valid parameter. |
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21:50.32 | ghost75 | when in sip.conf faxdetect=yes then it should go to exten fax on incoming t.38 fax right? |
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22:16.43 | m0sphere | do i have to restart asterisk after changing the voices.scm or festival.scm? |
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23:10.16 | ghost75 | is the udptl port range so large because of a previous bug? |
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23:28.12 | danfromuk | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR says that I can add my own fields to CDR(fieldname). Do I need to set this up somewhere, because it doesnt seem to log the new field. |
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23:39.14 | danfromuk | Sorry, i timed out. If there were any replies, please repost. |
23:39.45 | WIMPy | Eat. Sleep. Rave. Repeat. |
23:40.07 | WIMPy | Have you taken a look at cdr*.conf? |
23:42.23 | danfromuk | Yes, nothing seems to be documented in the sample files. |
23:42.30 | danfromuk | I think its something to do with alias => |
23:44.30 | WIMPy | Don;'t know what backend you use, but at least cdr_custom.conf and cdr_syslog.conf have format specifiers. |
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23:44.48 | danfromuk | I use cdr_csv and odbc |
23:46.10 | danfromuk | Actually, ive found something on google that details adaptive odbc. I'll switch to it and give it a try. |
23:51.54 | danfromuk | Didnt seem to help. Although the sample conf file seems to say that all that is requires is the column needs to exists. |
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