| 00:00.00 | [o_o]owl | and -31 in the other dirction |
| 00:00.06 | [o_o]owl | I mean -2^31 |
| 00:00.17 | x86 | so it's a signed int? |
| 00:00.32 | x86 | citywok: it's not a memory limit heh |
| 00:00.37 | x86 | citywok: it's a data structure limit |
| 00:00.38 | [o_o]owl | shouldn't be but looks like |
| 00:00.51 | citywok | The number 2,147,483,647 is also the maximum value for a 32-bit signed integer in computing |
| 00:00.58 | [o_o]owl | oh |
| 00:00.59 | [o_o]owl | ok |
| 00:01.01 | citywok | http://en.wikipedia.org/wiki/2147483647 |
| 00:01.04 | [o_o]owl | lol |
| 00:01.16 | citywok | i googled the number. that was the first result :) |
| 00:01.17 | [o_o]owl | they hav entries for everything |
| 00:01.28 | x86 | *nods* |
| 00:01.28 | [o_o]owl | it's also the amount of $$ they need this year :P |
| 00:01.37 | citywok | hahahaha. is it a personal appeal? |
| 00:01.40 | [o_o]owl | heheh |
| 00:01.48 | x86 | no, Wales needs paid, so at least 10x that amount hah |
| 00:02.09 | [o_o]owl | :D |
| 00:02.14 | x86 | doesn't understand why they need so much money |
| 00:02.20 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 00:02.45 | x86 | they could get a bunch of VPS accounts and cut out hardware costs completely |
| 00:02.59 | citywok | lmao |
| 00:03.01 | x86 | and co-location costs too |
| 00:03.02 | citywok | ec2 :P |
| 00:03.07 | [o_o]owl | hehe |
| 00:03.08 | x86 | ec2 is pricey |
| 00:03.15 | x86 | but not that much :) |
| 00:03.32 | citywok | yea it's not cheap. but it's reliable and somebody elses problem! :P |
| 00:04.58 | x86 | a bunch of VPS servers rsync'ing static content to each other constantly across multiple providers throughout the world, with multiple database instances running multi-master replication -- just as reliable for thier needs |
| 00:05.17 | outtolunc | [o_o]owl: this looks like you are creating and disconnecting in under 1 sec, have you tested > 1 sec? |
| 00:05.20 | citywok | yea they don't need epic reliability. just a good load balancer. |
| 00:05.32 | citywok | and a big memcached cluster lol |
| 00:05.52 | [o_o]owl | outtolunc: no but is disconnecting within a sec not allowed by libpri ? |
| 00:06.22 | outtolunc | i am pretty sure i seen a changelog/commit about 'under 1 sec' issues |
| 00:06.28 | [o_o]owl | aah |
| 00:06.30 | [o_o]owl | nice ! |
| 00:06.34 | [o_o]owl | thanks a lot :) |
| 00:06.39 | [o_o]owl | I'll check them out then |
| 00:06.48 | outtolunc | k, let me know |
| 00:07.01 | [o_o]owl | sue |
| 00:07.03 | [o_o]owl | sure* |
| 00:11.32 | outtolunc | bbiaf |
| 00:20.33 | *** part/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 00:43.43 | [o_o]owl | this is happening with > 1 second calls as well :( |
| 00:43.48 | [o_o]owl | infact always :( |
| 00:45.18 | Freeaqingme | If in my dialplan I can do "Background(/home/kndlr/pbx/ivr/mainMenu.WAV)", why cant I do something like that over agi? |
| 00:47.43 | Freeaqingme | hmm, guess its alternative would be stream file. I dont understand however why they couldn't use the same commands |
| 00:48.51 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 00:48.53 | ghenry | ha http://www.nepalnews.net/story.php?rid=41563642&ht=Four-VOIP-racketeers-in-police-net |
| 00:50.29 | JunK-Y | Freeaqingme: or use GET OPTION |
| 00:51.32 | Freeaqingme | I c. tnx. JunK-Y but can you explain why all the dialplan commands aren't available through AGI? |
| 00:53.46 | JunK-Y | i always use STREAM FILE and GET OPTION, since they're AGI native |
| 00:54.04 | JunK-Y | EXEC BackGround /home/kndlr/pbx/ivr/mainMenu doesnt work? |
| 00:55.02 | Freeaqingme | oh cool! |
| 00:55.34 | *** part/#asterisk wizard171 (~wizard171@h25.199.91.75.dynamic.ip.windstream.net) |
| 00:56.08 | Freeaqingme | JunK-Y, works. I'm a huge noob to asterisk (reading lots of docs though), but I had missed the exec function |
| 00:57.20 | JunK-Y | k, the reason why is with background, you dont specify the file type, so it should be: Background(/home/kndlr/pbx/ivr/mainMenu) |
| 00:57.40 | Freeaqingme | yeah, found out while doing a strace |
| 00:58.36 | JunK-Y | AGIs are great, enjoy them! |
| 01:17.04 | *** join/#asterisk carndt (~carndt@173-24-26-213.client.mchsi.com) |
| 01:50.21 | *** join/#asterisk Sigfawn (42be509a@gateway/web/freenode/ip.66.190.80.154) |
| 01:53.34 | Sigfawn | good evening =) |
| 01:59.27 | *** part/#asterisk Sigfawn (42be509a@gateway/web/freenode/ip.66.190.80.154) |
| 01:59.37 | *** join/#asterisk Sigfawn (42be509a@gateway/web/freenode/ip.66.190.80.154) |
| 02:11.32 | Iron_Chef | how do i know when an extension that matches _. is incoming or outgoing? |
| 02:11.42 | Iron_Chef | i'm having troucle with the concept |
| 02:12.05 | Iron_Chef | and i can either make them dial out, or dial into an existing extension but not both |
| 02:12.12 | citywok | Iron_Chef: in the eyes of asterisk, it's always incoming, b/c the call is coming in to asterisk. |
| 02:12.53 | citywok | Iron_Chef: it's best to have an [incoming] context assigned to your peer, and a [from-phones] context for calls coming from your sip phones. |
| 02:13.05 | Iron_Chef | citywok: yes, i get that now, someone pointed that out last night, but i'm still confused as to how to tell if it has originated from the outside worlk |
| 02:13.07 | citywok | name them whatever you want. i use sipinbound and from-[staff/agent] |
| 02:13.10 | Iron_Chef | *world |
| 02:13.35 | citywok | if you set a context for a peer, you KNOW where the call originated from. |
| 02:13.50 | Iron_Chef | ah |
| 02:13.51 | citywok | unlessy ou use the same context for multiple peers, in which case it could be any of them. |
| 02:14.14 | citywok | which is why i suggest a peer for calls coming from the pstn/itsp, and another for calls coming from your internal phones |
| 02:14.18 | Iron_Chef | i don't have a context set on my vsp sip entry |
| 02:14.30 | citywok | i think freepbx uses from-pstn and from-internal |
| 02:16.02 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 02:16.19 | Iron_Chef | citywok: are those lables arbitrary if i'm using plain asterisk? |
| 02:17.08 | sawgood | I have a GREEN light on my Digium TE110 T1 PCI card (and I believe the DAHDI settings are correct), but when I call the main DID number, I see nothing at the Asterisk CLI (verbose = 5) |
| 02:17.19 | sawgood | any suggestions to see if the call is arrving at the IP PBX? |
| 02:18.30 | sawgood | I can call the T1/PRI vendor, but the less I have to be on hold with them the better! |
| 02:19.22 | citywok | Iron_Chef: yes |
| 02:19.26 | Iron_Chef | citywok: ah thanks for that bit of explanation, it was contexts :-) |
| 02:19.28 | Iron_Chef | got it now |
| 02:19.32 | citywok | you can set them to whatever you'd like, as long as you use the same one in both places |
| 02:19.43 | Iron_Chef | excellent |
| 02:19.55 | citywok | sawgood: make sure you set verbosity in the cli |
| 02:19.57 | Iron_Chef | i have it working, AND i understand why, i love this moment |
| 02:20.10 | sawgood | I have verbose 5 |
| 02:20.11 | citywok | you're welcome :) |
| 02:20.29 | citywok | sawgood: ah, okay. in that case that's the last of my knowledge with dahdi. i used zaptel a couple years ago but that's it. |
| 02:20.37 | sawgood | cool ... |
| 02:21.07 | sawgood | The IP PBX was running 1.2, but got hosed, so I rebuilt it with 1.6 today |
| 02:21.38 | sawgood | I saved the old 1.2 files (but most of them have some 'damage' from the system crashing) (some of the text was lost I think) |
| 02:24.11 | citywok | yea, having backups and an installation manual are useful :P |
| 02:26.49 | *** join/#asterisk fofware (~Fabian@host99.190-31-21.telecom.net.ar) |
| 02:32.37 | WIMPy | sawgood: 'pri set debug 2 span 1' and see if messages are going in both directions. |
| 02:34.25 | sawgood | Hi WIMPy: I'll try that right now |
| 02:35.30 | sawgood | a bunch of stuff flies by on the screen |
| 02:35.48 | sawgood | this is neat! |
| 02:35.52 | WIMPy | Both starting with < and > ? |
| 02:36.07 | sawgood | each line starts with a > |
| 02:36.20 | WIMPy | That's not good. |
| 02:36.25 | sawgood | I do not get any lines starting with a < |
| 02:36.39 | sawgood | > TEI: 0 State 5(Awaiting establishment) |
| 02:36.44 | sawgood | That is an example |
| 02:37.12 | WIMPy | You are not receiving anything then. |
| 02:37.26 | sawgood | ok ... I'll open a trouble ticket with the vendor now |
| 02:38.54 | WIMPy | I had that situation once where some (otherwise not visible) PCI conflict caused that. |
| 02:38.55 | sawgood | what is the best way to stop this debug process? |
| 02:39.07 | WIMPy | pri set debug 0 span 1 |
| 02:39.23 | sawgood | ty |
| 02:39.39 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 02:39.59 | sawgood | Well, this box was running fine (* 1.2) it crashed (I reinsalled 1.6 today) |
| 02:40.12 | sawgood | it had been running 1.2 since probably about early 2004/05 |
| 02:44.34 | sawgood | I have an open trouble ticket with the vendor now |
| 02:44.40 | sawgood | Thanks for your help WIMPy |
| 02:47.02 | [o_o]owl | WIMPy: any help with the call DISCONNECT/RELEASE/RELEASE COMPLETE issue please ? |
| 02:47.37 | WIMPy | [o_o]owl: Well, as I already said, I haven;t seen that before. |
| 02:49.07 | [o_o]owl | WIMPy: it's happening all the time with my calls - whether they're zero seconds or 30 or 60 - the RELEASE COMPLETE message always contains "Invalid call reference number" |
| 02:49.41 | WIMPy | Yes, it seems to be invalid indeed. |
| 02:50.10 | WIMPy | Have you tried another libpri? Maybe there was a bad version. |
| 02:51.26 | [o_o]owl | I've changed it to the latest as well |
| 02:51.28 | [o_o]owl | 1.4.11.5 |
| 02:51.29 | [o_o]owl | same error |
| 02:51.58 | WIMPy | Did you rebuild asterisk? |
| 02:52.34 | [o_o]owl | I'm actually not using asterisk but another application |
| 02:52.41 | [o_o]owl | perhaps I need to rebuild it |
| 02:52.46 | [o_o]owl | thanks will try and see |
| 02:53.07 | WIMPy | You probably have to. |
| 02:53.32 | [o_o]owl | k, will do :) |
| 02:53.36 | WIMPy | What are you using? |
| 02:53.45 | [o_o]owl | FreeSWITCH |
| 02:54.43 | WIMPy | I guess I could have guessed :-) |
| 02:55.06 | [o_o]owl | hehe :D |
| 02:56.44 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 03:23.43 | *** join/#asterisk sigfawn (42be509a@gateway/web/freenode/ip.66.190.80.154) |
| 03:24.21 | sigfawn | Does anyone have any experience installing 1.8 on centos 5.5 w/ cpanel? |
| 03:24.58 | raden | sigfawn, what does cpanel have to do with it ? |
| 03:25.49 | sigfawn | I've read cpanel takes over apache and which messes up asterisk and you have to use another webserver. Regardless I've read it's a huge pain in the ass to deal with asterisk and cpanel |
| 03:26.28 | *** join/#asterisk fofware (~Fabian@host99.190-31-21.telecom.net.ar) |
| 03:26.32 | WIMPy | Why would Asterisk care about your apache? |
| 03:27.17 | thehar | well no one likes cpanel |
| 03:27.21 | thehar | so i'm sure asterisk hates it as well |
| 03:27.23 | raden | my pap2t is set to register on port 5060 its doing it on 1024 anyone have a idea why ? |
| 03:27.25 | thehar | there's issue 1 |
| 03:27.35 | WIMPy | ok |
| 03:28.25 | sigfawn | No idea. I just read about running apache as the asterisk user for some install. But I have no clue. One person said they can't install it on . cPanel / WHM servers so I said ok |
| 03:28.52 | Freeaqingme | hmm, Been trying AGI for hours now. When using 'GET DATA' with my hardphone (grandstream 2020) I get no data, I thought of giving it a try with a softphone, it works right-away. What could cause that? |
| 03:29.10 | *** join/#asterisk thansen (~thansen@c-174-52-84-4.hsd1.ut.comcast.net) |
| 03:29.30 | WIMPy | raden: Define 'on'. |
| 03:29.52 | raden | <PROTECTED> |
| 03:30.00 | raden | that should be using 5060 |
| 03:31.32 | ChannelZ | That's its return port |
| 03:33.13 | raden | ChannelZ, well every other registry is at 5060 and since this happened that phone is not callable |
| 03:33.58 | ChannelZ | That's probably more of a firewall issue. But you can change the port in the SPA config |
| 03:34.09 | ChannelZ | on the Line 1 or PSTN Line tab, whatever it is |
| 03:35.08 | raden | ChannelZ, Yes sir its set to 5060 !!!! |
| 03:35.20 | raden | Fing Netgear FVX538 GRRRRR |
| 03:35.40 | raden | I think i will spend new years putting the ASA in place |
| 03:36.43 | WIMPy | Ah, still searching for challenges? |
| 03:37.20 | raden | ChannelZ, why would it be registering on 1024 ? |
| 03:37.21 | WIMPy | Trying the "Cisco challenged network"? |
| 03:37.26 | ChannelZ | I don't have a PAP so maybe it's just goofy |
| 03:37.30 | raden | WIMPy, working on my CCNA |
| 03:37.44 | raden | ChannelZ, I have over 100 of them out there never had a issue like this before |
| 03:37.50 | ChannelZ | or you've got something sniffing SIP traffic and playing games |
| 03:38.18 | WIMPy | Smells like NAT. |
| 03:38.21 | ChannelZ | or being masqueraded switching the port numbers |
| 03:38.58 | ChannelZ | I'm assuming this thing is remote to your Asterisk box |
| 03:39.36 | raden | ChannelZ, yes we have like 10 of them and today this one started going bonkers on wrong port |
| 03:40.05 | ChannelZ | I'd look at whatever router or firewall it is behind |
| 03:40.18 | ChannelZ | or it is just not configured right |
| 03:40.46 | raden | well its the one that sits at my house |
| 03:41.20 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 03:41.54 | WIMPy | And you tried using a soft phone or something round about the time it happened. |
| 03:42.09 | raden | its doing it right now |
| 03:44.27 | *** part/#asterisk lanning (~lanning@208.87.235.224) |
| 03:47.06 | raden | brb |
| 03:47.31 | *** join/#asterisk sol326 (~studyofli@c-71-56-135-99.hsd1.wa.comcast.net) |
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| 03:51.48 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 03:51.51 | raden | its fixed |
| 03:52.06 | raden | Stupid WRT54GL ROUTERS |
| 03:52.27 | raden | the only freaking one with linksys firmware on it tooo |
| 03:54.48 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
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| 04:01.20 | Freeaqingme | guys, any idea why all my phones (a softphone and an external number) work in my IVR (the GET DATA commands), except for my grandstreams? |
| 04:01.37 | *** join/#asterisk fofware (~Fabian@host99.190-31-21.telecom.net.ar) |
| 04:14.18 | xSmurf | wth... if I have Dial(X,0,d) rather than Dial(X) the state of the inbound call becomes up right away |
| 04:14.25 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 04:14.37 | xSmurf | so my monitoring thinks it's bee answered |
| 04:16.48 | xSmurf | also I can't redirect the channel anymore with those options |
| 04:16.50 | WIMPy | You can't get DTMF from the caller without answering the call. |
| 04:17.14 | xSmurf | ah true |
| 04:17.19 | xSmurf | fawk |
| 04:17.35 | xSmurf | any ideas of how else I should do thise? |
| 04:17.38 | xSmurf | *this |
| 04:18.07 | xSmurf | I have a silent ivr going in with some DTMF entry, but I also have a monitoring app that tracks those inbound calls |
| 04:18.26 | WIMPy | You can't. Put some user thing ito your CDRs or check the application. |
| 04:18.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@dsl093-155-020.phx1.dsl.speakeasy.net) |
| 04:19.36 | xSmurf | blarg |
| 04:19.45 | xSmurf | loss |
| 04:20.51 | xSmurf | thx though WIMPy |
| 04:21.24 | WIMPy | How do you get the calls? |
| 04:22.14 | xSmurf | from a sip trunk |
| 04:22.24 | WIMPy | no go |
| 04:23.00 | xSmurf | what if it were iax |
| 04:23.45 | WIMPy | That might work. |
| 04:23.48 | WIMPy | With restrictions. |
| 04:24.33 | WIMPy | You could use overlapping extensions and a timeout. But that would actuallt delay the Dial as well. |
| 04:33.19 | xSmurf | I can spare a second |
| 04:33.29 | xSmurf | but I can't get IAX at the moment :/ |
| 04:36.48 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 04:38.30 | xSmurf | also, maybe you have an idea... I'm issuing a redirect on the inbound call and right away I get a hangup event and it doesn't redirect. but it keeps ringing |
| 04:39.08 | WIMPy | To the same SIP account? |
| 04:39.41 | xSmurf | to a different internal extension |
| 04:40.02 | WIMPy | Ok, back one step: Who redirects what? |
| 04:41.25 | xSmurf | fair enough, inbound calls are directed to a context where I do some processing and then Dial() some ring-group (I used to have an exit dialplan but that's gone), some script monitors the inbound call and can issue a redirect to a different internal extension |
| 04:41.52 | WIMPy | Via AMI, I suppose? |
| 04:42.05 | xSmurf | lol shiw |
| 04:42.22 | WIMPy | sh-what? |
| 04:42.23 | xSmurf | I think it's just some moronic bug in the code, hold on |
| 04:42.28 | xSmurf | yeah via AMI |
| 04:42.37 | xSmurf | (in NodeJS :) |
| 04:43.11 | xSmurf | yeah no it still doesn |
| 04:43.12 | xSmurf | t work |
| 04:45.56 | WIMPy | I never tried to redirect a call that's not established yet, but at least from the console that doesn't seem to work for me, either. |
| 04:46.16 | xSmurf | I had it working before, not sure what's changed O.o |
| 04:46.49 | WIMPy | Interesting. Soemtimes it goes back to the original extension. |
| 04:49.04 | WIMPy | Strange things happening. |
| 04:49.40 | xSmurf | I do get a response: success too |
| 04:49.40 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 04:49.55 | WIMPy | Strange things happening when the call is established as well. |
| 04:50.15 | xSmurf | is there a better way to redirect a call? |
| 04:50.33 | xSmurf | + that's not established? |
| 04:50.37 | WIMPy | It always redirects the call to the original extension. |
| 04:50.48 | WIMPy | Smells like a bug. |
| 04:51.02 | xSmurf | I can't believe this |
| 04:51.04 | xSmurf | I had it working |
| 04:51.07 | xSmurf | hold on |
| 04:51.17 | WIMPy | Wat version are you using? |
| 04:55.39 | WIMPy | I have no idea what's going on, but I end up at the s extension now. |
| 04:57.08 | xSmurf | 1.6.x |
| 04:57.10 | WIMPy | And sip is dead now. |
| 04:57.28 | xSmurf | wow wtf my previous version of this script doesn't work anymore either!! |
| 04:58.05 | WIMPy | It's completely dead now. |
| 04:58.22 | xSmurf | I swear, I didn't do it! |
| 04:58.41 | WIMPy | Looks like the redirect did it. |
| 04:59.11 | xSmurf | this used to work, I demoed it to a half dozen people! |
| 04:59.12 | WIMPy | Could have something to do with MixMonitor, however. |
| 05:01.52 | xSmurf | WIMPy: sssssh don't tell anyone, but sometimes Asterisk really pisses me off ;p |
| 05:02.16 | WIMPy | No, removong MixMonitor makes no difference. |
| 05:02.38 | WIMPy | Well, it does so to me quite a lot, but what's the alternative? |
| 05:02.48 | WIMPy | I don't see any. |
| 05:03.31 | xSmurf | one day I'm gonna say "fuck all these endless hours I've lost making shit work with asterisk, I'll do it all over again with bluebox" |
| 05:04.23 | WIMPy | That's why I split the task. LCR for telephony, Asterisk for IP. |
| 05:04.29 | xSmurf | arggg this is really pissing me off though, it was working just fine a week ago!!! |
| 05:04.43 | WIMPy | What changed since then? |
| 05:04.55 | xSmurf | nothing that I can think of! |
| 05:05.20 | WIMPy | "nothing" sounds unlikely. |
| 05:05.48 | WIMPy | Unfortunatly it's beend quite some time since I used redirects. |
| 05:05.51 | xSmurf | or at least I reverted the only two things I did, the options for Dial() and one Set(CALLERID(name)..) |
| 05:06.53 | sol326 | hello room... |
| 05:07.00 | xSmurf | other stuff still works |
| 05:07.14 | sol326 | I need some help setting up my VIOP for the first time.... I'M VERY NEW... |
| 05:07.32 | sawgood | sol326: what version of Asterisk are you using |
| 05:07.39 | WIMPy | that's the bad thing about many features: I can take a while until you realize one of them don;t work any more. |
| 05:07.45 | sol326 | get my numbers transferred tomorrow and need to make sure I don't have any downtime |
| 05:07.58 | sol326 | 1.8 |
| 05:08.11 | sol326 | using the FreePBX interface |
| 05:08.23 | sol326 | got the preloaded centos last week |
| 05:08.40 | sawgood | oh ... you have to go to #freepbx for help then |
| 05:08.52 | sol326 | oh |
| 05:08.55 | WIMPy | sol326: You'd better see 'freepbx then. We find it pretty hard to find out what's going on insige fraapbx. |
| 05:09.12 | WIMPy | Argl. |
| 05:09.19 | *** join/#asterisk kieppie (~jaco@ip-58-28-154-35.static-xdsl.xnet.co.nz) |
| 05:09.27 | WIMPy | I think I'm getting too old to type :-( |
| 05:09.36 | WIMPy | MFBS |
| 05:10.29 | xSmurf | I reverted the dialplan to a single Dial() call |
| 05:10.36 | xSmurf | and I still get the same! |
| 05:10.53 | WIMPy | But at least your Asterisk still works. |
| 05:11.59 | xSmurf | ;/ |
| 05:18.38 | xSmurf | man this uber frustrating |
| 05:18.50 | xSmurf | I've been working on this for like two weeks |
| 05:19.12 | WIMPy | The whole voip shit is, but no way around it. |
| 05:19.56 | WIMPy | Well, maybe there is. |
| 05:20.10 | xSmurf | teletypes? |
| 05:20.16 | xSmurf | and telegrams |
| 05:20.52 | WIMPy | No, but with ever increasing costs for electricity at least voip phones become too expensive to run. |
| 05:21.01 | xSmurf | lol |
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| 05:21.23 | xSmurf | you'd be amazed how low power they could make the device if they wanted |
| 05:21.24 | WIMPy | That could give a big boomerang. |
| 05:21.37 | WIMPy | But they don't. |
| 05:21.50 | xSmurf | cause the customers are still being |
| 05:22.08 | WIMPy | And ethernet interfaces are known to consume quite some power, even if I have no idea why that is. |
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| 05:48.09 | citywok | does anybody use phoneprov+users, or do people create their own configs & use sip.conf? |
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| 05:49.25 | WIMPy | It's on my list to find out, what phoneprov actually does. Using Apache so far. |
| 05:49.41 | citywok | I just tested it out, got it working with my aastra's |
| 05:49.58 | citywok | i've never used users.conf, always just sip.conf |
| 05:50.25 | WIMPy | I used users.conf in the beginning, but don't any more. |
| 05:50.34 | citywok | any reason? |
| 05:51.11 | WIMPy | You can't put everything in users.conf, you can put in the specific configs. |
| 05:51.47 | citywok | like? |
| 05:52.21 | citywok | hmmm. according to users.conf.sample "Note that creating individual items (e.g. |
| 05:52.21 | citywok | ; custom SIP peers, IAX friends, etc.) will allow you to override specific |
| 05:52.21 | citywok | ; parameters within this file." |
| 05:52.26 | WIMPy | Can't remember what I wanted to use, but couldn't. But I think most options don't work in users.conf. |
| 05:52.38 | citywok | i wonder if that means you can specify additional parameters in sip.conf with a like named peer, or if you would have to respecify all settings |
| 05:53.01 | WIMPy | That does not sound very convenient. |
| 05:53.52 | citywok | it would be nice if you could use hte mac= parameter in sip.conf for phoneprov |
| 05:54.22 | WIMPy | Can't commetn on that (yet?). |
| 05:55.21 | WIMPy | But those parts are generated configs anyway, so it doesn't really matter to me. |
| 05:56.08 | citywok | do you use freepbx or something? |
| 05:56.13 | WIMPy | no |
| 05:56.14 | citywok | or home-brew generation? |
| 05:56.19 | WIMPy | yes |
| 05:56.31 | citywok | yea, that's how i do it |
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| 06:00.32 | citywok | i do like the phoneprov autoconfiguration, it's pretty slick if you specify the mac. |
| 06:03.28 | WIMPy | Is there good documentation on how it works somewhere? |
| 06:03.37 | citywok | not really |
| 06:03.42 | WIMPy | :-( |
| 06:03.59 | citywok | this isn't too bad https://wiki.asterisk.org/wiki/display/AST/Configuration+of+phoneprov.conf |
| 06:04.28 | citywok | some of the file structure and getting static vs dynamic files is not really well explained so it took a little bit of trial & error to make that work |
| 06:05.52 | WIMPy | That's not very informative. |
| 06:06.11 | citywok | Yea. I think i understand it now but it took me an hour to fumble my way through it. |
| 06:06.32 | citywok | I found a couple other tutorials that sucked equally as much, but filled in some gaps. |
| 06:06.38 | WIMPy | Well, that doesn't sound too bad. |
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