| 00:02.06 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 00:11.49 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 00:27.54 | *** join/#asterisk guitarHester (~guitarHes@97-82-49-84.dhcp.mtgm.al.charter.com) |
| 00:53.18 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 00:57.33 | calum_ | I have 2 freepbx systems connected via an IAX2 trunk. Both boxes are behind nats with iax port forwarded to the relevent box and both boxes have dynamic ip's via dyndns. Everything works fine for a while but upon a change of ip by either end the trunk fails to function and becomes unreachable. I understand that it will for a short time but It it like asterisk / freepbx is not re querying the dns for the new ip. I am wonderin |
| 00:57.33 | calum_ | g if I need a register string on each box and if this will help. Any clues???? |
| 01:02.47 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 01:03.12 | lvlinux | yes it should fix it if you turn down the register timer to a low enough value |
| 01:03.28 | WIMPy | Do you have dnsmgr enabled? |
| 01:03.43 | lvlinux | if you don't do it with register, then asterisk assumes the ip is the same as when it first resolved the name I think. |
| 01:04.01 | lvlinux | or maybe dnsmgr fixes that? |
| 01:04.08 | WIMPy | it does |
| 01:04.18 | lvlinux | oh ok---didn't know that. |
| 01:04.28 | WIMPy | If you configure a timeout. |
| 01:04.37 | lvlinux | ah |
| 01:08.47 | Penguin | What timeout are you talking about? |
| 01:24.33 | WIMPy | refreshinterval |
| 01:36.33 | *** join/#asterisk jsjc (~Adium@192.Red-79-151-33.dynamicIP.rima-tde.net) |
| 01:48.56 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 02:31.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:33.48 | *** join/#asterisk LiuYan (~LiuYan@222.125.137.217) |
| 03:00.26 | *** join/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 03:17.20 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 03:17.24 | *** join/#asterisk obesd (~obsed@ppp121-45-210-202.lns20.cbr1.internode.on.net) |
| 03:52.51 | lvlinux | I'm trying to get a very very basic conference room going and can't seem to get ConfBridge to work---do I not just need to answer the call to an extension and run Confbridge(123,M)? |
| 03:53.30 | lvlinux | when I dial the extension for it, the phone just hangs up and says it "exited non zero" |
| 03:53.33 | mjordan | lvlinux: what version of Asterisk? |
| 03:53.49 | Penguin | Answer first. |
| 03:53.55 | lvlinux | 11.5 |
| 03:54.12 | lvlinux | i have exten => 987,1,Answer(500) |
| 03:54.14 | mjordan | you aren't using the right documentation for that application |
| 03:54.25 | lvlinux | and then same => n,ConfBridge(123,M) |
| 03:54.26 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ConfBridge |
| 03:54.38 | mjordan | There is no 'M' option. You also need to provide a confbridge.conf |
| 03:54.46 | mjordan | You may want to read the documentation on that application: |
| 03:54.56 | mjordan | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 03:55.02 | lvlinux | ah, ok I'll check that out then---was it different with 1.8? |
| 03:55.08 | mjordan | substantially. |
| 03:55.14 | lvlinux | oh ok |
| 03:55.15 | mjordan | ConfBridge was rewritten in Asterisk 10+ |
| 03:55.30 | mjordan | If you're moving to Asterisk 11, you may want to take a look at what has changed in the intervening versions |
| 03:55.31 | lvlinux | i knew it had new features but didn't realize implementation was different |
| 03:55.36 | mjordan | Completely. |
| 03:56.01 | mjordan | Not just implementation, but the configuration of it is completely different than in 1.8. |
| 04:01.53 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 04:07.18 | lvlinux | ok got it working --- thanks! i didn'thave the config file in there. |
| 04:07.34 | lvlinux | the notification sounds are cooler than in meetme lol |
| 04:27.27 | LiuYan | hi, when executing asterisk with '-x' option, how can i get the output of the execute result? |
| 04:28.35 | Penguin | Use -rx |
| 04:29.15 | LiuYan | i mean, i want to execute 'asterisk -x "originate sip/xxxxx@iptel.org" extension' command from an IRC bot, and output the result to channel |
| 04:29.34 | LiuYan | Penguin: i tried -rxvvv, but no output |
| 04:30.42 | Penguin | What kind of output do you expect? |
| 04:31.57 | LiuYan | Penguin: the output like '-rcvvv' option, every dialplan are output the the console, |
| 04:33.02 | LiuYan | like '--Executing [s@default:1] Background(....)' |
| 04:33.49 | LiuYan | i want to get those outputs, and output them to IRC channel |
| 04:34.43 | LiuYan | to demonstrate how a command is executed by asterisk. |
| 04:37.47 | *** join/#asterisk smash` (~smash@c-50-139-7-160.hsd1.or.comcast.net) |
| 04:42.57 | obesd | are there good books on setting up a maximum security asterisk deployment too ? |
| 04:45.54 | smash` | wouldnt that be at the firewall? |
| 04:46.08 | obesd | a firewall is one component yes |
| 04:46.21 | smash` | a device at a lower layer |
| 04:46.23 | obesd | (of an overall system) |
| 04:47.06 | smash` | different distro have built in modules but its still linux and networking. |
| 04:47.19 | smash` | i wouldnt put a device on a real ip without any filters |
| 04:48.03 | obesd | yes that is correct |
| 04:48.22 | smash` | lots of failed auths, etc. |
| 04:48.43 | obesd | i wonder if there are any books, for security pros, who are not asterisk or voip pros, |
| 04:48.54 | smash` | im pretty sure hosting providers do their own sweeps or their is alot of scanners. |
| 04:49.52 | *** join/#asterisk mjordan (~matt@nat/digium/x-trjwkfikxkweqnuz) |
| 04:49.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:49.55 | smash` | just put a firewall that you know how to manage security in the path. |
| 04:50.14 | obesd | yes thats already handled/obvious |
| 04:50.14 | smash` | use only ACL |
| 04:50.25 | smash` | boom done |
| 04:50.53 | obesd | remember i asked about books on maximum security asterisk deployments |
| 04:50.59 | obesd | not lame/average security asterisk deployments |
| 04:51.34 | obesd | ;p |
| 04:52.13 | smash` | i was gonna say forward me that url bro. |
| 04:53.34 | obesd | well if find a book i'll tell ya |
| 04:57.27 | LiuYan | obesd: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Security.html ? |
| 05:00.20 | obesd | yeah i think i read most of that |
| 05:00.22 | obesd | good page |
| 05:08.45 | obesd | in the asterisk config |
| 05:09.35 | obesd | can you do things like say "ip 10.1.1.2 is only allowed to be sipperson2@mylan.net and only make calls to usernames@mylan.net" ? |
| 05:31.29 | *** join/#asterisk LiuYan (~LiuYan@222.125.137.217) |
| 05:36.01 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 05:36.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 05:36.03 | *** join/#asterisk hos7ein (~chatzilla@2.191.250.70) |
| 05:48.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:03.49 | kasanop | obesd: there has been a lot of talks about security at Astricon this year https://www.youtube.com/playlist?list=PLighc-2vlRgT3DhE9DkIgSmpUX6v2AtYo |
| 06:17.21 | obesd | thnx kasanop |
| 07:25.45 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 08:01.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:23.33 | *** join/#asterisk jsjc (~Adium@99.Red-79-150-184.dynamicIP.rima-tde.net) |
| 09:06.28 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 09:14.23 | *** join/#asterisk michael_work (~michael@bzq-112-168-31-118.red.bezeqint.net) |
| 10:06.59 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 10:16.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:19.07 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 11:05.19 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 11:17.42 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 11:26.07 | *** join/#asterisk hos7ein (~chatzilla@91.98.33.208) |
| 11:51.26 | *** join/#asterisk kleszcz (~tick@linuxmafia.pl) |
| 12:06.47 | *** join/#asterisk CeBe (~CeBe@89.204.135.230) |
| 12:18.19 | *** join/#asterisk CeBe (~CeBe@89.204.135.230) |
| 13:06.23 | obesd | hope im wrong |
| 13:06.32 | obesd | but so far seems like voip has fairly low security |
| 13:15.03 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:91:51ed:4796:65c) |
| 13:15.06 | obesd | Security Master Class - https://www.youtube.com/watch?v=N4BgOqu_JIo&list=PLighc-2vlRgT3DhE9DkIgSmpUX6v2AtYo&index=1 kasanop lol Change Your Default Password. i.e. absolute beginners security class |
| 13:17.10 | obesd | security for dumbasses class :) |
| 13:19.12 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:91:51ed:4796:65c) |
| 13:19.38 | file | and yet, people don't do it |
| 13:20.52 | obesd | yep there are alot of dumbasses around =) |
| 13:21.26 | obesd | well i mean i guess its part human nature, particuly if security is not really someones field |
| 13:22.47 | obesd | a friends asterisk box got hacked actually and used to make expensive calls; so he's come to me to setup a secure system; though i havent set up a voip system before |
| 13:34.25 | *** join/#asterisk calum_ (~calum_@92.40.249.160.threembb.co.uk) |
| 13:38.29 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 13:41.32 | obesd | file: im thinking of openbsd + asterisk + openvpn, then each voip mobile phone etc VPNs in and is firewalled to only talk to the bare minimum port asterisk needs to talk to sip clients |
| 13:41.36 | obesd | What do you think? |
| 13:42.28 | obesd | (client certificates and server certificates vpn mode) |
| 13:54.19 | *** join/#asterisk hehol (~Adium@ip-78-94-76-253.unitymediagroup.de) |
| 14:04.04 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:34.40 | *** join/#asterisk dar123 (~IceChat77@77.30.127.192) |
| 14:36.45 | *** join/#asterisk polysics (~Adium@tesla.alkaloid.net) |
| 14:36.50 | polysics | hello |
| 14:37.03 | polysics | is there a way to reliably list all SIP peers with status in real time? |
| 14:37.11 | polysics | I amo mostly interested in them being registered or not |
| 14:37.15 | polysics | *am |
| 14:37.26 | polysics | and if they are or not on a call |
| 14:41.51 | Martin` | sip show peers |
| 14:44.10 | polysics | is that relaible? |
| 14:44.29 | polysics | I seem to recall someone mentioning it was not the case |
| 14:46.53 | polysics | if it is updated realtime it is perfect |
| 14:57.44 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 14:57.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:01.36 | Martin` | I believe when you request the information is is uptodate :) |
| 15:04.10 | *** join/#asterisk LiuYan (~LiuYan@222.125.137.217) |
| 15:17.39 | *** join/#asterisk draco__ (~draco@unaffiliated/themrrobert) |
| 15:24.39 | draco__ | maybe someone here can help, my sip trunk isn't being picked up =/ http://pastebin.com/NicWX6nh |
| 15:24.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:24.59 | draco__ | however, sip show peers on both ends shows the trunk is online and OK, the IP address rejected matches the ip address in the trunk |
| 15:25.04 | draco__ | http://pastebin.com/NicWX6nh |
| 15:25.43 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 15:29.08 | Penguin | (2309.35) <obesd> can you do things like say "ip 10.1.1.2 is only allowed to be sipperson2@mylan.net and only make calls to usernames@mylan.net" ? <----- Yes. This is done with SIP ACLs and dialplan. |
| 15:32.07 | Penguin | obesd: Your friend's asterisk was essentially not configured to prevent anything bad. When you allow anonymous calls and you direct them into a part of the dial plan that has outbound dialing capability, that's your own fault. |
| 15:33.27 | draco__ | if i call from the server to my home machine, it works, the home machine recognizes the trunk, but calling from home to the server, the server rejects it, saying its unknown, using the default from-sip-external context.... wth am i doing wrong, i've done dozens of these this is insane |
| 15:33.50 | Penguin | obesd: Best practice is to not allow anonymous devices to make calls. Best practice if you must allow anonymous devices to make calls is to send them to an appropriate context with well-designed extensions that only allow very specific things to happen, and never allow any type of outside access. |
| 15:35.28 | polysics | no matter what I do, peers are reported as unmonitored - what controls that, please? qualify? |
| 15:37.15 | polysics | also, no, sip show peers does not report extension status (available, busy etc) |
| 15:38.05 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 15:39.21 | [TK]D-Fender | [10:37]polysicsalso, no, sip show peers does not report extension status (available, busy etc) <- that isn't its job. "core show channels" will show you what's in use |
| 15:39.29 | [TK]D-Fender | [10:35]polysicsno matter what I do, peers are reported as unmonitored - what controls that, please? qualify? <- yes |
| 15:42.05 | polysics | so to get the existing SIP peers with information about whether they are available for a call or not, I need both functions and cross-check the result? |
| 15:42.51 | Penguin | Of course sip show peers doesn't show extension status. It's SIP SHOW PEERS, which isn't related to extensions. |
| 15:42.59 | [TK]D-Fender | polysics: that is an assumption, and a bad one. |
| 15:43.24 | [TK]D-Fender | polysics: Just because I am on a call does not mean I am incapable of taking another |
| 15:43.44 | polysics | [TK]D-Fender: in this specific use case it is what is wanted, since it is a predictive dialer |
| 15:44.34 | [TK]D-Fender | polysics: Using qualify should let you know if the peer is at least acknolwdging those requests and should be capable of responding to others. Th response to an inbound call might be "busy" ... or NOT. Asterisk won't know until it tries |
| 15:45.43 | polysics | that is ok with me, I understand "nothing" can possibly know |
| 15:46.14 | polysics | in my use case, peers only have two states, "available" and "busy" |
| 15:46.21 | polysics | so I need to get that info somehow |
| 15:46.52 | polysics | DEVICESTATE does that but for 1 device |
| 15:47.06 | Penguin | Qualify? ChanIsAvail()? |
| 15:47.09 | polysics | and it is also a dialplan function |
| 15:48.06 | polysics | ideally, I think I should be listing all extensions with state, but I do not see that as an option - does something even exist? |
| 15:48.12 | Penguin | ChanIsAvail() should check if the channel is available or not. If it is not, you can go do something else instead of Dial() it. |
| 15:48.24 | [TK]D-Fender | [10:46]polysicsso I need to get that info somehow <- get it WHERE? |
| 15:48.53 | polysics | get it through AMI to my application |
| 15:49.54 | [TK]D-Fender | polysics: Then look at the AMI command list |
| 15:50.34 | polysics | CoreShowChannels |
| 15:51.23 | polysics | that shows channels, again, not extensions |
| 15:51.30 | polysics | ExtensionState only shows one |
| 15:53.35 | polysics | and I would like to avoid having to maintain state using PeerStatus |
| 15:53.47 | polysics | although that is looking more attractive now |
| 15:53.50 | [TK]D-Fender | Channel holsd the peer-name |
| 15:54.02 | [TK]D-Fender | extensions = lines in extensions.conf |
| 15:54.44 | [TK]D-Fender | So the channel-list will tell you who is on a call. |
| 15:54.45 | polysics | [TK]D-Fender: what would you suggest I do? |
| 15:54.57 | polysics | but it will not tell me who is not on a call, but logged in |
| 15:55.00 | [TK]D-Fender | if you set up HINTS you can dump those as well if you want something easier to aprse |
| 15:55.14 | *** join/#asterisk dar123 (~IceChat77@77.30.127.192) |
| 15:55.40 | [TK]D-Fender | 2 lists. 1 to scan who is "responding to qualify", the other to see who is "on a call" |
| 15:56.02 | polysics | thinking of it, I do not have extensions here. I always just dial peers |
| 15:56.11 | dar123 | how can i create a custom loadzone |
| 15:56.20 | [TK]D-Fender | How do you not have extensions. What is * without a dialplan? |
| 15:56.28 | Penguin | You can't dial without extensions. |
| 15:56.44 | polysics | there is only one extension that goes to AsyncAGI. And you can dial peers :) |
| 15:56.48 | [TK]D-Fender | Penguin: I can imagine 1 way... retarded as it may be... |
| 15:57.02 | polysics | AMI Originate |
| 15:57.44 | Penguin | ... application Dial SIP/your-device? |
| 15:57.53 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 15:57.56 | polysics | both of those work |
| 15:59.28 | polysics | and they do not look too wrong |
| 15:59.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:59.56 | [TK]D-Fender | gah |
| 16:00.00 | dar123 | http://tny.cz/16e158af |
| 16:01.55 | dar123 | how can i use the parameters in asterisk |
| 16:09.01 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 16:10.09 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 16:23.51 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:23.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:35.04 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 16:41.47 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 17:05.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:14.03 | *** join/#asterisk felipealmeida (~user@177.41.16.8) |
| 17:16.57 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 17:26.09 | *** join/#asterisk mjordan (~matt@nat/digium/x-lazsiigoduqzrfmt) |
| 17:26.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:35.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:39.32 | *** join/#asterisk navaismo (~navaismo@189.241.125.73) |
| 17:42.40 | *** join/#asterisk CeBe (~CeBe@82.113.121.85) |
| 18:04.03 | lvlinux | Hey what is the equivalent of MEETME_EXIT_CONTEXT with ConfBridge??? |
| 18:04.43 | lvlinux | ie how do I determine where (what context) a person gets dropped off in after ConfBridge runs? |
| 18:16.19 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 18:24.10 | *** join/#asterisk nix8n82 (~AndChat28@2600:100e:b021:940f:85c2:1bf1:4e4c:e8bd) |
| 18:27.07 | *** join/#asterisk ibercom (58196178@gateway/web/freenode/ip.88.25.97.120) |
| 18:49.24 | *** join/#asterisk jpoz_ (~jpoz@184.169.152.1) |
| 18:49.46 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 18:50.40 | *** join/#asterisk chuckf_ (~chuckf@fedora/chuck) |
| 18:52.35 | *** join/#asterisk dar123 (~IceChat77@77.30.127.192) |
| 18:54.56 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 18:55.39 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 18:58.09 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 18:59.00 | *** join/#asterisk cian1500ww (~cian@109.200.20.155) |
| 19:00.15 | *** join/#asterisk cian1500ww (~cian@unaffiliated/cian1500ww) |
| 19:07.58 | *** join/#asterisk foolsh (~foolsh@c-24-14-134-34.hsd1.in.comcast.net) |
| 19:16.16 | *** part/#asterisk ibercom (58196178@gateway/web/freenode/ip.88.25.97.120) |
| 19:21.50 | *** join/#asterisk cian1500ww (~cian@unaffiliated/cian1500ww) |
| 19:24.18 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 19:47.38 | *** join/#asterisk obesd (~obsed@ppp121-45-210-202.lns20.cbr1.internode.on.net) |
| 19:48.36 | obesd | Penguin: interesting, you seem to already know exactly how it got hacked |
| 19:48.36 | obesd | leet |
| 19:49.16 | obesd | Penguin: |
| 19:49.38 | obesd | i notice asterisk being very defensive =) *whistles* wonder why |
| 19:54.36 | Penguin | *shrug* |
| 19:54.45 | Penguin | I guess I don't see what you're talking about. |
| 19:56.52 | Penguin | Care to elaborate? |
| 19:58.19 | obesd | i just woke up |
| 19:58.26 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 19:58.32 | obesd | i will be sure of what im talking about after coffee =) |
| 19:58.46 | Penguin | That'll be fine. |
| 20:30.20 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 20:33.53 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 20:35.26 | Cubber | I am trying to use ekiga with asterisk. Ekiga client registers fine and even can accept inbound calls from other extensions, however when I try to dial any extension on the system there is no response. There is also no indication of the call in the asterisk CLI. The call window pops up and then just dissappears. This happens on two separate systems. |
| 20:36.21 | Penguin | Calls don't come from extensions. |
| 20:36.56 | Penguin | If your call fails without getting to asterisk, your phone is probably misconfigured. |
| 20:37.29 | Cubber | Penguin I think it is codec related |
| 20:37.47 | Penguin | Enable sip debug and try your call again. |
| 20:37.58 | Penguin | That will show you if it is a codec problem... IF the call makes it to asterisk. |
| 20:38.11 | Cubber | how can I enable debugging? |
| 20:38.26 | Penguin | sip set debug on |
| 20:38.31 | Cubber | thanks |
| 20:38.32 | Penguin | Also, core set verbose 3 |
| 20:38.35 | Penguin | Then make the call. |
| 20:38.45 | Cubber | I am in CLI with asterisk -rvvvvv |
| 20:38.50 | Penguin | That'll be fine. |
| 20:39.12 | Cubber | That is how I have been observing for the past 2 hrs while trying to debug this, so I guess it is already been tested that way |
| 20:39.28 | Penguin | You've had the sip debug enabled? |
| 20:39.31 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 20:40.23 | Cubber | Just did that, I see a bunch of spam when the phone registers, but nothing when I go to make the call |
| 20:40.29 | Penguin | That's not spam. |
| 20:40.46 | ChannelZ | It is delicious real meat. |
| 20:40.56 | Penguin | Spam is what you find in your email inbox. SIP debugging shows the SIP traffic. |
| 20:41.39 | Cubber | Penguin ok you got me there. |
| 20:41.59 | Cubber | This is deffinitly an ekiga issue since my polycom works fine |
| 20:42.02 | Penguin | If the phone successfully registers to asterisk, that leads me to believe that the networking between the phone and asterisk is in good shape. |
| 20:42.17 | Penguin | Is ekiga on the same computer as asterisk? |
| 20:42.29 | Cubber | and the ekiga phone can accept a call from the polycom just not call it |
| 20:42.35 | Cubber | no asterisk is a dedicated server |
| 20:43.06 | Cubber | This happens to the ekiga client on my desktop and laptop so it is duplicated |
| 20:43.37 | Cubber | it used to work fine, but I think when it was updated to version 4+ the issue started |
| 20:44.16 | [TK]D-Fender | <PROTECTED> |
| 20:44.19 | Penguin | Check for a sip "server" address setting which is separate from the registration setting. |
| 20:44.24 | [TK]D-Fender | Don't assume your registration alone is correct. |
| 20:44.27 | [TK]D-Fender | show us |
| 20:44.40 | [TK]D-Fender | and start clarifying the networking details |
| 20:44.49 | Cubber | I see some info on the ekiga site about DTMF modes. It defaults SIP to INFO I tried both that and RFC2833 |
| 20:45.02 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 20:45.19 | Penguin | Since registration is exclusive from making calls, there could easily be a different section for configuration of both sip calling and registration. |
| 20:45.55 | Cubber | There is only one registrar setting in ekiga, I have been using this phone for years with asterisk I understand how to configure it, seems the problem is deeper than plain configuration. |
| 20:46.07 | [TK]D-Fender | there is no "seems" |
| 20:46.12 | [TK]D-Fender | We aren't seeing debug and confgs |
| 20:46.24 | [TK]D-Fender | And I see assumptions flying |
| 20:46.32 | [TK]D-Fender | That is a ready-made formula for failure |
| 20:46.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.76) |
| 20:46.50 | Penguin | If you know everything there is to know about it, you don't need us. |
| 20:48.17 | [TK]D-Fender | [15:44]CubberI see some info on the ekiga site about DTMF modes. It defaults SIP to INFO I tried both that and RFC2833 <- DTMF has nothing to do witht he call placed from Ekiga to *. |
| 20:49.47 | Cubber | http://pastebin.com/8BQdxFfD |
| 20:49.49 | Cubber | sip debug info |
| 20:50.02 | Cubber | while the ekiga phone was registering |
| 20:50.09 | Cubber | nothing is produced when attempting the call |
| 20:51.02 | vlad_starkov | Question: Is Asterisk compliant to CALEA (http://en.wikipedia.org/wiki/Communications_Assistance_for_Law_Enforcement_Act)? |
| 20:51.06 | Cubber | output of sip show peers: desktop 192.168.42.41 D N 5060 OK (2 ms) |
| 20:51.09 | [TK]D-Fender | then call settings would be out of whack |
| 20:51.28 | [TK]D-Fender | Possibly a dialplan issue int he sofware, incorrect proxy, etc. |
| 20:51.51 | Cubber | no proxies, using a template for sip clients and polycom hardphone works |
| 20:52.38 | Cubber | I am not a newb at this, I actually teach VoIP using asterisk at a local college |
| 20:52.56 | Penguin | [tk]d-fender: It's clear he knows more about this than us. |
| 20:53.02 | Cubber | no I am not saying that... |
| 20:53.27 | [TK]D-Fender | You're not showing us and telling us a story instead. |
| 20:54.21 | Cubber | What other info would you like? It seems I am not giving you everything your looking for but your not also providing me with what you want? Ekiga used to work fine with the settings I have in place on my server. Something in the lastest version of ekiga is causing issues. |
| 20:54.50 | [TK]D-Fender | Show us your ekiga settings in full |
| 20:54.51 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 20:55.37 | Penguin | In most cases, people come here with a problem and ask for help. We don't know everything, so we often give ideas -- thoughts on what could be wrong -- based on what you show and tell us about your specific scenario. Sometimes it is apparent where the problem lies. In those cases, someone will say, "There's yer problem." |
| 20:56.22 | Cubber | sip.conf: http://pastebin.com/Dck3tv55 account info changed but other than that it is identical |
| 20:56.36 | Penguin | If the call never makes it to asterisk, there's a problem with the phone or the network between the phone and asterisk. If the phone can register successfully and receive calls, that tells me that the networking is in good shape. That leaves the phone with a problem. |
| 20:56.42 | [TK]D-Fender | sip.conf is not the issue |
| 20:56.50 | [TK]D-Fender | your soft-phone is |
| 20:57.01 | [TK]D-Fender | and most likely and specificalyl your configuration of it |
| 20:57.08 | Cubber | Yah that is what I have been saying too, but there are not alot of config options besides what I have already stated in ekiga |
| 20:58.17 | [TK]D-Fender | You are still talking about it and not showing it |
| 20:58.25 | Cubber | Edit Account has: Name: set to extension number since it is just a name, Registrar: set to IP of server, User: set to sip account, Auth user: set to sip account, Password: set to password, Timeout: 3600 |
| 20:58.36 | [TK]D-Fender | screen-shots, not stories |
| 20:58.49 | [TK]D-Fender | Every time you dodge this you are wasting everyone's time. |
| 20:59.23 | Cubber | I do not want to post screenshots of account information to the web, I just typed out the entire screen |
| 20:59.46 | [TK]D-Fender | This proves nothing |
| 20:59.56 | mjordan | vlad_starkov: Asterisk is open source. What do you think the answer to that question is? |
| 21:00.53 | Cubber | http://wiki.ekiga.org/index.php/Manual#Managing_accounts |
| 21:00.59 | Cubber | shows all relevent entries for SIP acounts |
| 21:01.11 | [TK]D-Fender | Cubber: This is a continuing waste of our time. |
| 21:01.48 | Cubber | Ok I am sorry for wasting your time I will figure it out |
| 21:02.24 | vlad_starkov | mjordan: As I don't know the CALEA requirements in US, that's why I'm asking |
| 21:02.39 | Penguin | vlad_starkov: I would say the presence of the ChanSpy() application makes it compliant. I am not an attorney. |
| 21:03.08 | Penguin | Having the ability to monitor calls in real time? ChanSpy() does that. |
| 21:04.01 | vlad_starkov | Penguin: ChanSpy was my first thought too, but it doesn't mean that Asterisk is fully CALEA compatible at the law's point of view |
| 21:04.42 | ChannelZ | Sometimes it's not possible if the media isn't flowing through your system |
| 21:04.47 | Penguin | vlad_starkov: Asterisk doesn't do everything that the law prescribes, so I would say it can't be "fully compliant." |
| 21:04.48 | [TK]D-Fender | People make typos no matter the level of experience. You are forcibly refusing a second set of eyes over something you've probably done wrong and we'd see in a second. This is a stubborn and stupid mistake. Showing us what the form looks like doesn't prove you filled it in right. You typing is here again also proves nothing because there's no proof you are continuing being cross-eyed... |
| 21:04.49 | [TK]D-Fender | ...about it. or missing something you didn't think was of importance. |
| 21:05.01 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 21:05.03 | vlad_starkov | Penguin: Ok) |
| 21:05.44 | Penguin | vlad_starkov: Within the scope of asterisk, asterisk allows for monitoring of calls in real time. |
| 21:05.54 | Penguin | vlad_starkov: If you need a legal answer, ask an attorney. |
| 21:06.23 | mjordan | CALEA are requirements levied against carriers and providers, not the software itself. How those entities fulfill their obligations is up to them. And yes, IANAL. And yes, what Penguin said. |
| 21:06.29 | mjordan | The fact that Asterisk is open source means you can find out what its capabilities are, and what it does. No one is hiding anything. |
| 21:10.05 | vlad_starkov | Penguin: I agree with your opinion. I just thought that someone is familiar with the subject and has practical experience on CALEA. |
| 21:11.11 | ChannelZ | So is pjsip doing the codec work in v12? I'm confused a bit by the build instructions |
| 21:11.31 | ChannelZ | s/v12/asterisk 12/ |
| 21:11.38 | mjordan | ChannelZ: Nope. We don't use anything in the third_party folder in pjproject |
| 21:12.04 | ChannelZ | it just compiles anyway? |
| 21:12.27 | mjordan | yes, unfortunately. The build system doesn't have a way to disable that folder |
| 21:12.28 | ChannelZ | or I guess you do --disable-sound and it doesn't |
| 21:12.31 | ChannelZ | ah |
| 21:12.37 | mjordan | that's something we've been thinking about modifying and pushing up stream |
| 21:14.35 | ChannelZ | OK thanks. Was thinking about starting to fart around with 12 a bit |
| 21:14.41 | mjordan | we'd appreciate it :-) |
| 21:14.49 | mjordan | the more feedback we get now, the better it will be |
| 21:15.17 | Penguin | vlad_starkov: Asterisk supplies a mechanism that enables the service provider to perform real time phone call monitoring, thus allowing for the service provider's compliance while using asterisk for phone calls. If the service provider is "fully compliant" is up to the service provider. |
| 21:24.16 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 21:24.30 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 21:24.36 | pzn | is there any way to limit the filesize (or the time) of a mixmonitor/monitor command? |
| 21:25.09 | lvlinux | how do i set dynamic features on an incoming call---ie person calls in on PSTN, and the office phone answers, and the office phone can access dynamic features. ? |
| 21:26.11 | lvlinux | or when you set DYNAMIC_FEATURES=whatever does it apply to both sides of the call? |
| 21:26.53 | [TK]D-Fender | lvlinux: the feature itself says who can trigger it. |
| 21:27.36 | lvlinux | oh, ok, so that would be set in features.conf right? |
| 21:27.44 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/features.conf.sample?revision=371121&view=markup |
| 21:27.56 | [TK]D-Fender | read the sample config .... |
| 21:28.01 | Penguin | pzn: If you are using the append method, you could check the file size on subsequent executions, but for the running instance, I don't know if a way. |
| 21:28.27 | Penguin | s/w if/w of/ |
| 21:28.46 | pzn | Penguin, ok, understood. I'm not using append |
| 21:29.16 | pzn | Penguin, is there any way to "drop a call" after it lasts for 10 minutes, or if it is locked on queue for 10 minutes? |
| 21:29.35 | Penguin | pzn: Sure. There is the TIMEOUT function. |
| 21:30.23 | Penguin | When the call reaches the duration set by the TIMEOUT function, it will go to the t or T extension. |
| 21:30.42 | mjordan | Also the S option in DIal |
| 21:31.11 | pzn | Penguin, the main problem is that due to buggy SIP devices, or network instabilities, for some weird reason, the recording stays writing forever, until whole disk is filled up... that is that I intend to do, to finish the calls and consequently finish the recording |
| 21:31.25 | Penguin | That's also useful. Does the call go to t or T just like when TIMEOUT() is used? |
| 21:31.49 | mjordan | IIRC, you just get hungup. |
| 21:32.12 | pzn | and in this PBX system, all calls are below 3 minutes. it is just used as an "teller system" (don't know the exact term in english)... so doing timeout at 10 min will not annoy anyone |
| 21:32.44 | Penguin | TIMEOUT(absolute)=600 |
| 21:33.41 | Penguin | core show function TIMEOUT |
| 21:37.35 | pzn | Penguin, did it... thanks! now lets see it working :-) |
| 21:38.14 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 21:38.37 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 21:39.27 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 21:43.08 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 21:58.25 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 22:07.39 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 22:33.19 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:35.59 | *** part/#asterisk mjordan (~matt@nat/digium/x-lazsiigoduqzrfmt) |
| 22:42.26 | *** join/#asterisk CeBe (~CeBe@89.204.135.219) |
| 23:02.35 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:04.45 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 23:12.53 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 23:31.29 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 23:38.53 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |