| 00:13.47 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 00:14.57 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 00:15.15 | Katty | evening |
| 00:15.24 | WIMPy | Good morning |
| 00:15.50 | ChannelZ | good god |
| 00:16.45 | carrar | mooska |
| 00:32.26 | volga629 | patch not really helped |
| 00:32.28 | volga629 | SSL certificate ok |
| 00:32.30 | volga629 | <PROTECTED> |
| 00:32.32 | volga629 | [2013-12-26 19:31:56] WARNING[6704]: tcptls.c:280 handle_tcptls_connection: FILE * open failed! |
| 00:32.58 | volga629 | How to determine which file |
| 00:38.25 | drmessano | What options do you have defined? |
| 00:39.48 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 00:40.43 | drmessano | volga629, you need tlsenable=yes, tlscertfile=FULLPATHTOFILE, tlscafile=FULLPATHTOFILE, and tlsclientmethod=tlvs1 for successful TLS |
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| 02:28.40 | volga629 | All that params I have in may sip.conf |
| 02:28.44 | volga629 | my |
| 02:29.55 | volga629 | I have |
| 02:30.13 | volga629 | tlsclientmethod=tlsv1 |
| 02:30.15 | volga629 | tlscadir=/etc/asterisk/keys |
| 02:30.18 | volga629 | tlscertfile=/etc/asterisk/keys/ott-bundle.pem |
| 02:30.20 | volga629 | tlscafile=/etc/asterisk/keys/ca.pem |
| 02:30.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:30.47 | volga629 | tlscipher=ALL |
| 02:31.20 | volga629 | tlsenable=yes |
| 02:31.29 | volga629 | that all what I have |
| 02:32.34 | volga629 | and this permission on ca file -rw--w---- 1 asterisk asterisk 1753 Dec 21 13:43 ca.pem |
| 02:36.16 | WIMPy | Does it include the certificate? |
| 02:36.51 | volga629 | ott-bundle.pem is private no passwd and cert |
| 02:37.42 | volga629 | the point TLS stops working after I extended ca |
| 02:38.22 | volga629 | it does matter what params I put message is |
| 02:38.24 | volga629 | ERROR[4779]: tcptls.c:228 handle_tcptls_connection: Certificate did not verify: self signed certificate in certificate chain |
| 02:38.28 | WIMPy | So you made a new certificate and that made it stop working? |
| 02:39.39 | volga629 | look like something is not like in ca, but it just assumption. I can't find how to check which file is actually in trouble. |
| 02:40.38 | WIMPy | There's just nothing to beat a sensible error message. |
| 02:41.02 | volga629 | but which file openssl is complain |
| 02:41.59 | volga629 | when verify from cli with openssl it working ok no complains |
| 02:46.11 | *** join/#asterisk Sean-Der (~sean@108-245-5-17.lightspeed.toldoh.sbcglobal.net) |
| 02:47.08 | Sean-Der | Does anyone know how to get the num CID for X channel, or maybe give all channels that are using X num CID |
| 02:47.22 | Sean-Der | I can get all channels with the channels func |
| 02:47.34 | Sean-Der | but I don't think the channel func can give me the CID |
| 02:47.58 | WIMPy | You want it in the dialplan? |
| 02:48.11 | Sean-Der | WIMPy: yep! |
| 02:48.20 | Sean-Der | Oh I didn't know about shared |
| 02:48.48 | Sean-Der | WIMPy: I am trying to get the amount of current channels (in dialplan) that share the same CID |
| 02:48.50 | Penguin | It sounds like you are after the CALLERID function. |
| 02:49.16 | Sean-Der | Penguin: Can I use that for other channels, or just the one I am in |
| 02:49.29 | Sean-Der | I could also call system, then asterisk+grep -c |
| 02:49.30 | WIMPy | What about using the GROUP functions? |
| 02:49.32 | Penguin | CallerID is carried on each channel. |
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| 02:50.35 | Sean-Der | WIMPy: that should work perfectly! |
| 02:50.48 | Sean-Der | so at call start add it to the group by CID num, then do a group count |
| 02:52.27 | Sean-Der | WIMPy: Penguin: thanks for your help guys! It has been a while since I have done this, I have missed it |
| 02:52.39 | Sean-Der | I am still running 1.8 on my box at home |
| 02:52.44 | Sean-Der | WebRTC stuff looks really cool |
| 02:53.07 | volga629 | asterisk trying load old cert SSL error loading cert file. <asterisk.pem> ? |
| 02:53.32 | WIMPy | Old? |
| 02:53.36 | WIMPy | What kind of old? |
| 02:54.06 | volga629 | this cert is expired and not exist |
| 02:54.28 | WIMPy | So where does it come from? |
| 02:55.16 | volga629 | That interesting , I checked sip.conf and all look ok |
| 02:55.33 | volga629 | no kind file |
| 02:55.57 | WIMPy | This may sound too obvious, but did you restart Asterisk? |
| 02:56.13 | volga629 | yes a min ago |
| 02:56.49 | WIMPy | And you're editing the right config file? |
| 02:58.00 | volga629 | yes, tried grep on file name and nothing |
| 02:58.20 | drmessano | Do you have multiple instances of Asterisk running? |
| 02:58.31 | WIMPy | Or multiple installations. |
| 02:58.39 | volga629 | no |
| 02:58.54 | volga629 | one instants I even check PID |
| 02:58.56 | WIMPy | core show settings |
| 02:58.58 | *** join/#asterisk Sean-Der (~sean@108-245-5-17.lightspeed.toldoh.sbcglobal.net) |
| 02:59.11 | drmessano | There may be a default for one of those config options that points to asterisk.pem. Make sure the syntax is correct on the one you defined |
| 02:59.19 | WIMPy | Any #includes #execs or realtime stuff? |
| 02:59.52 | drmessano | Show where it's trying to load asterisk.pem |
| 03:02.58 | volga629 | https://fpaste.networklab.ca/QPLf/ |
| 03:03.13 | volga629 | that config is no asterisk.pem |
| 03:05.11 | WIMPy | Ok, so what if you remove the bad files? |
| 03:05.29 | drmessano | Show where you're seeing it try to load asterisk.pem |
| 03:05.45 | volga629 | on sip reload |
| 03:05.52 | drmessano | Show me |
| 03:05.58 | drmessano | I dont want a play by play |
| 03:07.05 | volga629 | https://fpaste.networklab.ca/XSSG/ |
| 03:07.44 | volga629 | I restarted again with wait of 15 sec and on cli I see SSL certificate ok |
| 03:08.31 | drmessano | Wait a min |
| 03:08.34 | drmessano | This is a FreePBX install. Where do you have these defined? |
| 03:08.59 | drmessano | the tls options |
| 03:09.39 | volga629 | yes I do |
| 03:09.53 | volga629 | I pasted config |
| 03:09.56 | drmessano | ... |
| 03:10.09 | drmessano | In WHICH FILE do you have the TLS options defined? |
| 03:11.10 | volga629 | sip_general_additional.conf |
| 03:11.30 | drmessano | uh |
| 03:12.08 | drmessano | You have them in the GUI, yes? |
| 03:12.28 | volga629 | yes |
| 03:12.56 | drmessano | ok |
| 03:13.38 | drmessano | Something isn't adding up here |
| 03:17.00 | volga629 | this message all the time ERROR[9903]: tcptls.c:228 handle_tcptls_connection: Certificate did not verify: self signed certificate in certificate chain |
| 03:17.18 | volga629 | only if I set verify to yes is go on |
| 03:17.47 | volga629 | even if ca file in config |
| 03:19.10 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 03:19.26 | volga629 | I mean for this option tlsdontverifyserver |
| 03:24.59 | drmessano | volga629, what exactly is in your cert file? |
| 03:25.18 | volga629 | key and cert |
| 03:25.25 | volga629 | key no passwd |
| 03:29.20 | volga629 | Ok I will continue look on it tomorrow, thank you Everyone today help |
| 03:30.05 | drmessano | volga629, still here? |
| 03:30.15 | volga629 | yes |
| 03:30.24 | drmessano | What started the failure again? |
| 03:30.44 | volga629 | I updated expired ca |
| 03:31.26 | drmessano | This is a self-signed cert, right? |
| 03:31.32 | volga629 | yes |
| 03:32.08 | drmessano | Are you just using this for TLS clients or server to server? |
| 03:33.18 | volga629 | yes I have SIP trunk with TLS layer, but second box will get TLS client ( bria 3 ) too |
| 03:33.56 | drmessano | Ok, so this is the CA cert from the other box? |
| 03:34.14 | volga629 | no 2 boxes have own CA |
| 03:34.42 | drmessano | tlscafile=</path/to/certificate> - If the server your connecting to uses a self signed certificate you should have their certificate installed here so the code can verify the authenticity of their certificate. |
| 03:35.07 | drmessano | You have THEIR (other) CA cert to verify THEIR cert |
| 03:35.33 | volga629 | yes, oposite way A ca to B and B ca to A |
| 03:35.46 | drmessano | Ok, I asked you that and you said no |
| 03:36.02 | volga629 | maybe wasn't clear, sorry |
| 03:36.52 | drmessano | the tlscafile=/some/cert on BOX A is the CA cert from BOX B, and vice versa? |
| 03:38.23 | drmessano | There are no other keys in /etc/asterisk/keys that would be problem, like the expired ones? |
| 03:39.03 | drmessano | Because IIRC you need either tlscafile or tlscadir which is either one cert or a directory of certs |
| 03:39.51 | volga629 | no, but I ma going double check again |
| 03:42.39 | volga629 | no everything exactly as described |
| 03:43.02 | drmessano | remove the tlscadir directive and see what happens |
| 03:45.33 | volga629 | ok have leave, I will continue dig around tomorrow, but I am at the point create new set of certs |
| 03:45.50 | volga629 | again thank you for patches and help today |
| 03:46.01 | volga629 | to |
| 03:46.25 | drmessano | ok np |
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| 05:24.06 | drmessano | Thats just weird |
| 05:24.22 | drmessano | SILK was failing every time I tried to load it under Asterisk 11 |
| 05:24.31 | drmessano | Now it's fine |
| 05:25.15 | drmessano | When I say "Every time", I mean with every new version of 11 I tried and it would fail |
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| 07:15.28 | dar123 | guyz i have my sip clients on my home network behind nat, i have a hosted asterisk on the cloud but unable to establish RTP |
| 07:16.01 | dar123 | i believe its caz of nat, any tips |
| 07:16.16 | carrar | ~sipnat |
| 07:16.16 | 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. |
| 07:17.59 | dar123 | thanks carrar |
| 07:18.12 | carrar | that may not help but might be a start |
| 07:18.16 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 07:18.27 | carrar | might be fw rules on your server |
| 07:18.30 | dar123 | one more thing, i keep seeing people attacking my sip server. with attempts to connect, how can i get rid of them |
| 07:18.34 | carrar | might be your home natting device |
| 07:18.56 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 07:18.58 | carrar | block everthing except for the IP's you know of |
| 07:20.09 | carrar | You can be a little more open with the RTP ports, but lock down SIP if you can |
| 07:20.38 | carrar | and of course everything else that isn't required lock down too |
| 07:20.48 | carrar | goes without saying though |
| 07:30.22 | dar123 | thanks again |
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| 08:05.42 | *** mode/#asterisk [+o pabelanger] by pratchett.freenode.net |
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| 08:11.22 | hayden_ru | Good day |
| 08:12.05 | ChannelZ | farts |
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| 08:21.46 | mirela666 | waving hand in front of the nose |
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| 09:14.39 | LiuYan | 'kmod-dahdi-linux-2.8.0.1-1_centos6.2.6.32_279.14.1.el6.x86_64.x86_64.rpm', huh, dahdi-linux build for centos 6.3, not for 6.5? |
| 09:19.44 | *** join/#asterisk hayden_ru (~hayden_ru@80.250.214.230) |
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| 09:55.03 | *** join/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 09:55.07 | DelphiWorld | hi asteriskers |
| 09:55.34 | DelphiWorld | anyone know the dialplan of the SPA? i want my SPA to send (*ALL*) to the server and not only the digits |
| 10:00.20 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
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| 10:01.31 | drmessano | DelphiWorld, yeah |
| 10:01.39 | DelphiWorld | drmessano: help so :P |
| 10:02.14 | drmessano | I would think (.) would do it |
| 10:03.16 | DelphiWorld | drmessano: the dp is in system or what tab of the SPA? |
| 10:03.29 | drmessano | Dialplan is set on each extension |
| 10:03.36 | drmessano | At the bottom for most firmware |
| 10:03.43 | DelphiWorld | i dont see... would i need to be in advanced view? |
| 10:03.48 | drmessano | Probably |
| 10:04.02 | DelphiWorld | lol |
| 10:04.16 | drmessano | Been a long time since i've modded one in the GUI |
| 10:04.23 | drmessano | But that sounds correct |
| 10:04.36 | DelphiWorld | drmessano: i realy loved SPA:P |
| 10:04.40 | DelphiWorld | i missed them for years |
| 10:04.57 | drmessano | Yeah... I still have about 30 PAP2s and a dozen or so SPA941 phones |
| 10:05.28 | DelphiWorld | drmessano: lol, this is SPA 8000 |
| 10:05.52 | drmessano | Nice |
| 10:06.01 | DelphiWorld | drmessano: remotely moding it |
| 10:08.13 | DelphiWorld | drmessano no... (.) gives busy signal on any fucking numbers |
| 10:08.16 | *** join/#asterisk marlinc (~marlinc@ip565fa73c.direct-adsl.nl) |
| 10:08.54 | drmessano | You said you want ALL to go to the server and not just the digits.. What more than the digits? |
| 10:09.17 | DelphiWorld | drmessano: yes but if i dial any *DIGIT* its give busy without going to the server |
| 10:09.19 | DelphiWorld | using (.) |
| 10:09.30 | drmessano | I understand that |
| 10:10.12 | drmessano | But I am asking you, what else other than the digits would go to the server. Why would (X.) not work? |
| 10:10.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 10:10.43 | DelphiWorld | i dont know, even * and # give just busy |
| 10:11.27 | drmessano | Ok, so lets clarify here.. you want NUMERALS + * and #? |
| 10:11.53 | DelphiWorld | offcource |
| 10:11.54 | DelphiWorld | all |
| 10:13.02 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 10:14.02 | drmessano | (x.|*.|#.) should give you any number of numerals, * or * + numerals, and # or # + numerals |
| 10:14.28 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:15.03 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:16.51 | DelphiWorld | lol |
| 10:16.53 | DelphiWorld | i'm trying |
| 10:18.05 | TimeRider | DelphiWorld : hey dude, long time no see |
| 10:18.13 | DelphiWorld | TimeRider: you, not me:P |
| 10:18.18 | DelphiWorld | TimeRider: how are you steve? |
| 10:18.35 | DelphiWorld | TimeRider: lol, let me remember your funy name on phone? :) |
| 10:18.44 | TimeRider | DelphiWorld : I'm good dude :) |
| 10:18.52 | DelphiWorld | lol |
| 10:19.05 | TimeRider | DelphiWorld : http://tube.timerider.co.uk MANY new ones now :) |
| 10:19.25 | DelphiWorld | LOL |
| 10:19.50 | TimeRider | Latest one I call m$ for LINUX support :p |
| 10:20.03 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 10:20.09 | TimeRider | .. and I speak to micro$oft Sam |
| 10:20.15 | DelphiWorld | if someone want to shit telemarketers... pay TimeRider :P |
| 10:20.44 | TimeRider | hehe |
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| 10:58.46 | DelphiWorld | thank drmessano for your help |
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| 11:09.04 | mirela666 | hello, on sip info: Record:on |
| 11:09.19 | mirela666 | i get 403 forbidden |
| 11:09.47 | mirela666 | what am i missing? |
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| 11:54.13 | mirela666 | where are the recordonfeature=automixmon and off |
| 11:54.47 | mirela666 | http://doxygen.asterisk.org/trunk/Config_sip.html |
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| 12:05.31 | DelphiWorld | hahahahahahahahaha! youjelly ! |
| 12:10.50 | mirela666 | damn stuppid mistake |
| 12:11.33 | mirela666 | i enabled automixmon and no automon |
| 12:12.11 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 12:13.08 | mirela666 | :q |
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| 14:11.25 | gusto | so |
| 14:11.28 | gusto | hi |
| 14:11.43 | gusto | does someone have experiences with app_confbridge on asterisk1.8 |
| 14:11.54 | gusto | ha? |
| 14:12.20 | WIMPy | Try in #history or |
| 14:12.22 | WIMPy | ~ask |
| 14:12.22 | 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. |
| 14:15.25 | gusto | ok |
| 14:17.20 | gusto | here my dialplan for confbridge http://pastebin.com/KsxenT4s and here is the thing that it produces http://pastebin.com/6tniBKp8 |
| 14:19.34 | gusto | so, now i do not have any confbridge.conf because there was none and of course, somewhere is written that asterisk 1.8's confbridge does not support any configuration through conf files anyway |
| 14:19.54 | gusto | so everything needs to be done through the dialplan, i understand |
| 14:20.36 | gusto | now here https://wiki.asterisk.org/wiki/display/AST/Application_ConfBridge is the man page for confbridge on asterisk1.8 and that is not very informative, sounds like it should work out of the box, but apparently it does not |
| 14:21.26 | WIMPy | Yes, the old ConfBridge was very limited. |
| 14:21.33 | gusto | i know |
| 14:21.55 | gusto | but i do not care about limitations like ... user management and so on |
| 14:22.09 | gusto | i just wanted a trivial conference |
| 14:22.43 | gusto | my telephone can do it ... eh ... acutally it's not the telephone, it's the telephone adapter (SPA112), but somehow i wanted to have that done by asterisk |
| 14:22.51 | gusto | that would save some traffic |
| 14:23.10 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 14:23.51 | gusto | and ... also i notice that he wants to create that confbridge first, that may not be a good idea, i think it would be more productive to have that already running |
| 14:24.12 | gusto | in worst case one would just need to activate and deactivate it by the dialplan |
| 14:24.24 | gusto | something like tunron and turnoff number |
| 14:24.44 | gusto | but, that should not be a problem, first we are not able to do any conferencing yet |
| 14:29.40 | *** join/#asterisk hayden_ru (~hayden_ru@80.250.214.230) |
| 14:29.59 | gusto | btw. i am in slovakia no |
| 14:30.00 | gusto | now |
| 14:30.32 | gusto | and my 120cm satellite dish witheld the wind in the meantime while i was away from here |
| 14:30.54 | hayden_ru | Can I set a fallback pattern somehow in extensions.conf, so if i don't hit any patterns I call through a set trunk? |
| 14:31.23 | gusto | you can do if |
| 14:31.36 | WIMPy | Something like _X. or i. |
| 14:33.45 | hayden_ru | say i have 1-800-XXXXXXX, 1-734-XXXXXXX and i don't hit them, how do i fallback to 1-XXX-XXXXXX to trunk |
| 14:34.07 | *** join/#asterisk johnmilton (~johnmilto@ool-44c5f03d.dyn.optonline.net) |
| 14:34.16 | WIMPy | _1XXXXXXXXX |
| 14:34.40 | hayden_ru | but then all calls would go through 1XXXXXXXXX pattern |
| 14:34.49 | WIMPy | No |
| 14:34.58 | WIMPy | That's not the way the matching works. |
| 14:35.17 | resist0r | www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 14:36.13 | WIMPy | To make it short it will do a most specific matching. |
| 14:39.53 | hayden_ru | oh okay, got it, so it's like IP routing |
| 14:40.45 | WIMPy | A little. |
| 14:41.03 | gusto | ??? |
| 14:41.04 | gusto | ah |
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| 14:45.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:50.16 | gusto | so |
| 14:50.22 | gusto | my problem seems to be in AST_BRIDGE_CAPABILITY_MULTIMIX |
| 14:50.46 | gusto | he refuses to answer it because he says that there is no way how to do audio muxing |
| 14:55.55 | gusto | ok |
| 14:55.57 | gusto | i got it |
| 14:56.07 | gusto | but my pulse crashed |
| 14:56.11 | gusto | always something ... |
| 14:57.31 | mjordan | huh |
| 14:57.50 | mjordan | If only I were paying attention a minute or two ago. |
| 14:58.05 | mjordan | (How did you even get AST_BRIDGE_CAPABILITY_MULTIMIX to do anything?) |
| 15:00.04 | resist0r | I am curious to know more about the issue as well |
| 15:00.50 | mjordan | I'm not really sure what the issue is. |
| 15:00.59 | resist0r | Right |
| 15:01.24 | mjordan | But if you're seeing AST_BRIDGE_CAPABILITY_MULTIMIX, that means you somehow managed to get bridge_multimix to do something. Since that was never used in Asterisk 11 and prior versions and was removed in Asterisk 12, you're pretty far off the reservation |
| 15:02.52 | mjordan | bridge_multimix is part of the Asterisk bridging framework, which is only fully used in Asterisk 12. In earlier versions, the only mixing technology that was used was bridge_softmix, and that was used by ConfBridge. ConfBridge is locked into using that mixing technology. |
| 15:02.58 | mjordan | hence my "huh"? |
| 15:08.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 15:08.29 | gusto | so |
| 15:08.35 | gusto | it seems to work now |
| 15:09.19 | mjordan | gusto: what version of Asterisk are you running? |
| 15:12.09 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:17.19 | gusto | Asterisk 1.8.13.1~dfsg-3+deb7u1 built by buildd @ build01.raspbian.lan on a armv7l running Linux on 2013-09-02 22:05:50 UTC |
| 15:17.27 | mjordan | k. Are you running ConfBridge? |
| 15:17.42 | gusto | yes |
| 15:17.42 | gusto | on that raspberry |
| 15:18.00 | mjordan | So, ConfBridge in 1.8 isn't hugely reliable. Even so, if you want to use it, I'd recommend unloading all bridge_* modules except for bridge_softmix. |
| 15:18.21 | gusto | i find it quite upsetting that they have only version 1.8 on there, but with openwrt routers its the same, so OK anyway |
| 15:18.25 | mjordan | You can think of ConfBridge in 1.8 as a proof of concept. It got completely redone in Asterisk 10, and from that version forward, is the preferred conference app |
| 15:18.35 | gusto | maybe i can even get my openwrt router to work with confbridge |
| 15:18.35 | mjordan | gusto: what do you mean? |
| 15:18.52 | mjordan | I run Asterisk 12 on a pi :-)] |
| 15:18.57 | mjordan | so you can run something other than 1.8 |
| 15:19.05 | gusto | yes |
| 15:19.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:19.15 | gusto | but i did not find it in this raspbian repositories |
| 15:19.33 | *** join/#asterisk edge (~edge@74-84-85-54.client.mchsi.com) |
| 15:19.34 | mjordan | ah, well that's possible. I build from source |
| 15:20.17 | edge | Is it possible to forward a voicemail from a user on Asterisk Server 1, to another user on Asterisk Server 2? |
| 15:20.26 | mjordan | anywho, if you do run into that again, try unloading the other bridge modules (other than softmix, which ConfBridge needs). Since they have to register themselves in order for their capabilities to be used, that should prevent anything weird from happening |
| 15:20.46 | gusto | you did build it from source on a raspberry? isnt that too slow for compiling stuff? |
| 15:20.58 | mjordan | it was slow, but it still worked |
| 15:21.11 | mjordan | type 'make', get a sandwich, etc. |
| 15:21.29 | mjordan | have a sword fight |
| 15:21.31 | mjordan | http://xkcd.com/303/ |
| 15:21.54 | gusto | i am not able to get a sandwich, i am in slovakia now |
| 15:22.05 | WIMPy | As long as it's not somethign you should start before going to bed... |
| 15:22.12 | mjordan | edge: I'm assuming the two Asterisk servers don't have a networked file system and aren't using IMAP/ODBC? |
| 15:22.37 | mjordan | IIRC, it took about an hour and a half |
| 15:22.48 | gusto | however |
| 15:22.59 | gusto | i see that wheezy does have a backported version of asterisk11 |
| 15:23.02 | gusto | that could be fun to install |
| 15:23.10 | gusto | http://packages.debian.org/wheezy-backports/armhf/asterisk/download |
| 15:23.17 | gusto | <PROTECTED> |
| 15:23.22 | gusto | gets around compiling |
| 15:23.24 | WIMPy | The local LUG has a RPi night tonight. |
| 15:23.37 | gusto | where? |
| 15:24.00 | WIMPy | St. Knudsborg, Flensburg |
| 15:24.02 | gusto | btw. is the c3 conference running already? |
| 15:24.12 | WIMPy | yes |
| 15:24.29 | gusto | well |
| 15:24.32 | gusto | i have to watch it |
| 15:24.42 | gusto | are there some streams? do you have links? |
| 15:24.56 | gusto | in the meantime i am going to try that backport there |
| 15:25.02 | WIMPy | events.ccc.de |
| 15:26.05 | WIMPy | hasn't watched last years talks, yet :-( |
| 15:26.21 | gusto | isnt that website down? |
| 15:26.24 | edge | mjordan, that is correct. They're both using simple flat file system storage for voicemails |
| 15:26.29 | WIMPy | no |
| 15:26.32 | gusto | somehow it takes too long to load |
| 15:27.24 | mjordan | edge: than nope. You'd need some other process to accomplish it. |
| 15:28.02 | edge | mjordan, if i used ODBC, that would be centralizing the storage right? |
| 15:28.58 | mjordan | correct, as well as the configuration. |
| 15:32.21 | edge | mjordan, any good documentation for sending Asterisk variables to a script? like a bash/shell ? |
| 15:38.41 | mjordan | I'm not sure I understand the question well enough to answer it. How are you attempting to launch the script? |
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| 15:49.23 | edge | mjordan, my idea would be to build an extension to playback voicemails , but allow the user to select a remote extension number, then use scp or ftp to send the file to the other server's file system |
| 15:49.59 | mjordan | ah. If you're doing that from the dialplan, then variable substitution should just "work" |
| 15:50.20 | mjordan | so using System or SHELL would substitution out the variables in the parameters passed to that application/function |
| 15:50.56 | mjordan | i.e., System(my_fancy_script.sh ${my_vm_location}) would first put in the value of ${my_vm_location} |
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| 16:00.15 | gusto | well |
| 16:00.22 | gusto | this backported asterisk segfaults on start |
| 16:00.29 | gusto | so that was not much of a big help |
| 16:03.59 | gusto | however, i can use archlinux instead, that has 11.6 too |
| 16:04.56 | LiuYan | gusto: or try install asterisk-11.7 from debian-testing? |
| 16:05.28 | gusto | i do not want to wreck this even more |
| 16:05.45 | gusto | archlinux seems to be a more stable solution |
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| 16:10.28 | WIMPy | That reminds me that I should try to get some current versions going for te HorstBox. |
| 16:11.23 | gusto | lol, not even ssh works any more |
| 16:11.38 | gusto | ok, i ll definitely have to rewrite it with something else |
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| 16:36.58 | keebs | anyone know how to redirect screen popping to a custom url? |
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| 17:26.14 | ChannelZ | huh? |
| 17:26.44 | gusto | so |
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| 17:55.45 | AndrewsEnterpris | Hello everybody whats the best way to get asterisk to add data numerical data to mysql via a incoming call? |
| 17:56.34 | navaismo | for me an agi or appmysql |
| 17:56.49 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 17:56.54 | AndrewsEnterpris | which one is more secure navaismo? |
| 17:57.32 | navaismo | but the 2nd is deprecated so an AGI is maybe the option. |
| 17:57.37 | mjordan | func_odbc. |
| 17:57.48 | saint_ | switching from 1.8 to 12.0 tonight. should be fun.. |
| 17:57.53 | navaismo | well the agi script is up to you so you can make it seure as you want |
| 17:58.02 | mjordan | saint_: that's quite the leap. Any reason you're going from an LTS to a standard release? |
| 17:58.04 | AndrewsEnterpris | Goodluck saint_ |
| 17:58.20 | Penguin | mjordan: I was just about to ask the exact same question. |
| 17:58.37 | Penguin | 1.8 to 11 makes more sense to me. |
| 17:59.03 | AndrewsEnterpris | any idea when 11 is going to be available via Aptitude? |
| 17:59.06 | mjordan | It does unless you're using 12 for a particular reason. I don't want to discourage anyone, but I'd hate for you to jump into 12 without knowing what you're getting. |
| 18:00.17 | file | func_odbc ftw |
| 18:00.25 | saint_ | mjordan i m interested into the REST API.. |
| 18:00.29 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:00.50 | mjordan | that's cool. Are you planning on putting 12 on a vm or some other test system? |
| 18:00.59 | saint_ | mjordan prod... |
| 18:01.11 | saint_ | i read it was tested in alpha and beta .. |
| 18:01.16 | [TK]D-Fender | [12:59]AndrewsEnterprisany idea when 11 is going to be available via Aptitude? <- next eventually, SHARP |
| 18:01.19 | saint_ | so if you guys release it , it should be good to go, hu ? |
| 18:01.22 | mjordan | saint_: it has been, but do you know what all has changed? |
| 18:01.32 | AndrewsEnterpris | Cheers D-Fender |
| 18:01.43 | saint_ | mjordan i read the wiki and other docs. i will spend more time reading it all before i do it tonight though. |
| 18:01.54 | saint_ | mjordan do you recommend 11 for now ? |
| 18:02.16 | mjordan | saint_: I don't make those kind of recommendations. :-) |
| 18:02.47 | saint_ | well.. nobody should be in the office so it should be nice and calm. worse cast scenario, i roll back to 1.8 |
| 18:02.59 | Penguin | Unless 12 has something specifically that you want to use that 11 does not have, I would suggest 11. |
| 18:03.06 | mjordan | saint_: Asterisk 12 is a Standard release. As such, it has some serious major changes in it, some of which were in fundamental portions of Asterisk. With anything of that size and scope, I would personally first do some testing on it. |
| 18:03.11 | navaismo | he said ARI |
| 18:03.17 | navaismo | Penguin, ^ |
| 18:03.30 | mjordan | And you can put Asterisk 12 on a VM and play around with ARI in a test environment |
| 18:03.36 | mjordan | (it's what we do) |
| 18:04.08 | saint_ | i guess i could try that . |
| 18:04.24 | saint_ | i wanted to clean my plate of todo list before the end of the year .. |
| 18:04.28 | file | or set up a SIP device connected to http://neutron.jcn-labs.net, connect to the ARI stuff, dial, and play with the different operations available #shamelessplug |
| 18:04.40 | mjordan | We do a lot of testing, but we don't test with your ITSP or your phones. We don't test with your dialplan. So... it's always good to make sure that Asterisk works in your environment and in your configuration before rolling the dice :-) |
| 18:05.10 | saint_ | file that s cool, i did not know about jcn-labs ! |
| 18:05.24 | file | it's an experiment. |
| 18:07.40 | AndrewsEnterpris | Would it be wise to make asterisk fully mysql for conf? |
| 18:08.47 | bsdice | I wait on 12 until it's a year old or so, I am skeptical of pjsip integration/stability |
| 18:08.49 | Penguin | Unless you prefer Postgres... |
| 18:09.07 | mjordan | bsdice: well, with that kind of help, it's bound to improve ;-) |
| 18:09.35 | AndrewsEnterpris | Penguin never played with postgres |
| 18:09.44 | saint_ | mjordan btw, would the latest dpma + firmwares prevent to have 2 identical "first name / last name" in the D70 feature key display ? |
| 18:10.19 | bsdice | mjordan rofl :) yeah, I wait for Digium to iron out all the easy shit, then come in and save everyone with the hard to find showstopper bugs in crypto |
| 18:10.48 | mjordan | saint_: I don't know the answer to that, unfortunately. It wouldn't be a DPMA thing, it would be a firmware update. |
| 18:11.54 | mjordan | that time I spent Sunday in the lab determining why re-INVITEs were messing up the SRTP session for old Aastra phones but not SNOMs was pretty easy. |
| 18:12.58 | mjordan | lunch! |
| 18:13.02 | mjordan | Qwell: taco truck? |
| 18:13.03 | saint_ | mjordan ok, cause I have the issue, and I can reproduce - I opened a ticket, but the guy at support thinks it s because of the new blf_items feature |
| 18:13.32 | file | mjordan, awwww taco truck |
| 18:13.40 | bsdice | I patched ssldump because friends had tls problems, to debug on the machine the SSL handshake |
| 18:14.06 | bsdice | file where are you guys located? |
| 18:14.19 | file | Digium is located in Huntsville, Alabama. |
| 18:14.27 | file | With another office in San Diego, California. |
| 18:14.36 | mjordan | And file is Digium, Canada. |
| 18:14.48 | mjordan | mourns the Canadians who went before |
| 18:14.56 | file | thus why I awwww because I can not go to the taco truck |
| 18:19.51 | bsdice | Legalize Cana |
| 18:19.53 | bsdice | ...da |
| 18:21.11 | *** part/#asterisk LiuYan (~LiuYan@222.125.137.85) |
| 18:38.10 | saint_ | ha. i have no more warranty on my car. Just reached 100k. Damn it. |
| 18:44.15 | ChannelZ-Wk | cue wheels falling off in 3.. 2.. |
| 19:00.57 | Penguin | You forgot to disconnect the speedometer? |
| 19:14.56 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.59) |
| 19:22.47 | bsdice | disconnecting the instrument cluster usually leads to car not starting any more, because it is connected via CAN bus to immobilizer ;-) |
| 19:23.01 | bsdice | unless saint_ drives a very old car |
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| 19:24.19 | saint_ | nah. i m exited to see the mileages go pass 700K and have no pb with my car. the one guy I read online went around the counter back to 0000000 |
| 19:24.46 | bsdice | counter will display ---------- |
| 19:24.55 | Penguin | I remember back when there was a mechanical cable driving the speedometer, you could just disconnect it to put your mileage on hold. |
| 19:25.07 | bsdice | those days are long gone |
| 19:25.10 | saint_ | Penguin you are that old ? :D |
| 19:25.42 | bsdice | also the engine control unit holds power on times etc. so in factory or good CAN software you can find frauds |
| 19:25.54 | Penguin | If you disconnect the wire from the VSS, what happens? I haven't tried it. |
| 19:26.04 | bsdice | VSS? |
| 19:26.22 | saint_ | vehicle speed sensor |
| 19:26.36 | Penguin | The speedometer gear used to drive a cable. Now it spins electronics. |
| 19:26.51 | bsdice | I think its integrated with ABS sensor |
| 19:27.08 | Penguin | But I'd guess if you disconnect the wire, you don't see speed and miles don't increase. |
| 19:27.10 | bsdice | disconnecting that from bus will make alot of stuff blink red in instrument cluster |
| 19:27.32 | saint_ | i think it would screw your car up. vss is not only telling how fast you are going, but manage your engine also at idle time for example . |
| 19:27.45 | saint_ | ha. here is a good article: http://www.jagsthatrun.com/Pages/SpeedSensors_Speedometer.html |
| 19:27.56 | saint_ | says you can have stall problems and other |
| 19:28.39 | Penguin | I know the reluctor rings can help determine loss of traction, but I've never heard of using them for the speedometer. |
| 19:29.55 | Penguin | ABS reluctor rings, I mean. Tone rings? |
| 19:30.44 | Penguin | The VSS probably uses a reluctor ring on the transmission or transaxle, depending on the type of vehicle and the age. |
| 19:37.47 | gusto | so |
| 19:37.57 | gusto | looks like i am up and running at this new asterisk now |
| 19:38.11 | gusto | but of course ... confbridge is not working |
| 19:38.28 | gusto | so i have to figure out now how to use that new one and then bring that to work again |
| 19:38.50 | gusto | hi Penguin |
| 19:38.59 | gusto | Penguin, how is your mirror? |
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| 20:22.23 | *** join/#asterisk thetooth (~thetooth@27-33-38-113.static.tpgi.com.au) |
| 20:23.29 | thetooth | i have direct dail enabled on a public facing IVR, how can i prevent a caller from dailing every possible number to try and map out our numbering system? |
| 20:25.19 | thetooth | we have about 10,000 calls, from 9999 to 0 being made within 13 seconds from a single number, rather than just blocking them, it would be nice to just limit the number of invaild calls you can make |
| 20:25.48 | thetooth | because of how rapid the calls are being placed they call went to congesion but that really doesn't solve the problem |
| 20:29.07 | [TK]D-Fender | So count the # of invalid calls. |
| 20:32.06 | thetooth | i should also mention i use freepbx so i don't know what im doing |
| 20:34.10 | [TK]D-Fender | Well if they are hitting the IVR, you're accepting the call. |
| 20:34.18 | [TK]D-Fender | At which point... too bad |
| 20:34.49 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:34.58 | [TK]D-Fender | You'll need to do manual dialplan to properly log off # of inappropriate choices. |
| 20:35.40 | [TK]D-Fender | At which point the "don't know what im doing" = game over if you don't have the proper understanding of * to do this. |
| 20:35.46 | thetooth | thats not the issue since they're only making a single call through the inbound route(from the SIP trunk provider), they are then managing to return to the IVR after a failed call and dail again |
| 20:35.56 | thetooth | ah right |
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| 20:36.20 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:40.25 | [TK]D-Fender | They shouldn't be getting identified as being from your SIP provider |
| 20:40.51 | [TK]D-Fender | This looks more like you're allowing un-athed calls to hit your system. |
| 20:40.58 | [TK]D-Fender | Which you shouldn't be |
| 20:44.27 | thetooth | oh, you're right, had that enabled for getting the trunking to work in the first place |
| 20:44.53 | thetooth | going to test that it works with it disabled, so that should take care of the hammering? |
| 20:45.12 | [TK]D-Fender | Almost certain that your trunk was not set up properly and thisattempt to compensate just left the door wide open |
| 20:46.00 | [TK]D-Fender | They can't hammer an IVR... that their lack of auth prevents from ever hitting the dialplan in the first place |
| 20:49.07 | thetooth | well there you go, it works fine with guest auth disabled |
| 20:50.05 | thetooth | but yeah, that makes perfect sense, i would think the caller would have been disconnected after entering an invailed number, but that wasn't the case if they were authenticated |
| 20:50.59 | thetooth | yep, "you're having problems, goodbye" |
| 20:51.04 | thetooth | love that message haha |
| 20:51.39 | [TK]D-Fender | They shouldn't be getting that message at all. |
| 20:51.52 | [TK]D-Fender | don't just use the silly checkbox for that. |
| 20:52.20 | [TK]D-Fender | if you're using a suitably modern version of FreePBX, in Asterisk SIP Settings you should be able to set general parameters. |
| 20:52.35 | [TK]D-Fender | If so add "allowguest = no". |
| 20:53.01 | [TK]D-Fender | If you aren't and can't get that module, then set it in sip_general_custom.conf |
| 20:53.21 | [TK]D-Fender | or if not available, another file that gets merged in before your peers |
| 20:53.42 | thetooth | yeah i have allowguest set to no |
| 20:54.02 | thetooth | at the IVR, when dailing an invaild ext it plays that message |
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| 20:55.14 | [TK]D-Fender | It chould never reach your IVR in the first place |
| 20:55.29 | [TK]D-Fender | [15:53]thetoothyeah i have allowguest set to no <- doesn't sound like it's in the right place |
| 20:57.00 | thetooth | no no, i have an inbound route that sends all calls calling from a DID to this IVR, this is intentional |
| 20:59.25 | thetooth | the layout is, PSTN > SIP trunk provider > my sip trunk > inbound route > IVR with a welcome message, the user is then prompted to enter an extension or wait for it to time out and enter a queue |
| 20:59.31 | thetooth | is there something wrong with this? |
| 20:59.40 | [TK]D-Fender | Some random person on the internet should be allowed to dial into your system directly. |
| 21:00.21 | [TK]D-Fender | Are you still seeing these attempts? |
| 21:00.55 | thetooth | no, the CDR was from september, i haven't investgated it until today |
| 21:02.04 | [TK]D-Fender | Calls from un-authed sources should get NO message whatever and not get processed in the dialplan |
| 21:02.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 21:03.27 | thetooth | hmm this is not good news |
| 21:04.04 | thetooth | i can give you my public facing address, can you try to log in and call a phone? |
| 21:04.34 | thetooth | i still don't beleive someone can auth now since disabling guests |
| 21:04.58 | thetooth | and it should be impossible from over the PSTN network |
| 21:05.47 | [TK]D-Fender | [16:04]thetoothi still don't beleive someone can auth now since disabling guests [16:04]thetoothand it should be impossible from over the PSTN network <- this has nothing to do with PSTN |
| 21:06.22 | [TK]D-Fender | If someone calls you using a DID you're getting from your ITSP then that is "jsut a call" and not directed at your server from someone directly. |
| 21:06.31 | [TK]D-Fender | That would be a NORMAL call delivered from your provider |
| 21:06.46 | thetooth | yes, this is where the hammering came from |
| 21:08.19 | thetooth | they did not authenticate with SIP over the internet if thats what you're saying |
| 21:08.55 | [TK]D-Fender | I'd expect to see some hard proof on that first (which we aren't getting since this was months ago), and there is littel you can do about that short of hoping they list a callerid you can block them by which still requires manual coding in the dialplan. |
| 21:09.18 | ThoMe | hihio |
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| 21:10.25 | ThoMe | have asterisk 11.7 in use and I would like use cdr in mysql. have also read "The below cdr_mysql module has been deprecated in 1.8.". but in my config dir is the file "cdr_mysql.conf". I have edit this but doesnt works. |
| 21:10.29 | ThoMe | in messages is only "[Dec 27 22:09:06] NOTICE[18468] cdr.c: CDR simple logging enabled." |
| 21:10.53 | ThoMe | in the modules-verzeichnis of asterisk is only "res_config_mysql.so". |
| 21:11.00 | ThoMe | what is missing? |
| 21:11.04 | [TK]D-Fender | ThoMe: Do what you should have been doing since 1.8 and use ODBC to get it there |
| 21:14.28 | thetooth | [TK]D-Fender: thanks for the help, i just checked our router config, the firewall prevents any traffic from the internet to even reach this machine, it white lists the SIP trunk providers ip range for both incoming and outgoing traffic, its certainly not IP based since i can't even ssh into this box from the internet |
| 21:14.55 | thetooth | the problem still remains is how did they manage to dail so many numbers at that IRV |
| 21:14.58 | thetooth | IVR* |
| 21:15.26 | [TK]D-Fender | SSH does not imply any other forwarded ranges./ Do not mix PROTOCOLS being forwarded to the actual ranges allowed |
| 21:15.30 | [TK]D-Fender | They are not the same |
| 21:16.05 | [TK]D-Fender | Anyway... you don't have anything current to show us, so lets leave the speculation out for now.... |
| 21:16.42 | [TK]D-Fender | let us know when it's happening again.... |
| 21:18.02 | thetooth | will do, i am on a different provider now after all, my last one was flynumber, it wouldn't be completely unreasonable to suspect them for pulling this rubish |
| 21:18.44 | [TK]D-Fender | I'm also not going to start placing blame without proper backup... |
| 21:18.56 | carrar | hi |
| 21:19.01 | thetooth | lel |
| 21:19.04 | carrar | let the blaming begin! |
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| 21:40.32 | ThoMe | [TK]D-Fender: emm. |
| 21:40.34 | ThoMe | [TK]D-Fender: ok. |
| 21:40.56 | ThoMe | [TK]D-Fender: I have now: odbc show asterisk2mysql andODBC DSN Settings .. bla bla: Connected: Yes |
| 21:41.17 | ThoMe | [TK]D-Fender: this file /etc/asterisk/cdr_mysql.conf i need never or? |
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| 21:41.33 | [TK]D-Fender | forget that file. |
| 21:41.37 | [TK]D-Fender | that is NOT ODBC |
| 21:43.27 | ThoMe | [TK]D-Fender: ok. works fine. logs now call. |
| 21:44.25 | ThoMe | [TK]D-Fender: I should also change res_config_mysql.conf to odbc? |
| 21:44.53 | file | is forgotten |
| 21:45.17 | [TK]D-Fender | No , not THIS file :p |
| 21:45.51 | [TK]D-Fender | ThoMe: Forget about ANY direct MySQL modules for Asterisk altogether. |
| 21:46.01 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.59) |
| 21:51.46 | ThoMe | [TK]D-Fender: When i Recieve a call then i jump to a other context, jump s@hauptnummerZeitabhaengig; |
| 21:52.25 | ThoMe | [TK]D-Fender: i my database is now in the field "dst" a "s". is it posible write in this field the called number (example 1234567) ? |
| 21:52.32 | ThoMe | have: |
| 21:52.33 | ThoMe | <PROTECTED> |
| 21:52.33 | ThoMe | <PROTECTED> |
| 21:52.33 | ThoMe | <PROTECTED> |
| 21:52.33 | ThoMe | <PROTECTED> |
| 21:52.43 | mjordan | ThoMe; no |
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| 21:52.46 | ThoMe | example in dst=callToNumber |
| 21:52.49 | ThoMe | mjordan: hm. ok. |
| 21:52.54 | ThoMe | mjordan: hello. |
| 21:53.05 | mjordan | ThoMe: You can use the CDR function to write custom data to a custom field. You can't alter the default CDR fields. |
| 21:59.31 | ThoMe | mjordan: hm ok. which file i need for custom fields. cdr_odbc.conf (i use this at the moment) or cdr_adaptive_odbc.conf ? |
| 21:59.43 | mjordan | cdr_adaptive_odbc |
| 22:00.41 | ThoMe | mjordan: cdr_odbc.conf < this file I can remove or ignore? |
| 22:01.49 | mjordan | Well, I wouldn't use both. You'd end up writing to the same database (potentially, anyway) twice |
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| 22:07.40 | ThoMe | mjordan / [TK]D-Fender: works perfectly. merci i gracias! |
| 22:07.44 | drmessano | Does Asterisk 11 still puke on the NAT settings when you enable ipv6? |
| 22:10.46 | drmessano | Lovely |
| 22:11.07 | drmessano | I see one ticket in JIRA, closed due to "lack of activity" |
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