| 00:37.00 | *** part/#asterisk Jamuel (~Adium@c-67-180-156-186.hsd1.ca.comcast.net) |
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| 00:58.58 | alexoc | Hello! Anybody already use asterisk in vmware, with about 500 simultaneous calls? |
| 01:00.39 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-150-222.dsl.stlsmo.sbcglobal.net) |
| 01:00.49 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-150-222.dsl.stlsmo.sbcglobal.net) |
| 01:00.52 | leifmadsen | alexoc: 500 simultaneous calls for a single machine is quite a bit already |
| 01:04.48 | alexoc | How many simultaneous calls are supported in an asterisk in vmware (Assuming a hardware xeon quadcore 2.4GHz, 16GB Mem and SAS hard drives)? There are many success stories asterisk virtualized? |
| 01:09.28 | leifmadsen | alexoc: well that's a bit of a loaded question due to what asterisk is doing. Is there transcoding? Is there call recording? What is the disk i/o? etc.... |
| 01:10.36 | leifmadsen | alexoc: the only way to really know is it actually test it and see if you can really push that many calls through in your scenario with automated scripts and such (SIPp is a good tool for that kind of thing) |
| 01:12.28 | alexoc | leifmadsen: SIPp thanks a lot, I will test and report the results for you! :) |
| 01:12.37 | leifmadsen | alexoc: ok thanks :) |
| 01:15.47 | *** join/#asterisk acidfu (~nib@184.175.13.164) |
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| 01:26.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-021.mobile.uci.edu) |
| 01:31.08 | leifmadsen | Qwell: ping? |
| 01:34.33 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
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| 01:54.35 | FiReSTaRT | hey guys where can i specify for a user not to auth by extension number, but by a unique username? |
| 01:55.35 | leifmadsen | FiReSTaRT: please elaborate |
| 01:57.12 | FiReSTaRT | puts on a shameful face.. i'm trying to create extensions in freepbx (yeah yeah i know...) and i can't seem to figure out how to forbid auth by extension # as that's a security vulnerability |
| 01:58.00 | FiReSTaRT | so i wanna try to create extensions where you can only auth by a username such as gigasethome, unclealex, motherinlaw etc etc etc |
| 01:59.17 | FiReSTaRT | is considering completely doing away with that web-based crap and doing the administration the tried tested and true way - by editing config files :P |
| 02:09.26 | leifmadsen | FiReSTaRT: sorry, might have to try #freepbx, I don't know how to do things with that GUI system, just asterisk config files |
| 02:09.39 | FiReSTaRT | leifmadsen: that's what i'm looking for |
| 02:09.47 | FiReSTaRT | i gave up on trying to set it up in the gui |
| 02:09.56 | FiReSTaRT | so i wanna set one up by editing the config files instead |
| 02:12.00 | leifmadsen | FiReSTaRT: well the way authentication is done for non-ip matching is via type=user |
| 02:12.05 | leifmadsen | in sip.conf |
| 02:13.25 | *** join/#asterisk ks3_ (ks3@2600:3c02::f03c:91ff:fedf:e474) |
| 02:14.30 | FiReSTaRT | leifmadsen: thanks.. i see it.. even dug up some docs on it.. time to do a bit of reading :) |
| 02:14.39 | leifmadsen | ~book |
| 02:14.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:17.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 02:19.54 | FiReSTaRT | leifmadsen: read through it a while back but a lot of it wasn't current when i tried to get a box up and running so i gave up after a while |
| 02:20.14 | leifmadsen | FiReSTaRT: ya I updated asteriskdocs.org fairly recently to match the new book |
| 02:20.21 | leifmadsen | it was 1.4 based before that |
| 02:21.03 | leifmadsen | aha |
| 02:23.58 | FiReSTaRT | even though that's not the ideal solution, i'm willing to hack it just to move on.. still it would be nice if i could figure it out from the web interface.. not for my sake, but if i deploy it in a setting staffed by non-technical people where i just occasionally ssh in to run updates, i'd like for them to still be able to create extensions that won't be easily hackable |
| 02:25.48 | leifmadsen | FiReSTaRT: well once you figure it out on the vanilla install, then you might know better what you're looking for and can determine if freepbx can actaully generate it how you need |
| 02:26.06 | *** join/#asterisk ks3_ (ks3@2600:3c02::f03c:91ff:fedf:e474) |
| 02:29.17 | FiReSTaRT | leifmadsen: good point.. i might just do it from scratch.. haven't even touched the sip stuff other than to quickly play with it.. so far i've mostly been doing some security tweaks like disabling root ssh, altF9, set alwaysauthreject=yes, etc etc etc |
| 02:29.40 | FiReSTaRT | basically wanted to get the security down pat before playing with the fun stuff |
| 02:29.52 | FiReSTaRT | and pretty much all of that was through the cli anyway lol |
| 02:30.03 | leifmadsen | FiReSTaRT: take your existing sip.conf per freepbx and load it into your vanilla, and then tweak from there |
| 02:39.14 | *** part/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
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| 03:19.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 04:05.13 | FiReSTaRT | leifmadsen: they don't seem to offer that feature.. now i'm considering whether alwaysauthreject=yes (with strong passwds) is enough of a protection or to just forget about it and work with vanilla as you suggested |
| 04:06.08 | FiReSTaRT | and btw i just realized who you were.. thought your nick sounded familiar from another channel here lol |
| 04:37.36 | FiReSTaRT | in any case all the help is much appreciated |
| 04:57.12 | *** join/#asterisk alucardx-matt (~alucardx@75-163-11-16.chyn.qwest.net) |
| 04:57.42 | alucardx-matt | Hi everyone |
| 04:58.58 | alucardx-matt | I have a question that I cannot find a good answer to |
| 04:59.55 | alucardx-matt | Is it okay to ask it here? |
| 05:00.09 | [TK]D-Fender | Depends what it is you are intending to ask, so just ask it |
| 05:02.26 | alucardx-matt | I need to replace an old voiceworks voicemail system that is hooked up to a comdial dxp. I need to know what signals the asterisk box needs to recognize on an FXO port to answer calls that are transferred to it |
| 05:02.45 | alucardx-matt | I need to make sure I can work this before I order a telephony card |
| 05:05.01 | alucardx-matt | and, if I sound ignorant I don't mean to and do apologize |
| 05:11.38 | [TK]D-Fender | alucardx-matt: Where are you located? And clarify "transferred to it". And FXO typically "just rings" Are you referring to some unual circuit or state? |
| 05:14.29 | alucardx-matt | where am I located? like country or what? |
| 05:14.54 | [TK]D-Fender | yes |
| 05:15.06 | alucardx-matt | united states |
| 05:15.23 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 05:15.32 | alucardx-matt | so the comdial is currently interfacing to the voiceworks using a dxist board |
| 05:15.48 | alucardx-matt | the dxist is analog, I know that much |
| 05:16.24 | alucardx-matt | so calls get transferred to one of the four voicemail extensions programmed into the comdial system. |
| 05:17.24 | alucardx-matt | the voiceworks is picking up the call on one of those lines, but how does it know to pick up the line? what signals the call? I connected an analog phone to it an could hear voice both ways but the phone never rang |
| 05:17.50 | [TK]D-Fender | So you're looking to use * as a Vm system for another PBX that has signalling to take calls in and also bounce them out freeing up the channel? |
| 05:20.04 | alucardx-matt | yes, I am looking to use * as a VM system on a comdial DXP PBX. I don't know the terminology well enough to say it would free up the channel, but it would allow outside callers to leave voicemail messages on * |
| 05:20.14 | alucardx-matt | I hope I'm explaining what I'm trying to do well enough |
| 05:21.40 | [TK]D-Fender | alucardx-matt: Well the call coming in is probably just your typical dumb analog FXO which means it'd ring like normal and some interface would ahve to answer it. |
| 05:21.58 | [TK]D-Fender | Now what kind of hardware you pick is up to you as long as itll cooperate with * |
| 05:22.30 | alucardx-matt | you mean as far as the telephony card? |
| 05:23.13 | [TK]D-Fender | yes |
| 05:23.39 | [TK]D-Fender | Could be a card. Could be soe other sort of gatway device, etc |
| 05:23.48 | alucardx-matt | do you think there will be a problem with the telephony card interfacing to the ports on the DXIST voice card in the PBX? |
| 05:24.15 | alucardx-matt | I mean, what signal comes through there that will tell the FXO port that a call is coming in? |
| 05:24.25 | alucardx-matt | that's the part I don't fully understand and want to be clear about |
| 05:25.23 | [TK]D-Fender | I think you'd better find out just what that card of theirs does. |
| 05:25.43 | [TK]D-Fender | Plug a boring phone into it. Does it ring like noral? |
| 05:25.45 | [TK]D-Fender | noral* |
| 05:25.56 | [TK]D-Fender | Can you just pick up? Does it work? |
| 05:26.08 | [TK]D-Fender | Go figure all this out before juping to any other conclusion |
| 05:26.11 | alucardx-matt | no, it doesn't ring. But, if you pick it up when a call comes through that port you get voice communication |
| 05:26.26 | alucardx-matt | I'm not sure how to figure this out |
| 05:26.48 | [TK]D-Fender | So that eans it does some other electrical signalling to indicate the call. this is in "good luck" territory. |
| 05:26.56 | [TK]D-Fender | Time for some serious reading... |
| 05:27.08 | alucardx-matt | what should I read? |
| 05:27.27 | [TK]D-Fender | Information on that card and it's signalling... |
| 05:27.51 | alucardx-matt | yeah |
| 05:27.57 | alucardx-matt | I was afraid you'd say that |
| 05:28.13 | alucardx-matt | I'm having one hell of a time finding good material that is clear and concise about how this card functions |
| 05:28.18 | [TK]D-Fender | alucardx-matt: How big is this PBX of yours? |
| 05:28.18 | alucardx-matt | it's old and proprietary |
| 05:28.31 | alucardx-matt | just under 30 lines |
| 05:28.36 | [TK]D-Fender | 30 analog? |
| 05:28.37 | alucardx-matt | and it will be replaced, just not yet |
| 05:28.51 | alucardx-matt | I think it's a proprietary signaling to each terminal |
| 05:29.20 | [TK]D-Fender | How many phones? What kind of wiring? Single pair digital set? 4 pair RJ45? What kind of lines? How many sets? |
| 05:29.21 | alucardx-matt | only the four vm ports are straight analog and the others, well they go to comdial impact phones |
| 05:30.15 | alucardx-matt | single pair and I"m not sure if it's digital or not |
| 05:30.30 | alucardx-matt | I'd say about 27 lines and they are mostly going through cat 5 |
| 05:30.57 | alucardx-matt | my building to building wiring is sketchy at best which puts about 7 phones in a bad situation for digital |
| 05:31.19 | [TK]D-Fender | Little confused about "27 lines". Can you clarify Lines as TELCO line (and what signalling), and how any PHONE and what the wiring is like... |
| 05:31.28 | alucardx-matt | oh, sorry |
| 05:31.34 | [TK]D-Fender | many* |
| 05:32.14 | alucardx-matt | I have six analog lines from the telco, they go into the PBX. I have about 27 phones from the PBX and I have four lines directly to the current (and dying) voicemail system |
| 05:33.15 | [TK]D-Fender | Ok, 27 phones effectively have CAT5 from your telco room to the station? |
| 05:34.40 | alucardx-matt | yes |
| 05:34.47 | alucardx-matt | well, about 21 of those phones do |
| 05:34.57 | alucardx-matt | the others are in another building with wiring challenges |
| 05:36.10 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001688.dsl.bell.ca) |
| 05:37.28 | alucardx-matt | my ultimate plan is to overhaul the whole system with * but for now I just need functioning voicemail |
| 05:37.50 | [TK]D-Fender | alucardx-matt: Ok, well you have a proble and an opportunity. Proble is yuo have an issue for a V you have to replace, but any oney spent on that approach is wasted keeping that old crap alive and might not be reusable. Also I expect it won't work as well as their intended solutions. |
| 05:38.03 | dijib | hows everybody doing? |
| 05:38.21 | [TK]D-Fender | alucardx-matt: However your setup is far from "large" and you seemed to have it on the block to be replaced. This is a good reason not to fuck around with it |
| 05:38.42 | [TK]D-Fender | alucardx-matt: These aren't the nicest ters, but certainly motivated ones |
| 05:38.54 | [TK]D-Fender | terms* |
| 05:39.28 | dijib | if you have a server asterisk is an easy build |
| 05:39.35 | dijib | i could have mine running in 45min. |
| 05:39.50 | dijib | alucard * all night long. |
| 05:40.06 | alucardx-matt | I agree |
| 05:40.49 | alucardx-matt | well, I may have to accelerate plan for replacing the whole PBX system with * if I want to have VM then I guess |
| 05:41.09 | dijib | r asterisk is an easy build |
| 05:41.15 | dijib | how many users? |
| 05:41.30 | alucardx-matt | about 21 within reach of cat5, that's the easy part |
| 05:41.31 | [TK]D-Fender | alucardx-matt: You wanted to anyway... now you have a need for spending, better not to waste it on a dead-end patch |
| 05:41.36 | dijib | do you have an ip system currently? |
| 05:41.43 | alucardx-matt | no |
| 05:41.50 | dijib | ah i see |
| 05:42.00 | dijib | users? |
| 05:42.06 | dijib | extensions? devices? #? |
| 05:43.21 | alucardx-matt | about 27 users, give er take. I have the bulk of them within reach of cat5 to the com closet. The others are in another building across some nasty copper |
| 05:43.45 | dijib | conduit? overhead? |
| 05:43.46 | [TK]D-Fender | alucardx-matt: Where you have cat 5 forcast this : $80/phone, $350 phone switch, $800 tops worth of PSTN line interface card |
| 05:44.15 | dijib | you will need an intermediary qos/bitshaping going on. be warned |
| 05:44.18 | alucardx-matt | conduit. One end is accessible the other is buried in a wall |
| 05:44.31 | dijib | how many meters feet? |
| 05:44.40 | alucardx-matt | I was thinking of seperating this from my current data network |
| 05:44.51 | dijib | sure. |
| 05:45.01 | alucardx-matt | it's a short distance (directly) but when I test the copper it reports 624 feet |
| 05:45.23 | alucardx-matt | and getting an ethernet link is almost impossible |
| 05:45.34 | alucardx-matt | I have one device that will link up on the far end, an old 3com switch |
| 05:45.58 | dijib | wireless hop. n? a? |
| 05:46.09 | WIMPy | long reach ethernet or just sdsl. |
| 05:46.28 | alucardx-matt | I will be going wireless |
| 05:46.49 | alucardx-matt | easiest solution I think and I have experience (extensive, including tower) setting up wireless |
| 05:46.50 | dijib | you better test the spectrums before investing |
| 05:46.53 | WIMPy | Usually not a good idea for realtime applications. |
| 05:47.01 | dijib | ok then |
| 05:48.00 | alucardx-matt | and...being point to point, probably on 5.7Ghz in wyoming (sparsely populated) it should be good |
| 05:48.05 | alucardx-matt | I'll test though |
| 05:48.15 | dijib | gotcha |
| 05:48.44 | dijib | where you buiding * on an appliance, 1u? VM? |
| 05:48.59 | alucardx-matt | lu? |
| 05:49.04 | dijib | 1u |
| 05:49.05 | dijib | 2u |
| 05:49.05 | dijib | 3u |
| 05:49.06 | dijib | 4u |
| 05:49.16 | alucardx-matt | oh, that was a 1 |
| 05:49.20 | dijib | the chassis size and mounting standards |
| 05:49.29 | dijib | current server? |
| 05:49.32 | dijib | or extra box? |
| 05:49.35 | alucardx-matt | I was leaning toward a rackmount with proper power supply and everything |
| 05:49.52 | dijib | batteries. |
| 05:50.02 | alucardx-matt | maybe an extra box. I have a 64-bit amd box lying around but there would be no redundancy there |
| 05:50.03 | [TK]D-Fender | alucardx-matt: My watch could just about handle your PBX needs.... and it's ANALOG. |
| 05:50.14 | alucardx-matt | lol |
| 05:50.27 | dijib | you but for the time being if you wanted to get something up you could put one together pretty quick |
| 05:50.48 | dijib | whats the bandwidth needed for 27 concurrent calls. or how do you guys do that math #asterisk? |
| 05:50.49 | alucardx-matt | yeah, that is true and then my costs would be phones and upgrading infrastructure in to the other building |
| 05:51.12 | dijib | whats your budget? |
| 05:51.22 | alucardx-matt | I would probably have no more than 10 concurrent calls, and that's really going to be a rare high |
| 05:51.35 | [TK]D-Fender | dijib: I'd lke to know where you campe up with "27" in the first place |
| 05:51.40 | dijib | so like 200kb up and down? |
| 05:52.00 | dijib | i thought someone said it previously no? |
| 05:52.03 | alucardx-matt | yeah |
| 05:52.09 | [TK]D-Fender | No. |
| 05:52.09 | dijib | ef iirc memes also |
| 05:52.12 | alucardx-matt | I said 27 but I meant phones, total |
| 05:52.17 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 05:52.37 | dijib | ok 27 extensions. what is cool with * is wireless extensions. |
| 05:52.43 | [TK]D-Fender | And those phones are local to the building and at worst a few across some better than boring wifi bridge. |
| 05:52.43 | dijib | i like that ability |
| 05:52.46 | [TK]D-Fender | NOT an issue |
| 05:53.01 | dijib | well you could always buy a set of ATA's |
| 05:53.07 | dijib | if you want to keep budget low |
| 05:53.08 | [TK]D-Fender | And that majority of WifI phones are junk |
| 05:53.11 | [TK]D-Fender | And pricey |
| 05:53.29 | dijib | wait you running digium cards? |
| 05:53.35 | [TK]D-Fender | alucardx-matt: Careful, he's about to start selling fridges to eskimos.... |
| 05:53.37 | dijib | an ipod is a wireless voip client to me |
| 05:53.40 | dijib | good enough |
| 05:54.00 | dijib | snakeoils actually, are you interested |
| 05:54.13 | alucardx-matt | lol |
| 05:54.38 | [TK]D-Fender | dijib: Ok, you've done it.... you have just gone full-retard. BACK TO YOUR CORNER! :) |
| 05:54.51 | dijib | dude *=/VideoConferencing, my brain. |
| 05:55.09 | dijib | why is this [TK]D-Fender i havent been here in a while |
| 05:55.43 | dijib | so k lets see some schematics, bring up a whiteboard. |
| 05:55.47 | dijib | what devices where? |
| 05:55.51 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 05:56.02 | alucardx-matt | well, I will be going asterisk pbx that's no question |
| 05:56.21 | dijib | what the budget? |
| 05:56.24 | dijib | s |
| 05:56.46 | alucardx-matt | well, that's not specific. I'm simply working to beat a quote by a company selling proprietary solutions |
| 05:56.56 | alucardx-matt | I think the low quote was $14,000 |
| 05:57.46 | dijib | http://www.digium.com/en/products/analog/24-port/ |
| 05:57.53 | dijib | for legacy conversion |
| 05:58.05 | dijib | http://www.digium.com/en/products/digital/quad-span/ |
| 05:58.22 | dijib | for up to date tech, am i right? |
| 05:58.43 | alucardx-matt | but I don't know if the phones, which are comdial impacts can communicate |
| 05:58.50 | WIMPy | What makes you think he's got standard phones now? |
| 05:58.57 | [TK]D-Fender | alucardx-matt: Call up your local telco's See how much they can offer you a PRI for (either full 23 or partial to the nuber of channels you find cost effective and sufficint to your needs) |
| 05:59.19 | alucardx-matt | PRI? |
| 05:59.27 | WIMPy | And what makes you think you need a quad PRI for 10 calls? |
| 05:59.48 | dijib | future growth |
| 05:59.57 | WIMPy | Err, no. It was 6 lines, wasn;t it? |
| 05:59.59 | [TK]D-Fender | alucardx-matt: From there you'll have an anssfer for what kind of link to plan for. Then confirm your wiring. That will help pin down where you can directly place SIP phones as-is. Then consider in-line phones whre you have PC's but no suitable phone system dedicated jacks |
| 06:00.32 | alucardx-matt | so here was my plan, tell me what you think (the long term plan that is) |
| 06:00.36 | [TK]D-Fender | WIMPy: see "full-retard" :) |
| 06:00.44 | WIMPy | yes |
| 06:01.04 | WIMPy | Well for the rest of the world that would be a quad BRI card. |
| 06:01.05 | [TK]D-Fender | MORE FRIDGES!!! |
| 06:01.08 | alucardx-matt | I was going to put a patch panel in the comm closet, I have 66 blocks now. I am going to wire up all 4 pair on each CAT5 to the patch panel |
| 06:01.09 | [TK]D-Fender | MORE COWBELL! |
| 06:01.10 | alucardx-matt | from there... |
| 06:01.26 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 06:01.36 | alucardx-matt | I was going to connect a switch with POE capabilities so I can just plug in some SIP phones for each extension |
| 06:01.47 | alucardx-matt | this is for the easy rooms with good wiring |
| 06:02.21 | alucardx-matt | in the other building I was going to go with just a regular switch and power each phone at each location |
| 06:02.23 | [TK]D-Fender | alucardx-matt: that is the ideal. Polycom IP 321 is the best quality entry-level phone @ 80$~ each where you have a dedicated jack to a phone. |
| 06:02.39 | [TK]D-Fender | alucardx-matt: Ok, You seem to have a grasp on this then. |
| 06:02.40 | alucardx-matt | I'll write that down |
| 06:02.44 | dijib | yeah but thats a cost there, the phones. |
| 06:02.48 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 06:02.51 | dijib | why not slowely transition? |
| 06:02.58 | [TK]D-Fender | Just take the VM as a serious motivating push to brace management for your assessment. |
| 06:03.43 | [TK]D-Fender | dijib: because he isn't just transitioning "just because" there is a failure he has to deal with |
| 06:03.58 | alucardx-matt | right |
| 06:04.03 | alucardx-matt | the VM component failed |
| 06:04.07 | [TK]D-Fender | dijib: And that helps validate going further |
| 06:04.25 | dijib | what flying spegetti noodle cut all the lines? the infrastructure is still intact |
| 06:04.35 | dijib | the hardware phones have already been purchased |
| 06:04.45 | alucardx-matt | well, they haven't |
| 06:04.55 | [TK]D-Fender | Which was on the table for a little later. Better to move it up now and not wait for the next piece of legacy gear to break and be discovered as painful or impossible to replace |
| 06:04.56 | alucardx-matt | our hardware phones are old comdial impact |
| 06:05.17 | alucardx-matt | well, I was thinking of doing the major overhaul this summer, but hoping for voicemail now |
| 06:05.22 | dijib | phone lines last a while |
| 06:05.31 | alucardx-matt | I just need to make this wretched comdial talk to an FXO |
| 06:05.59 | [TK]D-Fender | alucardx-matt: Yeah, most proprietary system have odd signalling that standard commodity gear just won't do. This is how they own you in the end. |
| 06:06.20 | dijib | i would test that before defeat |
| 06:06.24 | [TK]D-Fender | alucardx-matt: it''s not "just" FXO". it's FXO + "just enough *special* to wreck your day" |
| 06:06.31 | alucardx-matt | yeah |
| 06:06.45 | alucardx-matt | in the manual for this gear it says these are industry standard telephone ports |
| 06:06.54 | alucardx-matt | I don't know if that's a stretch or not |
| 06:07.08 | [TK]D-Fender | the port yes... the "special sauce" however is something else |
| 06:07.12 | dijib | its a comdial system? |
| 06:07.21 | WIMPy | Just sike sip. That's also not just sip but sip plus various extensions to give you some headaches. |
| 06:07.26 | WIMPy | Some things never change. |
| 06:07.27 | [TK]D-Fender | Maybe it's wink-start or similar |
| 06:07.48 | [TK]D-Fender | You could experiment a bit but you risk buying euipment that will sit in a bin when you finally upgrade. |
| 06:08.34 | alucardx-matt | well, for just $600 to get the telephony card, and I may well use it in another location later, but if it won't work now then I won't buy it |
| 06:08.41 | alucardx-matt | I don't like the comdial special sauce |
| 06:08.42 | dijib | it is rj11 |
| 06:08.43 | alucardx-matt | damn them |
| 06:09.07 | alucardx-matt | let me send you this pdf about the comdial and see what you think |
| 06:10.20 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 06:10.24 | dijib | here you go mr president, http://www.adamtelco.com/media/catalog/product/cache/1/image/9df78eab33525d08d6e5fb8d27136e95/g/x/gxv3175.jpg |
| 06:11.52 | alucardx-matt | is it offering you a download? |
| 06:13.46 | alucardx-matt | now, the promising thing about the ports on this comdial is that I plugged in a plain analog phone and could hear voice over it |
| 06:14.00 | alucardx-matt | it wouldn't ring though which is why I question this whole vm thing |
| 06:14.28 | dijib | have you tried sending dtfm through it? dialing out? |
| 06:14.45 | dijib | ring is a little concerning |
| 06:14.46 | alucardx-matt | no go |
| 06:14.50 | dijib | no go |
| 06:14.52 | dijib | uh oh |
| 06:15.12 | dijib | im smelling a proprietary rat |
| 06:15.16 | dijib | dirty rat |
| 06:15.29 | dijib | what you guys saying? [TK]D-Fender WIMPy |
| 06:15.41 | WIMPy | What else? |
| 06:16.20 | alucardx-matt | here's a quote from the comdial manual: |
| 06:16.36 | alucardx-matt | "You can choose the DTMF digits that the DXP Plus sends to a voice mail system. A voice mail |
| 06:16.36 | alucardx-matt | system uses these DTMF digits to determine system and station status so that it can properly process a call." |
| 06:17.07 | WIMPy | Doesn't sound too bad. |
| 06:17.25 | dijib | DXP Plus is the phone? |
| 06:17.27 | alucardx-matt | do you think the original installer of this dirty system might now how this was configured and be able to help out? Further, can * be flexible with it? |
| 06:17.32 | alucardx-matt | DXP |
| 06:17.37 | [TK]D-Fender | alucardx-matt: Almost sounds like it's expecing an "always conencted" state where DTMF signals "on" and "off" |
| 06:17.53 | dijib | no i doubt that fully, im sure its on default settings |
| 06:18.13 | alucardx-matt | so can * be flexible here do you think? |
| 06:18.18 | WIMPy | Or it expects an FXS? |
| 06:18.26 | alucardx-matt | that crossed my mind |
| 06:18.36 | alucardx-matt | would a handset work to the port for voice though? |
| 06:18.53 | WIMPy | Yes, but only for voice. |
| 06:19.03 | WIMPy | No dialling or ringing. |
| 06:19.13 | alucardx-matt | well it's not getting dialing or ringing |
| 06:19.20 | alucardx-matt | so you think it expects fxs? |
| 06:19.39 | WIMPy | But if you connect a phone to your VM thing, dialling may cause something to happen. |
| 06:20.03 | alucardx-matt | I can try that |
| 06:21.17 | alucardx-matt | so in theory I should be able to dial connected to the VM box if it is fxs? |
| 06:21.42 | WIMPy | Yes. |
| 06:22.03 | WIMPy | But you don't know what to dial, so it's a bit hit or miss. |
| 06:22.24 | WIMPy | You could try to listen in while the two are connected and talking to each other. |
| 06:22.37 | [TK]D-Fender | alucardx-matt: Mine was a guess... there may be more, and this is a tricky thing.... |
| 06:22.55 | alucardx-matt | you're telling me |
| 06:23.09 | alucardx-matt | I very much dislike proprietary things |
| 06:23.32 | WIMPy | Who doesn't? |
| 06:23.45 | [TK]D-Fender | The souls of the bought :) |
| 06:23.51 | WIMPy | Althoug I have to admit that they can do a very good job sometimes. |
| 06:24.11 | alucardx-matt | yeah, but it means little when it's controlled by someone else |
| 06:25.53 | alucardx-matt | well, I'll give it my best shot. If I can't get it to work I'll have to accelerate plans for overhauling the whole system |
| 06:28.04 | alucardx-matt | how much can * log about incoming DTMF once I have this system plugged in? |
| 06:28.14 | alucardx-matt | would I then be able to see what asterisk needs to respond to? |
| 06:28.19 | alucardx-matt | then write my dialplan from that? |
| 06:30.33 | [TK]D-Fender | alucardx-matt: Go read up on that cicuit spec. I'm not sure if * has a way to keep it always up & listening |
| 06:32.28 | alucardx-matt | finding good docs, that's the hard part |
| 06:32.40 | alucardx-matt | I've been doing nothing but trying to find good docs on that circuit |
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| 06:34.38 | alucardx-matt | listen though, I really appreciate all of your help |
| 06:35.29 | alucardx-matt | I think I'll come back to this channel again sometime |
| 06:35.58 | WIMPy | You can quit any time you like, but you can never leave. |
| 06:36.14 | alucardx-matt | nice |
| 06:36.21 | alucardx-matt | wait a minute though |
| 06:36.42 | alucardx-matt | so let me learn a bit about you guys. what are your professional backgrounds? |
| 06:37.12 | WIMPy | Nerd |
| 06:37.25 | alucardx-matt | I am too! |
| 06:37.26 | alucardx-matt | lol |
| 06:38.18 | WIMPy | just took the wrong way somewhere. |
| 06:39.13 | alucardx-matt | do you work with phone systems professionally or something else? |
| 06:39.24 | WIMPy | Sometimes. |
| 06:40.47 | alucardx-matt | well, I'm just a nerd too |
| 06:41.02 | alucardx-matt | most recently into phone systems but I've had many other little projects as well |
| 06:41.20 | alucardx-matt | finished a samba server not too long ago for work |
| 06:41.23 | alucardx-matt | that was fun |
| 06:41.45 | WIMPy | Asterisk will take over your whole life. |
| 06:42.12 | alucardx-matt | well, I find it to be really bad ass |
| 06:42.25 | WIMPy | In all possible ways. |
| 06:43.09 | alucardx-matt | I'm a big fan of free software too |
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| 06:46.08 | alucardx-matt | well guys, I think I'm going to go. Thanks again for all of your help and input |
| 06:47.11 | [TK]D-Fender | checkout time, later all |
| 06:47.48 | alucardx-matt | later |
| 06:47.52 | alucardx-matt | oh, he left |
| 06:47.56 | alucardx-matt | er quit |
| 07:06.12 | jmwpc | I was hoping someone could help me with an X100p FXO card... I suspect an interrupt issue. I have changed pci slots, it's on its own IRQ, i have set smp_affinity so it is on a different cpu core than the drive controllers, and tried changing LATENCY_TIMER. http://pastebin.com/Tjm4BtC7 --- I can't make or receive calls. |
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| 07:16.59 | IsUp | morning |
| 07:17.45 | IsUp | is that possible to use different external IP per peer? |
| 07:18.38 | kaldemar | IsUp: no |
| 07:18.40 | IsUp | a provider gaves us a fiberoptic link to their end. they assigned a public IP for us. its called "vpn voip" i think. |
| 07:19.26 | kaldemar | IsUp: external IP as in the setting when you are behind NAT. it is a general setting. |
| 07:19.47 | IsUp | kaldemar: yes i already have an external IP setting. but it dnesnt match on the provider |
| 07:20.00 | IsUp | kaldemar: because they want their IP in SIP packets |
| 07:20.17 | IsUp | kaldemar: so i cant change my external ip for them. i am connected to other providers/peers |
| 07:21.36 | kaldemar | then you need to avoid setting the external ip in calls that go through a VPN to them. try to add the addrss space in a localnet option. |
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| 08:07.08 | Diffen | Hello. When im trying to do a attendant transfer (im using ## but *# is the default way of doing it) to an extension that have no registererd device connected to it i get chanunavail in the asterisk cli. Is it possible to let me override that so the call is transfered anyway. We are using follow me a lot and a couple of users doesnt have registered devices so its not possible to transfer a call to them using ##. |
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| 09:29.42 | nunne | Im getting : "__sip_xmit: sip_xmit of 0x1fb5be0 (len 835) to 192.168.100.254:5060 returned -1: Operation not permitted" when dialing a group on my embedded asterisk 1.4 |
| 09:30.15 | nunne | the group has 9 members. and it seems i get this error on 4-5 of the members. and this in return makes it impossible for them to answer |
| 09:31.38 | nunne | does anyone know what this means?? 9 members shouldnt be alot.. and the only think i can think of that is ALOT of the subscriptions (19 phones subscribing to each each of these members).. Could there be something there? and is it possible from the dialplan to disable notifyringing etc?? |
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| 09:32.56 | FlashDeluxe | hi! i got a question, whats the best way to reroute my outgoing callers over sipgate if all lines are busy? |
| 09:35.45 | nunne | FlashDeluxe: Using ${DIALSTATUS} |
| 09:36.00 | nunne | if it's busy just jump to another label and dial sipgate from there |
| 09:36.27 | FlashDeluxe | nunne: ahh ok thank you :) |
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| 10:10.16 | mechbangirc | I lost link of a free web service. You feed it email, ip address and time duration of test (from hours to weeks) and it would at the end of test send you detailed report. Any idea? |
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| 10:18.18 | kaldemar | mechbangirc: by all means, do tell what it tests. |
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| 10:19.02 | mechbangirc | kaldemar: connection health for voip |
| 10:19.35 | mechbangirc | I used it 2 years ago, now I have no idea where to find it |
| 10:20.04 | mechbangirc | it would give detail reports with graphs, nice presentation |
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| 11:05.48 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 11:06.41 | eduzimrs | Hi, anyone knows this issue when I perform an "Inconditional transfer" on xlite? "Span 1: Channel 0/2 got hangup request, cause 127" the call hangs up |
| 11:14.09 | kaldemar | eduzimrs: that's not an issue by itself, only a hangup request with cause 127 which is an indication of end of an interworking call. |
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| 11:35.05 | eduzimrs | kaldemar: ok but why its hanging up my call? I need call forwarding by the clientes like to an cel phone for example. |
| 11:36.55 | eduzimrs | kaldemar: weird cause, when I set up to this number for example (08006340014) it works fine, but when its to a phone number it hangs up the call |
| 11:37.11 | eduzimrs | *cel phone |
| 11:37.55 | kaldemar | eduzimrs: hanging up what call? you're not giving much of a description of the issue. |
| 11:38.00 | eduzimrs | kaldemar: something about the number of digits dialed ? |
| 11:38.30 | eduzimrs | ill show u |
| 11:41.58 | eduzimrs | kaldemar: http://pastebin.com/b5KnWKU0 |
| 11:42.25 | eduzimrs | kaldemar: take a look at line 7 |
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| 11:43.45 | eduzimrs | app_dial.c: Not accepting call completion offers from call-forward recipient Local/88292571@internacional-1b5d;1 |
| 11:43.56 | kaldemar | and? still not seeing an issue. |
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| 11:44.19 | eduzimrs | kaldemar: im still learning about |
| 11:44.50 | eduzimrs | kaldemar: I don't know whats that mean |
| 11:47.09 | eduzimrs | kaldemar: something about call transfer in dahdi channel |
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| 11:48.28 | kaldemar | eduzimrs: that has nothing to do with any DAHDI channel. it's just a notice about call control tracking and completely normal. |
| 11:49.32 | eduzimrs | kaldemar: ok, but did u understando that the call does not work? saw the logs? |
| 11:49.55 | eduzimrs | kaldemar: in the end of log the call hangs up |
| 11:50.24 | eduzimrs | kaldemar: it works to other numbers |
| 11:50.43 | eduzimrs | kaldemar: but to cell phones number doesnt |
| 11:51.06 | kaldemar | there are no issues in what you showed me. and nothing that would even match your original description. |
| 11:53.06 | eduzimrs | kaldemar: if u see the log you`ll realize that there is no Ring statement |
| 11:53.31 | eduzimrs | kaldemar: there is just the dial |
| 11:53.50 | eduzimrs | "-- Local/88292571@internacional-1b5d;1 is making progress passing it to SIP/411-0000000b" |
| 11:54.18 | eduzimrs | and does not ring, It hangs up the call after a few seconds |
| 11:56.48 | nunne | is the number in the correct syntax? |
| 11:57.13 | nunne | how does the dialplan for internacional look |
| 11:57.22 | eduzimrs | sure its always worked before |
| 11:58.04 | eduzimrs | same => n,Dial(${TRUNKGVT}/${EXTEN},60,TwW) |
| 11:58.13 | eduzimrs | that's the dial |
| 11:58.20 | kaldemar | eduzimrs: enable sip debug for a call and you'll see what there is. those verbose prints are not the whole truth, not even close. |
| 11:59.03 | eduzimrs | ok |
| 11:59.05 | nunne | and 88292571 is a valid number? you dont need to add 0, 00XX before it? |
| 11:59.15 | kaldemar | then you might also enable debug for your DAHDI side of the call to see what actually happens, and pastebin a WHOLE call. |
| 11:59.57 | eduzimrs | nunne: no, its a valid exten |
| 12:00.01 | eduzimrs | kaldemar ok |
| 12:09.24 | eduzimrs | kaldemar: http://pastebin.com/ihpMsVcm |
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| 12:24.09 | kaldemar | eduzimrs: and what is the issue? who calls who and who gets hung up? |
| 12:25.10 | kaldemar | 411 calls 485 who has forwarded calls out of your box and 411 gets a hangup? |
| 12:26.36 | kaldemar | next time paste what you see in CLI. easier to read without all the extra stamps. |
| 12:29.20 | kaldemar | all i see is an incoming DISCONNECT with cause 16 which is normal clearing. |
| 12:30.15 | eduzimrs | 411 calls 484 who forward to dahdi/88292571 |
| 12:30.23 | eduzimrs | 485* |
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| 12:34.15 | Lantizia | Anyone with experience of getting a patton smartnode to send calls to an asterisk system as a peer that's already registered? (asterisk keeps sending back Unauthorized for some reason) |
| 12:34.42 | kaldemar | eduzimrs: and whatever is behind the DAHDI and that number just hangs up. there's nothing you can do about it if the number is correct. |
| 12:35.37 | kaldemar | Lantizia: it's supposed to send unauthorized when authentication is required. does the smartnode not answer that with a new invite with credentials? |
| 12:36.41 | Lantizia | kaldemar, well it's really odd... if I tell the patton to use the username in the from field the call passes to asterisk dial plan - but i don't want that i want the patton to send whatever callerid it wants |
| 12:38.05 | Lantizia | kaldemar, so in short yeah I think it's authing - so i'm not sure why asterisk is saying unauthorised |
| 12:39.06 | kaldemar | then it matches something you don't expect it to or nothing at all. a call attempt with verbosity and sip debug will tell. |
| 12:39.22 | Lantizia | kaldemar, does this help? http://pastie.org/3262938 |
| 12:39.38 | Lantizia | that's all I get when I make an outbound call _to_ asterisk _from_ the patton |
| 12:39.54 | Lantizia | the patton has registered with asterisk as a peer perfectly OK and can receive calls normally |
| 12:41.12 | Lantizia | kaldemar, from that log it's obvious that asterisk _knows_ it's come from Grenke (the peer name in sip.conf) but still disallows it... but if the from: field is Grenke@192.168.121.2 (that ip is the patton) instead of callerid@192.168.121.2 it passes through |
| 12:46.45 | *** join/#asterisk nonlinearly (~nonlinear@relate.ath.forthnet.gr) |
| 12:46.58 | nonlinearly | Hi |
| 12:47.49 | nonlinearly | we have an alcatel pbx with 300 lines to pstn and we want to get rid of it |
| 12:48.01 | nonlinearly | so we want to move to Voip |
| 12:48.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:48.16 | nonlinearly | but I have a question |
| 12:48.56 | nonlinearly | Microsoft Lync Server and asterisk has the same target group |
| 12:49.13 | nonlinearly | Microsoft Lync Server and asterisk have the same target group |
| 12:49.17 | nonlinearly | ? |
| 12:50.20 | kaldemar | Lantizia: a previous registration from a device has nothing to do with authentication of actual calls. they are two separate things. |
| 12:51.19 | kaldemar | Lantizia: after that 401, the smartnode is supposed to send another invite that has a WWW-Authenticate-header with credentials. that's how authentication works. |
| 12:52.11 | Lantizia | kaldemar, hmm ok so it is just a question of finding that setting in the patton - somewhere |
| 12:52.19 | kaldemar | nonlinearly: not knowing what "target group" means, i'll just say that lync and asterisk are not comparable. |
| 12:52.40 | nonlinearly | market maybe? |
| 12:53.24 | nonlinearly | does it make any sense the comparison between Lync Server 2010 and Asterisk? |
| 12:53.31 | kaldemar | Lantizia: yes. or if you have a closed environment, you can always turn authentication off on the asterisk side with insecure=invite. but it's definitely better if you get the smartnode to authenticate properly. i used one many years ago and don't remember having trouble with it when configured. |
| 12:54.01 | kaldemar | nonlinearly: as much as any other comparison. that depends on your needs. |
| 12:54.54 | *** join/#asterisk Gaiax (beb64b33@gateway/web/freenode/ip.190.182.75.51) |
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| 12:55.31 | nonlinearly | I mean that if someone search for Voip solution can the Lync Server 2010 be such a solution? |
| 12:56.33 | nonlinearly | As the asterisk is? |
| 12:57.33 | kaldemar | "voip solution" is quite ambiguous. list your requirements for a system and you'll have easier time comparing solutions. |
| 12:58.34 | nonlinearly | So the Lync Server 2010 maybe is an alternative to Asterisk... |
| 13:01.45 | nonlinearly | Are Lync Server 2010 and Asterisk combeting products? |
| 13:02.04 | kaldemar | they are very different. |
| 13:02.39 | kaldemar | lync is more like a service, asterisk is a telephony toolkit that can be used to build solutions. |
| 13:05.14 | nonlinearly | Ok thanks Kaldemar... So can not someone choose the one or the other... |
| 13:05.59 | nonlinearly | like windows or linux... |
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| 13:16.47 | nonlinearly | this question (Lync Server 2010 VS Asterisk) has a sense for us because we will change our CRM (Siebel) to Microsoft Dynamics CRM that we know there are smoothly colaboration |
| 13:17.27 | nonlinearly | with Lync |
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| 13:34.31 | vassilux | hi, I use B410P with dahdi, my telco brings layer 2 and layer 1 down on BRI lines on idle. I found HAVE_PRI_L2_PERSISTANCE(chan_dahdi.c). Can I activate this option to keep my lines UP ? :-) If yes How I can do it ? |
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| 14:16.47 | Katty | drags in |
| 14:30.45 | singler | drags Katty out |
| 14:34.07 | Alborracho | Has anyone know what this means? WARNING[10163]: l4isup.c:442 mtp_enqueue_isup_packet: MTP send fifo not ready, lsi=0 I lost signaling few days ago and i cant recover it |
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| 14:59.18 | tech | good mornign/afternoon all.. |
| 14:59.44 | tech | i am trying to implment asterisk (freepbx install) to avaya r5.1 |
| 15:00.02 | tech | i am able to establish a chanenel over my signaling group but i am unable to make a call in either direction. |
| 15:00.45 | tech | on the avaya side I am getting 1191 Network Failure |
| 15:00.50 | tech | when trying to call |
| 15:00.54 | kaldemar | ~freepbx |
| 15:00.54 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:00.59 | tech | oh thnx |
| 15:01.01 | tech | and very sorry |
| 15:01.31 | kaldemar | it's ok. that's common. |
| 15:01.45 | [TK]D-Fender | And that isn't an * error and this isn't #avaya. |
| 15:01.59 | *** join/#asterisk wheeler_32 (~wheeler_3@macauth-160-245.resnet.mtu.edu) |
| 15:02.18 | [TK]D-Fender | So lets leave "FreePBX" out of this for a while. It's ace-able on so many other terms already :) |
| 15:02.57 | wheeler_32 | Any other users in here from SAT4240? |
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| 15:46.32 | TimeRider | uh oh |
| 15:46.57 | Curdie | Morning all. I have a newbie problem, but I've read everything and I cannot solve it on my own. I've created a call file http://pastebin.com/GNvvmrey which I copy to /var/spool/asterisk/outgoing to play a system recording on a schedule. It's worked for months, but now it's calling and not playing anything. I don't think I've changed anything that should effect it. I don't know what to do, or |
| 15:46.57 | Curdie | even what to check. |
| 15:47.16 | Curdie | The extension works if I call it manually |
| 15:47.42 | [TK]D-Fender | Curdie, first you never "copy" a call-file in, you need to MOVE it from within the same filesystem |
| 15:48.34 | Curdie | Ok, thanks. I'll do it that way and see if it helps. |
| 15:48.37 | [TK]D-Fender | Curdie, On the 2 sides of "next", you need to be looking at * CLI to see what it is actually doing when you do do this, and next you should asing in #freepbx as that's what the context would generally indicate you're using. |
| 15:49.33 | Curdie | Mm. I am looking at the CLI and had some questions, but I'll take them over to #freepbx. Thanks! |
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| 16:01.36 | wheeler_32 | Kind of new at this but how do I check the configuration of or status of a hardware card? |
| 16:02.27 | WIMPy | What card? |
| 16:02.40 | wheeler_32 | Digium TDM411B. |
| 16:04.01 | WIMPy | dahdi_hardware was tehe a dahdi_status or somethign, or cat /proc/dahdi/* |
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| 16:18.53 | Katty | Nugget: and telnet. |
| 16:18.54 | Nugget | telnet is eeeeeeevil! |
| 16:19.03 | Katty | Nugget: we should be able to telnet into cars |
| 16:19.34 | WIMPy | Like in to Airplanes? |
| 16:21.04 | Nugget | heh |
| 16:21.09 | Nugget | huggles katty |
| 16:21.34 | [TK]D-Fender | Katty, http://www.snopes.com/humor/jokes/autos.asp |
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| 16:35.59 | anonymouz666 | Katty: I am able to see how my car is working through my computer :-) |
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| 16:40.01 | Katty | hugs Nugget |
| 16:40.09 | Katty | anonymouz666: :< |
| 16:40.11 | Katty | err |
| 16:40.13 | Katty | anonymouz666: :> |
| 16:45.05 | jsarrel | I need some help or guidance debugging a PRI. In a nutshell, incoming works fine, outgoing is the problem. |
| 16:45.15 | jsarrel | pb of the pri debug http://pastebin.com/3MjCJzFC |
| 16:46.05 | jsarrel | When you try to make a call, it hangs up immediately. It has worked flawlessly for several years now and all of sudden this happens with no changes made to the system. |
| 16:46.33 | WIMPy | Did you pay the bill? |
| 16:46.34 | jsarrel | Windstream is telling us that no dtmf is being sent once the line picks up. |
| 16:46.44 | jsarrel | haha...i'm sure they have |
| 16:46.57 | WIMPy | What would DTMF do on a PRI? |
| 16:47.19 | jsarrel | Ok, maybe not DTMF, but they arent getting the number to be dialed. |
| 16:47.52 | WIMPy | yes, that's what your debug says as well. |
| 16:47.53 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 16:47.57 | WIMPy | called number '' |
| 16:50.09 | jsarrel | hmm, I'm working on cleaning up the full log a little bit, but I do indeed see that * is getting the number |
| 16:50.18 | jsarrel | so i guess it's getting lost between the two |
| 16:50.46 | WIMPy | It's not sending any. As if you Dial(dahdi/g1/) |
| 16:51.33 | jsarrel | [Jan 27 11:07:58] VERBOSE[16760] logger.c: -- Called G0/wXXXXXXXXXX |
| 16:51.43 | jsarrel | replace the x's with my cell number |
| 16:51.49 | WIMPy | w? |
| 16:51.59 | WIMPy | What does that do there? |
| 16:52.04 | [TK]D-Fender | <PROTECTED> |
| 16:52.04 | [TK]D-Fender | q931.c:3134 q931_setup: call 32855 on channel 6 enters state 1 (Call Initiated) |
| 16:52.13 | [TK]D-Fender | jsarrel, Called number = BLANK |
| 16:53.02 | WIMPy | But still interesting that the call gets disconnected. I'd expect you'd hear a dialtone. |
| 16:53.35 | jsarrel | according to windstream, that is why they are disconnecting is because it's blank |
| 16:53.35 | WIMPy | Anyway. Get rid of that w. |
| 16:53.57 | WIMPy | That's no reason. |
| 16:54.00 | [TK]D-Fender | w is an ANALOG delay |
| 16:54.01 | jsarrel | k |
| 16:54.13 | WIMPy | But it's a reason for not reaching anyone. |
| 16:54.22 | [TK]D-Fender | That is a reason. Just not one valid for anything other than DAHDI analog channels |
| 16:54.47 | jsarrel | Full log, tried to clean out other calls http://pastebin.com/J2b037ps |
| 16:55.45 | WIMPy | smells some freepbx |
| 16:55.49 | jsarrel | lol, maybe |
| 16:56.00 | [TK]D-Fender | jsarrel, -- Executing [s@macro-dialout-trunk:19] Dial("SIP/sip.vowdata.com-08d4ea50", "ZAP/G0/w8647063524|300|wW") |
| 16:56.11 | WIMPy | The go to #freepbx and ask there how to get rid of that w. |
| 16:56.11 | [TK]D-Fender | jsarrel, Still a "w" in front |
| 16:56.24 | jsarrel | that was from earlier |
| 16:56.34 | [TK]D-Fender | jsarrel, that is from the PB you just gave us |
| 16:56.37 | WIMPy | It's nonsense. |
| 16:56.57 | [TK]D-Fender | Stop showing us the past. This isn't the Time Travel Network. |
| 16:57.10 | [TK]D-Fender | That show is on at 11. |
| 16:57.14 | jsarrel | No settings have changed between now and then. |
| 16:57.22 | [TK]D-Fender | Every eleven. Ever. They're that good... |
| 16:57.31 | [TK]D-Fender | jsarrel, Show us a real call now. |
| 16:57.35 | WIMPy | isn't so sure about the time travel. I often feel like that here. |
| 16:57.57 | [TK]D-Fender | WIMPy, Regrettably 1s/s here.... |
| 16:58.40 | WIMPy | I've often got this "retro" feeling here. |
| 16:59.11 | jsarrel | "w" seemed to do it. Now I know why you guys get paid the big bucks ;) |
| 16:59.17 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 16:59.40 | jsarrel | I don't understand why it seemed to break it after it has been working for so long. |
| 16:59.45 | WIMPy | And where did that come from? |
| 16:59.56 | [TK]D-Fender | Analog <- |
| 17:00.02 | jsarrel | Most likely been there from the start. |
| 17:00.04 | [TK]D-Fender | that's a 500ms pause before dialing |
| 17:00.14 | [TK]D-Fender | "been there from the start' really doesn't say anything |
| 17:01.10 | jsarrel | In other words, it was there last week and worked fine up until now. |
| 17:01.21 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:01.35 | WIMPy | Until upgrading something? |
| 17:02.07 | jsarrel | Nope, we haven't heard from them in a long time. If it isn't broke, I don't fix it. |
| 17:02.32 | jsarrel | Maybe a setting or something changed on the other end? Timeout for waiting for digits perhaps? |
| 17:02.38 | jsarrel | Other end being Windstream |
| 17:02.40 | WIMPy | is pretty sure it didn't change by itself. |
| 17:03.23 | WIMPy | You didn't send any digits, so there is nothing they could use. |
| 17:05.15 | *** join/#asterisk Psych0 (~quassel@host-173-247-25-174.JENOLT1.epbfi.com) |
| 17:05.51 | Psych0 | Hey guys, is there any way to implement allow deny rules for SIP connections like in apache? |
| 17:06.17 | jsarrel | Welp, now I know to be on the look for the w on non analog. THANK YOU VERY MUCH FOR THE HELP!!!! |
| 17:06.23 | WIMPy | Psych0: Have you ever taken a look at the config file? |
| 17:07.03 | Psych0 | :facepalm: Actually, I didn't think to look thorugh it and look at the comments |
| 17:07.11 | WIMPy | jsarrel: I'd call it a bug that it's even possible. |
| 17:07.20 | Psych0 | sheepishly goes to the SSH terminal |
| 17:08.09 | Katty | aww |
| 17:08.15 | Katty | hugs Psych0 |
| 17:08.18 | Katty | Psych0: there there, it's ok. |
| 17:08.25 | Katty | Psych0: they are often brash here at times, don't take it personally |
| 17:09.01 | Psych0 | No, I actually feel dumb for not thinking of that first |
| 17:10.09 | carrar | BRASS MONKEYS |
| 17:10.23 | carrar | It's Funky Funky |
| 17:11.07 | Psych0 | OK, confession time.... I'm running a Trixbox..... could someone give me a pointer as to the config file (or menu item) at which I need to be looking? |
| 17:11.18 | carrar | #trixbox |
| 17:11.58 | Psych0 | Nobody answers there. |
| 17:12.26 | Psych0 | Hence, I choose the wrong channel and straighten my shoulders for whatever beating is administered |
| 17:12.26 | xavia | when you edit the sip phone.. there is a field there for it. |
| 17:12.48 | xavia | trixbox just uses freepbx |
| 17:13.09 | WIMPy | Beating? Ok. |
| 17:13.13 | WIMPy | ~trixbox |
| 17:13.13 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 17:13.14 | navaismo | Psych0: Permit/Deny fields |
| 17:13.17 | [TK]D-Fender | No, trixbox uses an outdated version of FreePBX they've FORKED from the main |
| 17:13.18 | WIMPy | ~freepbx |
| 17:13.18 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:13.37 | xavia | [TK]D-Fender, thats great. don't care. |
| 17:14.05 | Psych0 | WIMPy: I was being metaphorical. You guys have been most helpful in the past |
| 17:14.58 | Psych0 | And the explaination as to why is helpful now |
| 17:15.26 | [TK]D-Fender | xavia, Correct... the people in #freepbx won't care... it isn't their problem either... |
| 17:15.36 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 17:15.49 | xavia | and I am trying to help him myself in this channel where he asked the question |
| 17:15.51 | xavia | so if you don't want to help him? don't answer? |
| 17:16.42 | navaismo | cover eyes |
| 17:16.47 | [TK]D-Fender | xavia, and I was correcting a singular piece of misinformation given here. Who said I was talking to him? |
| 17:17.03 | Micc | Does anyone have experience with yealink t28 remote provisioning and sip notify? My phones seem to only check for the config when I hit auto provision, reboot and sending sip notify don't make it check even though it is set to check on reboot. |
| 17:17.10 | [TK]D-Fender | What's all this talk about narcissism ... and what does it have to do with ME? |
| 17:17.13 | [TK]D-Fender | :) |
| 17:17.58 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:18.16 | xavia | they ship an outdated version of freepbx.. but I have updated through the web interface freepbx from inside trixbox.. so it's hardly a fork.. just a skinned version of the normal freepbx. |
| 17:18.40 | p3nguin | We don't support Trixbox nor FreePBX here. |
| 17:19.23 | Qwell | xavia: The skinning requires source changes. If it's still skinned, it's not FreePBX. |
| 17:19.31 | [TK]D-Fender | xavia, xaviaNo, it is a fork, long documented over the years as Fonality didn't like the FreePBX's development pace. |
| 17:19.32 | Psych0 | TBH, I didn't know if it was something that could be done at the text config (base asterisk) level |
| 17:19.33 | xavia | ok ok fine I will stop |
| 17:19.47 | Qwell | PS, we don't support FreePBX here anyways, so your point is completely moot. |
| 17:19.49 | Psych0 | Sorry for the aggravation. |
| 17:20.06 | [TK]D-Fender | Psych0, I still wasn't talking about you :) |
| 17:20.37 | p3nguin | psych0: If you control the system using the GUI, the GUI controls Asterisk's config files. |
| 17:20.41 | [TK]D-Fender | Psych0, but do look if they offer "permit/deny" in your extnsion config. You have have to enable some sort of "advanced settings" option for it to be listed if it is even offered |
| 17:20.46 | navaismo | Psych0: look for fields like permit deny in the extension or sip settings if you have it |
| 17:20.56 | [TK]D-Fender | may have* |
| 17:21.33 | Psych0 | Thx guys. also [TK]D-Fender, I take very little personally. |
| 17:21.43 | Psych0 | Except my own ineptitude. |
| 17:21.47 | Psych0 | :-D |
| 17:22.38 | [TK]D-Fender | Psych0, So if you wanted to restrict the who server (not just per extension), then that is IPTABLES job, not even *'s (and your GUI by proxy |
| 17:22.51 | [TK]D-Fender | whole* |
| 17:23.30 | *** join/#asterisk skrull (~skrull@189-10-230-246.smace701.dsl.brasiltelecom.net.br) |
| 17:23.33 | Psych0 | Gotcha.... BTH, I was hoping to throw that info into SIPDefault and let it ride per extension |
| 17:24.34 | p3nguin | How can I stop the jabber messages "Got presence packet from PEOPLE I HAVE NEVER HEARD OF, someone not in our roster!!!!"? It is periodically flooding the console and I want to make it so I do not receive those packets. |
| 17:25.19 | [TK]D-Fender | Psych0, last I checked permit/deny were peer-only options, not usable in [general] |
| 17:25.31 | jsarrel | Psych0: have you looked at fail2ban to ban brute force attempts? |
| 17:25.55 | Psych0 | I'm not seeing permit/deny or advanced options anywhere in the new extension screen.... looks like it's time to dig into IP tables rules |
| 17:26.25 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 17:26.30 | [TK]D-Fender | Psych0, Psych0They may be there but hidden. There is an "advanced features" flag in the admin for FreePBX. See if their's has it |
| 17:26.42 | Psych0 | jsarrel: No, I'm about as green as they come. I got SSH and httpd access under control and thought I had it "good enough" |
| 17:26.50 | xavia | Psych0, they won't show up until after you added the extension |
| 17:27.50 | Psych0 | OH!, I thought it was avaiable at th "add extension" screen |
| 17:28.02 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 17:28.25 | [TK]D-Fender | Psych0, No, look at one you have already |
| 17:29.49 | Psych0 | I see it now. |
| 17:30.00 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:30.20 | [TK]D-Fender | Psych0, Not sure if the GUI permits you multiple entires though.... |
| 17:30.51 | Psych0 | That's fine, all the peers are at one IP |
| 17:31.04 | Psych0 | Thank you so much guys |
| 17:31.13 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 17:32.19 | Psych0 | So just to confirm deny: 0.0.0.0/0.0.0.0 Allow: myip/mynetmask will deny everything besides that one IP from connecting to that extension, yeah? |
| 17:33.06 | [TK]D-Fender | Psych0, Of if you're protecting LAN phone then you'er set |
| 17:33.09 | [TK]D-Fender | Oh* |
| 17:33.23 | [TK]D-Fender | typing is becoming all too sloppy... TGIF |
| 17:34.00 | Psych0 | Alright. Well, the company owns 4 IPs, the Trixbox is one one and the LAN where the phones reside is on another, but same concept |
| 17:34.16 | Psych0 | It's amazing the IMPORTANT boxes we overlook |
| 17:44.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 18:26.57 | blizzow | I have a couple of users who have been complaining about call quality. I think I figured it out that when the microphone hears a fair bit of noise (ambient, the agent interrupts, or they blow into the microphone...) the distant side cuts out for a second. I was looking around in my softphone for a full duplex setting and don't see one. Is there a server side setting I can fiddle with to make sure conversations are full duplex? |
| 18:27.22 | blizzow | The headsets they're using are GN2000 usb headsets and do full duplex just fine. |
| 18:27.45 | WIMPy | It can only be the phones. |
| 18:28.15 | WIMPy | But quite possibly the ones at the other end. |
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| 18:34.13 | blizzow | WIMPy: It happens for just about any phone number I call. |
| 18:34.19 | blizzow | :( |
| 18:34.29 | blizzow | So I can't say it's at the distant end. |
| 18:36.46 | *** part/#asterisk Psych0 (~quassel@host-173-247-25-174.JENOLT1.epbfi.com) |
| 18:38.25 | autofsckk | hello, can somebody recommend me a VoIP provider for a call center? looking for good prices and good service too |
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| 18:48.36 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 18:49.16 | hudony | hi there. Having problem getting "block caller id per-call" working using asterisk and spa504G phones. Anyone willing to help me a bit? |
| 18:50.12 | [TK]D-Fender | hudony, Show us the problem and we might be able to show you a solution. |
| 18:51.54 | hudony | Well.. When I dial *67, I can hear something special as I am offered a second dialtone. I can then dial the number but my caller id isn't blocked. In fact, the *67 isn't grabbed by my dialplan even with *67. as an extension. It seems like the phone is interpreting it |
| 18:52.05 | hudony | I found information about this on the net |
| 18:52.24 | hudony | So I disabled the appropriate functionnality on the phone |
| 18:52.26 | hudony | but still the same thing |
| 18:52.33 | hudony | What could help you? |
| 18:52.47 | leifmadsen | hudony: sounds like the phone itself is doing something with the *67 |
| 18:52.52 | hudony | yes |
| 18:53.02 | [TK]D-Fender | It was. |
| 18:53.03 | leifmadsen | so it's not being passed to asterisk, and nothing that the phone is doing is causing somethign idffernet on the asterisk box |
| 18:53.09 | [TK]D-Fender | And not necessarily anything * cares about. |
| 18:53.24 | leifmadsen | even if asterisk isn't given callerid information, that doesn't mean your dialplan isn't setting it |
| 18:53.45 | [TK]D-Fender | Or that it is even happening in dialplan... |
| 18:53.55 | [TK]D-Fender | Or tell us what's in your peer... |
| 18:54.34 | hudony | hold on |
| 18:56.16 | hudony | This is teh appropriate section of my dialplan : http://pastebin.com/64qmyAfy |
| 18:56.27 | hudony | I will also show you the phone config |
| 18:57.23 | *** join/#asterisk goddva (~glarsen@cm-84.209.37.238.getinternet.no) |
| 18:58.00 | hudony | This is what I have changed about the caller id settings: http://pastebin.com/cQfyAZZn |
| 18:58.27 | hudony | Actually, I didn't change the phone dialplan. I just posted it so you can see it |
| 18:58.43 | hudony | I have remove *67 from the config and put CID blocking service to no |
| 18:58.50 | [TK]D-Fender | hudony, show us the call. |
| 18:59.00 | hudony | ok |
| 19:02.06 | eduzimrs | edar |
| 19:02.09 | *** join/#asterisk alexoc (~alexoc@201.82.173.134) |
| 19:05.29 | hudony | well |
| 19:05.50 | hudony | Dunno what I have changed but now, when I dial *67, I get beep beep beep etc |
| 19:05.59 | hudony | I am not receiving the second dialtone anymore |
| 19:06.35 | hudony | nothing is being displayed on the console though |
| 19:06.51 | hudony | verbose is set to 100 |
| 19:07.05 | hudony | It's like the phone won't allow me do dial it since I removed it from the configuration |
| 19:07.53 | [TK]D-Fender | hudony, Verbose != sip debug |
| 19:08.10 | [TK]D-Fender | hudony, and * has no match for *67 anyway |
| 19:10.01 | hudony | What about the extension I created? Shouldn't it match ? |
| 19:10.10 | [TK]D-Fender | hudony, Nope |
| 19:10.14 | hudony | oh |
| 19:11.05 | hudony | Can you explain me why? I am with the manual and the dot is supposed to match one or more character |
| 19:12.37 | [TK]D-Fender | *67 + one or more characters = how many total? |
| 19:13.37 | hudony | ... |
| 19:13.42 | hudony | you are right |
| 19:13.45 | hudony | :S |
| 19:14.59 | hudony | ok |
| 19:15.12 | hudony | When removing the dot, * now interpret the *67 :D |
| 19:15.26 | [TK]D-Fender | hudony, This is a good learning experience for you. Seeing how tiny little snafus can catch you. Trust as little as possible and start with the most immediately suspect pieces. |
| 19:16.24 | hudony | Ya... I have talked a couple of time to you and often, I'm having problem cause of false assumption |
| 19:16.39 | hudony | I guess I want to go a bit too fast |
| 19:16.41 | hudony | :S |
| 19:17.37 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 19:18.22 | *** join/#asterisk Sarcast (~Sarcast@e26216.upc-e.chello.nl) |
| 19:18.27 | Sarcast | Evening! |
| 19:18.30 | [TK]D-Fender | hudony, Think about the scope of the problem you're working on. * isn't processing what you dial. First prove * is even getting a call in the first place "SIP debug". If not, then the device isn't doing what you expect. Fix that. If the call is arriving and looks like the number you suspect, SIP DEBUG will show what peer it patched. Was it right? Ok, if so, is the call accepted? |
| 19:18.57 | Sarcast | you guys mind if I join the channel and dump a question on your doorstep right away? |
| 19:19.26 | [TK]D-Fender | hudony, It will say looking for [number] in [context]. Go prove you have the proper context named and pin down the precise exetns you feel should match it. break every little part of any pattern match apart a few times over. |
| 19:20.16 | p3nguin | ~ask |
| 19:20.16 | 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. |
| 19:20.27 | Sarcast | ah fine |
| 19:20.34 | [TK]D-Fender | hudony, Even that fails? Did you make changes you forgot to reload for? Ok, you checked the reload, still no go? use * CLI to dump the dialplan. Don't see your context & exten in there? Do I even trust that the right config file was there and readable? |
| 19:20.42 | Sarcast | normally you get blasted to crap if you ditch questions and leave hehe |
| 19:21.04 | Sarcast | Situation: I've set up an Asterisk server a while ago on a Linux Debian machine. Works fine. |
| 19:21.16 | Sarcast | now trying to get the webgui going so I can manage it more easily |
| 19:21.16 | [TK]D-Fender | hudony, in a hurry to create from scratch did you get the permissions wrong? Are you in the right folder at all? Didi you somehow misspell the config file name (happens a LOT) |
| 19:21.29 | Sarcast | gui works, can login, can add users and the gui adds the stuff to the users.conf file |
| 19:21.34 | Sarcast | except the sip.conf doesn't get touched |
| 19:21.44 | [TK]D-Fender | hudony, This is a "debugging thought process" |
| 19:21.46 | p3nguin | There's no GUI in Asterisk. |
| 19:21.52 | Sarcast | Asterisk webgui |
| 19:21.53 | hudony | Ya thank your for all these toughts |
| 19:21.55 | Sarcast | ah |
| 19:21.59 | Sarcast | wrong channel probably |
| 19:22.06 | p3nguin | So it seems. |
| 19:22.09 | Sarcast | sigh |
| 19:22.11 | Sarcast | thanks :) |
| 19:22.17 | hudony | I'll try harder being more systematic from now on |
| 19:22.22 | [TK]D-Fender | Sarcast, AsteriskGUI doust touch sip.conf AFAIK. Everything is users.conf |
| 19:22.31 | p3nguin | #asterisk-gui or #freepbx perhaps? |
| 19:22.40 | Sarcast | TK yea, that's what I figured out eventually too |
| 19:22.50 | Sarcast | but asterisk only picks up the users in the sip.conf file |
| 19:23.02 | Sarcast | or seems to |
| 19:23.35 | [TK]D-Fender | Sarcast, make sure your PBX is loading all the right modules.... |
| 19:23.51 | *** join/#asterisk timahvo1 (~rogue@41.81.19.107) |
| 19:23.53 | [TK]D-Fender | Sarcast, and that the config files are properly named and the permissions correct |
| 19:24.28 | Sarcast | well I haven't changed any of the config files, just the default make install stuff |
| 19:24.52 | [TK]D-Fender | Sarcast, I would recommend really looking and proving that. |
| 19:24.54 | Sarcast | permission could be it, but everything seems to work if I set things up manually and reload asterisk itself |
| 19:25.14 | *** join/#asterisk Crowb4r (u1012@gateway/web/irccloud.com/x-bbfeksmvamabwcic) |
| 19:25.15 | [TK]D-Fender | "I haven't changed anything" doesn't mean you might not be mistaken or that something may have changed behind your back. |
| 19:25.19 | Sarcast | I did a chown asterisk.asterisk -R yesterday on the etc/asterisk folder and the /var/lib/asterisk |
| 19:25.34 | Sarcast | before that it didnt' work, after that, it didn't either |
| 19:25.38 | [TK]D-Fender | And if the GUI changes the permissions by modding it you could be DOA |
| 19:25.40 | Sarcast | but made it more secure I guess :) |
| 19:25.53 | Sarcast | hm |
| 19:25.55 | Sarcast | good point |
| 19:26.47 | Sarcast | on second thought, I dont think it's a permission problem |
| 19:26.52 | Sarcast | users.conf gets updated just fine |
| 19:27.05 | Sarcast | just not sip.conf, permissions/ownership is the same on files |
| 19:27.08 | [TK]D-Fender | Sarcast, Start showing us stuff.. pastebin is your friend... |
| 19:27.11 | [TK]D-Fender | ~pb |
| 19:27.11 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:31.41 | Sarcast | hm, asterisk shows the users.conf file when I throw a 'config list' command at it |
| 19:31.46 | Sarcast | seems it loads it just fine |
| 19:32.00 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:38.12 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.108.33.dsl.dyn.forthnet.gr) |
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| 19:48.22 | Sarcast | hm, when I ditch the info from users.conf in the sip.conf, everything works dandy |
| 19:48.54 | Sarcast | so why isn't asterisk reading the users.conf, even though the 'config list' command shows it is loading it :? |
| 19:49.39 | Sarcast | -rw-r--r-- 1 asterisk asterisk 715 Jan 27 20:37 users.conf |
| 19:49.39 | Sarcast | -rw-r--r-- 1 asterisk asterisk 65K Jan 27 20:47 sip.conf |
| 19:49.53 | Sarcast | permissions are same, and it reads the sip.conf just fine when I manually set things up |
| 19:50.07 | Qwell | What makes you think it isn't reading it? |
| 19:51.04 | Sarcast | my phone can't connect if the user informatie is just in the users.conf |
| 19:51.18 | Qwell | So what makes you think the information in users.conf is correct? |
| 19:51.26 | Sarcast | when I copy the data from users.conf over to sip.conf and reload asterisk, user connects just fine and everything works |
| 19:51.34 | Sarcast | *phone |
| 19:51.56 | Sarcast | litterally copy/paste the content from users.conf to sip.conf |
| 19:52.08 | Qwell | so then it isn't correct in users.conf |
| 19:52.43 | Qwell | Did you set hassip=yes? |
| 19:52.56 | Qwell | (hint, you didn't) |
| 19:53.14 | Sarcast | lol |
| 19:53.26 | Sarcast | well asterisk gui didn't then |
| 19:53.37 | Qwell | Because you didn't tell it to. |
| 19:53.38 | Sarcast | im just a noob user wanting to easily config the thing ;) |
| 19:53.46 | Sarcast | I'll check it out, thanks for the heads up :) |
| 19:57.30 | leifmadsen | Qwell: figured out my build issue with CentOS -- had to tell it a KSRC value of /usr/src/kernels, not /lib/modules/<somethingorother> |
| 19:57.42 | Qwell | weird |
| 19:57.46 | leifmadsen | not sure how you build CentOS stuff, because I've never seen kernel-devel installed there |
| 19:57.53 | leifmadsen | always in /usr/src/kernels for me |
| 19:58.01 | Qwell | it's all magic! |
| 19:58.11 | Qwell | Did you have kernel-devel? |
| 19:58.19 | Qwell | That should create the build/ link |
| 19:58.27 | WIMPy | There should be a link from /lib/modules/version |
| 19:58.36 | Qwell | err, is it version or build? |
| 19:58.38 | Qwell | You might be right. |
| 19:59.18 | WIMPy | /lib/modules/version/build to be exact |
| 19:59.32 | Qwell | oh, version isn't literal. right. |
| 19:59.32 | leifmadsen | Qwell: ya kernel-devel installed, no build/ link that I saw |
| 19:59.53 | leifmadsen | didn't see anything in modules that I remember, but it was getting late last night |
| 20:00.05 | leifmadsen | I hacked up the spec file to handle the KSRC value |
| 20:00.22 | leifmadsen | after that everything went to happy |
| 20:00.53 | Qwell | So what is the source/ symlink there pointing to? Or does that not exist? |
| 20:01.03 | leifmadsen | there is no symlink |
| 20:01.10 | Qwell | huh |
| 20:01.14 | Qwell | those actually are owned by kernel |
| 20:01.22 | leifmadsen | just exists in /usr/src/kernel/`uname -a`-target_arch |
| 20:01.22 | Qwell | Your distro done broked it. |
| 20:01.23 | leifmadsen | KSRC=%{_usrsrc}/kernels/%{kversion}${kvariant:+-$kvariant}-%{_target_cpu} |
| 20:01.34 | leifmadsen | Qwell: it's a flat centos install, so it does whatever it does |
| 20:03.07 | Qwell | rpm -qf /lib/modules/2.6.18-*/build |
| 20:03.07 | Qwell | kernel-2.6.18-274.3.1.el5 |
| 20:03.07 | Qwell | kernel-2.6.18-274.7.1.el5 |
| 20:03.10 | Qwell | shrugs |
| 20:03.12 | leifmadsen | Qwell: strange how on centos6 itls all in the right spots |
| 20:03.32 | leifmadsen | not sure if installing in a chroot has anything to do with it, but it's just a 'yum install kernel-devel' that is running |
| 20:03.37 | leifmadsen | so that rpm would handle all that |
| 20:04.05 | Qwell | Do you get anything weird from rpm -V kernel-`uname -r` ? |
| 20:04.42 | leifmadsen | not sure, I'd have to rebuild again and leave the chroot environment intact to find out what it does |
| 20:05.04 | leifmadsen | it gets created and destroyed each time I build unless I say not to |
| 20:09.03 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:09.04 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:09.05 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:23.20 | p3nguin | How can I stop the jabber messages "Got presence packet from PEOPLE I HAVE NEVER HEARD OF, someone not in our roster!!!!"? It is periodically flooding the console and I want to make it so I do not receive those packets. |
| 20:26.01 | Sarcast | disable the jabber module :? |
| 20:26.12 | leifmadsen | remove that notice line from the code? |
| 20:27.36 | p3nguin | Removing the notice does not eliminate the packet. I want a FIX, not a workaround for the message. |
| 20:29.23 | Sarcast | ..well technically you asked how to stop the message ;) |
| 20:29.48 | leifmadsen | p3nguin: block that user on the jabber side, you're getting a presence packet from the jabber server |
| 20:29.51 | leifmadsen | it's not an asterisk thing |
| 20:30.06 | p3nguin | I think I just solved it. |
| 20:30.11 | leifmadsen | asterisk just handling it and telling you about it |
| 20:30.11 | ChannelZ | Jabber spam |
| 20:30.24 | p3nguin | I logged in to gtalk with pidgin and removed those spammers/users from my list. |
| 20:30.54 | p3nguin | I hope that takes care of it permanently. |
| 20:31.51 | p3nguin | <query xmlns='jabber:iq:roster'> |
| 20:31.51 | p3nguin | <item jid='cheerdayana2@aol.com' subscription='remove'/> |
| 20:31.53 | p3nguin | </query> |
| 20:32.29 | p3nguin | Where do they all come from, G+? |
| 20:33.48 | Sarcast | most IM programs have a block on unkown contact or unsollicitated messages |
| 20:33.54 | Sarcast | probably same thing |
| 20:34.13 | p3nguin | The problem is that I do not use any IM programs. |
| 20:34.39 | Sarcast | you just mentioned gtalk and pidgin |
| 20:34.49 | Katty | hello. i am not dave. |
| 20:34.54 | p3nguin | Asterisk uses jabber/gtalk for the Google Voice stuff, but these turds somehow weaseled only my roster. |
| 20:35.06 | p3nguin | s/only/onto/ |
| 20:35.52 | p3nguin | Because they weaseled onto my roster, asterisk complains about the presense packets of these users that I DO NOT KNOW. |
| 20:36.37 | p3nguin | They are not my "friends." I do not know them and I do not want to see notices from asterisk regarding them. |
| 20:37.41 | p3nguin | Since asterisk does not have any type of interface to control such things, I decided to see what pidgin would show me. It showed those turds on my list... so I removed them. |
| 20:38.18 | Sarcast | it's spam... |
| 20:38.20 | Sarcast | what can you do |
| 20:38.22 | p3nguin | No shit. |
| 20:38.28 | Sarcast | you can't cure spam |
| 20:38.34 | Sarcast | you can block it or drop the message |
| 20:38.46 | p3nguin | Apparently I just did by using the remove command. |
| 20:39.06 | ChannelZ | You might benefit from autoprune |
| 20:39.13 | p3nguin | I had it set to yes. |
| 20:39.16 | p3nguin | Didn't help. |
| 20:39.24 | p3nguin | I changed it back to no, now. |
| 20:39.36 | leifmadsen | autoregister=yes ;;Auto register users from buddy list. |
| 20:39.38 | ChannelZ | then not sure how those people got onto your list in the first place |
| 20:39.43 | leifmadsen | may or may not help |
| 20:39.47 | ChannelZ | if you're saying you didn't add them |
| 20:40.12 | p3nguin | What's the default for autoregister? I have it commented out. |
| 20:40.44 | p3nguin | I don't want to be subscribed to spammers and I don't want spammers subscribed to me if I have a choice... |
| 20:41.06 | p3nguin | I've kind of assumed that autoregister would actually agree to subscribe them. |
| 20:41.26 | ChannelZ | no idea. I've never seen this problem |
| 20:41.52 | p3nguin | It seems pretty popular, but no one has ever expressed a real solution. |
| 20:42.00 | ChannelZ | Do you have a separate account for your Asterisk or is it a personal google account you use for everything else? |
| 20:42.26 | p3nguin | I use my personal google account for asterisk/gtalk because it is tied to my gvoice number. |
| 20:44.01 | p3nguin | While trying to find answers to the problem, I found a plethora of people experiencing a similar problem where this happens to them in their gmail chat. I never use my gmail chat, but all those spammers' names were on that list as well. |
| 20:45.32 | p3nguin | My daughter's gvoice is registered with asteirsk, too, and when I do jabber show buddies, I see spammers on her account too... but I never get the presence packet notices on those (it was always the ones from my list, and it just started a few days ago). |
| 20:46.07 | p3nguin | She also does not use her gmail chat. |
| 20:46.19 | p3nguin | nor any gtalk IM. |
| 20:46.31 | p3nguin | We just use gmail and asterisk/gtalk. |
| 20:48.13 | leifmadsen | p3nguin: there is likely a way to make it so it's not as easy for people to get on your list, but from the google settings side |
| 20:48.38 | p3nguin | I made a change to the chat settings in gmail, hoping that stops it from happening again. |
| 20:49.09 | p3nguin | Something about not auto-adding people. It seemed relevant. |
| 20:54.12 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:56.25 | Sarcast | I think I found it.. |
| 20:56.47 | Sarcast | Asterisk GUI only adds users to the file, the first time you update the config after a restart |
| 20:56.55 | Sarcast | the 2nd update doesn't get pushed for some reason |
| 20:57.06 | p3nguin | *shrug* No clue. |
| 20:57.26 | Sarcast | maybe I should just ditch the whole thing and start from scratch -.- |
| 20:57.28 | p3nguin | ~users.conf |
| 20:57.29 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 20:57.36 | Sarcast | interesting learning way though |
| 20:57.47 | Sarcast | lol |
| 20:57.48 | p3nguin | Learning a GUI, you mean. |
| 20:58.08 | Sarcast | no, the GUI is just a way of easily adding people to the server when Im at work |
| 20:58.13 | p3nguin | I'd rather learn how to use Asterisk. |
| 20:58.19 | Sarcast | and dont want to get shell access to my server and dig my way through config files |
| 20:58.30 | Sarcast | fyi, asterisk works fine on my end |
| 20:58.43 | Sarcast | using my cellphone to call people via wifi, works all dandy |
| 20:58.53 | Sarcast | just screwing with the GUI atm, which doesn't seem to work :P |
| 20:59.07 | p3nguin | Which sip phone do you use on your mobile? |
| 20:59.20 | [TK]D-Fender | Sarcast, GUI needs a certain amount of basic configuration to be in place to operate as intended. |
| 20:59.24 | Sarcast | 3cxphone on iphone |
| 20:59.28 | p3nguin | I like iSip a lot. |
| 20:59.30 | Sarcast | not a clue what's on my htc p3600 |
| 20:59.39 | Sarcast | and couldn't find a free one for my blackberry |
| 20:59.49 | Sarcast | open for BB suggestions btw |
| 20:59.51 | [TK]D-Fender | Sarcast, Also it is virtual unmaintained and you can double the population of their support channel just by entering... |
| 21:00.02 | Sarcast | lol |
| 21:00.47 | p3nguin | This one time, at band camp, I tried to use the Asterisk GUI just to see how it worked... |
| 21:00.52 | p3nguin | It didn't. |
| 21:01.07 | p3nguin | All kinds of problems. |
| 21:01.18 | p3nguin | I gave up, since config files are actually much easier to deal with. |
| 21:04.31 | *** join/#asterisk hmmhesays (~hmmhesay@174-126-194-60.cpe.cableone.net) |
| 21:06.18 | *** join/#asterisk timahvo1 (~rogue@41.81.169.10) |
| 21:19.36 | *** join/#asterisk krotos (~d3v1l@host26-34-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:19.40 | krotos | hi all :) |
| 21:21.41 | krotos | i'm reparing an old asterisk box, that use this old ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 21:21.48 | krotos | can i use dadhi with this card? |
| 21:22.56 | WIMPy | Generelly, yes. |
| 21:23.23 | WIMPy | If you need NT mode, you may need a patch, depending on the exact type of card. |
| 21:25.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:25.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:27.14 | krotos | WIMPy: thankyou for reply, i use this card in TE mode , so i hope is "ready to run |
| 21:27.57 | WIMPy | That will work, yes. |
| 21:40.59 | *** join/#asterisk jdavidow (~justin@rx0nr-brewers-ans-backup.wp.bigpipeinc.com) |
| 21:49.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:53.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:53.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:21.02 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:23.37 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 22:35.25 | *** part/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 22:38.23 | Kobaz | what's it mean when you have a triangle with an ! inside it, on a polycom line (showing on the display) |
| 22:38.45 | *** join/#asterisk chasing`Sol (~cS@197.132.141.223) |
| 22:49.33 | p3nguin | I think it means DANGER!!! |
| 22:59.30 | *** join/#asterisk Goldwing (~Goldwing@84.245.46.83) |
| 23:03.08 | Sarcast | hum |
| 23:03.16 | Sarcast | DANGER DANGER Will Robinson! |
| 23:03.19 | Sarcast | *runs* |
| 23:03.32 | Sarcast | well ditched the GUI, set up stuff manually, working fine |
| 23:03.54 | Sarcast | now trying to strangle my firewall to get iptables to nat the stuff from outside to inside.. or reverse |
| 23:05.01 | p3nguin | Are you talking about two different systems? |
| 23:05.10 | p3nguin | one firewall, one with asterisk? |
| 23:05.21 | [TK]D-Fender | Sounds more like he wants his server to be a router as well |
| 23:06.00 | ChannelZ | that's easy |
| 23:06.11 | p3nguin | I'm trying to determine if that's really the case or not. |
| 23:06.32 | p3nguin | If there is already a gateway, there probably isn't any need to make the asterisk system a router, too. |
| 23:06.36 | Sarcast | one firewall, one asterisk |
| 23:06.43 | Sarcast | internal lan works like a charm |
| 23:06.52 | Sarcast | outside, phone rings, connection made.. ..no audio |
| 23:06.58 | p3nguin | NAT is pretty easy to configure in iptables. |
| 23:07.06 | Sarcast | usually yea.. |
| 23:07.22 | Sarcast | not entirely sure if my cellphone provider allowes voip traffice |
| 23:08.13 | p3nguin | Sounds like two completely different things to me. |
| 23:08.24 | ChannelZ | backs away slowly |
| 23:08.57 | Sarcast | drops an oil tanker on ChannelZ |
| 23:08.59 | Sarcast | wins! |
| 23:09.35 | Sarcast | p3nguin trouble always comes two's |
| 23:09.51 | Sarcast | twos |
| 23:09.56 | Sarcast | or threes |
| 23:10.19 | p3nguin | Maybe, but what does your cell provider have to do with voip? |
| 23:10.51 | Sarcast | them blocking voip traffic? |
| 23:11.57 | p3nguin | It's really hard for me to not repeat the question to try to get a more sensible answer. |
| 23:12.41 | Sarcast | slapping is allowed I guess |
| 23:12.52 | Sarcast | long work day, not enough food, not enough coffee |
| 23:13.07 | *** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.1.0 (2012/01/27), 1.8.9.0 (2012/01/27), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:13.23 | Sarcast | ghe |
| 23:13.45 | Sarcast | visits the official Asterisk wiki |
| 23:50.29 | *** join/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com) |
| 23:50.43 | patrickod | when using asterisk realtime configuration is it possible to still use templates? |
