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| 00:12.01 | Coburn | Hi guys. Just wondering, does Asterisk transmit VoIP packets encrypted? |
| 00:12.48 | Coburn | For example, I have my Asterisk box at my parent's work, connected to my VoIP service via Internode. I want to have a remote port opened so I take calls from abroad. |
| 00:13.30 | Coburn | I have changed the default Asterisk passwords |
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| 07:33.01 | lmkone | hello all - yesterday i resolved my 'Failed to authenticate on INVITE' error by supplying fromuser parameter in sip.conf, I have moved the configuration to another server and I back to the same error, any ideas on where to look now? http://pastebin.com/7pfXkpna |
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| 07:42.47 | Ast001 | Hello can you point me to some location with good examples of using MeetMe and MeetMeAdmin applications for Asterisk version 1.8. I am trying to find a way how to let admin to mute/unmute/kick other users by pressing numbers on his telephone. |
| 07:49.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:54.25 | Ast001 | From documentation I think MeetMe(1234,aDFx,01) should make conference 1234 dynamicly, allow sending dtmf, leave when last participiant leave and set admin mode which should mean that this user is able to do MeetMeAdmin commands by dialing digits defined in same context like exten => 202,1,MeetMeAdmin(1234,m,02) which should unmute user with pin 02 in conference. Am I right ? |
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| 10:15.10 | XATRIX | Hi, i have tons of such messages, why it doesn't show the actuall IP of the attacker ? |
| 10:15.18 | XATRIX | [2014-08-08 13:06:18] NOTICE[2781][C-000001a8]: chan_sip.c:25682 handle_request_invite: Failed to authenticate device 801<sip:801@176.111.63.xxx>;tag=0687ce5e |
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| 11:36.26 | babak | WIMPy hi, In our coubtry we use Euroisdn and irdinary call to asterisk ove rPRI is ok |
| 11:37.23 | babak | but I want to test ECT service , it seems Asterisk just support ETSI but ECT acording to ITU is Q.952.7 |
| 11:38.12 | babak | usually many of our telecom standards ar aslso ETSI like caller ID... |
| 11:38.45 | babak | I already think ITU and ETSI are similar |
| 11:39.04 | WIMPy | Yes, they are mostly identical. |
| 11:39.08 | babak | but it seems for ECT at least they are different |
| 11:39.34 | babak | do you think it is easy to add itu q952.7 to asterisk ? |
| 11:40.10 | WIMPy | Do you have ECT available on your line at all? Or are you connected to a PBX? Or is it perhaps something else you want to do? What's your scenarion like? |
| 11:40.22 | babak | this is my try https://issues.asterisk.org/jira/browse/PRI-175 |
| 11:40.32 | babak | yes |
| 11:40.56 | babak | i am sure Huawei C&C08 Local exchange support it |
| 11:41.09 | babak | but they support itu version |
| 11:41.59 | WIMPy | brb |
| 11:42.55 | babak | I want to use Asterisk in first of call setup and after that bypass Asterisk and LX connect B and C |
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| 11:51.01 | WIMPy | Ok, read that ticket. |
| 11:51.49 | WIMPy | First for the easy part: CD works AFAIR if you just Dial() the same interface before answering the call. |
| 11:55.13 | babak | I used this some years ago https://issues.asterisk.org/jira/browse/ASTERISK-19708 |
| 11:55.22 | babak | but not successful :( |
| 11:55.42 | babak | for CD |
| 11:56.54 | babak | I will try dial also |
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| 12:07.00 | slackie | hey fellas, compiled dahdi, libpri and asterisk and there is no pri command :-F any tip ? is there a compile sequence? |
| 12:07.17 | slackie | tried to compile dahdi, then libpri and finally asterisk 11.10.2 |
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| 12:19.57 | sekil | can one setup * for b-leg cdr logging? |
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| 12:37.54 | babak | WIMPy,you are sure we can invoke CD by using Dial command befor answer ? here http://lists.digium.com/pipermail/asterisk-users/2012-March/271016.html Richard suggest using DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) |
| 12:40.45 | WIMPy | Sorry, been on the phone... |
| 12:41.24 | WIMPy | This is what a working ECT request from a phone looks like: 00 81 12 0a 08 01 01 62 1c 09 91 a1 06 02 01 07 02 01 06 |
| 12:41.53 | WIMPy | As far as CD goes, I'd have to try. |
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| 12:48.13 | babak | your output for ECT is ETSI 300 369-1 ? |
| 12:49.00 | WIMPy | That should be the ETSI way, yes. |
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| 12:49.46 | WIMPy | And that's also the one dahdi understands in recent versions. |
| 12:50.23 | babak | what is your local exchange model ? |
| 12:51.45 | WIMPy | None. |
| 12:52.28 | WIMPy | And while we had them, ECT wasn't enabled anyway. But that's what PBXs do accept, ans current dahdi versions. |
| 12:53.21 | babak | I mean Siemens EWSD or Alcatel S12 may support it ? |
| 12:53.59 | WIMPy | I'm pretty sure they do. But I don't know anywhere it is enabled due to billing issues. |
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| 12:56.54 | babak | thx for your information |
| 12:58.15 | WIMPy | And using Dial() diesn't seem to do a CD, but DAHDISendCallreroutingFacility works. |
| 12:58.52 | WIMPy | ... on a line connected to an EWSD. |
| 12:59.36 | babak | ok I will try it |
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| 13:41.18 | paolo_ | is there a way to chance the a default value in chan_pjsip for all endpoints ? |
| 13:42.08 | file | you can apply a template if using pjsip.conf |
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| 13:42.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:44.34 | paolo_ | file: thanks. but i can't find an example in pjsip.conf.sample |
| 13:45.03 | file | templates aren't pjsip.conf specific, they apply to all .conf files |
| 13:45.19 | paolo_ | file: ahh ok |
| 13:45.41 | paolo_ | file: thanks again |
| 13:45.57 | XATRIX | Any idea how they do it ? |
| 13:46.10 | XATRIX | [2014-08-08 16:44:55] NOTICE[2781][C-000001f1] chan_sip.c: Failed to authenticate device 1000<sip:1000@176.111.63.187>;tag=89712f34 |
| 13:46.22 | file | paolo_, http://pastebin.com/TCZ343iF |
| 13:46.22 | XATRIX | 176.111.63.187 - it's my server's IP |
| 13:46.47 | file | paolo_, defaults is the template and file inherits from it, so unless you specify those values in file the values from defaults will be used |
| 13:47.31 | [TK]D-Fender | XATRIX: That is normal. They are identifying as being an address at your domain (IP) |
| 13:47.47 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 13:47.54 | [TK]D-Fender | XATRIX: Whena phone you configure registers to your server it also shows your server's IP that way |
| 13:48.07 | [TK]D-Fender | XATRIX: Perfectly standard formatting. |
| 13:48.12 | paolo_ | file: i used templates in sip.conf before. anyway thank you for the link |
| 13:48.37 | XATRIX | [TK]D-Fender: They are hackers. They are not my users |
| 13:48.46 | XATRIX | How can i restrict such connections ? |
| 13:49.08 | XATRIX | I don't have such extension. They do bruteforce |
| 13:49.38 | [TK]D-Fender | go firewall them |
| 13:49.53 | XATRIX | How ???? I have to firewall my own IP addr ? |
| 13:50.14 | [TK]D-Fender | No, you need to actually look at the packet and see where it's coming from |
| 13:50.42 | DruidZ | XATRIX: You are restricting them. Note that they failed to authenticate. |
| 13:51.08 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 13:51.13 | XATRIX | And how am i supposed to ? fail2ban examine asterisk's log file, which logs my own server's IP as an incoming connection |
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| 13:51.38 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.44 | [TK]D-Fender | You should be running * 10 or higher to get proper security logging for things like that |
| 13:51.50 | XATRIX | I understand that the L3 packet has the different IP. But i can't make fail2ban to use tcpdump or whatever |
| 13:51.56 | DruidZ | They are hoping that you were stupid enough to put a live server on the net without deleting the default entries or at least changing the passwords. |
| 13:52.14 | XATRIX | I'm using asterisk 11.11.0 |
| 13:52.26 | [TK]D-Fender | Then you'll have to configure your logging properly |
| 13:52.32 | [TK]D-Fender | And fail2ban |
| 13:53.35 | XATRIX | Still can't get it. |
| 13:53.44 | XATRIX | How should i configure logging ? |
| 13:54.35 | XATRIX | fail2ban is only able to read iterated entries from the log file. |
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| 13:55.53 | [TK]D-Fender | yes, and there may be MULTIPLE files and you're not using the right one. |
| 13:55.56 | [TK]D-Fender | logger.conf <- |
| 13:55.57 | DruidZ | XATRIX: Are you (or anyone else running 11.11.0) seeing the problem that I posted about in http://lists.digium.com/pipermail/asterisk-users/2014-August/284067.html? |
| 13:56.45 | XATRIX | DruidZ: nope, i didn't |
| 13:58.08 | [TK]D-Fender | duiGo replicate the problem and show the actual debug for the call... |
| 13:58.14 | [TK]D-Fender | DruidZ: Go replicate the problem and show the actual debug for the call... |
| 13:58.29 | [TK]D-Fender | DruidZ: That post doesn't actually show anything useful for it |
| 14:00.27 | XATRIX | Stiil can't fix it |
| 14:00.42 | XATRIX | I added alwaysauthreject=yes - but it doesn't solve this crap |
| 14:01.33 | [TK]D-Fender | that isn't logger.conf .... |
| 14:02.04 | XATRIX | Yeap, but i suppose it to give him the same error every time it tries to |
| 14:02.22 | XATRIX | So , it's gonna be a bit hard to guess the correct extension ID |
| 14:02.36 | [TK]D-Fender | You are not looking in the right place/ |
| 14:03.19 | XATRIX | Ok, let me show you the contents of |
| 14:03.31 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aumqvlmzifbmatbz) |
| 14:03.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:06.18 | DruidZ | [Aug 6 10:28:09] WARNING[-1] chan_sip.c: Autodestruct on dialog 'b7b83535-c8359d23@192.168.207.104' with owner SIP/4164251212-0000000d in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 14:06.39 | DruidZ | I got 1450 of those for that number alone. |
| 14:07.17 | DruidZ | Actually 4435 yesterday. |
| 14:07.58 | DruidZ | I had to revert to 11.10.2 so I can't test any more. I will try to reproduce it on a test server. |
| 14:08.22 | [TK]D-Fender | We need to see actual debug for a full call. |
| 14:11.09 | *** join/#asterisk Quastor (d44700fe@gateway/web/freenode/ip.212.71.0.254) |
| 14:12.44 | Quastor | Hi, anybody knows why I can't load digium's fax module : loader.c:423 load_dynamic_module: Error loading module 'res_fax.so': /usr/lib64/asterisk/modules/res_fax.so: undefined symbol: ast_verb_sys_level |
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| 14:13.05 | Quastor | <PROTECTED> |
| 14:13.16 | Qwell | Quastor: Did you install it for the right version of Asterisk? |
| 14:14.02 | Quastor | Qwell: We always use the same image, but I'll check with benchfax |
| 14:14.15 | Qwell | benchfax has nothing to do with anything |
| 14:14.50 | Quastor | it gives you the right version of the fax you can install no? |
| 14:15.00 | Qwell | no |
| 14:15.32 | Qwell | http://my.digium.com/en/docs/FAX/faa-download/ |
| 14:15.46 | Qwell | You can use benchfax to get the right flavor |
| 14:17.24 | Quastor | Ok so not the right version, sorry for bothering you, Qwell, Tnx! |
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| 14:44.07 | paolo_ | can qop be disabled in pjsip for authorization. do all sip clients support that ? i'am not so experienced with that kind of authorization.... |
| 14:47.40 | Stefan27 | i noticed there is no "devstate" manager (AMI) action in asterisk 12. is there any drawback in using a lot of AMI-'Command'-actions which calls the CLI-"devstate change x y" (used by a java application to change custom device states in asterisk - is there a better way?) |
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| 14:49.23 | file | Stefan27, use SetVar and DEVICE_STATE() |
| 14:49.28 | file | that's what I did for a test |
| 14:50.11 | Stefan27 | from dialplan or from actions sent to AMI? |
| 14:50.25 | file | AMI |
| 14:50.51 | file | the PJSIP dialog-info+xml test uses that to transition a custom device state through the various states, and then look at the resulting SIP NOTIFY |
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| 14:51.47 | Stefan27 | what's that: "PJSIP dialog-info+xml test" |
| 14:52.06 | Stefan27 | setvar seems to be a manager command, but device_state is a dialplan func? |
| 14:52.06 | file | it's a test in the test suite that runs against Asterisk which is used to confirm functionality and make sure we don't break things |
| 14:52.09 | [TK]D-Fender | [10:49]fileStefan27, use SetVar and DEVICE_STATE() <- that does require a target channel though, no? |
| 14:52.19 | file | [TK]D-Fender, it doesn't |
| 14:52.24 | [TK]D-Fender | O |
| 14:52.42 | Stefan27 | where can i locate that test suite? |
| 14:52.50 | file | Stefan27, SetVar allows you to set variables or execute dialplan functions |
| 14:53.05 | Stefan27 | aha |
| 14:54.36 | file | without a channel it will set global variables and invoke dialplan functions, but not give them a channel to act on |
| 14:54.45 | file | so if you try to invoke something that requires a channel it won't work |
| 14:54.55 | Stefan27 | corresponding to Set(DEVICE_STATE(Custom:mystate1)=BUSY) ? |
| 14:54.57 | [TK]D-Fender | file: Good to know.... |
| 14:55.21 | file | Action: SetVar, Variable: DEVICE_STATE(Custom:mystate1), Value: BUSY |
| 14:55.52 | Stefan27 | right, from here i think ill be ok, ill just telnet test it, thank u |
| 14:55.52 | Nugget | telnet is eeeeeeevil! |
| 14:59.26 | Stefan27 | by the way - file - is the cli solution bad? |
| 15:01.24 | file | I wouldn't say... bad... |
| 15:03.41 | file | I just say what I know works ^_^ |
| 15:05.30 | [TK]D-Fender | Stefan27: I'd bet that the SetVar approach would deal better with escaping, etc and certainly look more natural. Both probably work fine in the short term though |
| 15:07.19 | Stefan27 | thanks |
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| 15:13.05 | DruidZ | So, I have a dial command that calls two extensions, an internal one and an external one. There is a timeout on the dial command that is accurate. |
| 15:13.48 | DruidZ | However there is a five second delay before the next number is called and I can't remember where that value gets set. |
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| 15:14.23 | DruidZ | I know that I wanted that delay and probably set it somewhere but I can't recall where. |
| 15:14.36 | [TK]D-Fender | DruidZ: There is no setting for a delay in Dial() unless you're dialing an analog DAHDI channel with "w"'s |
| 15:14.47 | DruidZ | I would like to make it variable if I can. |
| 15:14.59 | WIMPy | And what "next" number? |
| 15:15.28 | [TK]D-Fender | DruidZ: And dial doesn't do "next". It does "simultaneous" |
| 15:15.43 | DruidZ | Really? The tests are pretty consistient. Once the first extension starts ringing the second one starts five seconds later. |
| 15:16.20 | [TK]D-Fender | DruidZ: Something else may be slowing the process down like a slow DNS lookup, etc |
| 15:16.28 | WIMPy | That's caused elsewhere then. |
| 15:16.29 | [TK]D-Fender | DruidZ: Then again, we have nothing to go on.... |
| 15:23.29 | Katty | we can go on broomsticks! |
| 15:23.31 | Katty | low gas mileage. |
| 15:23.41 | Katty | unless you ate mexican for dinner. |
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| 15:34.26 | paolo_ | in pjsip.conf.sample the is the outbound-auth in the registrations section. but this option belongs to the endpoint section. is the sample wrong ? |
| 15:34.53 | DruidZ | [TK]D-Fender: You're right. The five seconds must be the PSTN provider and/or the cell company consistiently adding the five second delay. |
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| 15:40.22 | DruidZ | It's Freeswitch that has that feature. That's where I remember setting it. |
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| 16:16.18 | darkdrgn2k | any suggestions for decent ip phones for use with asterisk |
| 16:17.51 | [TK]D-Fender | darkdrgn2k: Haven't you already asked this repeatedly over the years? |
| 16:17.57 | [TK]D-Fender | Polycom > all |
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| 16:18.12 | darkdrgn2k | i never remember cause i always cheep out.. im gonna try not to cheep out this time .. |
| 16:18.13 | darkdrgn2k | LOL |
| 16:18.19 | darkdrgn2k | an particular set you can recommend |
| 16:18.34 | [TK]D-Fender | Depends on what you want. |
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| 16:18.39 | [TK]D-Fender | Go stare at their lineup |
| 16:19.27 | darkdrgn2k | rkight now thye are running Avayas (formlay nortals) 1120e |
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| 16:44.20 | darkdrgn2k | <PROTECTED> |
| 16:45.10 | mjordan | yes |
| 16:46.13 | [TK]D-Fender | 1.6.0+ |
| 16:49.36 | darkdrgn2k | when they say "two line phone" that means only 2 programmable lines (2 lnes buttons) not only 2 calls at a time. |
| 16:49.39 | darkdrgn2k | ie you can put calls on hold |
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| 16:54.58 | [TK]D-Fender | IP 335 does it then |
| 16:55.30 | darkdrgn2k | 335 vs 331 is just hd voice |
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| 16:57.44 | [TK]D-Fender | No, is RJ9+EHD headset, plus backlight, plus newer firmware supported, etc |
| 16:58.22 | darkdrgn2k | whats the diff between 321 and 331.... |
| 16:58.39 | darkdrgn2k | oo sinle port |
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| 17:01.18 | [TK]D-Fender | 30X/32X/330/331 ARE ALL DISCONTINUED |
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| 17:48.58 | darkdrgn2k | any one know if the 335s can do PAGE mode? |
| 17:49.25 | [TK]D-Fender | All of their IP phones can |
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| 18:22.38 | Katty | ohhh where is my hair brush?! |
| 18:25.36 | Chainsaw | I swear I haven't used it. |
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| 18:48.44 | Qwell | Nugget: HALP! |
| 18:49.34 | Qwell | Nugget: NEVERMIND! I can't tooltip. |
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| 18:50.31 | Nugget | heh |
| 18:52.42 | Qwell | I was going to ask whether FA times listed were pushback/takeoff, landing/gate arrival |
| 18:53.44 | Nugget | depends on the airline |
| 18:54.06 | Qwell | I finally read the tooltip, clicked and found more detail :D |
| 18:54.16 | Nugget | yay |
| 18:54.21 | Qwell | <SMRT |
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| 19:55.46 | VultureZ | pfft |
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| 20:10.10 | keiths_ | anyone know if on Asterisk or a cisco SPA, a way to have the phone physically ring on call waiting instead of a beep? |
| 20:10.59 | keiths_ | asterisk 1.6 if it matters |
| 20:12.08 | [TK]D-Fender | It's up to the device, not Asterisk |
| 20:12.35 | keiths_ | figured maybe I could do it with a SIP header |
| 20:12.39 | [TK]D-Fender | It may be under their tones setup |
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| 20:27.27 | keiths_ | [TK]D-Fender: It seems this may not be completely up to the handset. Its an otion in a UC enviroment. Call- |
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| 23:23.00 | MycroftMk2 | Good evening all |
| 23:24.48 | MycroftMk2 | I'm looking to see if it is possible to add the IP address of an incoming caller to the CDR. |
| 23:25.11 | MycroftMk2 | I found the following post: http://lists.digium.com/pipermail/asterisk-users/2012-October/275193.html |
| 23:25.34 | MycroftMk2 | but since I've never touched the dialplan before, I was hoping to discuss it with someone here first |
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| 23:36.28 | MycroftMk2 | Bueller |
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| 23:42.12 | [TK]D-Fender | This would have to be added via the dialplan |
| 23:42.32 | [TK]D-Fender | "core show function CDR" <- UserField or some custom value if you can configure it |
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