| 00:00.30 | darksmurf | http://pastebin.com/m7b655d64 |
| 00:01.32 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:02.07 | darksmurf | updated http://pastebin.com/m7b655d64 |
| 00:03.22 | *** join/#asterisk RoPBX (n=nickserv@200.93.61.20) |
| 00:03.33 | RoPBX | hi all |
| 00:04.35 | RoPBX | please, i have an issue when i record calls the dtfm tones stop working.... |
| 00:04.53 | RoPBX | any ideas? |
| 00:05.55 | aurax | dtmf? |
| 00:06.10 | RoPBX | yeap, sorry... |
| 00:06.31 | aurax | what do you mean by stop working? |
| 00:06.48 | aurax | only in ivr when you try to generate a tone ? |
| 00:07.36 | RoPBX | yes, when i call to an IVR the tones doesn't arrive there |
| 00:08.24 | aurax | hmm, can you try instead of directing to IVR then direct exactly to extension and check weather it's just DTMF issue or you cant hear the other end at all |
| 00:12.50 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-195.mia.bellsouth.net) |
| 00:13.18 | geneticx | sup you all. |
| 00:14.54 | RoPBX | well, calls often are to banks or voicemenus |
| 00:25.16 | *** join/#asterisk Tim_Toady (n=moi@adsl274-56.kln.forthnet.gr) |
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| 01:01.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:59.54 | timeshell | Greetings |
| 02:00.46 | timeshell | Has anyone had success in setting up video call on asterisk in this progression? X-Lite <-->asterisk <-->asterisk <--> X-Lite |
| 02:03.53 | box2 | i don't use x-lite because it is not in debian repositories |
| 02:03.56 | box2 | i use linphone or ekiga |
| 02:12.16 | timeshell | Have you had success in the video->asterisk->asterisk-video then? |
| 02:12.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:13.18 | *** join/#asterisk BeeBuu (n=beebuu@59.38.96.35) |
| 02:14.41 | russellb | it should work in theory ... what version, though |
| 02:14.48 | box2 | never used video :( |
| 02:14.53 | box2 | been meaning to try |
| 02:15.10 | russellb | try 1.4 if you're using 1.6 ... |
| 02:15.39 | russellb | video in 1.6 got broken at some point, i've asked someone to work on it this coming week, though. |
| 02:16.25 | Tim_Toady | video in 1.4 and in 1.6.1 works for me, tested with ekiga a few days ago |
| 02:16.44 | russellb | ooh, cool. :-) |
| 02:17.31 | Tim_Toady | just make sure u have videosupport=yes and h261,h263,h263p in ur allowed codecs in sip.conf |
| 02:19.30 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 02:22.21 | Tim_Toady | app echo works with video too, so its a good starting point for testing |
| 02:29.33 | *** join/#asterisk propellerhead (n=yogurt2u@host68.190-31-73.telecom.net.ar) |
| 02:31.00 | leifmadsen | I tried video once, but it didn't work for me |
| 02:31.05 | leifmadsen | one of these days I'll try again |
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| 02:52.58 | BBHoss_ | can anyone recommend some home automation equipment that works well with linux? i' |
| 02:53.06 | BBHoss_ | i'm planning on using it with asterisk |
| 02:59.27 | Qwell | BBHoss_: like x10 stuff? |
| 02:59.35 | Qwell | Somebody wrote something for that, IIRC. |
| 03:00.07 | BBHoss_ | Qwell: yeah thats what i remember from looking at it years ago, i was just wondering if anything new had come out |
| 03:00.19 | BBHoss_ | like wtf is zigbee |
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| 04:24.36 | linagee | interesting. the way to get the absolute least amount of latency with an ITSP is to use DISA? hah.... |
| 04:24.57 | linagee | have a colo in the same data center as your ITSP and use DISA |
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| 04:29.39 | [TK]D-Fender | linagee: Huh? |
| 04:29.52 | [TK]D-Fender | linagee: What does DISA have to do with latency? |
| 04:30.49 | linagee | [TK]D-Fender: telephone lines have a fairly low latency, right? typically less than 4ms or something? |
| 04:31.32 | linagee | [TK]D-Fender: 4ms + 4ms < 4ms + 100ms + even more? |
| 04:31.40 | [TK]D-Fender | linagee: What does DISA have to do with anything? |
| 04:32.10 | linagee | [TK]D-Fender: i suppose that's true. If I had an OC3 to my colo it would also have low latency. lol |
| 04:32.34 | linagee | [TK]D-Fender: maybe I should restate that. lowest cost per latency. :) |
| 04:32.49 | [TK]D-Fender | linagee: ..... |
| 04:33.07 | [TK]D-Fender | linagee: Put. Down. The. Crack. Pipe. (c) JerJer |
| 04:33.15 | linagee | [TK]D-Fender: ? |
| 04:33.24 | linagee | [TK]D-Fender: why do you think I have a crack pipe |
| 04:33.36 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 04:33.36 | jaytee | [TK]D-Fender, ahahahaa I was just thinking about the crack pipe retort!!!! |
| 04:33.38 | [TK]D-Fender | linagee: Cost? Latency? What are you rambling about? |
| 04:34.01 | jaytee | DISA, latency? what does one have to do with the other? NOTHING!!!! |
| 04:34.27 | linagee | jaytee: call in + call out = low latency because I have a small latency to my ITSP from my colol. |
| 04:34.30 | linagee | colo |
| 04:34.41 | linagee | jaytee: so in my particular case, this works better. :) |
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| 04:35.54 | [TK]D-Fender | linagee: You don't DISA to an ITSP, and clearly that is 1 extra bridging step as it is. |
| 04:36.01 | linagee | [TK]D-Fender: I never said you do |
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| 04:36.16 | jaytee | [TK]D-Fender, is it just me or does it seem like there's some kind of pandemic spreading that has no noticable symptoms other than a sudden drop in intellectual capacity? |
| 04:36.25 | [TK]D-Fender | linagee: It looks like english but reads as gibberish |
| 04:37.00 | *** join/#asterisk mosburn (n=mosburn@24-56-179-224.co.warpdriveonline.com) |
| 04:37.09 | [TK]D-Fender | reads the list of ingredients on linagee's back label...... "mixed nuts" |
| 04:37.31 | linagee | [TK]D-Fender: maybe I should have started this as "I found a way to get lower latency to my conversations" which you would have promptly ignored everything I said after that. |
| 04:37.47 | linagee | [TK]D-Fender: or trolled on like this too. I could be wrong. |
| 04:38.33 | [TK]D-Fender | linagee: Well, you've got to be good at something I guess :) |
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| 04:38.40 | jaytee | way to go, Tonto! |
| 04:38.52 | [TK]D-Fender | linagee: I fail to see DISA's relevance on jsut about anything really... |
| 04:38.53 | linagee | gives [TK]D-Fender a special star |
| 04:39.17 | [TK]D-Fender | linagee: I was referring to your being WRONG :p |
| 04:39.28 | linagee | gives [TK]D-Fender another special star |
| 04:43.28 | [TK]D-Fender | sharpens the edges and tosses them back as shuriken |
| 04:48.22 | jaytee | "NEWS BULLETING: FDA announced today that several studies have proven that Laetrile does not cure cancer. In an unrelated note the FCC announced that claims that DISA could prevent or alleviate latency in VOIP communications had no basis in fact" |
| 04:50.20 | linagee | jaytee: replace that with "using telephone lines has a lower latency than internet connections". and yes this seems a bit obvious. |
| 04:50.36 | linagee | throws jaytee a star too |
| 04:50.56 | blkry | does anyone know of a guide to install res_snmp? it requires recompiling asterisk, but when can res_snmp be added? |
| 04:51.51 | [TK]D-Fender | LinDepends what in between those "lines" |
| 04:52.13 | *** join/#asterisk manxpower (n=EWieling@47.sub-70-223-3.myvzw.com) |
| 04:52.22 | blkry | install res_snmp then recompile?? |
| 04:54.36 | linagee | [TK]D-Fender: allow me to clarify. a regular 1950s bell telephone line with no crappy "cox digital telephone" / "comcast digital telephone" / other |
| 04:54.58 | linagee | only god knows what sort of latency and codecs cable providers introduce. |
| 04:55.36 | manxpower | lin20-30ms latency usually |
| 04:55.46 | linagee | manxpower: ah |
| 04:56.51 | [TK]D-Fender | LinWell if you are reffering to POTS to telco and high BW SS7 to the PSTN, then sure, why not |
| 04:57.21 | [TK]D-Fender | manxpower: HI GOD! |
| 04:57.28 | linagee | manxpower: and codecs they use? G.729? PCM? You'd *think* PCM, but there's an assume there. |
| 04:58.35 | linagee | [TK]D-Fender: I'd bet you $5 that a cox/comcast sales rep would refer to their service as "POTS" (as incorrect as that may be) |
| 05:00.41 | linagee | [TK]D-Fender: I have an image for you. http://img27.imageshack.us/img27/9774/murderturtle.gif |
| 05:00.58 | manxpower | linagee: the codecs don't matter all THAT much for latency |
| 05:01.32 | manxpower | I expect G.711 ulaw is the codec. Causes much less issue with things like fax |
| 05:01.34 | linagee | manxpower: true, but I've heard some pretty crappy cox digital phone service. :( sounded worse than a 1950s bell POTS |
| 05:02.00 | manxpower | linagee: I bet that is all in their network, not in their actual codecs/standards |
| 05:02.09 | linagee | manxpower: hrm |
| 05:02.30 | LiNeTuX | <PROTECTED> |
| 05:02.46 | manxpower | You can have the best ultra-wideband codec, but if your network sucks it's not going to sound good |
| 05:03.00 | linagee | manxpower: indeed. worse point along a path and that sort of thing |
| 05:03.38 | [TK]D-Fender | linagee: OLD |
| 05:04.18 | manxpower | [TK]D-Fender: I'm *really* liking my new job. I started on Wed, so far it's mostly learning their TLAs, procedures, etc. |
| 05:04.45 | [TK]D-Fender | manxpower: What & where? |
| 05:04.51 | manxpower | They have the basic design of their network the same as I would have loved to have the budget to set up my customers like that. |
| 05:05.21 | manxpower | [TK]D-Fender: a CLEC in NY/NJ, provisioning, 2nd tier support, PBX setup |
| 05:10.53 | [TK]D-Fender | Any impact on where you're living? |
| 05:10.55 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 05:12.04 | manxpower | [TK]D-Fender: Both job offers were similar in salary and benefits, but this one let me work from whereever I wanted to live. That was a major deciding factor. Getting up to speed will be harder remotly, but after that I don't think it will be much of a problem. |
| 05:12.48 | [TK]D-Fender | manxpower: Well at least it eases up on your time for commute, etc |
| 05:12.58 | manxpower | I worked remotly for 10 years. |
| 05:19.58 | manxpower | [TK]D-Fender: *nod* Takes some willpower to start working 8am M-F, but since I like the job, it's not that difficult. |
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| 05:24.02 | mosburn | How much trouble am I going to have getting a polycom phone to remote into the network over VPN? |
| 05:25.02 | linagee | mosburn: not much |
| 05:25.43 | mosburn | linagee: any specific models to look into to make my users a bit happier |
| 05:25.56 | linagee | mosburn: not much, YMMV. :) |
| 05:26.14 | manxpower | mosburn: Polycoms don |
| 05:26.15 | manxpower | ' |
| 05:26.33 | manxpower | 't "VPN", but they should work over an existing VPN. Latency and jitter are things to be concerned about |
| 05:27.12 | blkry | moshburn what devices are the VPN on? |
| 05:27.23 | linagee | manxpower: when you get to manxpower level, latency and jitter are things to always worry about though, even when not VPNed. :) |
| 05:27.35 | mosburn | manxpower: Thanks, it is only for a few remote users that dont want to use their cell phones all the time |
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| 05:27.41 | *** mode/#asterisk [+o denon] by ChanServ |
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| 05:28.23 | mosburn | blkry: right now they are using the cisco software client, but I can mandate a router with VPN connection for their offices |
| 05:29.28 | linagee | mosburn: cisco software client = ? |
| 05:29.43 | blkry | for the PC's |
| 05:29.57 | blkry | yeah you'll have to |
| 05:30.01 | linagee | a SIP client? VPN client? |
| 05:30.07 | mosburn | linagee: yeah, on the windows boxes, or VPNC on the linux laptops |
| 05:30.22 | mosburn | linagee: its a vpn client |
| 05:30.46 | linagee | mosburn: I wouldn't trust windows to do my dishes. :) If you get hacked, watch a bunch of international phone calls on your phone bill. :) |
| 05:31.33 | mosburn | linagee: Unfortunately, the only users that still use it here are the C*O level |
| 05:32.53 | mosburn | linagee: and they refuse to change that |
| 05:33.17 | linagee | mosburn: gotta have Microsoft Powerpoint, right? :) |
| 05:33.41 | linagee | so they can run all their spreadsheets? |
| 05:34.15 | mosburn | linagee: no, its some ancient reporting software they run, circa 1989 by the way it looks |
| 05:34.26 | mosburn | dos based |
| 05:34.29 | linagee | (The Webserver Is Down) |
| 05:34.47 | manxpower | DOS emulation under Linux has been very good |
| 05:35.13 | coppice | freedos works far better than DOS ever did |
| 05:35.22 | mosburn | manxpower: trying next week with the new laptop rollouts to get them to switch |
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| 06:41.35 | jkroon | which fax level does {Send,Receive}Fax in asterisk 1.6.1.1 support? |
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| 07:45.41 | VooDooNOFX | Why does the dialplans in the astbook make frequent use of NoOp at the beginning of many routines? |
| 07:46.06 | box2 | MAGIC!!!!!!!!!! |
| 07:46.13 | box2 | disappears in a cloud of smoke |
| 07:46.27 | VooDooNOFX | lol |
| 07:48.14 | VooDooNOFX | I assume it's simply for debugging purposes |
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| 08:05.52 | iksik | morning |
| 08:06.31 | iksik | I've got a "little" problem with configuration. Don't know why, I don't have incoming caller ID presentation - I don't see CID in asterisk logs (higher debug level) either... |
| 08:06.40 | iksik | in asterisk console* |
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| 08:23.31 | BeeBuu | hi,all |
| 08:23.41 | BeeBuu | Callee cannot use dynamic features! how can i do? |
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| 08:49.11 | jkroon | VooDooNOFX, mostly yes. |
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| 09:10.48 | VooDooNOFX | after editing the mailbox.conf file, how do I reload it into asterix? |
| 09:27.58 | BeeBuu | Callee cannot use dynamic features in 1.4, anyone know is it fix in 1.6 ? |
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| 09:29.32 | VooDooNOFX | When leaving a voicemail, does the callee have to press # when done, or can they just hang up? |
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| 09:34.46 | VooDooNOFX | oh, nm :D |
| 09:34.58 | VooDooNOFX | I like asking stupid q's that I eventually find the answers to |
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| 10:17.31 | VooDooNOFX | I need some help setting up my callerid |
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| 11:13.13 | Zhad | lo coppice |
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| 11:14.39 | coppice | hi |
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| 11:28.03 | yziquel | hi. how can one create a meetme conference from the cli? |
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| 12:33.53 | Faustov | hello, is ther a migration howto for asterisk 1.4 to 1.6? |
| 12:34.27 | Faustov | or in particular, looks like my extensions that had "goto" in them are no longer recognized as extensions for some reaosn |
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| 13:03.08 | Corydon76-dig | Faustov: UPGRADE*.txt |
| 13:06.57 | *** join/#asterisk puzzled (n=foobar@535335A1.cable.casema.nl) |
| 13:10.06 | Faustov | damn |
| 13:10.31 | Faustov | doesn't give me any idea, i thought it could be the change with Macro() but it says it is still tehre |
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| 13:31.08 | Faustov | oh god |
| 13:31.12 | Faustov | i had a rubbish character |
| 13:31.40 | Faustov | comment from me: would be nice if asterisk reported a problem with rubbish character in extensions.conf rather than just ignore it and say nothing |
| 13:32.55 | Pan3D | Hi |
| 13:33.06 | Pan3D | in 1.6 , you can't use "|" anymore |
| 13:33.24 | Pan3D | that will cause many functions to fail |
| 13:33.40 | Faustov | yup, luckily asterisk reports when it finds a pipe |
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| 14:58.49 | jkroon | [TK]D-Fender, just wanted to say thanks for yesterday re the fax rotation stuff. managed to get stuff outside of asterisk to pre-rotate and rescale things to sanity before passing off to asterisk as indeed rotation of a tiff is non-trivial due to the way it's encoded. |
| 14:59.04 | jkroon | once more you prodded me in the right direction. |
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| 15:04.31 | [TK]D-Fender | jkroon: If you can trust portrait for that I guess. |
| 15:04.51 | [TK]D-Fender | jkroon: You should put a width threshold on it |
| 15:05.04 | jkroon | hehe, it just gets the shape right, and no, it was a tad more involved (eg, the width threshold) |
| 15:05.35 | jkroon | using convert from imagemagic to detect the resolution, to then rotate if it's wider than long, and to then clamp the width. |
| 15:12.25 | [TK]D-Fender | jkroon: because not all pages are in fact long. |
| 15:13.00 | [TK]D-Fender | jok on an old fax I used to fax the confirmation slip which coming from a roll was in fact rather sharply landscape |
| 15:13.35 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 15:14.08 | jkroon | on a roll the length is variable, so it doesn't matter. mostly other fax machines use letter or a4, so i'm hoping my reasoning is good enough. time will tell. |
| 15:14.36 | jkroon | imho it's better to print something like a ppt thing sideways to the fax machine so that you get a more decent size out of it. |
| 15:15.03 | jkroon | anyway, it works now and unless somebody complains that's the way it's going to stay. |
| 15:15.11 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-195.mia.bellsouth.net) |
| 15:17.26 | [TK]D-Fender | jkroon: Sounds like you got the 95%+ solution... |
| 15:17.34 | [TK]D-Fender | jkroon: Rest probably wouldn't matter much |
| 15:17.50 | jkroon | exactly: good enough. |
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| 15:25.42 | leifmadsen | dances |
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| 15:41.25 | [netman] | I would need some advices about oh323, please. I'm always getting a "transport error" when trying to make a call. |
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| 15:45.04 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/x-eb6422b6450b63dc) |
| 15:48.43 | yziquel | hi. how can i check if the ztdummy module works fine with respect to the meetme application. (I have only voip communications, no real phones). |
| 15:48.55 | yziquel | [TK]D-Fender: thanks for the help. it was the routeur. |
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| 17:10.00 | manxpower | yziquel: if meetme works then ztdummy is working |
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| 17:27.29 | manxpower | [netman]: have you searched the mailing lists? very few people use H323 with Asterisk |
| 17:29.44 | *** join/#asterisk nixer (n=nixer@78.154.233.235) |
| 17:30.33 | [netman] | manxpower: I used google search |
| 17:30.47 | [netman] | The trouble is that error could have a lot of different causes :( |
| 17:31.34 | [netman] | thank you for your answer anyway |
| 17:31.52 | nixer | Hello. I'm using Asterisk 1.6 along with dahdi 2.2. I got the PBX to *almost* work. I can't make any outgoing calls until I remove the phone cable and plug it back to the analog socket, otherwise I keep getting "circuits are busy" |
| 17:32.24 | nixer | I've been trying to search for asterisk "circuits are busy" for the past 5 hours on Google but no luck finding a solution... |
| 17:34.40 | manxpower | I assume that is an "circuits are busy" AUDIO message? |
| 17:35.17 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 17:35.18 | nixer | Yes. |
| 17:35.43 | manxpower | nixer: from the telco? |
| 17:36.05 | nixer | I don't even reach the telco. It's from Asterisk. |
| 17:36.22 | nixer | When I replug the phone cable, I can dial normally. |
| 17:36.25 | manxpower | Asterisk does not play audio messages by default. Are you running some sort of GUI? |
| 17:36.40 | nixer | I'm running freepbx. |
| 17:36.49 | manxpower | ~freePBX |
| 17:36.55 | 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 |
| 17:36.55 | manxpower | You know better, nixer |
| 17:37.40 | hesco | Can anyone who knows sox advise me how to convert a .WAV file from the Voicemail() app into some format which can be easily played from a cli with aplay? |
| 17:37.42 | nixer | It's not a freepbx issue. The sound doesn't matter. It's the fact that asterisk won't detect the trunk until I replug the phone cable. |
| 17:38.06 | nixer | hesco: aplay can't play gsm. sox can play them. |
| 17:38.13 | manxpower | nixer: Maybe so, but it's impossible to debug since FreePBX is so terribly complicated. |
| 17:38.35 | drmessano | Its impossible to debug dahdi, which is configured outside FreePBX? |
| 17:38.54 | manxpower | drmessano: Thank you for volunteering to help nixer |
| 17:39.37 | hesco | nixer: are you saying that sox can be used like aplay to play an audio, not just convert it? |
| 17:39.40 | manxpower | I can't really think of a way to debug DAHDI without Asterisk running. At least not for stuff that makes calls fail on analog. |
| 17:39.44 | nixer | I went through a lot of forums to finally have asterisk identify my card's channels. Now I'm having this tiny annoying problem.. :/ |
| 17:39.49 | drmessano | manxpower: Someone has to, since you continue to tell someone with even a molex connector question that its a "freePBX problem" since they run FreePBX |
| 17:40.02 | hesco | I just tried running sox msg0072.WAV on a cli and got its help/usage message |
| 17:40.12 | manxpower | drmessano: so help the poor sod. |
| 17:40.28 | nixer | hesco: Because you have to tell sox what to do with the file. Read the help info it showed you. |
| 17:40.58 | nixer | hesco: sox <input file> -d if you just want to play the file :) |
| 17:41.39 | drmessano | nixer: Is the last call not hanging up? |
| 17:42.39 | nixer | drmessano: I can't dial in the first place, until I replug the phone. I did that and I called the line, when the voicemail finished, it didn't hang up. It stayed for 30 seconds I think. |
| 17:43.01 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 17:43.01 | *** join/#asterisk SkywaIker (i=pirch@113.53.162.48) |
| 17:45.23 | hesco | thanks nixer! worked like a charm. Never knew sox could be used that way. That should come in handy |
| 17:45.45 | nixer | hesco: In the command line, type: man sox |
| 17:45.57 | nixer | It will tell you how to use sox. (man -> manual) |
| 17:47.32 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:51.46 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:02.08 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 18:03.01 | jkroon | nixer, https://issues.asterisk.org/view.php?id=14577 |
| 18:03.03 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 18:03.31 | nixer | jkroon: Checking. |
| 18:04.07 | jkroon | the other fix is making a call into the system. |
| 18:04.34 | nixer | Yeah, that worked too... but that's not a solution :/ |
| 18:04.41 | nixer | It's a dirty trick! |
| 18:04.44 | jkroon | apparently this isn't a "critical" issue, so i'm custom patching each and every asterisk install at the moment. |
| 18:04.54 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:05.16 | jkroon | the patch i've supplied there works, so far without any reported issues. |
| 18:05.36 | nixer | oh! I did read that bug report! |
| 18:06.03 | jkroon | i see other patches has now been attached too. |
| 18:06.05 | nixer | I'm using PBX in a Flash, which compiles Asterisk for me. If I'm to patch it, would I have to recompile it myself? |
| 18:06.06 | manxpower | jkroon: Impressive. It's not often someone manages to find such a bad bug. I wonder where/when this bug was introduced. |
| 18:06.24 | jkroon | somewhere between 1.6.0.1 and 1.6.0.3 |
| 18:06.26 | nixer | Well duh, of course I'm gonna have to recompile it... |
| 18:06.33 | nixer | I'm using 1.6.1.1! |
| 18:06.51 | jkroon | patch applies |
| 18:07.19 | nixer | But I'm still having the problem :/ |
| 18:08.09 | yziquel | manxpower: meetme works, but I get a whole load of 'unmonitored' clients to the meetme conf. It seems they do not leave properly, and I've also got a whole bunch of timer.c warnings. Cannot reproduce now, unfortunately. |
| 18:09.30 | manxpower | yziquel: "unmonitored" as in "sip show peers" shows Unmonitored? |
| 18:09.53 | nixer | jkroon: I'm using the wctdm24xxp driver for my TDM410 card. |
| 18:10.38 | yziquel | manxpower: as in 'meetme list n°ofconf' |
| 18:10.51 | jkroon | nixer, the same issue applies to any card supported by wctdm24xxp. which is what i use on the TDM400, 800 and 2400 cards. |
| 18:10.53 | manxpower | yziquel: weird |
| 18:11.10 | jkroon | nixer, apply the patch, recompile and go on with your live. |
| 18:11.26 | manxpower | I thought wctdm was used for tdm400? |
| 18:11.46 | *** join/#asterisk dandate2 (n=mangy@173-11-82-122-SFBA.hfc.comcastbusiness.net) |
| 18:14.38 | nixer | Someone commented: "I have the same issue with 1.6.2 todays SVN version, do you need any info to move this issue forward?" |
| 18:14.54 | nixer | They haven't applied the bloody patch. I'm not gonna recompile.. I'll go back to asterisk 1.4 and zaptel. |
| 18:15.49 | nixer | Well, at least now I know where the problem is. Thank you jkroon! |
| 18:16.03 | jkroon | i'm commenting again now. |
| 18:16.33 | dandate2 | talntid u around? |
| 18:16.51 | nixer | The status of the bug is confirmed, but it's still open, right? So it hasn't been committed to the code? |
| 18:17.35 | jkroon | tzafrir_laptop just reworked my patch for 1.6.2 from the looks of it, and i tried something down the lines of the dahdi patch of not resetting state back to INITIAL, which failed for me. the battery hack I don't know about, haven't tested that but I doubt it'll work since the alarm status is correct. |
| 18:18.19 | jkroon | so I'm personally in favor of just applying the tzafrir patch to 1.6.2 and my variant to 1.6.[01] |
| 18:19.11 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:19.17 | nixer | The problem is that if I do recompile asterisk, it may as well break PBX in a Flash and its binary scripts... so I'd rather go back to asterisk 1.4 for now until this issue is resolved. |
| 18:23.52 | jkroon | manxpower, just looked again at the code, looks like it was never correct in 1.6.x and that it worked whilst I was using the zaptel drivers, but that the hookstate thing changed during migration to dahdi. The dahdi behaviour seems more sane IMHO, to report the FXO availability via alarms, thus I don't think the issue should be fixed in the kernel. |
| 18:24.30 | jkroon | ok, need some help with call files and cdr generation stuff. my head is spinning a bit. |
| 18:25.33 | jkroon | what i've got at the moment: .call file has Channel: CAPI/groupname/${number} and then Context: Local/${number}@mycontext |
| 18:26.08 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 18:26.30 | jkroon | mycontext issues a Dial() in order to get stuff logged into a CDR(). however, if after a bunch of retries CAPI/groupname/${number} still doesn't answer the whole thing goes lost. Thus I need a different approach. |
| 18:27.55 | jkroon | the obvious change is to set Channel: Local/${number}@mycontext and then use Application: Dial with Data: CAPI/groupname/${number} ... which gets the CDR logged (properly even in failure cases) but without any custom Set(CDR(varname)=fneh) stuff that gets set in mycontext. |
| 18:27.56 | nixer | jkroon: Thanks again for the link. I've been going crazy over that. |
| 18:28.01 | *** part/#asterisk nixer (n=nixer@78.154.233.235) |
| 18:30.00 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 18:30.21 | *** join/#asterisk mrbnet (n=mrbnet@c-75-73-142-28.hsd1.mn.comcast.net) |
| 18:30.59 | mrbnet | Is it safe to setup endpoints over the internet or is it best to run through VPN |
| 18:44.46 | jkroon | that's useful ... wait for a very short time for an answer ... |
| 18:56.19 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 19:04.13 | *** join/#asterisk carl0s- (n=carl@87-194-108-252.bethere.co.uk) |
| 19:05.51 | carl0s- | I have an irritating problem. If I end an outbound call while it is ringing, before it has been answered, every time I see "chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call xxx". Is there a simple fix for this? Only happens when the call isn't answered. Host is sipgate. |
| 19:06.45 | carl0s- | the symptoms are that the number I rang carries on ringing for ages after I ended the call at the originating end. |
| 19:13.17 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 19:14.11 | _Raptor_ | ok, what is tonezone and what do i have to satisfy the dependency? i have installed dahdi and called: ./configure --prefix=/local/asterisk-1.6.1.2/ --with-dahdi=/local/asterisk-1.6.1.2/src/dahdi |
| 19:14.54 | jkroon | tonezone is a config setting. |
| 19:15.01 | thehar | carl0s-: what version of asterisk? |
| 19:17.30 | _Raptor_ | jkroon: ok, so what do i have to do? i am in menuselect and is see DAHDI Telephony\n.Depends on: res_smdi(M), dahdi(E), tonezone(E) dahdi is found (app_meetme now can be compiled) |
| 19:18.03 | jkroon | no clue. i always just set tonezone in /etc/dahdi/system.conf |
| 19:18.20 | *** join/#asterisk Elwell (i=elwell@freenode/staff/elwell) |
| 19:18.27 | carl0s- | thehar, the version that is apparently bad and has this problem ! 1.4.23.1 . I am just looking into how I can update. .maybe atrpms. I am running on Druid OSE y'see .. |
| 19:18.34 | Elwell | anyone speak spa3102 dialplans here? |
| 19:18.40 | _Raptor_ | jkroon: what version of asterisk? |
| 19:18.53 | jkroon | >=1.6.0 |
| 19:18.57 | thehar | carl0s-: yeah, appears to be a bug in that branch |
| 19:19.53 | _Raptor_ | jkroon: and you use chan_dadhi? |
| 19:20.03 | jkroon | _Raptor_, i do. |
| 19:21.16 | _Raptor_ | i think this is the problem: http://pbot.icip.de/2714bd677046f48c501ade0115596093 |
| 19:21.23 | _Raptor_ | but i have no clue what to do |
| 19:22.14 | jkroon | do you have dahdi properly installed? |
| 19:22.30 | _Raptor_ | make;, sudo make install |
| 19:22.38 | Elwell | assumes he's on his own (with google) for the dialplan Q then... |
| 19:23.39 | *** join/#asterisk Lis (n=Lis@p5B2076F1.dip.t-dialin.net) |
| 19:23.42 | *** part/#asterisk Lis (n=Lis@p5B2076F1.dip.t-dialin.net) |
| 19:24.25 | jkroon | Elwell, why would you need a specific dialplan for the spa3102? |
| 19:25.05 | *** join/#asterisk toresbe (n=toresbe@4.84-48-119.nextgentel.com) |
| 19:25.09 | toresbe | Hello :) |
| 19:25.39 | Elwell | jkroon: cos I want it to route to PSTN by default, and only pass to * when I prefix by # |
| 19:26.03 | jkroon | ah, config on the spa itself. check the thing's manual. |
| 19:26.16 | toresbe | Hey guys, I have a conundrum and I wasn't sure where to turn; I'll ask the question here and maybe someone here knows a good place to find out about this... |
| 19:26.19 | toresbe | A neighbour gave me some of his old ADSL crap; not only a standard ADSL/WLAN router, but also a telephone adapter for SIP VoIP service. This is great, because I have a cool 1970s phone I really wanted to hook up to Asterisk. |
| 19:26.23 | toresbe | However, when I boot it up, it acts a bit strange. It sends some form of UDP packet to a server called "apsv3.nsc.no" which belongs to the VoIP vendor. The only plaintext in the packets is "mac=...;alg=1". |
| 19:26.27 | toresbe | They are sent - four in rapid succession - with an extended timeout. Some of the packets receive a reply from a server on a completely different subnet, which has an ICMP header, but the checksum inside it doesn't match. |
| 19:26.31 | toresbe | The datasheet (in quite broken English) by the Chinese hardware vendor mentions "SecureAPS", which is a "secure auto provisioning system". |
| 19:26.34 | toresbe | This matches up to the server name, certainly; it seems to be some kind of configuration and firmware update deployment system. Has anyone had any experience with this? Has anyone managed to spoof a SecureAPS system, or know anything about that protocol? |
| 19:26.38 | thehar | O.o |
| 19:26.59 | toresbe | (I spent a lot of time phrasing the question somewhere else, so I just copy-pasted the question :) |
| 19:27.31 | Elwell | jkroon: yeah, the manual is crap tho |
| 19:27.36 | Elwell | np anyway |
| 19:28.29 | jkroon | toresbe, figured, however, i highly doubt anybody here would know. and an engrish manual doesn't exactly help either. |
| 19:28.55 | jkroon | you don't even mention exactly what the device is. |
| 19:29.22 | toresbe | jkroon: it's a Tecom GR1000ST; there is no manual, only a datasheet |
| 19:29.40 | thehar | toresbe: a lot of cpe devices we order from the manufacturer includes a custom firmware/provisioning to help us deploy them easier. |
| 19:29.44 | jkroon | toresbe, what does google say about the device? |
| 19:29.56 | toresbe | jkroon: little else than the data sheet. |
| 19:30.03 | toresbe | jkroon: I was hoping the problem wasn't specific to the device, that it was a known protocol or something. |
| 19:30.13 | toresbe | thehar: Does SecureAPS ring a bell? |
| 19:30.34 | thehar | no but sounds like it's just phoning home |
| 19:31.09 | toresbe | problem is, it doesn't send out any SIP data or anything, so even though it supports SIP in *principle* I haven't gotten that to work yet |
| 19:31.36 | _Raptor_ | make clean && ./configure now says: The TONEZONE installation appears to be missing or broken |
| 19:32.42 | toresbe | thehar: What I hoped to do is to spoof the config data if that's what it fetches... |
| 19:49.51 | *** join/#asterisk Vutral (n=vutral@nathan.66h.42h.de) |
| 19:49.53 | Vutral | hallo |
| 19:50.03 | Vutral | how are yoou |
| 19:51.22 | toresbe | Die wählte Nummer ist nicht in Betrieb; aufhangen und neuwählen. |
| 19:51.29 | Vutral | jaja |
| 19:51.35 | Vutral | ich brauch assistenz |
| 19:51.43 | Vutral | ich will nen board fixen |
| 19:51.49 | Vutral | aber irgendwie is die anleitung unverständlich |
| 19:51.57 | toresbe | Ich wähl die Nummer, rufe Bildschirmtext... |
| 19:52.03 | Vutral | steckdosenleiste ist da powerstrip |
| 19:52.07 | Vutral | o_O |
| 19:52.20 | toresbe | Surfen surfen durch die Welt, mit multimedia-aa! |
| 19:52.42 | Vutral | nich mit nappe |
| 19:52.50 | Vutral | *g* |
| 19:52.51 | toresbe | I don't command German well enough to give technical support. But I can recite song lyrics. |
| 19:52.57 | Vutral | oh nice |
| 19:53.06 | Vutral | i like aerials |
| 19:53.32 | toresbe | . o O http://www.youtube.com/watch?v=yFPQbnraeVg |
| 19:54.07 | Vutral | sorry i would like to look |
| 19:54.15 | Vutral | but my spdif is disfunctionally |
| 19:54.19 | Vutral | so no audio |
| 19:54.21 | Vutral | no fun |
| 19:55.14 | Vutral | what is a peremjutschkamii |
| 19:55.18 | Vutral | *g* |
| 19:56.08 | Vutral | rofl polski for mainboard is ANAKART |
| 19:57.15 | Vutral | so i have to go |
| 19:57.16 | Vutral | working |
| 19:57.41 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 20:00.59 | *** join/#asterisk ratdeptrai (n=dfavro@d14-69-67-36.try.wideopenwest.com) |
| 20:02.19 | ratdeptrai | Is there any way to play a sound-file into a meetme conference-room from within a dialplan? |
| 20:03.22 | Vutral | what is meetme |
| 20:03.41 | manxpower | ratdeptrai: In theory yes, in practice it would be complicated. Create a .call file to call into the conference and then make the other leg of the call jump to the dialplan and play the sound. |
| 20:03.56 | manxpower | Vutral: "core show application meetme" |
| 20:04.19 | Vutral | mhm |
| 20:04.24 | ratdeptrai | manxpower: yeah, that's what I was thinking too, but I was hoping there might be an easier way. |
| 20:04.36 | Vutral | i get payne in my eyes if i read words like ball grid array |
| 20:04.37 | manxpower | ratdeptrai: not that I'm aware of |
| 20:06.27 | manxpower | ratdeptrai: you might be able to use a Local/ channel to Dial into the Meetme, then use the M() or other dial option to play a sound file. |
| 20:11.00 | jkroon | ratdeptrai, create a .call file that uses Local/ to get into the Meetme channel, Application: Playback, Data: soundname |
| 20:11.11 | jkroon | that's pretty much the only option i can think of. |
| 20:11.19 | Vutral | playback hooray |
| 20:12.04 | jkroon | Vutral, i hear you re the ball grid array stuff. |
| 20:14.27 | ratdeptrai | Yeah, that's what I'll do. |
| 20:15.06 | ratdeptrai | Thanks. |
| 20:19.47 | yziquel | hi. when i have an incoming call from my sip provider, i get an INVITE sip:s@myip. asterisk tries to match for extension s, and doesn't match. How can i tell the dialplan to send the calls on a given extension regardless of the INVITE field? |
| 20:20.16 | *** join/#asterisk lizone (n=zenst@user-0ccejib.cable.mindspring.com) |
| 20:23.13 | jkroon | register => user:pass@provider/exten |
| 20:23.19 | Vutral | i am testing these strange patches for q931 support |
| 20:23.26 | Vutral | anybody knows why asteriks keeps failing |
| 20:23.33 | jkroon | if they still come in to s, just add a line exten => s,1,Goto(${wanted_exten},1) |
| 20:23.38 | Vutral | maybe its true alphaware |
| 20:25.33 | karleeto | leave |
| 20:25.37 | *** part/#asterisk karleeto (n=karl@server.nashvilleproweb.com) |
| 20:25.57 | yziquel | jkroon: oh!the /exten part of the register... thanks. |
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| 20:27.33 | *** part/#asterisk boscop (n=boscop@e181241025.adsl.alicedsl.de) |
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| 20:43.40 | *** join/#asterisk kiwi_ (n=kiwi@ks359129.kimsufi.com) |
| 20:43.56 | kiwi_ | asterisk is so great ... :D |
| 20:44.21 | Vutral | yeah i know |
| 20:44.28 | Vutral | even without batteries |
| 20:44.45 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 20:45.52 | *** join/#asterisk sack (n=sack@102.Red-79-148-189.dynamicIP.rima-tde.net) |
| 20:47.26 | drclue | Howdy all. I'm writing some FastAGI tools from scratch. I've reached the point where I can interactively issues commands. What I'm trying to figure out is what commands I would need to issue to after playing a message to a caller , forward their call to another number. Any clues? |
| 20:47.43 | kiwi_ | no batteries |
| 20:51.49 | ratdeptrai | drclue: how about "EXEC Playback" followed by "EXEC Dial" ? |
| 20:52.00 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 20:53.53 | drclue | That seemed to be a thought , although maybe I did not do it right. I can try again real quick since I phrased my FastAGI to dynamically load "plug ins" of a sort to keep from having to start and stop my FasAGI server. So lets see. |
| 20:55.28 | drclue | "cannot be completed as dialed" message. I've been programming for decades , but I've only been messing with asterisk for a month , so I'm still learning |
| 20:55.38 | ratdeptrai | drclue: have you been able to successfully execute any simple commands to start with? What language are you working in? If you're having problems, try getting an asterisk console running and turn on some AGI debugging to see WTF is happening. |
| 20:57.24 | drclue | I have been able to use the ANSWER,Flite and other commands , and I have the responses from same displaying in my console window. I also have rasterisk running in another window |
| 21:00.09 | drclue | After the FastAGI call completes , the next line in rasterisk says NoCDR, then Wait , then "cannot complete as diald" |
| 21:06.27 | drclue | As to the language I'm writing the FastAGI in , I'm using PHP to prototype the tools and will probably translate it to C++ later. I really doubt that that is the issue though. I tend to believe I'm just issuing the commands improperly. Right now I do an [ANSWER], followed by an [EXEC Flite] to acknowledge the user with a message and then I'm attempting to [Dial] another number , but I'm not sure if I have that right, |
| 21:06.51 | ratdeptrai | drclue: EXEC Dial |
| 21:07.01 | drclue | Ya EXEC Dial |
| 21:08.43 | drclue | To be more precise "EXEC Dial \"2000\"" |
| 21:09.04 | ratdeptrai | drclue: those quotes might give problems, I'm not sure. |
| 21:09.27 | Vutral | it wont if you do it right |
| 21:09.34 | Vutral | i did that multiple times in many codes |
| 21:10.08 | ratdeptrai | drclue: What's the problem, I mean, what makes you think that it isn't working? Also, what's the result-code? If your FastAGI does an EXEC Dial (which is answered), I believe that it will not get a 200 back until one party or the other hangs up. But I could be mistaken. At any rate, be aware that (per http://www.voip-info.org/wiki/view/Asterisk+AGI ) "If the AGI application dials outward by executing Dial, control over the call |
| 21:10.13 | drclue | I just tried this "EXEC Dial 2000" and that don't seem to float either |
| 21:11.20 | ratdeptrai | drclue: try turning on AGI debugging in an asterisk console, I think that might illuminate a lot. |
| 21:12.49 | Vutral | mhm |
| 21:13.46 | drclue | Well , what makes me think it is not working is that when I ask it to dial the 2000 |
| 21:13.46 | drclue | Did not notice a special command in rasterisk for debugging AGI (just turned it on) |
| 21:13.46 | drclue | Does not seem to illuminate much |
| 21:14.55 | ratdeptrai | drclue: what's the result line that you get back after EXEC Dial 2000 ? |
| 21:15.21 | drclue | 200 -1 |
| 21:16.18 | ratdeptrai | drclue: maybe you need more that just 2000? How about EXEC Dial Local/2000@your_context ? |
| 21:16.27 | drclue | I think I just figured it out SIP/2000 vs 2000 |
| 21:16.36 | Vutral | ahaha |
| 21:16.49 | Vutral | someone knows howto attach a S0 bus to asterisk |
| 21:16.57 | Vutral | query me for details |
| 21:17.14 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 21:19.20 | drclue | Actually this Asterisk stuff is a lot of fun for me. It has taken a month to go from someone hiring to setup an Asterisk machine to where I'm at, but it's entertaining |
| 21:25.03 | drclue | So lets say that instead of simply forwarding the call straight away , that I wanted to implement a group or sequential hunt from inside the FastAGI while perhaps placing the caller on hold? |
| 21:26.32 | *** part/#asterisk sack (n=sack@102.Red-79-148-189.dynamicIP.rima-tde.net) |
| 21:30.52 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:38.37 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 21:38.40 | giesen | I need some help |
| 21:38.49 | giesen | I'm trying to setup a SIP trunk between a CME and Asterisk |
| 21:39.01 | giesen | authenticated only on IP (no registration |
| 21:39.08 | giesen | can't seem to do it though |
| 21:39.24 | giesen | all my calls into the asterisk box show: Received incoming SIP connection from unknown peer |
| 21:39.30 | giesen | (using trixbox 2.8) |
| 21:41.26 | jkroon | giesen, i've tried that one too, between asterisk boxen, and couldn't get it right. |
| 21:42.03 | jkroon | you still need to authenticate each call, the host= line only specifies where to pass outgoing calls to, |
| 21:42.56 | giesen | bah |
| 21:43.00 | giesen | retarded |
| 21:43.06 | giesen | I think I found a workaround |
| 21:43.14 | giesen | http://nerdvittles.com/index.php?p=208 |
| 21:44.03 | giesen | host= is supposed to allow you to use insecure=very |
| 21:44.06 | giesen | I believe |
| 21:44.13 | giesen | so you can authenticate calls solely on IP |
| 21:53.39 | Vutral | lol all paramedics are markoff models |
| 21:56.46 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 21:59.40 | drclue | Can Asterisk call ques be created on-the-fly from FastAGI? |
| 22:11.52 | *** join/#asterisk propellerhead (n=yogurt2u@host68.190-31-73.telecom.net.ar) |
| 22:21.51 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 22:24.08 | drclue | Maybe a different question. If I wanted to dynamically Dial multiple numbers from within a FastAGI and connect an incoming call to the first one that answered , what would be the best way to do that? |
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| 22:55.34 | yziquel | can i write 'permit=domain.net' in sip.conf for my SIP provider, or must i stick with network ip and masks? |
| 23:28.54 | *** join/#asterisk VooDooNOFX (n=Joe@pool-71-104-35-56.lsanca.dsl-w.verizon.net) |
| 23:29.27 | VooDooNOFX | Can anyone recommend a Canadian SIP trunk carrier that sells metered plans? |
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