| 00:00.22 | igcewieling | this doesn't work? SET VARIABLE x |
| 00:06.07 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
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| 01:16.55 | igcewieling | I'm a Realtime n00b, I have sip realtime setup correctly, I think because I can do a "sip show peer ipmax-nyigc load" and get the info from realtime for that peer, but none of my peers show up in "sip show peers" (even with rtcachefriends=yes) until do a "sip show peer <peername> load" Anyone know what I might be doing wrong? |
| 01:17.11 | igcewieling | BTW, none of my peers register |
| 01:17.27 | igcewieling | they are all setup with a host= and permit/deny lines |
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| 01:22.38 | ChannelZ | host=what? |
| 01:23.35 | ChannelZ | actual hostnames or IPs? |
| 01:23.43 | igcewieling | the host entry in the db for that field is the ip address of the device |
| 01:24.39 | ChannelZ | well if you've specified IPs for them then they don't need to register, unless their port numbers change for some reason |
| 01:25.16 | igcewieling | they should still show up in sip show peers, shouldn't they? |
| 01:25.26 | ChannelZ | for realtime, I have no idea |
| 01:25.46 | [TK]D-Fender | You actually have to "hit" them for something IIRC |
| 01:25.48 | igcewieling | *nod* I was hoping someone using realtime would be around. |
| 01:28.16 | igcewieling | [TK]D-Fender: define hit? Calls are being rejected with "Sending fake auth rejection" until I manually load them with sip show peer <peername> load, then calls will work from that peer. |
| 01:29.34 | igcewieling | ah, none of these are set as friends, would that make a difference in using rtcachefriends? |
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| 03:31.11 | keanne | good morning. i am testing/trying to integrate panasonic tde200 with asterisk 1.8.10.1 running on ubuntu server. I can call from asterisk local to tde200 local but it ends after exactly 1m and 33 secs. asterisk to asterisk call does not share same problem. |
| 03:32.51 | keanne | asterisk to panasonic tde200 is through sip trunk configured in asterisk. |
| 03:33.36 | keanne | protocol used is G729 as panasonic tde200 only uses this protocol for sip |
| 03:35.38 | keanne | can anyone point me in right direction what to look to troubleshoot this problem? |
| 03:35.55 | keanne | there is no time limit restriction in tde200 |
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| 04:02.18 | mattsl | I'm trying to troubleshoot the fact that I have no audio on a remote phone. After verifying all my settings are correct, include NAT settings (and I even ended up connecting a softphone on a PC connected directly to the internet with no firewall for the remote phone), I still have no dice. So, I ran a tcpdump on the server with a portrange set in rtp.conf and I get NO traffic. Any ideas? |
| 04:11.07 | igcewieling | maybe the audio does not even get set up. do a packet capture on port 5060/udp and see if there is anything weird about the INVITE packets. |
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| 04:24.29 | mattsl | igcewieling, what should I look for in the INVITEs? |
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| 04:26.51 | mattsl | the call does connect... |
| 04:27.01 | mattsl | just no audio |
| 04:27.09 | mattsl | which is why I was looking at RTP instead of SIP |
| 04:33.04 | igcewieling | look for asterisk sending the packets to the internal IP not the external IP of the NAT'd endpoint |
| 04:45.04 | *** join/#asterisk chang33 (~jimmy@114-32-7-232.HINET-IP.hinet.net) |
| 04:47.45 | chang33 | I'm running HelloEvents java example, why couldn't i get the DndStateEvent when I changed the sip client phone to DND state? |
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| 05:17.16 | keanne | here is the pastebin of the sip debug: http://pastebin.com/XT7JxEQv |
| 05:17.27 | keanne | if anyone can help, it is much appreciated. thanks. |
| 05:30.43 | edong23 | keanne: what is this? |
| 05:31.21 | keanne | edong23: that is related to my post this morning |
| 05:31.23 | keanne | good morning. i am testing/trying to integrate panasonic tde200 with asterisk 1.8.10.1 running on ubuntu server. I can call from asterisk local to tde200 local but it ends after exactly 1m and 33 secs. asterisk to asterisk call does not share same problem. |
| 05:31.38 | edong23 | keanne: i saw that |
| 05:31.44 | edong23 | but what is this 500 lines here |
| 05:31.55 | edong23 | can you get a more concise asterisk debug from the console |
| 05:32.30 | edong23 | sorry, 900 lines |
| 05:32.55 | keanne | edong23: sorry i am very new to asterisk - that is the entry from /var/log/asterisk/full |
| 05:33.09 | edong23 | can you just do asterisk -rv |
| 05:33.18 | edong23 | set debug for sip and then do your call |
| 05:33.30 | edong23 | then just get a cp and paste of that into pastie? |
| 05:33.59 | edong23 | this might very well have the information needed, but... its so long |
| 05:34.18 | keanne | edong23: ok, ill do that now |
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| 05:38.54 | stev3n | hi, for some reason I use Originate for calls, but MixMonitor don't work in this case. What should I do? Here the diagram: http://db.tt/WtpRgT7F |
| 05:39.37 | keanne | edong23: http://pastebin.com/yauhGBzP but its also 800 plus liner |
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| 07:01.10 | v0lZy | # uname -a |
| 07:01.11 | v0lZy | Linux TRITON 3.7.6-1-ARCH #1 SMP PREEMPT Mon Feb 4 10:21:12nux |
| 07:01.28 | ChannelZ | whoami |
| 07:01.35 | ChannelZ | ls |
| 07:01.36 | v0lZy | erch .. sory, wrong chan |
| 07:01.42 | ChannelZ | WTF this think is broken! |
| 07:01.57 | ChannelZ | :P |
| 07:02.05 | lorsungcu | idk i like the random command output game |
| 07:04.10 | keanne | how many failover trunk can be set? in gui only 1 can be configured |
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| 07:08.34 | ectospasm | We don't support any kind of GUI here. |
| 07:08.52 | ectospasm | keanne: in dialplan, you can set as many failover trunks as you want. |
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| 07:17.16 | keanne | ectospasm: ok i am very newbie to asterisk and right now i am using the gui. can you point me in correct document how to do it? |
| 07:17.39 | ectospasm | whaddya mean "the GUI?" |
| 07:17.59 | ectospasm | Which GUI are you referring to? There are several to choose from. |
| 07:18.28 | ectospasm | some more popular than others. |
| 07:18.38 | ectospasm | We do not support any GUIs in this channel, however. |
| 07:19.05 | keanne | the one from digium, i downloaded from trunk |
| 07:19.30 | ectospasm | that phrase makes no sense. |
| 07:19.35 | ectospasm | keanne: do you know its name? |
| 07:19.54 | ChannelZ | failover isn't a particular function of asterisk, so in your case you're at the mercy of the GUI to construct a dialplan that does it. If they only have one, well, that's what you're stuck with. |
| 07:21.06 | ChannelZ | and he's probably referring to AsteriskGUI |
| 07:21.37 | keanne | yes, asteriskgui from http://svn.digium.com/svn/asterisk-gui/trunk |
| 07:21.43 | keanne | ChannelZ: i understand |
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| 07:36.53 | ectospasm | AsteriskGUI has not been maintained in what, 2 years? |
| 07:36.59 | ectospasm | You should not use AsteriskGUI |
| 07:37.13 | ectospasm | Use FreePBX instead (see AsteriskNOW), and go to #freepbx |
| 07:37.18 | ectospasm | keanne: ^ |
| 07:38.16 | keanne | ah i understand |
| 07:39.15 | ChannelZ | They're sort of different things |
| 07:39.19 | keanne | ectospasm: thanks |
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| 08:03.06 | chang33 | Can I unregister a sip client from asterisk-java api? |
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| 08:06.08 | ChannelZ | you'd have to ask the API |
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| 08:08.50 | chang33 | asterisk-dev channel? |
| 08:09.10 | ChannelZ | asterisk-java is some 3rd party thing I assume |
| 08:10.18 | kaldemar | chang33: sure you can. if there is no action for it in the API, you can always use the "sip unregister" CLI command. |
| 08:10.19 | chang33 | ok,thx |
| 08:10.33 | ChannelZ | Can Manager unreg a sip peer? sure. Can this java wrapper you're using have a method for it specifically? I dunno. |
| 08:10.55 | ChannelZ | s/Can this/Does this/ |
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| 08:11.51 | chang33 | actually, i want to set client dnd from my java application |
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| 08:12.06 | chang33 | is that possible? |
| 08:12.32 | kaldemar | is the dnd something you have implemented yourself in the dialplan or a phone feature? |
| 08:13.25 | chang33 | no, just want to do it from a administrator by using java app |
| 08:13.37 | kaldemar | "no" on which part? |
| 08:14.09 | chang33 | not implement dialplan and phone feature |
| 08:14.43 | kaldemar | does it exist at all? |
| 08:14.51 | sekil | dnd is easily done imo either by phone or not routing the call on the pbx side |
| 08:15.42 | chang33 | yeah, just want to know is it possible from java |
| 08:16.41 | kaldemar | that depends on what your DND feature is. |
| 08:17.33 | kaldemar | there is no DND feature on the asterisk side unless you make one. controlling a phone depends on the phone. |
| 08:17.43 | chang33 | simply like NOTREADY as traditional pbx |
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| 09:06.17 | tonsit | s/hello/there all |
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| 11:01.21 | caterwaul | http://i.imgur.com/nn26a02.jpg |
| 11:05.11 | ghost75 | is it making problems when asterisk is loaded too early on bootup? i had application answer not loaded, then i restarted asterisk and everything was ok |
| 11:11.21 | wdoekes | ghost75: generally no.. unless dependencies are not started. for instance, in case of static realtime, it is imperative that the database is started. |
| 11:12.21 | ghost75 | could there be other reasons? |
| 11:12.41 | wdoekes | note that the Answer app is not in a module but in main/pbx |
| 11:13.08 | wdoekes | so I doubt that that's what you didn't load |
| 11:13.55 | ghost75 | also it was congested/busy when dialing out |
| 11:15.26 | wdoekes | you need to go back and check the exact problem before we can say anything useful |
| 11:16.10 | wdoekes | for all I know you modules weren't loaded because they're on a remote fs that wasn't mounted yet |
| 11:16.17 | wdoekes | s/you/your |
| 11:16.32 | ghost75 | all the same fs |
| 11:16.47 | wdoekes | that wasn't a question |
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| 11:17.23 | wdoekes | playing guessing games is inefficient |
| 11:17.46 | ghost75 | this was log: http://pastebin.com/6j2Fvkkx |
| 11:18.50 | wdoekes | "No application 'Answer ' for extension " <-- you have a space in there.. |
| 11:19.37 | wdoekes | that's a different problem altogether. |
| 11:20.17 | wdoekes | your first problem is on line 5, and that probably has to do with sip peers being unreachable a short while |
| 11:20.35 | ghost75 | that could be possible |
| 11:20.38 | wdoekes | if you reproduce, do 'sip show peers' and 'sip set debug on' you'll be able to identify the cause |
| 11:22.27 | ghost75 | hmm in default i have: Answer () |
| 11:22.35 | ghost75 | so it was in default probably |
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| 11:23.39 | ghost75 | maybe something in dialplan was wrong before restart that it went to default |
| 11:25.47 | wdoekes | ghost75: well yes.. missing registrations could cause inbound calls to not match the right peers (with certain configuration) |
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| 11:56.36 | ghost75 | thx for pointing the direction |
| 11:57.11 | eirirs | someone have 7975 phones with v9 firmware? |
| 11:58.56 | ghost75 | only 7941 |
| 12:02.57 | eirirs | if I run 7975 with v8 firmware, phone feature buttons like DND, CfwAll works, but if I run v9, these don't work |
| 12:05.05 | ghost75 | on mine 9 works |
| 12:05.33 | eirirs | but 7975 |
| 12:05.50 | eirirs | do you use .xml or .cnf file on your 7941 ? |
| 12:06.03 | ghost75 | xml |
| 12:06.21 | eirirs | you got BLF working too? |
| 12:06.55 | ghost75 | by patching * |
| 12:07.07 | eirirs | even version 11 ? |
| 12:07.13 | ghost75 | 1.8 |
| 12:07.41 | eirirs | I got it working by patching 1.6 , but upgraded to 11, and BLF still works, no change, and feature buttons still don't work |
| 12:07.42 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 12:08.30 | ghost75 | you use file for softkeys? |
| 12:09.00 | eirirs | no |
| 12:09.09 | eirirs | it worked out of box in v8 firmware |
| 12:09.16 | eirirs | as soon as I upgrade to v9, stop |
| 12:09.34 | eirirs | but the additional change is that I was forced to go to TCP on v9, while v8 I used UDP |
| 12:09.55 | ghost75 | 9-3-1sr1 i have |
| 12:10.46 | eirirs | cmterm-7975-sip.9-3-1SR2-1.zip |
| 12:10.48 | eirirs | on my phones |
| 12:10.50 | eirirs | 7975 |
| 12:11.38 | ghost75 | sip? |
| 12:11.40 | eirirs | ghost75: do you mind me looking on your XML file for comparing? |
| 12:11.41 | eirirs | yes |
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| 12:11.52 | ghost75 | which xml |
| 12:11.59 | eirirs | conf on your phone |
| 12:12.07 | ghost75 | the sep xml |
| 12:12.09 | eirirs | yes |
| 12:13.38 | ghost75 | http://pastebin.com/fVjGsqCK |
| 12:16.05 | eirirs | thanks |
| 12:17.41 | ghost75 | but i dont think this has to do with softkeys |
| 12:17.57 | eirirs | I see I don't have softkey.xml mentioned in my xml |
| 12:18.43 | eirirs | care about sharing it too, if you don't mind? :) |
| 12:21.07 | ghost75 | it worked without too |
| 12:21.35 | ghost75 | http://pastebin.com/FtiE1DB3 |
| 12:23.33 | eirirs | thanks again, I'll try it out when I'm back home from work |
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| 12:34.55 | sarthor | HI, How to check that USB GSM modem support voice calls or not. |
| 12:35.23 | sarthor | my lsusb shows this.. "Bus 001 Device 002: ID 1bbb:f000 T & A Mobile Phones" |
| 12:36.36 | caterwaul | I need an expert of Google Voice, Asterisk, FreePBX, Incredible PBX, PBXes.org, CSipSimple, Android, and Samsung Galaxy Note II. So far, everyone's been stumped by my problem of being able to make outgoing calls but not being able to receive them. Do any of you have the skillset? |
| 12:43.01 | ghost75 | i need to know how bluetooth pairing works on konsole :) |
| 12:44.35 | xuw | any fixes for this when using asterisk realtime? " '' is not a valid RTP hold time at line 0. Using default." ? |
| 12:44.47 | xuw | same goes for no value given for outbound proxy on line 0 in sip.conf |
| 12:45.06 | xuw | why is those messages even there, since this is realtime? any way to get rid off it, in a 'nice' way? |
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| 12:51.34 | kzoo | when yealink phones boot via auto-provisioning, do they define any special characteristics so that the web server can determine what model of phone it is (eg. distinguish between t28p, and t38g) |
| 12:53.26 | ectospasm | kzoo: if they do, they probably advertise it in the header message they send to the server |
| 12:53.31 | caterwaul | this is a question for everyone |
| 12:54.49 | caterwaul | Scientifically, objectively, what are the two best codecs to use for a cell phone in VoIP? Wifi/mobile. Right now, the best I've found is G.722 for Wifi (considered HD Voice) and G.729 for mobile. |
| 12:55.05 | kzoo | ectospasm: yes, I don't have one here to test with but am building the provisioning service |
| 12:55.27 | ectospasm | kzoo: sounds like a difficult proposition |
| 12:56.02 | ectospasm | caterwaul: which question are you most interested in? for the codec question, it depends on how you define "best" |
| 12:56.20 | ectospasm | best sound quality? Best use of bandwidth? |
| 12:56.27 | *** join/#asterisk hbonath (~hbonath@rrcs-98-102-200-228.central.biz.rr.com) |
| 12:56.31 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.66.207) |
| 12:57.02 | caterwaul | for wifi, sound quality. for mobile, a mix of sound quality and little to no jittering/garbled crap while driving for example |
| 12:58.02 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.241) |
| 12:58.49 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 12:58.52 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-147.red.bezeqint.net) |
| 13:03.31 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 13:06.45 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:11.38 | ectospasm | caterwaul: the jitter/garbled problem is more dependent on the local cell network than it is the codec. |
| 13:11.51 | ectospasm | changing the packetization will help some, though |
| 13:12.06 | caterwaul | oh? |
| 13:12.14 | caterwaul | well |
| 13:12.20 | caterwaul | I want to learn more about that kind of crap |
| 13:12.30 | caterwaul | but first I'd sure love to be able to receive a single call. That would be great. |
| 13:13.13 | WIMPy | The short answer is the same for both: Get a working network connection. |
| 13:13.21 | *** join/#asterisk hal-j (~Adium@cpe-67-240-99-71.nycap.res.rr.com) |
| 13:13.42 | caterwaul | T-Mobile IS a working network connection |
| 13:13.45 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:14.46 | WIMPy | Then, why doesn't it work? |
| 13:15.47 | caterwaul | Why can I make calls but not receive them? |
| 13:16.25 | WIMPy | NAT? Firewall? They are the usual suspects. |
| 13:16.34 | caterwaul | Despite everything set up exactly how Mrs. Doubtfire explains in his "tutorials" |
| 13:17.01 | caterwaul | Cisco E2000 router with tomato firmware with all of the ports properly opened. |
| 13:17.37 | caterwaul | here are all my settings in asterisk: http://i.imgur.com/I8fDcY6.jpg |
| 13:17.54 | caterwaul | here are my google voice settings that worked fine with pbxes.org: http://i.imgur.com/wnjGtRq.jpg |
| 13:18.01 | WIMPy | jpeg? |
| 13:18.02 | caterwaul | and here's what I think the real issue is: http://i.imgur.com/nn26a02.jpg |
| 13:18.27 | caterwaul | what's wrong with jpeg? |
| 13:19.14 | WIMPy | We don't do GUIs here. |
| 13:19.25 | caterwaul | haahah WHAT? |
| 13:19.40 | caterwaul | you know it's 2013, right? (oh boy) |
| 13:19.57 | caterwaul | Look, I'm a big fan of keyboardy things |
| 13:20.06 | WIMPy | They are impossible to support. |
| 13:20.55 | caterwaul | so maybe you can call Ward Mundy and tell him to stop promoting Incredible PBX to potential new users? |
| 13:21.08 | caterwaul | tell him to stop writing tutorials and stop recommending GUIs |
| 13:21.29 | WIMPy | No, but you have to ask him how it works. This is the wrong place. |
| 13:21.39 | caterwaul | in fact, maybe you should give me his number. he's going to get an EARFUL for his disingenuous bullshit and misleading information |
| 13:21.44 | file | why? GUIs are fine for a set of users, but this room is for the Asterisk project itself - not the GUI and how it uses Asterisk |
| 13:22.49 | file | so if individuals don't want to help you because they don't know how the GUI is configuring things, so be it |
| 13:23.38 | *** join/#asterisk davlefouAMD (~david@197.15.50.19) |
| 13:24.38 | file | and you should eliminate as many variables as possible for your issue... |
| 13:33.13 | *** join/#asterisk [TK]D-Fender (~Joe@216.191.106.165) |
| 13:55.45 | *** join/#asterisk serafie (~erin@nat/digium/x-mptktsmllqfkyenv) |
| 13:57.08 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 13:57.28 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 13:59.59 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:03.19 | caterwaul | variables, varibles.... |
| 14:03.24 | caterwaul | oh, I've got it! |
| 14:03.33 | caterwaul | VoIP on my phone worked PERFECTLY with PBXes |
| 14:03.45 | caterwaul | and then suddenly didn't work with Asterisk |
| 14:08.26 | caterwaul | ahaha just when I want to post for the first time, the forums are down. http://www.freepbx.org/forums |
| 14:08.29 | caterwaul | isn't that fantastic? |
| 14:08.41 | [TK]D-Fender | Well at least your nick is accurate |
| 14:09.38 | AkkerKid | bzao! |
| 14:10.07 | caterwaul | hey fender |
| 14:10.07 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:10.08 | caterwaul | http://home.comcast.net/~nirvgorilla/anydaynow.jpg |
| 14:10.33 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:10.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:11.30 | [TK]D-Fender | caterwaul, Cute.... |
| 14:11.50 | [TK]D-Fender | caterwaul, Any day now you might actually be looking for help and not just whining like an ass. |
| 14:11.59 | file | caterwaul, yeah I'm not going to help you with that attitude |
| 14:12.05 | [TK]D-Fender | caterwaul, Heck if you're really lucky those two days might even COINCIDE |
| 14:12.12 | caterwaul | why did I even need to come here with all the tutorials written online about this? |
| 14:12.21 | caterwaul | the information should be somewhere online but it isn't |
| 14:12.35 | caterwaul | and the experts in #freepbx (3 of them) were bamboozled by my problem |
| 14:12.45 | [TK]D-Fender | caterwaul, Maybe it's your eyes which are defective |
| 14:12.57 | caterwaul | they couldn't fatholm that xmpp debugging didn't show anything when receiving a call. they never seen that before |
| 14:13.03 | caterwaul | hey fender |
| 14:13.55 | caterwaul | you'd better make sure you've got an Elementium-Reinforced Bulwark shield if you're going to try to pull aggro and tank me |
| 14:14.07 | file | ... |
| 14:14.36 | caterwaul | http://www.wowhead.com/item=19349/elementium-reinforced-bulwark |
| 14:14.38 | hal-j | Ive seen this off and on too (if we're talking about incoming via GV) |
| 14:15.02 | [TK]D-Fender | WoW? Why would I want that? I HAVE a life.... |
| 14:16.14 | caterwaul | equating games with being a loser? Why don't you include movies, music, or any other form of entertainment, WEIRDO? |
| 14:16.24 | file | blinks |
| 14:16.26 | caterwaul | (or art) |
| 14:16.38 | caterwaul | oh I get it |
| 14:16.43 | caterwaul | this guy knows what a life consists of |
| 14:16.45 | [TK]D-Fender | I dhighly doubt it... |
| 14:17.25 | *** join/#asterisk Who-m3 (john@50.21.180.197) |
| 14:17.49 | caterwaul | Fender, are you going to post Skype, Vent, Mumble, or Teamspeak info so you can tell all about this incredible life you have that I don't? |
| 14:17.54 | caterwaul | I'm champing at the bit in anticipation |
| 14:17.57 | file | caterwaul, do you want help or not? |
| 14:17.58 | caterwaul | oh BOY |
| 14:18.03 | file | answer that. |
| 14:18.24 | caterwaul | let's put it this way, if I get this working, I'm making a video tutorial with competent commentary |
| 14:18.29 | caterwaul | I can't be the only one who wants this working |
| 14:18.35 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.86.80) |
| 14:18.36 | file | that's not what I asked |
| 14:18.44 | file | and there are numerous individuals who have this working |
| 14:18.50 | caterwaul | BULLSHIT |
| 14:18.55 | caterwaul | I've been searching for 24 hours |
| 14:18.58 | caterwaul | (or more) |
| 14:18.59 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xptituxfomdxicms) |
| 14:19.06 | file | that doesn't mean it doesn't work for other people |
| 14:19.19 | caterwaul | not A SINGLE person I've found using Incredible PBX/VirtualBox/CSipSimple/google voice has it working |
| 14:19.21 | file | now - do you want help or not? |
| 14:19.32 | WIMPy | Maybe noone writes abou it because it just works for them? |
| 14:19.47 | WIMPy | Maybe noone writes about it because noone cares. |
| 14:20.06 | [TK]D-Fender | Maybe that combination of 4 things isn't relevant. |
| 14:20.12 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 14:20.14 | hal-j | FWIW - on thurs I got asterisk running in centos running under virtualbox going through GV for both incoming and outgoing |
| 14:20.35 | hal-j | this morning incoming is being bitchy and not working, but I haven't looked into it yet. |
| 14:20.51 | [TK]D-Fender | hal-j, You aren't using CSipSimple and IncrediblePBX. Sorry, you are unfortunately disqualified... |
| 14:21.00 | hal-j | so be it :D |
| 14:21.04 | [TK]D-Fender | :) |
| 14:21.07 | file | hal-j, what version of Asterisk? |
| 14:21.16 | caterwaul | according to Mrs. Doubtfire's tutorial here, you can have Google Voice action in 5 minutes using PIAF/VirtualBox: http://nerdvittles.com/?p=4222 |
| 14:21.32 | file | caterwaul, you have yet to answer my question |
| 14:21.40 | caterwaul | you're asking a setup question |
| 14:21.43 | caterwaul | if I say no you're going to ban me |
| 14:21.50 | caterwaul | if I say yes, you're going to qualify it with some smarmy remark |
| 14:21.54 | caterwaul | "Then TUCK IN YOUR SHIRT" |
| 14:21.56 | caterwaul | I can see it now |
| 14:22.02 | file | if you say no I'm not going to ban you, unless your behavior continues against others in this channel |
| 14:22.07 | file | I just won't help you |
| 14:22.25 | file | I'll only help people who are willing to be helped, or else it's a waste of my time |
| 14:22.40 | *** mode/#asterisk [+b *!*whatapain@*.hsd1.mn.comcast.net] by leifmadsen |
| 14:22.40 | *** kick/#asterisk [caterwaul!~lmadsen@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (caterwaul) |
| 14:22.44 | leifmadsen | I will though |
| 14:22.46 | leifmadsen | screw that |
| 14:23.00 | file | leifmadsen, FINE! ruin my experiment |
| 14:23.06 | leifmadsen | I do that. |
| 14:23.07 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:23.26 | file | hal-j, if you can tell me your Asterisk version I'll put something up to see if it might have a very specific bug... |
| 14:23.29 | file | erm pull |
| 14:23.56 | hal-j | let me check. |
| 14:24.28 | hal-j | TBH - I haven't looked into the issue yet, so not actually looking for help, but I'll check (I just installed elastix and ran with it, so need to see which version of asterisk is baked in |
| 14:25.18 | file | nods |
| 14:27.52 | *** join/#asterisk jmetro (~nickserv@75.150.221.198) |
| 14:30.50 | hal-j | file: give me a few. need to jump on a call with a vendor now anyhow |
| 14:30.54 | file | k |
| 14:31.11 | hal-j | hadn't intended to be looking into this, this morning :D |
| 14:31.33 | file | I like to catch Motif issues as quickly as possible |
| 14:31.56 | file | unless the version is old and it's using chan_gtalk, in which case... not going there ^_^ |
| 14:34.03 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fdzhaghbywwrgali) |
| 14:34.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:35.06 | R1ck | wow, I didn't know ppl like caterwaul still exist, I thought we left those IRC trolls behind in the 90's |
| 14:37.37 | WIMPy | Now CeBIT has already started and the announced Digium phone features still don't seem to have been released. :-( |
| 14:37.47 | jmetro | r1ck: another relic of the 90's, http://bash.org/?top |
| 14:38.08 | *** join/#asterisk igcewieling (~igcewieli@98.183.25.31) |
| 14:44.54 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 14:48.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:48.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:49.26 | *** join/#asterisk TimeRider (~steve@81.136.216.215) |
| 14:51.17 | edong23 | caterwaul came back? |
| 14:52.29 | hal-j | file: looks like 1.8.20.0 |
| 14:52.43 | file | yeah that's chan_gtalk, it's... not good |
| 14:52.48 | hal-j | ahh |
| 14:52.58 | hal-j | recommended version? |
| 14:53.23 | [TK]D-Fender | hal-j, 11 |
| 14:53.30 | file | Asterisk 11 has a completely rewritten module for doing it that so far has had very few bug reports |
| 14:54.27 | hal-j | I'll get on that :) |
| 14:54.41 | hal-j | (that wasn't intended to be sarcastic, despite coming across as such) |
| 14:57.28 | *** join/#asterisk grummund (~user@unaffiliated/grummund) |
| 15:04.35 | grummund | Hi, are there any channels discussing generic VoIP/SIP on freenode? |
| 15:04.44 | hal-j | I'll dig up some motif docs and get something running in a VM tonight |
| 15:05.37 | igcewieling | I thought Motif was copyrighted by the Common Desktop Enviroment (CDE) people. |
| 15:08.22 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 15:11.03 | [TK]D-Fender | And Yamaha ... |
| 15:11.24 | [TK]D-Fender | grummund, perhaps #ser? |
| 15:11.32 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:11.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:11.42 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:12.01 | [TK]D-Fender | grummund, In as much as it's generally not bad suject matter here as well as many have used it outside of * as well |
| 15:13.21 | grummund | well i'm struggling to set up a SIP device but it's not asterisk :p |
| 15:13.47 | grummund | i could use a SIP test address if someone knows a good working one? |
| 15:15.32 | igcewieling | edong23: Who is this caterwaul you keep going on about? Is he anything like Cthulhu or Fred Phelps? |
| 15:16.51 | jmetro | i wouldnt consider Fred Phelps to be in the same league as Cthulu, the god of insanity |
| 15:17.35 | igcewieling | jmetro: You think fred phelps is not insane? |
| 15:18.09 | [TK]D-Fender | He isn't. |
| 15:18.41 | hal-j | guess I need to give up my pretty gui if I'm going to v11 :D |
| 15:23.29 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 15:23.34 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-147-150.user.veloxzone.com.br) |
| 15:23.56 | igcewieling | [TK]D-Fender: I consider anyone who pickets funerals to be either evil or insane. |
| 15:25.22 | [TK]D-Fender | igcewieling, Evil... yes I guess that could qualify. Supreme asshole, and troll all for legal compensation gains.... |
| 15:26.25 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:26.25 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:29.26 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 15:32.47 | Greenlight | I seem to be having some issues with manager connections. "manager show connected" does not seem to be clearing old disconnected sessions, and the Read/Write count is static at 2147483647. Also, more annoyingly the manage session seemed to hang and stop sending any data, like something was deadlocked somewhere. Are there any known issues with the manager/AMI interface performing oddly under |
| 15:32.48 | Greenlight | moderate load? |
| 15:35.57 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 15:48.45 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:48.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:52.35 | mjordan | Greenlight: are the sessions actually logging off, or just disconnecting their sockets? |
| 15:53.22 | Greenlight | mjordan: I'm not sure. I stopped receiving anything at all over the socket, although TCP didn't tell me that it had died. I then restarted by app, which would kill the socket, but not log off. |
| 15:53.50 | Greenlight | BUt the counters for the old session and the new one I just established are all stuck at 2147483647 |
| 15:54.45 | Greenlight | 32bit max, but it's not wrapping around just sticks |
| 15:55.05 | mjordan | which version? |
| 15:55.12 | Greenlight | 11.2.1 |
| 15:56.02 | mjordan | kk |
| 15:56.05 | mjordan | is looking |
| 15:56.16 | mjordan | but to answer the quicker question: no, I'm not aware of any issues |
| 15:57.09 | Greenlight | I did have a deadlock with asterisk AMI before, it was caused by a ConfBridge record. But I'm not doing any of that here |
| 15:57.45 | igcewieling | I'm also using 11.2.1 with a long running daemon connected to manager, no issues so far, but I only read from manager, never write |
| 15:58.19 | skrusty | afternoon |
| 15:58.20 | Greenlight | I've recently enabled "all" so get a spam of VarSets, and do a LOT of traffic over AMI. |
| 15:58.23 | mjordan | Greenlight: well, hm. The counters are definitely screwed up |
| 15:58.53 | mjordan | meeting time, but I'll look more into what's up with the read/write counters. |
| 15:59.03 | Greenlight | Okay, thanks |
| 16:01.25 | _Corey_ | Hmm, the Astricon website seems to have vanished... :( |
| 16:02.06 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 16:03.41 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 16:04.14 | Ice_Strike | Yo Greenlight |
| 16:04.16 | R1ck | does anybody know if there are asterisknow download mirrors? dl1.digium.com is sloooooooow |
| 16:05.07 | Greenlight | Hi Ice_Strike |
| 16:05.13 | *** join/#asterisk mrproper (9c23c002@gateway/web/freenode/ip.156.35.192.2) |
| 16:05.37 | Ice_Strike | Greenlight Whats up |
| 16:06.54 | Greenlight | Not a lot Ice_Strike, just wrestling with an AMI interface, and loosing :) |
| 16:08.03 | chris_n | so the latest fad in hacking attempts on our * server result in logs like this: http://pastebin.com/aus2TfiQ |
| 16:08.28 | chris_n | is there a way to grab the IP of the device attempting to register? |
| 16:09.15 | *** join/#asterisk menti (9c23c002@gateway/web/freenode/ip.156.35.192.2) |
| 16:09.23 | [TK]D-Fender | chris_n, Isn't that what you masked in the logs? |
| 16:09.45 | chris_n | [TK]D-Fender, nope, read carefully the note at the top of the paste |
| 16:10.08 | Ice_Strike | Greenlight Fun :P Do you know how HLR Lookup work? |
| 16:10.25 | WIMPy | Yes, same with registration attempts. The log gives you the SIP headers, not the source IP. |
| 16:10.29 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 16:10.35 | Greenlight | Ice_Strike: Only the basic concept - I beleive most telcos expose an API for it |
| 16:11.12 | Ice_Strike | Nah don't want reseller :) |
| 16:11.20 | Greenlight | Ice_Strike: ? |
| 16:11.30 | chris_n | WIMPy: I figure that the source ip is in the sip headers, but I don't see a way to access that info via the dialplan; so I'm wondering if there is some other way |
| 16:11.43 | Ice_Strike | Greenlight I meant HRL Lookup reseller service |
| 16:11.52 | Greenlight | chris_n: Isn't it in the actual log file; I thought that's what fail2ban read |
| 16:12.05 | Greenlight | Ice_Strike: So you want the actual database ? |
| 16:12.22 | chris_n | Greenlight: fail2ban log shows the same info as the full log |
| 16:12.32 | WIMPy | You can get the source IP in the dialplan with $CHANNEL, I think. |
| 16:12.38 | Ice_Strike | Greenlight Yea |
| 16:12.52 | igcewieling | chris_n: this is nothing new. Those are not registrations, a registration rejection will have the IP of the attacker. There are call attempts not regisration and for some reason Asteirsk does not log the attacking IP for those rejections |
| 16:12.54 | WIMPy | But that's useless for registration attempts. |
| 16:13.00 | Greenlight | Ice_Strike: I didn't realise that could be done; wouldn't it be outdated quickly ? |
| 16:13.30 | WIMPy | igcewieling: I see the same infor for registrations. |
| 16:13.32 | Ice_Strike | Greenlight This is not the way HLR lookup work. They don't relay on the database. |
| 16:13.50 | Greenlight | Ice_Strike: I presumed there was a central database ? |
| 16:13.50 | Ice_Strike | They use some kind of technology for lookup. |
| 16:14.06 | Greenlight | Ice_Strike: Yea, but that lookup must be against a database |
| 16:14.18 | igcewieling | WIMPy: you must be running a different version of Asterisk. On my asterisk registration rejection log messages include the IP of the attacker, non-registration attempts don't. |
| 16:15.17 | Ice_Strike | Greenlight Check this out - HRL Lookup status page http://www.cardboardfish.com/cgi-bin/hlr_status.cgi |
| 16:15.18 | chris_n | igcewieling: so short of doing some custom work, I'm stuck with no way to ban ips for this type of attack? |
| 16:15.25 | WIMPy | 11.2.0-rc1 and the logged IP is that of the server. |
| 16:16.03 | Greenlight | Ice_Strike: See it's a database.... |
| 16:16.35 | Ice_Strike | I need to find out where the database from |
| 16:16.56 | Greenlight | Ice_Strike: I thought it was a carrier type thing |
| 16:17.07 | Greenlight | Ice_Strike: Why not use an API ? |
| 16:17.39 | WIMPy | Ice_Strike: Maybe you should try in #osmocom if you want to discuss technical details. |
| 16:18.21 | *** join/#asterisk hbonath (~hbonath@rrcs-98-102-200-228.central.biz.rr.com) |
| 16:18.27 | igcewieling | chris_n: if this is not a busy server you could enable sip debug and use fail2ban. people often use fail2ban on rejected registration messages |
| 16:19.52 | igcewieling | our solution won't work for most people -- block all off-net packets with iptables |
| 16:20.05 | [TK]D-Fender | igcewieling, Would would create psycho-sized logs through... |
| 16:20.10 | Ice_Strike | Greenlight Hmm I chekc it out |
| 16:20.24 | Ice_Strike | WIMPy Cool!! |
| 16:20.32 | chris_n | igcewieling: yeah, that would kill my service providers :) |
| 16:21.03 | Greenlight | chris_n: Do you need to accept SIP calls and registrations from worldwide ? |
| 16:21.25 | chris_n | Greenlight: not worldwide, no |
| 16:21.25 | igcewieling | chris_n: we put in exceptions for the 5 or 6 which are off-net and we need to allow. |
| 16:21.49 | Greenlight | chris_n: Why not block IP's from countries you dont want to allow |
| 16:21.56 | Greenlight | chris_n: What country you in ? |
| 16:22.01 | chris_n | US |
| 16:22.12 | Greenlight | chris_n: I bet most hacking attempts are not from US IP's |
| 16:22.36 | chris_n | most are from gaza and brazil atm |
| 16:22.46 | Greenlight | chris_n: It's far from a full solution, but you can likely reduce 99% of attempts by blocking ranges you dont need to let thjrough |
| 16:22.57 | Greenlight | Yea, Gaza we used to see a lot |
| 16:23.22 | Greenlight | Since we allowed only UK IP's (since that's where our customers connect from), we've had ZERO attempts against us |
| 16:23.50 | chris_n | I'll try that and see how things run |
| 16:29.05 | hal-j | any specific docs on motif I should be reading before i dive in? |
| 16:31.49 | Qwell | all of them |
| 16:33.08 | igcewieling | hal-j: other than this? https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 16:33.30 | hal-j | actually, that's perfect |
| 16:34.15 | ghost75 | ectospasm here? |
| 16:34.31 | dr0ck | no |
| 16:36.09 | igcewieling | I'm a Realtime n00b, I have sip realtime setup correctly, I think because I can do a "sip show peer ipmax-nyigc load" and get the info from realtime for that peer, but none of my peers show up in "sip show peers" (even with rtcachefriends=yes) until do a "sip show peer <peername> load" and no calls are accepted from the peer until I did the manual load. My peers do NOT register, they all have static IP addresses configured in the host field in the |
| 16:40.58 | leifmadsen | igcewieling: that's because until the peer is used it isn't loaded into memory |
| 16:41.21 | leifmadsen | once authentication is completed though the data should be available via sip show peers |
| 16:43.38 | *** join/#asterisk Jinxed- (~DynamicFa@147.177.61.231) |
| 16:44.53 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-16357 this applies to all * versions? |
| 16:49.13 | igcewieling | leifmadsen: define "used". If the peer makes a call it gets rejected. |
| 16:49.48 | leifmadsen | igcewieling: then something isn't right, as when it goes to authenticate it should be getting looked up in the db |
| 16:49.58 | leifmadsen | check the sql statements being passed on the database side through logging |
| 16:50.20 | mjordan | ghost75: I'm not sure what you mean by applies - it was a bug in Asterisk up until 1.8.21/11.3 |
| 16:50.21 | leifmadsen | used == authenticated |
| 16:50.25 | mjordan | which are in RC status right now |
| 16:50.26 | igcewieling | leifmadsen: how would I enable logging in asterisk to log the database commands? |
| 16:50.34 | leifmadsen | igcewieling: you don't, you log it on the database |
| 16:50.37 | igcewieling | leifmadsen: our peers do not auth. |
| 16:50.39 | leifmadsen | because odbc hides that info |
| 16:50.58 | leifmadsen | in mysql you enable statement loggin |
| 16:51.03 | igcewieling | leifmadsen: oh! We are using My-we-don't-need-to-log-SQL. I'll look into it. |
| 16:51.16 | leifmadsen | yep, just enable in mysql.conf |
| 16:51.21 | *** join/#asterisk SuperNull (~FreeManof@24.148.101.238) |
| 16:51.24 | mjordan | Greenlight: the INT_MAX you're seeing is under the read/write columns, correct? |
| 16:51.27 | leifmadsen | errrrr mysql.cf I think? |
| 16:51.31 | Jinxed- | Anyone help debug this: http://paste2.org/p/3042705 It is a call from a CME into my asterisk box... the phone rings but audio doesn't go through |
| 16:51.33 | igcewieling | yep, I'll have to figure ou the right option to set |
| 16:51.51 | serafie | leifmadsen: my.cnf |
| 16:51.59 | Greenlight | mjordan: Yes, oddly even for "new" sessions (once any session has got up to that max level) |
| 16:52.03 | leifmadsen | igcewieling: you sohuld see a sql statement get shown when something comes in an attempts to auth |
| 16:52.07 | leifmadsen | serafie: something like that ya :D |
| 16:55.27 | mjordan | Greenlight: so, as it turns out, those aren't message counts by the integer value of the permission masks |
| 16:55.32 | mjordan | INT_MAX == All |
| 16:55.41 | mjordan | s/by/but |
| 16:55.47 | igcewieling | I never have been able to figure out Mysql logging. |
| 16:56.17 | igcewieling | there it goes. |
| 16:56.25 | mjordan | probably would have been much more useful to simply compute the text value of the permission masks and print that out |
| 16:58.07 | Greenlight | haha |
| 16:58.12 | Greenlight | Ok, that exaplains that |
| 16:58.25 | Greenlight | I wondered if I'd really send that many messages since reboot this morning |
| 16:58.44 | Greenlight | I'm *sure* they used to work correctly |
| 16:59.04 | Greenlight | In 1.8 iirc, I'm sure they used to count up as messages were sent/received |
| 16:59.25 | mjordan | looks |
| 17:00.22 | mjordan | been that way at least since r174764, which is 2009-02-10 |
| 17:00.37 | Greenlight | Hmm I must be thinking of something else thenm |
| 17:01.18 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:02.20 | Greenlight | Ok, so that being a red herring, should asterisk be detecting a TCP socket going dead, and removing the session? |
| 17:03.17 | mjordan | It's possible that the socket is orphaned and it's keeping the session up. If the socket itself disconnects, Asterisk will terminate the session |
| 17:03.33 | mjordan | you can emulate that pretty easily by connecting over telnet and then killing the telnet session |
| 17:03.33 | Nugget | telnet is eeeeeeevil! |
| 17:03.42 | mjordan | yes, yes it is |
| 17:03.52 | Jinxed- | how do you enable sip debuging now? |
| 17:04.05 | mjordan | "sip set debug on" |
| 17:04.07 | Greenlight | mjordan: Ok, thanks I'll go give that a shot |
| 17:04.15 | Jinxed- | mjordan, is that the same for: 1.8.10.1 |
| 17:04.57 | mjordan | Greenlight: if you run into it, I'd be interested in seeing what netstat thinks the state of the socket is |
| 17:05.56 | Greenlight | Hmm... closing telnet and asterisk cleans up properly. So that gives weight to this particular manager session being deadlocked (it just stopped giving any output at all, and my app still thought socket was up) |
| 17:06.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:06.27 | WIMPy | What kind of actions do you send? |
| 17:06.59 | Greenlight | All sorts - Originates, VarSets, Redirects, Bridges |
| 17:07.29 | Greenlight | mjordan: Two secs I'll check what netstat thinks the state of the socket is |
| 17:07.45 | *** join/#asterisk Who-m3 (~john@50.21.180.197) |
| 17:08.07 | Greenlight | mjordan: Netstat thinks it's no longer there (only shows the new, current one) |
| 17:09.16 | Greenlight | (http://pastebin.com/fv7WGMw9) |
| 17:10.42 | Greenlight | As I say, about 6 months ago I had an issue whereby I could deadlock an AMI session by doing something funny with a ConfBridgeRecord. But, hanging up the offending channel via CLI, resurected the AMI session. This time I hung up all channel, and that didn't help |
| 17:13.10 | mjordan | yeah, that's pretty odd. |
| 17:14.06 | Greenlight | I sorted the confbrdige deadlock thing (which would only happen maybe once a week - say a million or two calls) by adjusting the timing when I sent my events. BUt I can't see what might be causing it this time |
| 17:14.22 | Greenlight | Or, more importantly, how to diagnose it |
| 17:15.44 | Greenlight | Not even sure where to start when looking for how something could deadlock sending of AMI events |
| 17:17.47 | igcewieling | my bet is a deadlock in your application 8-) |
| 17:18.34 | Greenlight | tcpdump disagrees with you ;/ |
| 17:18.44 | Greenlight | That was my though initially too though |
| 17:19.12 | Greenlight | Also, we can see a stale/locked session in asterisk |
| 17:19.23 | Greenlight | Which would indicate the issue is there |
| 17:19.40 | mjordan | Greenlight: oo. Hm. The ConfBridge deadlock thing is real, and probably would mess up the calling thread, which is an AMI thread |
| 17:20.11 | mjordan | Greenlight: this just got fixed in r381702 |
| 17:20.23 | Greenlight | Ahh cool - It was around ConfBridgeRecord irrc |
| 17:20.33 | Greenlight | *iirc |
| 17:20.38 | mjordan | correct |
| 17:20.54 | Greenlight | I'm not using that action anymore, but I guess a similar thing could be happening |
| 17:21.08 | Greenlight | Just tracking the thing down |
| 17:21.09 | mjordan | well, it was very specific to ConfBridgeRecord |
| 17:21.52 | Greenlight | DOes the fact that the AMI session is sitting there "stale" when it's actually dead not indicate something is locked or wrong with it ? |
| 17:22.17 | mjordan | no, there's something wrong there. |
| 17:23.58 | Greenlight | So, what sort of things should I be looking at that could potentially make cause the AMI to get locked like that. For example, in the ConfBridgeRecord thing, does that mean any Action which calls a function that locks or dies, will stall the AMI ? |
| 17:24.23 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 17:25.45 | Greenlight | The last message I received before it hung was a "Dual Redirect Successful" response. However, I do remember with the ConfBridgeRecord thing it wasn't as cut and dry because you could get a few "other" messages after the confbrdige one |
| 17:28.06 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 17:28.21 | mjordan | Greenlight: each session has a thread. If an action blocks that thread, then yes, that session is pretty much up a creek. This is why we typically will marshal long running actions off onto another thread |
| 17:29.14 | Greenlight | mjordan: Ok, makes sence. I've been working back through the logs, and think I see what killed it |
| 17:30.26 | Greenlight | Basically I've two channels bridged, and want to turn that into a 3 way call. I firstly dual redirect the two channels to a ConfBridge, and then Originate a call from that ConfBridge to get the 3rd part in. I get a response to the Dual redirect, but NOT to the Origiante (usually get the "originate sucessfully queued" message). |
| 17:30.57 | mjordan | interesting. |
| 17:31.14 | Greenlight | SO *either* after sending the response, the dual redirect died. Or the Originate had a problem |
| 17:31.19 | mjordan | does the originated party answer? |
| 17:31.38 | Greenlight | Not sure, sorry |
| 17:31.41 | mjordan | hm. That shouldn't matter actually, since an Originate is one of those where things are dispatched off to another thread |
| 17:32.18 | Greenlight | With "async=yes" usually right away get a response saying it's queued |
| 17:32.44 | Greenlight | *async=true |
| 17:32.49 | mjordan | Greenlight: ah. If you didn't, it may have blocked until the originate completed. |
| 17:33.02 | mjordan | Greenlight: Also, we just fixed some issues with Redirect in 11.3 - ASTERISK-19948 |
| 17:34.37 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:f827:8d15:e04:7252) |
| 17:35.26 | Greenlight | Just checked the log, "async=true" was sent. (http://pastebin.com/UjUWFwHH) |
| 17:36.13 | Greenlight | In fact I remember you mentioning fixing some issues with Redirect when i last had a problem with it, wish I'd went with 11.3 |
| 17:37.50 | Greenlight | Then oddly, I get a message saying the redirect has worked |
| 17:37.59 | *** join/#asterisk psilikon (~joel@24.73.76.82) |
| 17:38.19 | Ice_Strike | Where you from Greenlight |
| 17:38.37 | Greenlight | Scotland, UK... |
| 17:38.50 | Ice_Strike | Ahh |
| 17:39.02 | Greenlight | ... ? |
| 17:39.14 | Ice_Strike | I meant Oh right lol |
| 17:39.19 | Greenlight | Heh :) |
| 17:39.22 | Ice_Strike | LOL! |
| 17:41.16 | Greenlight | I could delay the "Originate" after the "Redirect" by say 500ms; think that would help ? |
| 17:43.13 | Greenlight | Hmmm... looking at the code, *after* the "Dual Redirect Successful", if then tries to lock stuff |
| 17:43.20 | Greenlight | *it |
| 17:45.11 | Greenlight | Ahhh... this is where changes were made in ASTERISK-19948 |
| 17:46.28 | Ice_Strike | Yup |
| 18:04.06 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-vvyczsxqcwthzdxv) |
| 18:04.38 | *** join/#asterisk marksaitis (~marksaiti@86.28.107.165) |
| 18:07.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:09.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:13.39 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:27.45 | ghost75 | leifmadsen: the bluetooth connect, the os has to do this or asterisk can do this too? |
| 18:29.41 | leifmadsen | OS must support BT |
| 18:29.58 | leifmadsen | chan_bluetooth has received such little support, I would be surprised if it worked |
| 18:31.04 | BludSuckingFiend | chan_mobile or is there a separate chan? |
| 18:31.42 | BludSuckingFiend | I was disappointed I couldn't get chan_mobile to work on my rPi... but I think that's more my adapter's fault. |
| 18:32.06 | leifmadsen | oops, ya sorry meant chan_movile |
| 18:32.07 | leifmadsen | mobile* |
| 18:32.14 | hal-j | <PROTECTED> |
| 18:32.19 | leifmadsen | ya, the adapter is a big deal for that |
| 18:32.25 | leifmadsen | hal-j: no |
| 18:32.37 | leifmadsen | you should definitely create a local branch for your configurations and test though |
| 18:32.44 | leifmadsen | and fix the things that are different |
| 18:32.45 | hal-j | so I can drag my configs over, nuke the google stuff, configure motif, and tie the ends together |
| 18:32.58 | leifmadsen | something like that -- don't do it in production |
| 18:33.05 | carrar | click go |
| 18:33.06 | leifmadsen | setup a test environment and fix things as need be |
| 18:33.10 | BludSuckingFiend | nuke it from orbit... it's the only way to be sure |
| 18:34.23 | hal-j | heh |
| 18:34.25 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:34.29 | hal-j | I wasn't planning on doing it production :X |
| 18:34.43 | [TK]D-Fender | BludSuckingFiend, </ripley> |
| 18:35.33 | BludSuckingFiend | [TK]D-Fender: Yes, a classic. |
| 18:36.08 | leifmadsen | hal-j: wasn't planning doesn't mean it won't happen |
| 18:36.41 | hal-j | this is a hope setup, different VMs. I'm pretty safe here |
| 18:36.42 | hal-j | :D |
| 18:37.06 | hal-j | while I run a ton of prod systems in my real life, none of them are asterisk |
| 18:37.28 | BludSuckingFiend | What are they usually? |
| 18:38.08 | hal-j | hm? |
| 18:38.29 | BludSuckingFiend | I have an Avaya CM, Cisco CM, Nortel CS1000, Shoretel, eOn, Mitel, and aserisk box here at work I get to play with. |
| 18:38.37 | BludSuckingFiend | asterisk* |
| 18:38.53 | hal-j | in-house IT has not been one of my responsibilities |
| 18:39.08 | hal-j | I tend towards production operations, running the outward facing stuff |
| 18:39.13 | BludSuckingFiend | The Nortel has got to be the worst telecom techie torture device ever invented |
| 18:39.30 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.55.144) |
| 18:39.45 | BludSuckingFiend | Most of those above are Lab switches we use for CTI development |
| 18:39.59 | BludSuckingFiend | The Asterisk is what we use in production for our Internal company phones |
| 18:40.24 | ghost75 | leifmadsen: bluez is f0cked too |
| 18:41.32 | ghost75 | they are cowbow style proggers |
| 18:42.42 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:42.42 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:43.25 | hal-j | my plan is to move my house over to it |
| 18:43.31 | hal-j | still in the POC stage |
| 18:43.56 | BludSuckingFiend | I'd like to use chan_mobile for presence detection here at work |
| 18:44.12 | BludSuckingFiend | too bad bluetooth is range limited... I don't sit close enough to our data center :( |
| 18:45.57 | hal-j | goals are 1) convert the "normal" landline over to something (GV? other? dunno yet), run the house telephony through it (convert to IP phones? use a few ATAs? not sure yet) |
| 18:46.09 | lorsungcu | wonder if it could be reworked to use something like WoL |
| 18:46.10 | hal-j | 2) integrate IVR with the lighting / HVAC control |
| 18:46.16 | lorsungcu | @ BludSuckingFiend ^ |
| 18:46.45 | BludSuckingFiend | I was thinking a script that pings a reserved wifi IP for my phone |
| 18:46.52 | BludSuckingFiend | that way it will be based on the range of my wifi instead |
| 18:46.56 | lorsungcu | sure, or an http request |
| 18:47.16 | lorsungcu | if you're using Android, there's an app called Locale that would work nicely |
| 18:47.17 | BludSuckingFiend | I hate it when people call my cell first when I am sitting next to my desk phone |
| 18:47.29 | BludSuckingFiend | iphone unfortunately |
| 18:47.34 | BludSuckingFiend | not as flexible, but I like thenm |
| 18:47.38 | lorsungcu | probably something simialr |
| 18:48.00 | lorsungcu | has rules for what to do when connected to specific SSIDs |
| 18:48.14 | lorsungcu | just make a php script that changes call rules |
| 18:48.16 | eirirs | BludSuckingFiend: you could put iphone to auto forward all inc call to the desk phone |
| 18:48.20 | eirirs | :) |
| 18:48.21 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:48.22 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:49.25 | BludSuckingFiend | I could, but I'd rather have it detect my presence and do it automatically :) |
| 18:49.27 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 18:49.33 | eirirs | hehe |
| 18:49.34 | BludSuckingFiend | It just isn't as cool using forward |
| 18:49.41 | eirirs | get a rfid chip implanted |
| 18:49.42 | eirirs | :P |
| 18:50.01 | BludSuckingFiend | if you don't use at least 3 different exciting technologies to do something, it's not worth doing |
| 18:54.35 | igcewieling | cham_mobile is one of the "blue bells" features of Asterisk, it sounds awesome and you are excited to put this feature in place, but you eventually realize it won't do what you want. |
| 18:55.17 | BludSuckingFiend | It's fun to play with... not real practical |
| 18:55.35 | BludSuckingFiend | kinda like TDMoE |
| 18:55.43 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:55.55 | BludSuckingFiend | I keep trying to think of ways to use that, but there's always something more practical for the situation |
| 18:56.26 | *** join/#asterisk kchehab (~david@62.84.68.161) |
| 18:57.11 | kchehab | i was using digium skype to sip and now the licence expired ,is there a digium solution or recommended solution to call and recieve calls from skype |
| 18:57.34 | [TK]D-Fender | kchehab, No |
| 18:57.45 | [TK]D-Fender | kchehab, Microsoft bought out Skype and closed the doors |
| 18:57.55 | [TK]D-Fender | kchehab, Dead end as far as direct support is concerned |
| 18:58.10 | kchehab | [TK]D-Fender from day 0 i have microsoft |
| 18:58.13 | igcewieling | Skype does not want your business. It is not worth fighting companies which don't want your business |
| 18:58.15 | [TK]D-Fender | kchehab, Go look for a Skype > SIP conversion tool/site |
| 18:58.25 | BludSuckingFiend | I've noticed even pidgin can't support skype directly |
| 18:58.38 | BludSuckingFiend | they have to use an API to the installed Skype application -_- |
| 18:59.18 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 18:59.18 | lorsungcu | i've never installed skype |
| 18:59.22 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:59.24 | lorsungcu | and do not feel that I've missed anything |
| 18:59.44 | kchehab | BludSuckingFiend i find alot of solutions in voip.org for skype to sip ,i will try to test it ,and print my feed back if anyone interested |
| 18:59.49 | BludSuckingFiend | Yeah... Me too until recently. I know some people that only use MSN messenger |
| 18:59.54 | BludSuckingFiend | now I have to use Skype for them. |
| 19:00.29 | lorsungcu | eh, if they don't have a google account, I use email :D |
| 19:00.54 | [TK]D-Fender | kchehab, That wiki is dead. Just like Skype for Asterisk |
| 19:00.55 | BludSuckingFiend | I use AIM mostly |
| 19:01.32 | BludSuckingFiend | I was surprised I recently remembered my ICQ number |
| 19:01.47 | BludSuckingFiend | hadn't signed into that for years |
| 19:02.01 | kchehab | BludSuckingFiend how is that if a XXXXXXXXXXXXX number :) u have a good memory |
| 19:02.44 | BludSuckingFiend | Yeah, if I type a # 3-5 times I remember it for life... usually |
| 19:03.42 | BludSuckingFiend | phone numbers, ID #s, my CEO's credit card # |
| 19:04.27 | [TK]D-Fender | My ICQ was very early.. thus very short... |
| 19:04.54 | lorsungcu | think i had like 8 digits |
| 19:05.29 | BludSuckingFiend | same |
| 19:05.56 | BludSuckingFiend | My slashdot UID is relatively low too lol |
| 19:06.27 | kchehab | BludSuckingFiend did you find any one you know there ? |
| 19:07.35 | BludSuckingFiend | where? |
| 19:08.52 | hal-j | I'd have to dig for my ICQ. my slashdot has 4 digits tho :) |
| 19:11.18 | kchehab | BludSuckingFiend in ICQ |
| 19:15.24 | BludSuckingFiend | not online |
| 19:20.54 | edong23 | <BludSuckingFiend> Yeah, if I type a # 3-5 times I remember it for life... usually |
| 19:20.58 | edong23 | it goes away with age |
| 19:21.00 | edong23 | youll get there |
| 19:28.01 | paulc | I just went to look at my slashdot ID.. also a 4 digit number, starts with a 5 |
| 19:28.27 | paulc | and I remember my 7 digit ICQ number, though I couldn't tell you the last time I logged in.. (10+ years ago?) |
| 19:32.41 | hal-j | my /. starts with 8, so you beat me |
| 19:32.47 | hal-j | but probably only by weeks |
| 19:34.23 | paulc | Back in the day eh.. I used to be a regular reader.. these days I skim it through Google Reader once in a while. |
| 19:34.37 | hal-j | yeah, it's been a long while since it was a regular read |
| 19:34.39 | hal-j | (for me) |
| 19:38.36 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-219-104.pools.arcor-ip.net) |
| 19:42.49 | FLeiXiuS | I need to make a pseudo call, is there a better handler thats native to Asterisk which wouldnt require compiling/using DAHDI? |
| 19:45.04 | [TK]D-Fender | FLeiXiuS, Local |
| 19:46.29 | FLeiXiuS | [TK]D-Fender, IS there a way to have meetme use Local? |
| 19:47.18 | [TK]D-Fender | FLeiXiuS, Meetme doesn't care what channel types join it. |
| 19:47.31 | [TK]D-Fender | FLeiXiuS, And MeetMe requires DAHDI |
| 19:48.30 | FLeiXiuS | [TK]D-Fender, You're right, it won't load without it. Now off to alternative conference rooms than I guess... |
| 19:49.00 | [TK]D-Fender | FLeiXiuS, "core show application confbridge"\ |
| 19:51.11 | FLeiXiuS | [TK]D-Fender, Actually, this will work. Thanks man. |
| 19:52.11 | [TK]D-Fender | FLeiXiuS, in 11 it really outstrips MeetMe in it's functionality |
| 19:52.37 | FLeiXiuS | I'm on 1.8 atm, need the LTS support for the time being with product releases etc. |
| 19:52.52 | FLeiXiuS | But If I can integrate it now, we'll be in the $$ |
| 19:53.11 | FLeiXiuS | Figuratively speaking.. |
| 19:53.31 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:53.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:54.26 | mjordan | FLeiXiuS: 11 is LTS as well. And has a longer support window. |
| 19:54.55 | FLeiXiuS | You're 100% correct, I just looked at the website. |
| 19:55.05 | FLeiXiuS | Been a while since I've taken a look out of the ppa's |
| 19:55.39 | [TK]D-Fender | FLeiXiuS, 11 = LTS |
| 19:55.49 | [TK]D-Fender | ... and already said. |
| 19:55.53 | FLeiXiuS | ;-) |
| 19:56.01 | FLeiXiuS | Riddle me this, how did we go from 1.8 -> 11. |
| 19:56.08 | mjordan | there was a 10 in there |
| 19:56.13 | mjordan | 1.8 => 10 => 11 |
| 19:56.22 | [TK]D-Fender | FLeiXiuS, Arbitrary numbering :) |
| 19:56.33 | mjordan | http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 19:56.56 | [TK]D-Fender | FLeiXiuS, http://xkcd.com/670/ |
| 19:57.34 | FLeiXiuS | Ha ha |
| 19:57.43 | FLeiXiuS | Arbitrary numbering schemes ftw. |
| 19:58.10 | mjordan | you should try writing code that has to compare Asterisk versions for relative weight |
| 19:58.17 | mjordan | it sucks. A lot. |
| 19:58.52 | FLeiXiuS | Well 1.8 < 10 shouldn't be too bad ;-) |
| 19:59.03 | leifmadsen | FLeiXiuS: it's not arbitrary |
| 19:59.17 | leifmadsen | 1.10 == 1.^H^H10 |
| 19:59.34 | mjordan | http://svn.asterisk.org/svn/testsuite/asterisk/trunk/lib/python/asterisk/version.py |
| 19:59.36 | FLeiXiuS | leifmadsen, And 1.11 == 1.^H^H11 |
| 19:59.37 | FLeiXiuS | ;-) |
| 19:59.39 | leifmadsen | the 1 was never going to increment, so it was simply ${VERSION:2}'d |
| 19:59.47 | leifmadsen | well, there would have never been a 1.11 |
| 19:59.57 | FLeiXiuS | leifmadsen, What's the point of a major number if you're never incrementing ;-) |
| 19:59.58 | leifmadsen | but when you go to 10, it makes sense to just go to 11 instead of 12 |
| 20:00.03 | leifmadsen | FLeiXiuS: exactly. |
| 20:00.46 | FLeiXiuS | My versions are major.minor simple. Scheduled major releases, with minor improvements along the way |
| 20:01.15 | FLeiXiuS | Can't get much simpler, but that's scrum methodology rather OSS. |
| 20:01.41 | FLeiXiuS | I |
| 20:01.45 | FLeiXiuS | I'll brb * |
| 20:05.46 | *** join/#asterisk tonsit (~FtC@tonsit.com) |
| 20:13.00 | *** join/#asterisk Menti (5d9cdaba@gateway/web/freenode/ip.93.156.218.186) |
| 20:13.08 | Menti | Hi guys |
| 20:14.14 | Menti | Need help with some technical stuff, i swear that i googled it, but i can't find the solution :P |
| 20:14.44 | navaismo | ~ask |
| 20:14.44 | 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. |
| 20:15.52 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 20:16.10 | Menti | last week i installed Asterisk in my Raspberry Pi (Raspian distribution), everything looked fine until i lanched Asterisk |
| 20:16.38 | Menti | i made an netstat -tuplv and Asterisk weren't there, anything on port 5060 |
| 20:17.38 | Menti | and no errors during launching (sudo Asterisk -vc) any ideas? do you need more technical information i think just ask, i don't wanna flame with all the config :3 |
| 20:18.19 | navaismo | can you connect it with the command: asterisk -r |
| 20:19.35 | Menti | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 20:19.45 | Menti | never seen this before O__O |
| 20:19.46 | *** join/#asterisk NightMonkey_ (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:21.08 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:21.08 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:21.28 | Menti | That directory is totally empty btw |
| 20:21.32 | navaismo | if you run: ps -fea | grep asterisk, do you see something like: /usr/sbin/asterisk -f -vvvg -c |
| 20:22.15 | navaismo | dont know how ddid you installed it, but chek if you have the init script and try to run with: /etc/init.d/asterisk start |
| 20:22.25 | Menti | no, only my "grep" process |
| 20:22.45 | navaismo | otherwise run asterisk with: asterisk -vvvvvvvddcg and check for errors |
| 20:22.49 | Menti | i installed it following the "Quick Start Guide" from Asterisk main page |
| 20:23.32 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 20:24.55 | *** join/#asterisk tallest_red (~CNZ@ip98-169-207-41.dc.dc.cox.net) |
| 20:24.58 | navaismo | try starting via init scrip or with asterisk -vvvvvvcg, check if selinux is disabled |
| 20:25.21 | navaismo | and if nothing of the obove works you need to pastebin the output of asterisk -vvvvvvvcg |
| 20:26.07 | Menti | (don't have any script con init.d) i'm looking for errors with -vvvvvvvddcg |
| 20:27.57 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 20:32.19 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 20:32.19 | Ice_Strike | Hi |
| 20:33.06 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:35.45 | Menti | here you have the pastebin of the output http://pastebin.com/vn7Gq8gR |
| 20:37.30 | *** join/#asterisk menti_ (5d9cdaba@gateway/web/freenode/ip.93.156.218.186) |
| 20:42.12 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 20:45.27 | hal-j | ? |
| 20:45.31 | hal-j | damnit, ww |
| 20:48.14 | Tagor | I'm trying to get incoming calls working, but at the INVITE I get an 'SIP/2.0 401 Unauthorized' error. Does anyone know what could be wrong? |
| 20:48.33 | [TK]D-Fender | Tagor, Incoming from where? |
| 20:48.52 | Tagor | [TK]D-Fender: my SIP provider |
| 20:49.38 | navaismo | menti_, the pb show some errors how did you compiled asterisk |
| 20:50.18 | menti_ | i downloaded the code, ./controb/scripts/install_pereq |
| 20:50.23 | menti_ | ./configure |
| 20:50.26 | menti_ | sudo make |
| 20:50.37 | menti_ | sudo make install and that's it :S |
| 20:52.30 | navaismo | make config & make samples?? |
| 20:52.58 | menti_ | only make samples. didn't execute "make config" only ./config |
| 20:53.38 | navaismo | ok make config only copy the init script, not big deal |
| 20:54.25 | menti_ | can i make it now? we can solve a little issue :3 |
| 20:54.47 | navaismo | during compilation did you saw errors |
| 20:54.50 | navaismo | ? |
| 20:55.10 | *** join/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 20:55.41 | menti_ | if errors appears compilation should stop isn't it? |
| 20:56.24 | menti_ | i didn't see anything, i had to install some libraries but it was during ./configure phase, some stuff for xml.... |
| 20:57.41 | menti_ | "We could not install init scripts for your operating system" when i execute make config |
| 20:58.09 | Tagor | [TK]D-Fender: any idea what could be wrong? |
| 20:59.07 | [TK]D-Fender | Tagor, What provider? |
| 20:59.37 | Tagor | [TK]D-Fender: MTTM, you probably don't know them |
| 21:00.05 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:00.48 | [TK]D-Fender | Tagor, Excellent odds they are sending you un-authed calls and don't respond to *'s challenge. Add "insecure=port,invite" to your peer and retest |
| 21:09.44 | Tagor | [TK]D-Fender: already tried that, here's the debug log of the incoming call: http://pastebin.com/8DEzsxXh |
| 21:10.00 | navaismo | menti_, the pb show [Mar 4 20:30:08] WARNING[2150]: loader.c:409 load_dynamic_module: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No such file or directory |
| 21:10.12 | navaismo | menti_, its like some modules are missing |
| 21:11.10 | hal-j | or are elsewhere |
| 21:11.23 | Greenlight | How do the dialplan SRV functions fare (with a local dns server), to do proper SRV load balancing? |
| 21:11.31 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:13.10 | menti_ | navaismo how can i fix it? if i copy that files from other Asterisk should work? |
| 21:13.20 | navaismo | no |
| 21:13.24 | navaismo | you need to compile again |
| 21:13.34 | menti_ | or i must reinstall all? |
| 21:14.40 | igcewieling | Greenlight: you write the load balancing code yourself. |
| 21:15.14 | igcewieling | the SRV functions just expose the data. I have an AEL script I use for that, you can have it, but I don't support it. |
| 21:15.16 | Greenlight | igcewieling: Yea, but in the dialplan using the two FUNC's, yea? |
| 21:15.17 | hal-j | TBH - if they didn't compile, there may be deeper issues than themissing files |
| 21:15.39 | Greenlight | I was going to get the number of records, and RAND pick one |
| 21:15.48 | igcewieling | Greenlight: yes, though in hindsight writing an AGI to do pick the server it would have been easier |
| 21:15.51 | Greenlight | Just wanted to know if anyone else was using in production, under load |
| 21:17.00 | Greenlight | My telco keeps moaning that I don't balance my load over all the nodes ;/ |
| 21:17.36 | igcewieling | green http://pastebin.ca/2328184 |
| 21:17.41 | Greenlight | Many thanks |
| 21:19.02 | navaismo | menti_, no only asterisk |
| 21:19.34 | Greenlight | Wow looks like it does everything |
| 21:19.59 | igcewieling | Greenlight: if your isp has roundrobbin IPs for their server you can just enable dnsmgr and that might do it for you. |
| 21:20.08 | igcewieling | Greenlight: like I said, would have been easier in PHP 8-) |
| 21:20.24 | menti_ | navaismo all asterisk? and what if the problem persist? cause this is the second time that i install it, last time "/etc/asterisk" was empty lol |
| 21:20.35 | igcewieling | Greenlight: func_srv seems to sort records by priority so that might not have been needed. |
| 21:20.58 | hal-j | menti_: are you following some instructions somewhere? |
| 21:21.25 | menti_ | hal-j yes, que "quick start guide" from asterisk web |
| 21:22.22 | jmetro | i suggest trying to in-depth starters guide , which starts i think right after the quick-start guide |
| 21:22.32 | Greenlight | igcewieling: They just use SRV, no roundrobin IPs. Do you just call that macro from your normal dial plan then ? |
| 21:24.26 | [TK]D-Fender | Tagor, Pastebin your peer setup. |
| 21:24.38 | igcewieling | Greenlight: do you use FreePBX? |
| 21:24.39 | dr0ck | jmetro: he needs to compile successfully first |
| 21:25.07 | Greenlight | igcewieling: Naa, just a simple raw dialplan on this box that'll need this |
| 21:25.54 | igcewieling | Greenlight: should be able to do something like AELSub(sm_dial_srv,${EXTEN},srvhost.youtisp.net,sipconfpeertodialoutusing) |
| 21:25.54 | jmetro | dr0ck the indepth guide has that. |
| 21:28.13 | Greenlight | igcewieling: Excellent, that's so much, saves me a lot of time :) |
| 21:29.07 | igcewieling | You're welcome. it has only been tested with a very limited set of situations. We plan on releasing it as GPL some day but I want to document it and clean it up first. |
| 21:30.57 | Tagor | [TK]D-Fender: here's the peer setup: http://pastebin.com/8dNxZ0J6 |
| 21:31.06 | Greenlight | igcewieling: I'll let you know how it works out :) |
| 21:31.12 | menti_ | Guys the Quick Start Guide from Asterisk is fine? or you recomend someone different? Thank you ^_^ |
| 21:31.28 | [TK]D-Fender | Tagor, This was not done properly otherwise * wouldn't be giving a 401 as it did match - Found peer 'mttm' for 'anonymous' from 89.146.63.27:5060 |
| 21:31.35 | [TK]D-Fender | Tagor, You must not ahve applied your changes |
| 21:31.42 | [TK]D-Fender | checkout time, BBIAB |
| 21:32.32 | jmetro | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#Installing_id291070 |
| 21:32.35 | jmetro | Start there. |
| 21:33.58 | Tagor | [TK]D-Fender: I did save the changes and reloaded *. Let me try if it helps when I kill * and start it again |
| 21:34.19 | menti_ | Thank you jmetro |
| 21:35.36 | jmetro | menti_ no problem..all questions should first be researched in the asterisk book i just linked you to, for future reference. =) |
| 21:35.47 | Tagor | [TK]D-Fender: this is really weird, it appears reload didn't do the job. Killing * and starting it again did :/ |
| 21:36.00 | Tagor | [TK]D-Fender: thanks for your help!! |
| 21:40.13 | menti_ | jmetro one last question if i can :3 could be a problem if i install Asterisk in an ARM? (Raspbian, Raspberry Pi) Thank you :3 |
| 21:41.28 | jmetro | I dont think so, I've heard of many different Asterisk implementations on Raspberry Pi because its so darn neat. Just make the accomodations you need to make [filesize, minimization, where to install, etc] |
| 21:42.33 | menti_ | it's a Debian OS so i think that shoudn't be problem, thank you so much jmetro :D |
| 21:42.46 | navaismo | menti_, no there is no problem i have mein with asterisk 11.2.0 on fedora remix |
| 21:47.43 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:54.56 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:59.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 22:11.08 | zemmali-voip | hi guys, please a need yours help to install Astersk11.2.1 |
| 22:11.17 | zemmali-voip | " WARNING WARNING WARNING |
| 22:11.17 | zemmali-voip | <PROTECTED> |
| 22:11.17 | zemmali-voip | <PROTECTED> |
| 22:11.18 | zemmali-voip | <PROTECTED> |
| 22:11.18 | zemmali-voip | <PROTECTED> |
| 22:11.26 | Chainsaw | Please do not flood the channel again. |
| 22:11.32 | Chainsaw | You will find yourself evicted. |
| 22:12.42 | zemmali-voip | why?? |
| 22:13.15 | malcolmd | right, so what does it say there? it's a warning, and it says the modules installed in /usr/lib/asterisk/modules are from a different asterisk build. running asterisk against modules it wasn't built for is not generally a good idea (excepting digium's commercial modules). so, move the modules that are there to somewhere else, or delete them. then, run your install again. |
| 22:15.51 | zemmali-voip | thanks malcolmd i will try |
| 22:16.54 | malcolmd | and…flooding is bad. flooding is pasting lots and lots of consecutive lines of stuff. it fills up everyone's screen and kills any other conversation threads. it's better to get to the point in as few lines as possible |
| 22:26.10 | Greenlight | igcewieling: That SRV AEL script works a treat, a little spammy to the CLI, but seems perfect in distributing calls |
| 22:27.48 | igcewieling | Greenlight: good to hear it works for someone other than myself. Yes, it is rather verbose. |
| 22:29.02 | Greenlight | Yea, my colleague's gonna love the verbosity - he generally has the CLI output on one of his screens, as a quick indicator if something bad has happened. This box handled 114k calls today :) |
| 22:29.38 | igcewieling | Greenlight: generally you want your verbosity at 1 on a system which handles that many calls |
| 22:30.18 | Greenlight | Yea, but you can't beat that Matrix-style look of stuff flying past I guess ... |
| 22:31.15 | pabelanger | zemmali-voip: How could we improve the output you seen to help you understand better? |
| 22:31.45 | Chainsaw | Greenlight: Yes, I sit there all day with my verbosity & debug at 10. |
| 22:32.10 | Greenlight | Chainsaw: Glad I'm not the only one! |
| 22:32.11 | Chainsaw | Greenlight: It gives you the all ammo you need for that "I called X and Y..." complaint. |
| 22:32.18 | Chainsaw | Greenlight: "Well actually..." |
| 22:32.28 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 22:34.16 | igcewieling | I run dumpchan in my hangup extension which helps a lot as well |
| 22:38.15 | *** join/#asterisk R2E4 (~IceChat77@24.201.174.39) |
| 22:38.27 | R2E4 | Hi all |
| 22:38.50 | R2E4 | What is the best dist to get up and running no frills..... install and use.? |
| 22:39.05 | Qwell | Whatever you're familiar with. |
| 22:39.59 | R2E4 | I use to use without a dist package. But I have used freepbx and beofre Asterisk Now. |
| 22:40.46 | R2E4 | Does it still work with the older Digium cards? |
| 22:41.53 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:42.58 | TimeRider | anyone here use sipgate.co.uk, phones registered find to em earlier today, but now... nothing... is it just me? |
| 22:43.17 | Qwell | TimeRider: call them? |
| 22:43.36 | Chainsaw | Qwell: *unobtainable tone* |
| 22:44.30 | TimeRider | qwell dunno the number, lol... never had need to.. |
| 22:44.39 | Qwell | ~google sipgate |
| 22:45.42 | *** join/#asterisk navaismo (~navaismo@189.191.199.181) |
| 22:45.43 | TimeRider | not seeing their number on site, typical lol |
| 22:45.51 | TimeRider | anyone else use them so I can rule out an issue this side? |
| 22:46.45 | Chainsaw | Don't look at me. I use ISDN from BT and gateway boxes. |
| 22:46.59 | Chainsaw | It's about call routing & internal calling from me, not outbound call pricing. |
| 22:47.19 | TimeRider | eek, bt nice and pricey :p |
| 22:48.06 | Chainsaw | TimeRider: Hey, my lines work. Look who has the last laugh... |
| 22:48.31 | lorsungcu | burn. |
| 22:49.35 | Greenlight | Hmm one problem with that AEL script, my cdr's show the dst as "~~s~~". I presume that's because it's going to the "s" extension of a macro? |
| 22:54.57 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-nwwudpqsxgubcivh) |
| 22:56.45 | Greenlight | Can I access AEL contexts and macros from the "normal" dialplan as if they existed there? |
| 22:57.27 | WIMPy | AEL ist converted to normal dialplan. |
| 22:59.09 | Greenlight | is or isnt? |
| 22:59.20 | WIMPy | is |
| 22:59.24 | Greenlight | Oh, cool |
| 22:59.30 | WIMPy | sorry. languagelapsus |
| 22:59.36 | Greenlight | Heh, np :) |
| 23:00.22 | jmetro | ist IS is |
| 23:00.27 | jmetro | =D |
| 23:00.56 | WIMPy | If you speak german. |
| 23:01.52 | igcewieling | Greenlight: use AELSub() to call AEL macros from Dialplan. Yes AEL is converted into Dialplan, but the resulting dialplan is hard to read. |
| 23:02.38 | igcewieling | Greenlight: we get our CDRs from our wholesalers so I never noticed the ~~s~~ issue in our CDRs |
| 23:10.45 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:13.50 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:18.26 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 23:19.17 | Greenlight | igcewieling: I changed the AEL so that it's a context rather than a macro, and then just use GoTo from my existing dialplan. Works perfect, and CDR's now have correct dst, which we use to bill |
| 23:19.45 | Greenlight | I can't find the syntax used for Dial documented anywhere, but it's definetly works a treat |
| 23:22.00 | Greenlight | like Dial(SIP/peer/exten/overridepeerip,args) |
| 23:22.11 | Greenlight | Handy to know that can be done |
| 23:23.11 | igcewieling | Greenlight: sip.conf.sample |
| 23:23.47 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:25.13 | Greenlight | Oh Yea |
| 23:25.39 | Greenlight | I guess I overuse google and the wiki when stuff is right under my nose :) |
| 23:26.22 | Greenlight | Ahh added in 1.8 was it |
| 23:26.22 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 23:26.35 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:36.21 | igcewieling | poor guy. we have a customer who has a queue, last agent logged out and they were all sent to voicemail, including a person who has been waiting 111 mins |
| 23:37.17 | *** join/#asterisk |Physis| (b14168fe@gateway/web/freenode/ip.177.65.104.254) |
| 23:41.04 | paulc | Someone actually waited 111 mins? wow.. |
| 23:41.10 | paulc | kinda feel for the guy :) |
| 23:44.19 | igcewieling | they have very long queue times. |
| 23:44.40 | igcewieling | I assume this means 111 mins? " 1. SIP/vmax-0000dfef (wait: 111:04, prio: 0) " |
| 23:45.31 | navaismo | :S |
| 23:45.42 | navaismo | indeed poor guy |
| 23:46.31 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 23:46.54 | igcewieling | there were 3 people waiting over 50 mins, and 3 people waiting 30-50 mins, and two waiting less than 30 mins, all would have been sent to VM |
| 23:48.24 | navaismo | i wait 5 and then hangup |
| 23:49.39 | igcewieling | they have issues with queues -- mostly of their own making |
| 23:53.20 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 23:53.23 | *** part/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 23:53.26 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 23:53.46 | Tagor | I have multiple DIDs attached to 1 SIP trunk. Is there a way to route the calls based on the DID that was called? |
| 23:55.21 | navaismo | yep use your did to route the call |
| 23:57.14 | Tagor | navaismo: When I use ${EXTEN} it just returns the first DID that's attached to the trunk, no matter what DID I call |
| 23:57.25 | Tagor | navaismo: Is there another var? |
| 23:58.15 | navaismo | usually you use the complete numbre in your context to route the call |
| 23:58.38 | navaismo | like exten=> 99886677,1,goto(gg,s,1) |
