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| 00:09.45 | LiNeTuX | OMG /. is down |
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| 00:12.08 | mwalling | ha |
| 00:12.25 | mwalling | declares a national emergency... |
| 00:12.48 | mwalling | not for /. being down, but for the fact that the trolls might emerge from their mother's basements |
| 00:12.52 | LiNeTuX | files for bankruptcy |
| 00:12.59 | LiNeTuX | heh |
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| 00:13.09 | LiNeTuX | they have to comment somewhere |
| 00:13.54 | LiNeTuX | Nerds everywhere have felt a disturbance |
| 00:25.56 | jblack | It has been down a bit long for them |
| 00:26.02 | flitex_666 | I feel a disturbancce in the force.. |
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| 00:28.26 | UngaMan | hello... godd evening |
| 00:28.29 | UngaMan | *good |
| 00:30.05 | UngaMan | I've been reading Digium website and found some recommeded motherboard where to install the cards |
| 00:30.37 | UngaMan | my question is: could any computer with enough PCI slots available work with those cards? |
| 00:30.40 | UngaMan | thank you |
| 00:31.09 | deeperror | UngaMan, probably |
| 00:31.16 | LiNeTuX | UngaMan: there are very few motherboards that play nice with phone stuff |
| 00:31.54 | LiNeTuX | it'd be fine for testing, but i wouldn't use it in production |
| 00:32.12 | UngaMan | ooook |
| 00:32.42 | UngaMan | so I have to include those recommended motherboards in my Buy List |
| 00:32.48 | UngaMan | interesting |
| 00:32.58 | LiNeTuX | you can also buy from folks who specialize in * hardware |
| 00:33.13 | LiNeTuX | Rhino is one I've used, there's a lot of others. |
| 00:33.19 | UngaMan | other than Digium... |
| 00:33.25 | UngaMan | Rhino... :: taking notes :: |
| 00:33.28 | UngaMan | :) |
| 00:33.28 | LiNeTuX | I like Rhino and Redfone |
| 00:33.29 | UngaMan | ok |
| 00:33.47 | LiNeTuX | the guys at voipsupply.com have never let me down on recommendations |
| 00:33.48 | UngaMan | thank you so much for this tip |
| 00:34.08 | LiNeTuX | (and no, i don't work for them, but have spent about $200K with them) |
| 00:34.19 | UngaMan | hehe... enough proof |
| 00:34.55 | deeperror | be careful of rhino r4t1 had lots of issues on that one but the r1t1 cards worked 100% |
| 00:35.12 | UngaMan | oh! |
| 00:35.19 | UngaMan | is taking notes |
| 00:35.51 | LiNeTuX | UngaMan: just come back with your shortlist and I'm sure you'll get an opinon on it |
| 00:37.50 | ManxPower | The problem with using cards other than Digium and Sangoma is that you will find virtually no support for those other cards here, |
| 00:38.58 | UngaMan | ok |
| 00:39.09 | UngaMan | LineTux: will do! |
| 00:39.25 | UngaMan | ManxPower: ok |
| 00:40.01 | ManxPower | I can't comment on the QUALITY of the other cards, other than to say that most of them (not Rhino, I've been told) are based on the open source Zapata specification -- which is what the FIRST Digium T-1/E-1 cards used -- that is like 4 generations old |
| 00:40.29 | UngaMan | is installing * in to OpenSuSE Virtual Boxes... willing to test the Dual Box Connectivty |
| 00:40.32 | lmadsen | and no one sane would use that design |
| 00:40.35 | UngaMan | *two |
| 00:41.33 | ManxPower | That old design is why many people switched away from Digium in the early days |
| 00:42.17 | lmadsen | luckily they have changed it and re-engineered the card -- that design is no longer in use in any of the digium cards now |
| 00:42.54 | ManxPower | lmadsen: *nod* That design has not been used on Digium cards for several years. |
| 00:42.59 | lmadsen | agreed |
| 00:43.11 | lmadsen | it was more of a proof of concept |
| 00:43.16 | LiNeTuX | UngaMan - if you're thinking about HA, look into Redfone |
| 00:44.02 | UngaMan | LineTux: ok |
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| 00:44.18 | UngaMan | will check later tonite |
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| 01:11.46 | UngaMan | g'nite |
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| 01:32.22 | ngvoice | d |
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| 01:37.59 | BeeBuu | hello,all |
| 01:38.25 | BeeBuu | how to transfer a call when i as agent? |
| 01:48.51 | hsv-al | heh |
| 01:48.54 | hsv-al | i just caved into insanity again |
| 01:48.58 | hsv-al | ran another 4 miles at night, 8 for the day |
| 01:49.12 | hsv-al | all because i had taco bell last night - guilt trip :) |
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| 02:17.40 | l0verb0y | hey hows it going |
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| 03:53.19 | C4away | just out of curiosity, if a company was offering you piecemeal asterisk maintance work on ther servers, as a contractor, what would you consider a fair hourly rate? |
| 03:54.08 | C4away | average of about 20 hours per month, a project here, another project there... |
| 03:57.25 | C4away | with a promise of someday, maybe, a full-time position |
| 04:00.36 | C4away | oh, this is in the USA for reference |
| 04:02.12 | Strom_C | C4away: I do that kind of consulting work |
| 04:02.35 | C4away | I've been offered a job doing that and I'm wondering if it is a fair offer |
| 04:02.54 | Strom_C | may I PM you? |
| 04:03.04 | C4away | sure |
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| 05:01.45 | jblack | sigh. why does fun asterisk stuff turn into stuff like: http://jblack.linuxguru.net/~jblack/week_table.html |
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| 05:18.31 | drmessano | I am seriously starting to detest <insert pbx-on-a-cd distro here> users that know nothing about the CLI before installing |
| 05:18.52 | drmessano | Its one thing to want to save time to create a GUI install if you're just slinging GUI boxes out there |
| 05:19.14 | drmessano | Its another to expect a fully manageable system without ever having touched linux before |
| 05:19.29 | Maliuta | drmessano: s/about the CLI/anything/ |
| 05:19.50 | hsv-al | drmessano, i saved alot of $ |
| 05:19.54 | hsv-al | by switching to geico |
| 05:19.54 | Maliuta | drmessano: just starting ... I got over it years ago |
| 05:20.12 | drmessano | I am not a GUI nazi like some, but if you're going to use a GUI, don't be such a Vista user about it |
| 05:20.58 | jblack | I agree. "High end tools" that almost work are worse than no "high end" tools at all. |
| 05:21.05 | Maliuta | drmessano: I once demo'd a heap over a network that I had installed on the other side of uni campus to a newb, only to have him ask "where do I click to get that?" |
| 05:21.07 | drmessano | Not being able to open a CLI to do a simple verbose 8 to watch a call? Come the hell on, people |
| 05:21.45 | drmessano | or telling me you know the CLI but can't cat or nano a file |
| 05:21.49 | drmessano | CAT? |
| 05:21.52 | drmessano | I mean |
| 05:21.56 | Maliuta | a keyboard? how quaint? </scotty> |
| 05:22.13 | styelz | heh |
| 05:22.14 | jblack | Speaking of bitchin n moaning, I need some advice. |
| 05:22.22 | styelz | computer! |
| 05:22.38 | drmessano | I am gonna create an app called DOG that calls rm -Rf |
| 05:22.42 | Maliuta | jblack: the answer is "dog" |
| 05:22.43 | drmessano | dog * |
| 05:23.07 | jblack | A few months ago, I bid a job for $2k that was basically 'install * for me and a couple other things'. |
| 05:23.23 | jblack | And that was fine and good. But, I couldn't stop there. Just kept fiddling, adding stuff, having fun. |
| 05:23.30 | drmessano | Oh christ.. |
| 05:23.38 | Maliuta | drmessano: it's not like the old days where you could make them do bad stuff, then put a cat /dev/random > /dev/mem |
| 05:23.39 | drmessano | You slept with him.. didnt you? |
| 05:24.02 | jblack | They keep letting me letting me do whatever I want. I'm up to $15k of work for a $2k job, becuase I'm enjoying it so much. |
| 05:24.21 | Maliuta | jblack: have they paid you the $15k? |
| 05:24.47 | Maliuta | or have you done $15k of work for $2k of pay? |
| 05:24.49 | jblack | No, but they've paid the money they've promised. |
| 05:24.59 | Maliuta | fail! |
| 05:25.05 | drmessano | rut ro |
| 05:25.17 | jblack | I'm overachieving, and having a great time at it, but not getting extra money for the extra work. |
| 05:25.25 | jblack | I can see it from their side "OhhH! Freebies!". |
| 05:25.28 | Maliuta | your fault |
| 05:25.42 | jblack | Yeah, it's my fault. |
| 05:25.50 | drmessano | Don't expect to see that money, but I would see if they want to sign a support contract |
| 05:25.53 | Maliuta | you just raised the expectations on your next job for them |
| 05:26.05 | drmessano | Come up with a compromise and you can still do work for them and get paid regular |
| 05:26.06 | jblack | No, i don't expect them to send me a big wad of fifties or anything. |
| 05:26.12 | Maliuta | I would say you are never going to see the money out of them |
| 05:26.37 | drmessano | Use that work you've done as leverage of sorts.. and talk them into a contract |
| 05:26.48 | drmessano | then you can play and get paid |
| 05:26.50 | jblack | But seeing as how it's become a full time job that I'm enjoying, I'd like it to be full time pay. |
| 05:26.51 | Maliuta | _and_ you now have to support the freebies or it will be "why is this breaking our system, we never asked for it" |
| 05:27.09 | jblack | I don't have to support freebies. It's in the contract. |
| 05:27.17 | drmessano | Can they afford a full time PBX guy? |
| 05:27.40 | jblack | By the time they're done leveraging the info my extra work is providing them, I think they will be able to, yes. |
| 05:28.25 | Maliuta | jblack: tell them to forget you and hire me, I'll be cheaper and work from remote |
| 05:28.29 | drmessano | Considering what you posted earlier, it looks like you've gone beyond a PBX tech anyway.. you're helping them with statistical analysis and whatnot |
| 05:28.35 | drmessano | So yeah.. worth a shot |
| 05:28.51 | jblack | yeah. |
| 05:29.16 | jblack | I know the owner is drooling over the stuff I do |
| 05:29.30 | jblack | and this is really a taste of the analysis I really want to do. |
| 05:29.33 | drmessano | jblack: You are giving him tools to analyze SALESPEOPLE.. who wouldnt LOVE that? |
| 05:29.45 | jblack | I want to start putting machine learning on the task. |
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| 05:31.50 | jblack | What makes me nervous is if they're short sighted, they'll take what I provided, and say thanks, and move on. |
| 05:31.58 | jblack | I.E. not see this properly as an appitizer. |
| 05:33.07 | jblack | And if I don't push at all, it'll just continue being a sinkhole |
| 05:33.12 | drmessano | Well |
| 05:33.23 | drmessano | I would push it.. right now, you're getting $0 from it |
| 05:33.41 | jblack | That's a good point. Nothing to lose at this point |
| 05:33.43 | drmessano | If they dont want to invest, find out who his competition is, and sell them a PBX |
| 05:33.53 | drmessano | lol |
| 05:33.56 | jblack | Yeah, actually, since these are freebies, I own the copyright to them. |
| 05:34.58 | drmessano | When I worked in Radio, do you know what the #1 tool I found myself deploying every 6 months was? |
| 05:35.06 | jblack | a baseball bat? |
| 05:35.14 | drmessano | Some shit to track the salespeople.. who to fire, who to fire, who to promote.. |
| 05:35.23 | drmessano | who to hire* |
| 05:35.36 | jblack | That's exactly where this stuff leads, yeah |
| 05:36.05 | drmessano | Yeah, nobody wants a slacker on staff.. Everyone wants granular analysis of everything |
| 05:36.31 | drmessano | If you have a knack for that stuff and can tie to asterisk like you have, work it |
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| 05:37.03 | jblack | I love it. That's why I kept working on it long after the contract was satisfied and became worthless. |
| 05:37.47 | jblack | There's one more wrinkle to it. |
| 05:38.13 | jblack | Because of externalities, I wrote the contract up as a 4 month contract, which doesn't expire for another month and a half. |
| 05:38.15 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) [NETSPLIT VICTIM] |
| 05:38.15 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-116-111-122.rgv.res.rr.com) [NETSPLIT VICTIM] |
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| 05:39.49 | drmessano | Well, you've completed what was part of the contract, right? |
| 05:43.20 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:43.27 | TJNII | just got screwed on a RAID card |
| 05:43.36 | TJNII | Supported on all major OSes my ass..... |
| 05:44.24 | BeeBuu | drmessano: hello |
| 05:44.56 | BeeBuu | drmessano: would you teach me how to transfer a call when a agent answered? |
| 05:45.23 | TrentCreek | flash panel |
| 05:47.02 | BeeBuu | TrentCreek: i mean the agent press keys to transfer~~~ |
| 05:47.34 | TrentCreek | the BOOK has it |
| 05:48.17 | drmessano | BeeBuu: No, but I can PM you my PayPal info and do it for large sums of cash |
| 05:48.43 | BeeBuu | TrentCreek: would you tell me which page start? |
| 05:49.09 | BeeBuu | drmessano: you are kidding me..... |
| 05:49.13 | TrentCreek | not sure..been months since I looked at it...dial plans |
| 05:49.20 | TrentCreek | i think |
| 05:49.45 | BeeBuu | TrentCreek: i can't find the context that you said. |
| 05:50.07 | TrentCreek | you have to set each phone as an extension |
| 05:50.11 | TrentCreek | should be in there |
| 05:50.37 | drmessano | BeeBuu: I normally don't ask for Paypal donations, but for you, I will make an exception |
| 05:50.49 | TJNII | laughs |
| 05:50.58 | BeeBuu | :-( |
| 05:52.04 | TrentCreek | it should recognize the correct button on the phone for all that stuff |
| 05:52.53 | BeeBuu | i had tried #XXX,but it hangup... |
| 05:53.19 | BeeBuu | anything wrong? |
| 05:53.19 | TrentCreek | did you set the extensions for each phone? |
| 05:53.29 | BeeBuu | TrentCreek: yes,i did. |
| 05:53.56 | TrentCreek | you better look at the book |
| 05:54.15 | TrentCreek | it's all in there |
| 05:54.32 | BeeBuu | TrentCreek: i set up a queue ,and working... |
| 05:54.51 | TrentCreek | there are also many video tutorials online to check out |
| 05:54.55 | BeeBuu | but i want to make the agents can transfer call... |
| 05:55.05 | drmessano | TrentCreek: BeeBuu doesn't read the book or follow links.. thats why I refuse to help him |
| 05:55.28 | drmessano | TrentCreek: He'll waste a lot of your time and expect you basically spoon feed him |
| 05:55.43 | jblack | drmessano: Yeah, I've essentially completed the contract, and thrown in a good bit besides. |
| 05:56.18 | BeeBuu | drmessano: as your mean, i can teache anyone with a "read the book"? |
| 05:56.41 | drmessano | jblack: Then I would make them well aware that you're in support mode now... and ask them them if they want to pursue further with you |
| 05:56.46 | jblack | BeeBuu: Your agents should be able to transfer with the transfer button on the phone. |
| 05:56.51 | TrentCreek | ohhhh |
| 05:57.19 | jblack | Yeah. That's good advice. |
| 05:57.41 | TrentCreek | The book should cover how to set up extensions and dial them...I saw it myself |
| 05:57.43 | jblack | Talk to them about formalizing the impromptu deepening of our business relationship. |
| 05:58.18 | BeeBuu | jblack: i press #,and get voice,but when i pressed numbers,it hangup... |
| 05:58.29 | jblack | beebuu: hmm. Ok. |
| 05:58.55 | drmessano | jblack: Indeed.. Tell them you've completed all the work, you're in support mode.. you've done this extra work really as free work, and you would like to deepen or formalize a continuance if they are interested in seeing more |
| 05:59.17 | drmessano | Worse they can do is say no and give you the same $0.. as was already mentioned |
| 05:59.24 | TrentCreek | BeeBuu: Have you just tried dialing the extension? |
| 05:59.40 | jblack | Yup. |
| 05:59.45 | drmessano | I hear theres a good Asterisk book |
| 05:59.45 | jblack | excellent advice |
| 05:59.58 | jblack | BeeBuu: Does your phone not have a transfer button on it? |
| 06:00.30 | drmessano | jblack: They will either go for it, or show they're just a bunch of cheap asses looking for a handout, in which case, better to know now then later.. |
| 06:00.39 | BeeBuu | TrentCreek: yes.it work. |
| 06:01.02 | BeeBuu | jblack:i work in a zap channel. |
| 06:01.12 | TrentCreek | then press the transfer button and dial the extention number |
| 06:01.13 | BeeBuu | jblack: i using a normal phone |
| 06:01.28 | BeeBuu | TrentCreek: you mean #? |
| 06:01.34 | jblack | Yeah. I might enjoy the job, but not so much that I'd like to having the current situation continue indefinitely. Better to find where they stand. I can eitehr keep playing, or cut bait |
| 06:01.34 | TrentCreek | then press the FLASH button, dial 123 |
| 06:01.59 | BeeBuu | it still handup the line.... |
| 06:02.32 | drmessano | jblack: ... and that could free you up to pursue something putting money in your pocket.. right now you're holding your earning potential hostage to them . |
| 06:02.47 | jblack | True. |
| 06:03.03 | drmessano | LET MY BANK ACCOUNT GO |
| 06:03.44 | jblack | I go long times between employment for various reasons. |
| 06:04.45 | drmessano | Hiding from the CIA is a bitch.. Just remember to keep disguising yourself as a WMD |
| 06:04.46 | jblack | I know their wire monkey is worried that some day I'll fade away. ;) |
| 06:04.59 | BeeBuu | when i press #,get a voice : " transfer ", and i press numbers,but it hangup~~~ |
| 06:04.59 | TrentCreek | BeeBuu: wow this only took .00000001 seconds using google |
| 06:05.02 | TrentCreek | http://www.voip-info.org/wiki/view/Asterisk+PBX+functions |
| 06:05.22 | coppice | I have a PAP2T. I have the latest firmware on it. Contrary to what folks convinced me a few weeks ago, it has no T.38 support |
| 06:05.44 | BeeBuu | TrentCreek: thanks a lot,i'm reading.. |
| 06:06.21 | drmessano | TrentCreek: He's not |
| 06:06.24 | drmessano | TrentCreek: Not really |
| 06:06.28 | drmessano | TrentCreek: ;) |
| 06:06.28 | TrentCreek | LOL |
| 06:06.48 | drmessano | TrentCreek: Get SSH access if you really want to help him.. otherwise, go for a beer run real quick |
| 06:06.54 | drmessano | TrentCreek: he'll wait |
| 06:07.25 | TrentCreek | yeah..beeeeeer |
| 06:08.39 | drmessano | I'll shut up now.. I made my point. |
| 06:09.06 | drmessano | http://www.slash7.com/pages/vampires |
| 06:10.16 | TrentCreek | How do I install asterisk? |
| 06:10.28 | drmessano | ~helpvampire |
| 06:10.29 | jbot | Instead of consuming of ill-gotten hemoglobin, these vampires suck the very life and energy out of people. By nature they feed on generous individuals who tend towards helping others, and leave their victims exhausted, bitter and dispirited. See: http://www.slash7.com/pages/vampires |
| 06:11.37 | drmessano | That "How do I build a forum" is "How do I build a PBX" |
| 06:12.27 | drmessano | Good god |
| 06:12.35 | drmessano | I gave my wife a CD to install on her PC |
| 06:13.16 | drmessano | She had a CD already in the drive.. the install disk for her MP3 player... that I got her for a wedding present... in October |
| 06:13.25 | drmessano | It's been in there since |
| 06:14.03 | drmessano | That rocks.. |
| 06:15.11 | TrentCreek | I did the same here too,..iPod Shuffle...just sat it on the side |
| 06:17.06 | drmessano | coppice: The PAP2T does have T.38 support in some version.. Is it a 3.0.. You sure about the firmware? |
| 06:18.19 | coppice | I was told 3.3 had T.38, but that doesn't seem to be available now. I was also told 5.1.6 had it, but the documentation didn't really mention it. Well, if 5.1.6 has it, I think some pixie dust is needed to enable it |
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| 07:05.00 | Alpha_AI | Hello everyone |
| 07:05.17 | *** join/#asterisk CVirus (n=GoD@82.201.174.159) |
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| 07:09.20 | Alpha_AI | im looking to speak to someone so i can get some professional advice. Anyone can help me? |
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| 07:20.07 | l0verb0y | What kind of advice |
| 07:29.51 | Alpha_AI | im looking for implementation advice |
| 07:30.07 | Alpha_AI | integration advice |
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| 07:33.06 | ice_croft | hi all |
| 07:33.54 | ice_croft | have Asterisk 1.4.18 |
| 07:34.21 | ice_croft | 2. sip trunk |
| 07:34.33 | ice_croft | 3. allow=gsm on it |
| 07:35.06 | ice_croft | when i try to dial, * sez "no audio format avaliable" |
| 07:35.31 | ice_croft | but, when i set allow=gsm in [general], it works |
| 07:35.35 | ice_croft | where to dig? |
| 07:42.51 | synthetiq | sez? |
| 07:43.31 | ice_croft | says |
| 07:44.22 | ice_croft | reboot. later :( |
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| 07:45.53 | Thackyner | Hi ! |
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| 07:46.47 | Thackyner | I'm installing a SIP firmware on a cisco 7970 |
| 07:47.07 | Thackyner | But, the language must be french |
| 07:47.46 | Thackyner | Can anyone help me please ? (I've a file like CME-locale-fr_FR-4.0.2-2.0.tar) |
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| 07:49.30 | Thackyner | ok i've found the problem ^ it's a sccp language file |
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| 07:51.48 | BeeBuu | the book says: then transfer will happen only if the incoming call is of the same channel type. |
| 07:53.32 | BeeBuu | what's it real mean? |
| 07:54.20 | BeeBuu | a Zap channel only can be transfer to another Zap channel? |
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| 07:55.11 | bbryant | BeeBuu, it's refering to native transfers, which are when two peers start talking to each other without asterisk involving without the media and/or signaling |
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| 07:55.29 | bbryant | s/involving without/involved with/ |
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| 09:02.41 | Slashman | does someone knows if res_ldap is implemented in the classic package pf asterisk 1.4.20 ? |
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| 09:11.59 | xacatecas | ~book |
| 09:12.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 09:34.07 | mvanbaak | Slashman: download it and see for yourself |
| 09:35.12 | mvanbaak | Slashman: I dont see a res_ldap.c in 1.4.20.1 svn |
| 09:35.15 | mvanbaak | so I guess not |
| 09:35.25 | mvanbaak | http://svn.digium.com/view/asterisk/tags/1.4.20.1/ |
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| 09:35.39 | pputman | morning |
| 09:36.37 | Slashman | mvanbaak : does this mean that I need 1.6 to have res_ldap support ? or can I compile 1.4 with it ? |
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| 09:44.27 | mvanbaak | maybe there's a backport |
| 09:44.29 | mvanbaak | I'm not sure |
| 09:44.37 | *** join/#asterisk Zoup (n=Zoup@78.110.121.1) |
| 09:44.40 | Zoup | Hey all , |
| 09:45.09 | Zoup | which kernel options that asterisk depend on ? my Debian asterisk does not start ( seems freezing ) with customer 2.6.25 kernel |
| 09:45.19 | Zoup | s/customer/custom |
| 09:48.11 | pputman | Zoup, I wouldn't think that would be an asterisk issue, also though possibly a zaptel one. Do you have any zaptel hardware installed? |
| 09:49.55 | Zoup | pputman: No , i working on custom distribution with zaptel and Asterisk |
| 09:50.03 | pputman | If not, some common options I've had to try to get systems to boot are noapic, setting acpi=off, and pci=nomsi for my system here to get it to boot. |
| 09:50.06 | Zoup | pputman: by the way , this is not issue with debian kernel |
| 09:50.29 | pputman | Zoup, hrm not sure |
| 09:52.26 | Zoup | pputman: its 'service asterisk start' that stalls , can it be related to zaptel ? |
| 09:52.55 | pputman | Zoup, if it's causing it to kernel panic, I'd say it's a good possibility |
| 09:53.02 | pputman | what version of zaptel? |
| 09:53.22 | Zoup | No , its not a system freeze , start script just stay stalled |
| 09:53.38 | tzafrir_laptop | Zoup, asterisk freezing the kernel? can you try running it without -p? |
| 09:53.55 | *** join/#asterisk aksyn (n=aksyn@78.86.127.229) |
| 09:54.13 | Zoup | pputman: its 1.4.10.1 |
| 09:54.39 | tzafrir_laptop | Zoup, anyway, zaptel and asterisk are rather well-maintained. check http://updates.xorcom.com/rapid ('etch main') - automatic backports |
| 09:54.58 | tzafrir_laptop | and the live cd http://updates.xorcom.com/iso/ (live) |
| 09:55.07 | Zoup | tzafrir_laptop: nothing happens , it might be related to debian startup script ... |
| 09:55.07 | pputman | Zoup, and you don't have any zaptel hardware in the system? Like possibly any cards with an echo cancelation module on it? |
| 09:55.08 | tzafrir_laptop | needs to go |
| 09:55.20 | Zoup | pputman: no , theres no card |
| 09:55.50 | pputman | dunno |
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| 09:56.29 | Zoup | pputman: thanks :) |
| 09:57.29 | pputman | Zoup, but there is a zaptel 1.4.11 you could try to compile. |
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| 10:00.31 | Zoup | pputman: Thanks :) |
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| 11:18.06 | yang | Is there a way top tell CDR log to use local time instead of UTC ? |
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| 11:22.21 | pputman | yang, a quick look at cdr.conf has a setting: usegmtime=yes |
| 11:23.23 | yang | thanks !! |
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| 11:54.35 | viperdude | hi guys, anyone around who knows at Remote-Party-ID? |
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| 12:10.48 | FreezeS | hey guys |
| 12:10.58 | FreezeS | I've got a problem with cdr_addon_mysql |
| 12:11.10 | FreezeS | it's loaded, but the mysql cdr backend is not enabled |
| 12:11.31 | FreezeS | also there is no message from it in debug |
| 12:12.00 | FreezeS | (I'm using 1.4.20.1) |
| 12:12.10 | FreezeS | on amd64 |
| 12:14.37 | mvanbaak | what do you get with: module load cdr_addon_mysql |
| 12:14.45 | mvanbaak | add .so to that line :) |
| 12:14.48 | mvanbaak | what do you get with: module load cdr_addon_mysql.so |
| 12:15.02 | FreezeS | load_resource: Module 'cdr_addon_mysql.so' already exists. |
| 12:15.45 | mvanbaak | ok |
| 12:15.50 | mvanbaak | module unload cdr_addon_mysql |
| 12:15.53 | mvanbaak | and then: |
| 12:15.58 | mvanbaak | load it again |
| 12:16.04 | mvanbaak | look what it tells you |
| 12:16.57 | FreezeS | Unknown directive 'dbsock' at line 15 of /etc/asterisk/cdr_mysql.conf |
| 12:16.58 | FreezeS | ahaa |
| 12:17.24 | FreezeS | dunno how I could have skipped that line from the log |
| 12:17.26 | FreezeS | thanks |
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| 12:18.23 | FreezeS | however, the problem seems to persist |
| 12:18.42 | FreezeS | the module was loaded succesfully, but I still don't see mysql as a registered backend |
| 12:21.47 | FreezeS | and even funnier is that realtime mysql works perfectly |
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| 12:28.03 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 12:32.46 | FreezeS | anyone ? |
| 12:35.11 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:35.57 | *** join/#asterisk Datax (n=john@smirnoff.nurvnet.org) |
| 12:36.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:37.09 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 12:40.40 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 12:41.56 | viperdude | hi guys, anyone around who knows at Remote-Party-ID? |
| 12:42.55 | Datax | Hi all, anyone have an idea why the command "console" isn't recognized on my asterisk CLI ? |
| 12:43.09 | Datax | I've installed a new server and have decided to configure it from scratch, no model config files |
| 12:43.22 | [TK]D-Fender | viperdude: what about it? |
| 12:43.26 | Datax | So I'm certain that I've forgotten something but I don't know what :) |
| 12:44.20 | [TK]D-Fender | Datax: Odds are the syntax changed and that term is no longer a valid start of the command you're trying to use. "help" <--- |
| 12:44.20 | viperdude | [TK]D-Fender: i have set sendprid=yes in sip.conf but i need to set the privacy=yes/no on a per call basis |
| 12:44.37 | viperdude | depending on if i want to show or withhold CLI |
| 12:44.40 | Datax | [TK]D-Fender : I'm trying to use the command console dial |
| 12:44.57 | Datax | [TK]D-Fender : I'm using asterisk 1.4.20 |
| 12:45.25 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:45.30 | [TK]D-Fender | viperdude: I believe "core show application setcallerpres" should answer that. |
| 12:45.39 | Datax | when I type help there are no commands with the command "console" in them |
| 12:46.01 | viperdude | ok tanks |
| 12:46.03 | [TK]D-Fender | Datax: So you said NO config files huh? |
| 12:46.19 | Datax | [TK]D-Fender : I've created : extensions.conf logger.conf modules.conf sip.conf voicemail.conf |
| 12:46.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:46.54 | [TK]D-Fender | Datax: Something tells me your modules.conf is lacking. |
| 12:47.19 | Datax | [TK]D-Fender : here is my config file |
| 12:47.20 | Datax | [modules] |
| 12:47.20 | [TK]D-Fender | Datax: And you didn't mention an asterisk.conf either |
| 12:47.20 | Datax | autoload=yes |
| 12:47.23 | Datax | thats all |
| 12:47.26 | Datax | ah indeed ! ;) |
| 12:47.39 | [TK]D-Fender | it'd be nice if * had a clue where to FIND your modules... |
| 12:50.30 | Datax | mhhh, thought it would be clever since I left all of the default paths |
| 12:51.03 | *** join/#asterisk xrem1x (n=remi@bg-fw2out.monmouth.com) |
| 12:51.15 | [TK]D-Fender | Datax: You have outsmarted yourself quite well. Building your configs from scratch is a nifty idea, but a lot of stuff si best left alone. |
| 12:51.46 | *** part/#asterisk xrem1x (n=remi@bg-fw2out.monmouth.com) |
| 12:52.00 | [TK]D-Fender | Datax: Channel drivers, extensions, and one or two others you should do 100% by hand, but many should be left more or less stock for your own good. |
| 12:52.09 | Datax | [TK]D-Fender : building things from scratch is usally a fast way of learning how things work and the dependencies things have between each other |
| 12:52.44 | Datax | I'm looking at the asterisk.conf file on another server of mine and I understand what you mean ;) |
| 12:52.49 | Datax | all of the paths are there :) |
| 12:52.50 | [TK]D-Fender | Datax: Debateable. You will certainly learn everything you need as everything fails in sequence, but its a really bumpy ride you're asking for. |
| 12:53.06 | Datax | [TK]D-Fender : yes I agree :) |
| 12:54.38 | Datax | I now have an asterisk.conf file but still no console command |
| 12:54.46 | Datax | and I have performed a restart |
| 12:55.14 | viperdude | [TK]D-Fender: thanks got it working, you are a star! |
| 12:55.15 | *** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 12:55.33 | *** join/#asterisk ming_zym (n=ming_zym@222.241.208.70) |
| 12:59.39 | [TK]D-Fender | viperdude: You're welcome. |
| 12:59.50 | [TK]D-Fender | Datax: check which modules loaded, etc. |
| 13:02.51 | kannan | is it possible to record video also in *? |
| 13:03.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:05.40 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 13:06.05 | WildPikachu | hi guys .... to use pickupexten ... which is default *8, must I define this in my internal context or where does one use *8? |
| 13:12.03 | Datax | [TK]D-Fender : which module provides the console command ? |
| 13:12.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.87.172) |
| 13:12.22 | [TK]D-Fender | Datax: I believe app_dial.so |
| 13:12.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.35 | [TK]D-Fender | for Dial anyways |
| 13:12.44 | Datax | ok thxs |
| 13:12.54 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:12.59 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:13.00 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:13.34 | Datax | [Jun 9 15:12:56] WARNING[27331]: loader.c:647 load_resource: Module 'app_dial' already exists. |
| 13:14.40 | [TK]D-Fender | Datax: Hmm... |
| 13:15.40 | Corydon76-dig | Datax: you're running with embedded modules |
| 13:16.12 | Corydon76-dig | which means you turned off LOADABLE_MODULES in menuselect |
| 13:17.18 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:17.18 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:17.18 | Corydon76-dig | and the console command is provided by chan_alsa or chan_oss |
| 13:18.41 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 13:19.02 | hsv-al | hello fellow irc addicts |
| 13:19.10 | hsv-al | are we looking forward to another long & glorious week of irc ? :) |
| 13:20.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:20.42 | lmadsen | no way |
| 13:23.16 | *** join/#asterisk ManxPower (n=manxpowe@73.sub-70-220-213.myvzw.com) |
| 13:23.16 | hsv-al | I have like 80 hours of personal leave, and 40 of sick |
| 13:23.22 | hsv-al | so im blowing 8 sick today lulz |
| 13:26.22 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 13:30.00 | ManxPower | It's monday, right? Where are all the newbies? |
| 13:31.06 | jblack | sleepin in. |
| 13:31.32 | [TK]D-Fender | ManxPower: They come out on the weekend, don't you forget... |
| 13:32.02 | jblack | This weekend wasn't so bad. |
| 13:32.25 | jblack | messano and I had plenty of time this weekend to play dancing wits. |
| 13:33.03 | lmadsen | I'm just a consultant, so I take days off whenever I want :) |
| 13:33.39 | jblack | hmmm. |
| 13:33.43 | tzanger | yes, tis monday |
| 13:33.48 | anonymouz666 | lmadsen: this is dangerous :) |
| 13:33.56 | lmadsen | why for? |
| 13:34.06 | *** join/#asterisk quazzmarsh (n=quazzmar@62.8.93.2) |
| 13:34.12 | jblack | somewhere, it's not. I wonder if there, whether it's yesterday, or tomorrow. |
| 13:34.38 | hsv-al | jblack |
| 13:34.40 | hsv-al | http://img208.imageshack.us/img208/9974/motivatorjerseyguidossirv8.jpg |
| 13:34.43 | Rem| | hey, does anyone know how to disable sending cnam over the facility message when performing a two b-channel transfer? |
| 13:35.08 | jblack | Hey. I'm from Jersey. |
| 13:35.33 | jblack | Gah! I mean, I'm _not_ from Jersy |
| 13:36.04 | *** join/#asterisk CVirus (n=GoD@82.201.174.159) |
| 13:36.07 | jblack | I'm still pissed at them too. I got trapped in their bridge-tollbooth trap, the last time I was there. |
| 13:37.46 | jblack | Wait. Is that US New Jersey, or British Jersey? |
| 13:38.08 | [TK]D-Fender | jblack: ..... |
| 13:38.16 | [TK]D-Fender | jblack: New Jersey, duh. |
| 13:38.35 | jblack | turns pink |
| 13:39.07 | jblack | Almost as pink as that Serial killer one in the back |
| 13:39.16 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 13:39.36 | [TK]D-Fender | jblack: Easily distinguishable by their popped collars, fake craptastic tans, spiky hair, and "complete-package" douche-baggery. |
| 13:40.33 | lmadsen | [TK]D-Fender: you just described file |
| 13:40.48 | lmadsen | pops his colla |
| 13:40.54 | file | lmadsen: more like you |
| 13:41.04 | lmadsen | more like Kristian Kielhofner :D |
| 13:41.05 | jblack | And San Diego, and Boston, and most everyone under the age of 25 in the bottom half of florida. |
| 13:41.14 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 13:41.16 | lmadsen | LOL |
| 13:41.24 | lmadsen | oh man... you have no idea, hahahaha |
| 13:41.36 | jblack | If Jersey is hell on earth, then hell hath taken over |
| 13:42.14 | jblack | That pink guy in the back. Doesn't have that serial killer look something awful? |
| 13:42.41 | jblack | It's just that "I have a better idea of where your 14 year old's body is than I do" look that creeps me out |
| 13:43.46 | lmadsen | did anyone else find that statement confusing? :) |
| 13:44.03 | jblack | I did. |
| 13:44.06 | lmadsen | lol |
| 13:44.38 | hsv-al | what the |
| 13:44.40 | hsv-al | http://youtube.com/watch?v=UAbAIpZG7II |
| 13:45.35 | russellb | hey, this asterisk thing is cool |
| 13:45.38 | russellb | how about we talk about that |
| 13:45.55 | lmadsen | russellb: it's only partially cool |
| 13:46.00 | lmadsen | and it's not even 9am yet! |
| 13:46.16 | lmadsen | we should set some hours |
| 13:47.00 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
| 13:47.02 | hsv-al | its supposed to get to 98 here again today |
| 13:47.23 | pigpen | hi all, I am doing a queue with 8 members. About every 5 - 10 days, it stops functioning (ie: calls enter the queue, but members do not get dialed) |
| 13:47.28 | [TK]D-Fender | hsv-al: I beat the start of the heat-wave here on Thursday and installed the A/C |
| 13:47.28 | lmadsen | ok, here is an asterisk related question: for those of you who have read TFoT 1st and 2nd editions -- what would you like to see in a 3rd edition? |
| 13:47.35 | jblack | prepares to get rick rolled |
| 13:47.50 | jblack | wtf? |
| 13:48.03 | pigpen | Any ideas why, and what alternatives do I have (ie: ring group with 8 members) |
| 13:48.05 | jblack | How on earth does this have 224,000 views? |
| 13:48.07 | anonymouz666 | lmadsen: dundi, AEL, how it works the audio core system, etc. :) |
| 13:48.16 | anonymouz666 | j/k |
| 13:48.27 | lmadsen | anonymouz666: really? AEL would be a good one to add actually |
| 13:48.41 | lmadsen | and I already have the DUNDi section on my list for being re-written |
| 13:48.53 | jblack | lmadsen: Oh geeze, you're asking that at 10 am? |
| 13:49.08 | lmadsen | jblack: I'm trying to get us back on topic :) |
| 13:49.11 | lmadsen | and yes! |
| 13:49.15 | lmadsen | :) |
| 13:49.22 | [TK]D-Fender | lmadsen: Complete rewrite of context breakdown, extension sorting (specificity) ; Better compact complete samples for every tech (SIP for phones / ITSP's, analog Zap, PRI, BRI, Video support |
| 13:49.26 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 13:49.33 | [TK]D-Fender | lmadsen: dundi... FEH! |
| 13:49.53 | jblack | Dundi is such a... heh |
| 13:50.00 | lmadsen | I love DUNDi |
| 13:50.04 | lmadsen | I use it all the time in clustered systems |
| 13:50.28 | *** join/#asterisk greek_user (i=ath@adsl39-104.kln.forthnet.gr) |
| 13:50.28 | jblack | Granted, great for internal use for large organizations, etc. |
| 13:50.33 | lmadsen | I use it for passing and retrieving information from multiple servers -- best example is, "how many calls are you servicing right now? and how many can you service total?" |
| 13:50.45 | lmadsen | jblack: it's useful on as many as just 2 boxes |
| 13:50.49 | [TK]D-Fender | lmadsen: "The Book" is for newbs, not for people looking to build ITSP's |
| 13:50.53 | jblack | I used to use it! |
| 13:50.58 | jblack | Just don't have a use right now |
| 13:51.03 | [TK]D-Fender | lmadsen: You are missing your target audience. |
| 13:51.05 | lmadsen | [TK]D-Fender: i know -- and I'm not using it to build ITSPs |
| 13:51.15 | greek_user | can i ask a question here? |
| 13:51.19 | lmadsen | [TK]D-Fender: I don't think you understand how DUNDi can work besides as a routing protocol |
| 13:51.36 | lmadsen | it is an information gathering protocol -- it just doesn't get exposed as that very often |
| 13:51.37 | jblack | lmadsen: Did you keep the notes about odbc that I gave you a while back? |
| 13:51.39 | jaytee | lmadsen: is the 3rd edition going to cover version 1.6? |
| 13:51.43 | [TK]D-Fender | lmadsen: Seriously, newbs need to grasp enough to be useful by themselves, forget about trying to drag others into their incompetency. |
| 13:51.49 | lmadsen | jblack: I did not -- but I have a wiki now -- can you give them to me again? |
| 13:51.56 | lmadsen | jaytee: yes |
| 13:52.04 | jblack | Yeah, I can do that later today if I remember. |
| 13:52.14 | lmadsen | jaytee: actually -- it *may* cover 1.2, 1.4, 1.6 if I can figure out a good way of doing it |
| 13:52.33 | hsv-al | when is 3rd coming out? i just got the 1.4 like only 2 weeks ago |
| 13:52.33 | russellb | that's insanity |
| 13:52.37 | hsv-al | sigh |
| 13:52.39 | jaytee | lmadsen: great, then I'd suggest covering using SIP TCP |
| 13:52.47 | anonymouz666 | lmadsen: Ah I forgot to say. Talk about the libss7 in the 3rd edition. |
| 13:52.48 | lmadsen | hsv-al: we're just creating the outline -- don't worry... you have at least a year |
| 13:53.08 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 13:53.10 | [TK]D-Fender | lmadsen: Book should be able to get you functional with the reals stuff that matters. |
| 13:53.34 | [TK]D-Fender | lmadsen: And I'd add my vote for SS7. These are things real people care about. |
| 13:53.41 | jblack | I would have liked more about PRI protocols. I had a lot of confusion for a week before I realized that they were seemingly arbitrary. |
| 13:54.08 | pigpen | [TK]D-Fender, sorry to ask direct, but does the queue app bomb from time to time? (ie: does not ring the members)? |
| 13:54.10 | jaytee | maybe a half a chapter on SIP debugging? |
| 13:54.12 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 13:54.15 | [TK]D-Fender | pigpen: nope. |
| 13:54.36 | anonymouz666 | jaytee: you should read RFC3261 for that. |
| 13:54.44 | lmadsen | [TK]D-Fender: ok... so you think SS7 is a good technology to talk about, but not DUNDi |
| 13:54.44 | jblack | jaytee: If the book had that, then [tk] wouldn't have as many opportunities to type ~sipnat. :) |
| 13:54.45 | pigpen | [TK]D-Fender, man, I just got my ass reamed about a system that is bombing now for some time, across several versions. |
| 13:55.13 | lmadsen | jblack: PRI is one my things I wish to talk about -- I doubt we'll do BRI because the 3 authors are from North America, and we don't have BRI in N.A. |
| 13:55.43 | jblack | Sorry. I meant PRI |
| 13:55.51 | jblack | hurh? I said PRI |
| 13:55.53 | *** join/#asterisk fiddur (n=fiddur@78.82.254.164) |
| 13:55.54 | lmadsen | I know |
| 13:55.58 | lmadsen | someone else said BRI though |
| 13:56.41 | [TK]D-Fender | I did. |
| 13:56.55 | Datax | Corydon76-dig : does that mean that I need to recompile asterisk ? |
| 13:57.10 | [TK]D-Fender | lmadsen: And doesn't matter if you are from the Americas, You are quite well capable of outsourcing info for this. |
| 13:57.11 | lmadsen | we'll cover more about 'realtime' and database integration as well I hope |
| 13:57.18 | greek_user | [Question] In order to just call a sip destination, i.e. Dial(SIP/myfriend@sipcompany.com) , do I have to create a section in sip.conf? or the Dial() application is sufficient? My problem is that my call really rings the destination, but even when they answer the call, my side never gets informed about it, so eventually Asterisk times oute the connection. Any ideas? (yes, i am behind a NAT, my phone is behing, asterisk is behind, but a have 5 |
| 13:57.25 | lmadsen | [TK]D-Fender: we've tried for 2 versions now -- no one seemed to be interested in writing it |
| 13:57.32 | Datax | Corydon76-dig : how do you know that I'm using embedded modules ? (how amÃI supposed to know ? :p) |
| 13:57.34 | [TK]D-Fender | greek_user: Yes, you can dial direct like that. |
| 13:57.40 | jblack | Oh, AGIs... if the book mentions terminating properly, I never found it. |
| 13:57.49 | [TK]D-Fender | greek_user: And timeouts are likely a NAT/reinvite issue. |
| 13:57.52 | [TK]D-Fender | greek_user: ... |
| 13:57.54 | [TK]D-Fender | ~sipnat |
| 13:57.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:57.57 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:58.05 | lmadsen | jblack: terminating properly? |
| 13:58.08 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:58.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:58.22 | jblack | (i.e. use AGI_STATUS rather than expecting the return code to work) |
| 13:58.55 | lmadsen | ah -- you can't use the return code from a dialplan application -- that is internal |
| 13:59.02 | lmadsen | I will see about a status variable though |
| 13:59.06 | lmadsen | I don't use AGI too much |
| 13:59.23 | greek_user | i'll real aocomputing, and be back in a while, thank you for now. |
| 13:59.25 | lmadsen | loves his new wiki |
| 13:59.36 | jblack | It's a common catch. People expect to do a exit 1, die 1, etc, and end up surprised that the dialplan doesn't catch on that there's an error |
| 13:59.38 | lmadsen | making documenting these ideas so much easier |
| 14:00.25 | lmadsen | ok, 10am -- gotta go do some real work now |
| 14:00.30 | lmadsen | thanks all for the input! |
| 14:00.35 | lmadsen | got some good ideas there |
| 14:00.43 | lmadsen | feel free to ping me with other ideas as you come up with them |
| 14:00.52 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 14:00.55 | jblack | Fact is, the book is great once one gets a good grasp of what's where |
| 14:01.27 | jblack | lmadsen: formatting wise, I have trouble finding the sections for various configuration files. |
| 14:01.55 | [TK]D-Fender | jblack: problem is the book is supposed to TELL you what's where. |
| 14:02.18 | jblack | Yeah, but it doesn't. |
| 14:02.27 | [TK]D-Fender | I would gladly write for The Book, if there was some remuneration in it... |
| 14:02.37 | jblack | The bottom of each page could more clearly define what the page's definitions are a part of |
| 14:03.08 | lmadsen | [TK]D-Fender: Tilghman helped us a lot with the appendices, and he got a bit of a kickback and a nice thank you section in the opening credits :) |
| 14:03.36 | lmadsen | jblack: pages definitions? |
| 14:03.41 | lmadsen | please elaborate |
| 14:03.41 | jblack | Well, at least change "Application Reference" to "Dialplan Applications and FUnctions" |
| 14:03.50 | lmadsen | aha |
| 14:03.57 | lmadsen | I will make a note of it |
| 14:04.33 | jblack | I go looking for sip.conf settings in configuration files, only to find it's not there. |
| 14:04.37 | jblack | Stuff like that. |
| 14:05.03 | lmadsen | jblack: ya -- we need to definitely beef up some of the appendices |
| 14:05.14 | jblack | (sip.conf, iax.conf and such is in appendix a, under "VoIP Channels" |
| 14:05.39 | lmadsen | which should probably be labeled VoIP Channel Configuration Files |
| 14:05.43 | lmadsen | or something like that |
| 14:05.46 | jblack | Well, it's more like.... |
| 14:05.55 | jblack | voicemail.conf and such are way off in appendix d. |
| 14:06.05 | jblack | but sip.conf and iax.conf are way up front in appendix a. |
| 14:06.08 | *** join/#asterisk freddyk (n=freddy@79.43.94.193) |
| 14:06.17 | lmadsen | well -- they are being broken out by type of configuration file |
| 14:06.24 | freddyk | hi all |
| 14:06.38 | freddyk | can ask someone help on pickup implementation ? |
| 14:06.52 | lmadsen | ~ask |
| 14:06.53 | jbot | ask is probably 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. |
| 14:06.57 | freddyk | ok |
| 14:07.07 | freddyk | i'm developing with some other friends chan_sccp |
| 14:07.12 | jblack | A good question to ask is if people new to the book are going to understand the difference between channel config files and every other config file... |
| 14:07.12 | freddyk | fixing some bugs around |
| 14:07.13 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:07.23 | freddyk | but can't get the pickup function work as expected |
| 14:07.28 | jblack | Or if they just think that all the config files are in one config files group |
| 14:07.35 | freddyk | i got pickup |
| 14:07.54 | freddyk | using ast_pickup_call(mynewallocatedchannelwithpvt) |
| 14:08.01 | freddyk | then i use hangup |
| 14:08.14 | freddyk | to shutdown the masqueraded channel |
| 14:08.16 | freddyk | btw |
| 14:08.28 | freddyk | it hangs up my call instead of put on hold |
| 14:08.34 | jblack | But honestly, it's one of the best reference books I've seen. |
| 14:08.43 | lmadsen | jblack: pg xvi has a section on the organizing of the book -- but I agree that a description needs to go into the appendices sections |
| 14:08.58 | lmadsen | noted |
| 14:09.43 | freddyk | can someone help ? |
| 14:13.48 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:20.49 | ManxPower | freddyk: dev questions are usually on asterisk-dev mailing list or #asterisk-dev IRC channel. I suspect, however, that you should not hang up the MASQ channel |
| 14:21.41 | Rem| | so i guess no one here knows how I can disable sending cnam over facility message on a pri? |
| 14:22.57 | ManxPower | Rem|: I would guess that is correct. |
| 14:23.21 | ManxPower | Why not just set the Caller*ID name to something else or blank? |
| 14:23.48 | ManxPower | Of course, any CNAM info is unlikely to ever be passed to a far end PSTN connection |
| 14:24.13 | ManxPower | The receiving telco will look up the name in the telco database for what name is associated with that number. |
| 14:24.15 | Rem| | that won't work for me... I need to disable it |
| 14:24.47 | ManxPower | Rem|: then perhaps a question on asterisk-dev mailing list or the #asterisk-dev IRC channel? I expect you would have to edit the Asterisk source. |
| 14:24.53 | Rem| | yeah that is true... but in order to get another feature to work I need to disable it |
| 14:25.07 | Rem| | for some reason the switch doesn't like seeing the cnam |
| 14:25.16 | Rem| | ok thanks |
| 14:28.37 | ManxPower | Rem|: that would mean that your switch cannot be connected to the PSTN |
| 14:28.43 | ManxPower | And I doubt that is the case |
| 14:30.04 | tzafrir_laptop | Rem|, disable sending or send an empty one? |
| 14:30.13 | *** join/#asterisk quazzmarsh (n=quazzmar@62.8.93.2) |
| 14:30.33 | Rem| | well it doesn't like seeing cnam queued in the facility message when I am trying to perform a two b channel transfer |
| 14:31.30 | Rem| | tzafrir_laptop, disabling sending cnam in the facility message |
| 14:34.28 | kannan | hello, i am reently using centos 5.1 for Asterisk(with a quad port digium E-1 card). I noticed that if i needed to give sleep 15 bfore starting asterisk in rc.local, otherwise there are problems with the zap lines. Any one similar issue? i got asterisk 1.4.20.1 |
| 14:35.38 | [TK]D-Fender | kannan: you should be starting * under the standard services, not rc.local |
| 14:35.46 | [TK]D-Fender | kannan: "make config" for * and zaptel |
| 14:35.55 | [TK]D-Fender | kannan: and managing it the normal RH way |
| 14:36.14 | kannan | [TK]D-Fender , oh for * also, ok thanks |
| 14:36.23 | kannan | i did for zaptel |
| 14:36.39 | [TK]D-Fender | kannan: should for both. |
| 14:36.54 | kannan | [TK]D-Fender , thanks, will do |
| 14:39.19 | *** part/#asterisk fiddur (n=fiddur@78.82.254.164) |
| 14:41.28 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
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| 14:45.03 | Corydon76-dig | Datax: how are you supposed to know what? |
| 14:48.03 | *** join/#asterisk B1ST (n=overdose@d51A411BE.access.telenet.be) |
| 14:48.49 | Corydon76-dig | Datax: not to change defaults unless you understand what each option actually does? |
| 14:51.46 | ManxPower | I must have Datax on /ignore or something |
| 14:52.54 | [TK]D-Fender | ManxPower: Last comment was a while ago |
| 14:56.51 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 14:56.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:04.19 | ManxPower | [TK]D-Fender: 2 hrs, at leasr. |
| 15:04.29 | [TK]D-Fender | ManxPower: 1 |
| 15:05.18 | ManxPower | He pretty obviously doesn't want our help. |
| 15:08.17 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
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| 15:15.55 | [TK]D-Fender | ManxPower: His answer only came 1 hour later... |
| 15:16.08 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
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| 15:30.04 | *** join/#asterisk metfan2007 (n=jc@201.161.16.35) |
| 15:31.00 | metfan2007 | hi all!!! is there any tool can I use to test and analyse DTMF tones quality from my telco?? |
| 15:31.39 | metfan2007 | I have DTMF detection problems, and I need to know if it is in my telco side or Asterisk side, thanks!!! |
| 15:32.18 | *** join/#asterisk RoyK (n=roy@ip-150-21-149-91.dialup.ice.no) |
| 15:32.27 | jblack | I'm in a pickle. How do I tell asterisk which interfaces to listen on for iax? |
| 15:32.48 | Qwell | you can't specify an interface |
| 15:33.00 | Qwell | it's either 0.0.0.0 or the (one) IP address |
| 15:33.08 | Qwell | if you need interface level, use iptables or similar |
| 15:33.11 | jblack | 0.0.0.0 sounds dandy. |
| 15:33.30 | jblack | grabs the book |
| 15:33.49 | jblack | my frigging unreachable problem is back |
| 15:34.24 | jblack | however, it's not *'s problem itself |
| 15:35.56 | metfan2007 | any DTMF help? :S |
| 15:39.30 | jblack | metfan2007: What's your problem? |
| 15:43.08 | metfan2007 | jblack: The DTMF detection in my Asterisk IVR is not working at all, in some cases when a costumer dials the IVR number, he cannot browse inside IVR menus |
| 15:43.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:43.58 | metfan2007 | I have made some tests, in the IVR when I press some number, the CLI does not shows the DTMF debug, but in other cases it shows Ok |
| 15:44.09 | jblack | Ok. Turn on rfc2833. |
| 15:44.22 | metfan2007 | in the same trunk, same line, same number... I think is DTMF quality problem |
| 15:44.36 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 15:44.39 | jblack | are you using sip or iax? |
| 15:44.39 | metfan2007 | the issue here is that I'm using E1 line (MFCR2) |
| 15:44.48 | jblack | Oh. No idea then. Sorry. |
| 15:45.11 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:45.14 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:45.29 | metfan2007 | I want to guess if it is telco or asterisk side problem, I want to know if there is a tool or a way to detect DTMF quality |
| 15:45.31 | JT | metfan2007: wintone can decode tones as can some other software |
| 15:45.34 | [TK]D-Fender | metfan2007: try "relaxdtmf=yes" and see if that helps. |
| 15:46.34 | *** join/#asterisk ix33 (n=ix@2002:cede:da6:0:0:0:0:1) |
| 15:46.56 | ix33 | is there a mailing list for digium branded line cards? |
| 15:49.30 | metfan2007 | [TK]D-Fender: I already tried that, and the result is that Asterisk starts with DID recognition problems, I mean, If I dial 5590 Asterisk reads 5559, it repeats "5", remember that it is MFCR2, and DNI and ANI numbers pass via tones too... :S |
| 15:50.09 | [TK]D-Fender | metfan2007: Well, those are the 2 options you've got. You may be in TFB territory. |
| 15:50.19 | jeev | fender |
| 15:50.25 | jeev | BBB complaint on linksys. |
| 15:50.30 | jeev | they call 2 days later |
| 15:50.40 | jeev | cross ship WIP330, waiting for response.. they wont refund my ass |
| 15:51.00 | metfan2007 | [TK]D-Fender: TFB territory? |
| 15:54.11 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:54.12 | [TK]D-Fender | ~tfb |
| 15:54.13 | jbot | methinks tfb is Too #&^$ing bad.... |
| 15:54.33 | [TK]D-Fender | rather unfortunate. |
| 16:00.00 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:00.18 | jblack | Ok. I have a crashed soundpoint 550. The report I'm getting is that the buttons are frozen, including the menu button. Any suggestions? |
| 16:00.27 | nny_1 | so I am trying to simplify this system before i bestow it to the masses. I need to set it up so that on transfer (using *EXT) it 1.) checks to see if the intended recipient is using a ("line"/SIP channel) if they are (which is most likely),2.) it does a chanspy + whisper and allows the attendant to whisper the caller info to the recipient. After that, the call goes (?) to the phone on hold or some equivalent. |
| 16:00.34 | nny_1 | I have chanpsy + whisper working |
| 16:00.52 | nny_1 | can't figure out how to determine if the recipient is using their phone, i have hints set up |
| 16:02.24 | JT | not using the transfer button on the phones? |
| 16:02.38 | nny_1 | eh yeah i want to preserve caller id too |
| 16:02.40 | jblack | Even the menu button is being ignored. |
| 16:02.54 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 16:03.10 | nny_1 | i can blind transfer the calls, but at that point they have to menu over etc |
| 16:03.22 | [TK]D-Fender | nny_1: "core show function chanisavail" |
| 16:03.31 | [TK]D-Fender | nny_1: "core show application chanisavail" _ rather |
| 16:03.38 | lmadsen | I was gonna say :) |
| 16:03.47 | [TK]D-Fender | jblack: pull the plug. |
| 16:03.48 | JT | jblack: power cycle? |
| 16:03.58 | nny_1 | [TK]D-Fender: kk i'll check that out |
| 16:04.32 | JT | nny_1: you can't preserve callerid with transfer? |
| 16:04.57 | nny_1 | JT afaik only with blind transfer |
| 16:05.10 | nny_1 | attended transfer shows extension number doing the transfer |
| 16:05.11 | jblack | Power cycle didn't work, but I found something that did. |
| 16:05.17 | jblack | I called it, and it's unstuck. |
| 16:05.29 | jblack | Odd that it would be completely stuff over power cycling, but a call in fixed it |
| 16:06.00 | jeev | fender, i'll just bbb'ing their ass till they give me a lifetime supply of free linksys shit |
| 16:06.15 | jeev | then i'll say, i'm not happy with your linksys shit, give me everything cisco, i need a few GSR's. |
| 16:06.31 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 16:06.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:06.45 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 16:07.11 | CrashSys | Anyone ever had any issues with the older-generation polycom's (IP301/501) rebooting/hanging with Sip 2.1.2? |
| 16:08.44 | *** join/#asterisk greek_user (i=ath@adsl39-104.kln.forthnet.gr) |
| 16:09.20 | nny_1 | [TK]D-Fender: hmm i think there would be a channel available on the tech/resource still (say SIP/11) since they are 4 line phones. Trying to find out if *any* of the lines are being used. |
| 16:09.57 | [TK]D-Fender | jeev: Look at everything you're going through over 1 stupid phone, and one you were specifically warned about. SMRT |
| 16:10.16 | [TK]D-Fender | nny_1: Good... not this time READ the isntructions :p |
| 16:10.18 | [TK]D-Fender | now* |
| 16:10.31 | [TK]D-Fender | CrashSys: nope. |
| 16:10.56 | nny_1 | :D |
| 16:10.58 | CrashSys | seems to be localized to one side of the room... thinking a bad PoE switch |
| 16:11.07 | nny_1 | jeev: which phone? |
| 16:11.34 | CrashSys | I'll have them put a red sticker on the problem phones, then go over there and start pulling plugs on switches, see if they are all one 1 particular switch :) |
| 16:11.40 | jeev | yea i know fender |
| 16:11.42 | jeev | WIP330 linksys |
| 16:12.12 | CrashSys | the scarlett handset |
| 16:12.16 | CrashSys | sounds like a good book |
| 16:16.28 | greek_user | I had no luck: while the phone rings when I try to Dial(SIP/myfriend@remotesipserver.com) , when he answers it, my phone never gets infored, and still wait for a connection. However, when he hangsup, I immediately get SIP/remotesipserver.com is circuit-busy |
| 16:16.40 | nny_1 | eww |
| 16:16.43 | nny_1 | windows ce? |
| 16:16.45 | nny_1 | fuck that |
| 16:16.47 | nny_1 | :D |
| 16:17.30 | nny_1 | [TK]D-Fender: -s |
| 16:17.31 | nny_1 | :P |
| 16:17.52 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 16:18.44 | nny_1 | so when i have them do a chanspy + whisper, shoul di just give it a time frame before transfering? |
| 16:18.54 | nny_1 | i don't think I can end a chanpsy gracefully elsewise |
| 16:19.02 | nny_1 | apart from "hanging up" |
| 16:19.07 | [TK]D-Fender | greek_user: Disabe reinvites... |
| 16:19.32 | ManxPower | greek_user: as usual, the problem is NAT |
| 16:23.34 | `Sauron | looks away. |
| 16:23.38 | `Sauron | Manx said the evil word. |
| 16:23.40 | `Sauron | :) |
| 16:23.59 | nny_1 | `Sauron: how can you look away? You're one giant freaking eye... |
| 16:24.21 | `Sauron | On a completely unrelated note, is there anyone in here with good perl-fu that I could ask a non-* question? |
| 16:24.34 | [TK]D-Fender | With a single eye, should "looks" be pluralized? |
| 16:24.43 | nny_1 | hmm |
| 16:25.00 | `Sauron | "looks" is a verb, not a noun |
| 16:25.04 | nny_1 | i think it's a verb at that point |
| 16:25.06 | nny_1 | yah |
| 16:25.12 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 16:25.17 | `Sauron | So there's no pluralization going on. |
| 16:25.41 | [TK]D-Fender | mmmmm pluralizing... |
| 16:26.30 | ManxPower | `Sauron: the problem is not NAT, the problem is people not understanding SIP and not understanding NAT |
| 16:27.15 | *** join/#asterisk jamey-uk (n=jamey-uk@cpc3-cmbg1-0-0-cust158.cmbg.cable.ntl.com) |
| 16:27.50 | jamey-uk | Asterisk looks great but who do you pay money to for calls and so on? Sorry it's a very newbie question |
| 16:27.56 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 16:28.19 | *** join/#asterisk e` (n=IceChat7@38.102.196.202) |
| 16:28.31 | ManxPower | jamey-uk: a service provider, either a telephone company or an internet phone company |
| 16:28.43 | ManxPower | jamey-uk: you should read the Good Book |
| 16:28.44 | ManxPower | ~book |
| 16:28.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:29.01 | nny_1 | so in theory i have an idea of how this would work up to the point of the transfer, so i have asterisk checking to see if the channel is being used at all, if so, it does chanspy + whisper for 5 seconds. I am not sure if this is a good idea, because if the attendant has 3 calls going, how would the system know which one is intended for which recipient? I can use the transfer button, but the transfer would try to transfer to chanspy, which would be ugly |
| 16:29.40 | nny_1 | I could transfer the call to a parking lot and have the system whisper the number to the recipient |
| 16:29.49 | nny_1 | gah nm |
| 16:30.01 | nny_1 | still requires the system knowing which line is intending to be transfered |
| 16:30.16 | e` | Asterisk started to drop all calls when it was being flooded with "Received trunked frame before first full voice frame' warnings. restarting asterisk fixed the issue, but can someone explain the warning message to me? |
| 16:30.32 | jamey-uk | Can anyone recommend a service provider in the UK, particularly for mobile calls? Trying to figure out how much it will cost in comparison |
| 16:30.38 | `Sauron | Manx: Oh, I am quite aware of what the problem(s) are. |
| 16:30.57 | [TK]D-Fender | jamey-uk: www.gradwell.com |
| 16:32.25 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 16:33.40 | greek_user | how can I "show" my localnet setting from whithin the CLI ? |
| 16:34.54 | [TK]D-Fender | greek_user: AFAIK, you can't |
| 16:34.55 | nny_1 | Does anyone know of a way to retain caller ID on an attended transfer? |
| 16:34.59 | [TK]D-Fender | greek_user: go look at your configs. |
| 16:35.16 | greek_user | ok |
| 16:35.23 | [TK]D-Fender | nny_1: You don't, thats the point of an attended transfer. |
| 16:35.25 | spokra | ! vi sip.conf |
| 16:35.38 | greek_user | Is OK to specify 192.18.0.0/255.255.255.0 ??? or do i have to 192.168.0.0/24 ? |
| 16:35.38 | [TK]D-Fender | nny_1: If you're desperate, you've got the source... |
| 16:35.52 | nny_1 | rgr |
| 16:35.56 | [TK]D-Fender | greek_user: probably either |
| 16:36.00 | greek_user | yup.. |
| 16:39.00 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 16:39.14 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:39.22 | greek_user | My sip.conf is: sip.conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes externhost=my.dyndns.org externrefresh=900 localnet=192.168.0.0/255.255.255.0 nat=yes canreinvite=no |
| 16:39.44 | *** join/#asterisk uTx (n=unix@modemcable232.79-58-74.mc.videotron.ca) |
| 16:41.56 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:42.48 | DavidR2008 | I know that zaptel (at least) is tied to the kernel in some way. Is it a bad idea to install kernel updates? I'm running CentOS 5.1 and there is a popup on the desktop every so often about updates. Some times they are kernel updates. |
| 16:43.10 | jblack | I have a first serious argument against IAX |
| 16:43.13 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 16:44.38 | [TK]D-Fender | DavidR2008: If you upgrade your kernel you do have to recompile Zaptel to match |
| 16:45.02 | DavidR2008 | thx! |
| 16:45.03 | [TK]D-Fender | DavidR2008: So I might set an exclusion to kernel for yum, and do those when you have more time to reboot and take changes, etc. |
| 16:49.02 | jblack | Hans reiser is gonna show where he buried his wife |
| 16:50.01 | tzanger | jblack: ? |
| 16:50.51 | jblack | http://blog.wired.com/27bstroke6/2008/06/hans-reiser-off.html |
| 16:52.30 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 16:52.42 | freezey | quickly whats a good interview question i have a few looking for another one hard to thnk them up |
| 16:53.28 | greek_user | do i have to set matchexterniplocally when behind a nat? |
| 16:54.04 | freezey | is that for me or is that an actual question |
| 16:54.04 | freezey | lol |
| 16:54.07 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 16:54.26 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 16:55.34 | greek_user | sorry that was a general question |
| 16:56.03 | freezey | ha |
| 16:56.06 | greek_user | :) |
| 16:58.09 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:58.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:00.52 | *** join/#asterisk fainsys (n=fainsys@c-76-17-121-45.hsd1.ga.comcast.net) |
| 17:03.45 | greek_user | do you know where * stores its debug logs? |
| 17:04.05 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:06.03 | *** join/#asterisk ming_zym (n=ming_zym@222.241.208.70) |
| 17:08.47 | imcdona | anyone have any suggestions for a channel bank that could provide FXO ports to a legacy PBX via asterisk? I am putting Asterisk in front of a PBX and need to provide FXO ports |
| 17:08.57 | *** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net) |
| 17:09.00 | harryv | what's the best way to monitor my zapata spans? if one is not ok, but red/alarm i want to be notifiied by mail.. |
| 17:10.15 | *** join/#asterisk budmang (n=budman@adsl-75-22-52-27.dsl.irvnca.sbcglobal.net) |
| 17:10.16 | budmang | Anyone know a good provider alternative from teliax?(pay per minute unlimited channels) |
| 17:12.04 | B1ST | hmm |
| 17:12.04 | B1ST | grandstream1 81.164.17.189 5060 UNREACHABLE |
| 17:12.04 | imcdona | budmang: http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 17:12.05 | keith4 | why is the polycom SIP301 more expensive than the 320? |
| 17:12.16 | B1ST | anyone knows why it's unreachable? maybe wrong setting in sip.conf? |
| 17:12.29 | imcdona | B1ST: is it registerd? |
| 17:12.48 | B1ST | i did whit the grandstream conf page yeah |
| 17:13.16 | keith4 | ~itsp |
| 17:13.16 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:13.16 | imcdona | did you do a sip debug? |
| 17:13.22 | keith4 | ~itsplist-us |
| 17:13.23 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:13.26 | keith4 | budmang: ^^^ |
| 17:13.27 | B1ST | not yet |
| 17:13.44 | imcdona | do a "asterisk -r" then type "sip debug" |
| 17:13.49 | imcdona | see if you are getting an error |
| 17:14.02 | imcdona | is this phone behind nat by any chance? |
| 17:14.02 | B1ST | i have this also |
| 17:14.20 | B1ST | i don't think so |
| 17:14.50 | greek_user | can i have all the debug dialogs saved in a file? |
| 17:15.16 | imcdona | B1st: is the phone on the same subnet as your asterisk box? |
| 17:15.22 | B1ST | yes |
| 17:15.34 | B1ST | yesterday it worked great |
| 17:15.56 | imcdona | ok..so its not a nat issue. Next step is do a sip debug and see if you see packets from your phone. if nothing is coming through, check the settings on the grandstream |
| 17:16.18 | imcdona | B1st: it IS A GRANDSTREAM so you may have to powercycle it ;) |
| 17:17.33 | B1ST | booting now |
| 17:17.33 | B1ST | he's downloading stuff |
| 17:18.25 | imcdona | greek_user: its possible...mine logs to /var/lib/asterisk/full I am running freepbx though not sure how its done |
| 17:18.52 | B1ST | [Jun 9 19:18:43] WARNING[5888]: chan_sip.c:1785 __sip_xmit: sip_xmit of 0x819d1c0 (len 494) to 81.164.17.189:5060 returned -2: Bad file descriptor |
| 17:18.56 | B1ST | Really destroying SIP dialog '58cd255a62eecefa3e013e2e0fd2b563@81.164.17.190' Method: OPTIONS |
| 17:18.59 | B1ST | this is what i get now |
| 17:19.48 | *** join/#asterisk enemy^x (n=enemy@c213-158-248-202.static.sdsl.no) |
| 17:20.16 | imcdona | B1ST: WHOA! -2: Bad file descriptor Thats a new one....have you restarted Asterisk yet? I am almost thinking a filesystem issue |
| 17:20.29 | enemy^x | ${CALLERID} should contain the remote callers number in 1.4.17 also right? |
| 17:20.34 | B1ST | filesystem issue? |
| 17:20.42 | B1ST | i did, imcdona |
| 17:20.44 | B1ST | 3 times |
| 17:21.34 | imcdona | bad file descriptor is what is strange....go to pastebin.com and copy the full SIP dialog so I can have a look |
| 17:22.46 | B1ST | ok imcdona |
| 17:22.51 | styelz | i had an issue like that a while ago. something to do with vnodes ? |
| 17:23.00 | styelz | and too many files |
| 17:23.21 | styelz | on the fs |
| 17:23.28 | imcdona | inodes? |
| 17:23.32 | styelz | cant remember |
| 17:23.37 | styelz | something like that |
| 17:24.09 | B1ST | that's the only msg i got now, imcdona |
| 17:24.17 | B1ST | but earlier today i had more |
| 17:24.33 | B1ST | but i have a very clean sip.conf file now |
| 17:24.51 | B1ST | but i didn't messed today whit the sip.conf file and i got this issue |
| 17:24.55 | B1ST | very strange |
| 17:24.56 | styelz | maybe it was a permissions thing |
| 17:25.03 | B1ST | nah |
| 17:25.14 | B1ST | don't think so |
| 17:25.39 | [TK]D-Fender | enemy^x: No, it shouldn't. That variable was deprecated in 1.2 and removes in 1.4 |
| 17:25.53 | [TK]D-Fender | enemy^x: "core show application CALLERID" |
| 17:28.28 | lmadsen | [TK]D-Fender: "core show function CALLERID" |
| 17:28.32 | lmadsen | that's twice today, lol |
| 17:28.40 | [TK]D-Fender | lmadsen: yeah, I noticed... |
| 17:28.42 | [TK]D-Fender | aashdklshdsjadgfdasdtfg |
| 17:28.43 | lmadsen | hahaha |
| 17:28.46 | lmadsen | it's a Monday |
| 17:28.49 | lmadsen | all is forgiven |
| 17:29.43 | keith4 | is there a ~callerid factoid yet? |
| 17:31.39 | Qwell | keith4: why don't you ask the bot? |
| 17:32.05 | jaytee | ~callerid |
| 17:32.20 | keith4 | aww |
| 17:32.21 | Qwell | and there you have it |
| 17:32.30 | jaytee | bot's 'tarded |
| 17:34.20 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 17:34.45 | seanbright | ~cid |
| 17:34.46 | jbot | somebody said cid was CallerID, or a TCP client/server Caller-ID system, including server and Tk GUI client.. URL: http://www.tummy.com/cid |
| 17:34.58 | seanbright | that's just crazy talk |
| 17:35.13 | *** join/#asterisk dw (i=dmwdmw@unaffiliated/dw) |
| 17:35.35 | dw | hi there. can anyone recommend a method to hook skype up to asterisk? |
| 17:36.00 | Qwell | dw: You cannot. |
| 17:36.28 | dw | Qwell: i can think of at least one method involving some duct tape and a robot, so that statement is incorrect :P |
| 17:36.39 | Qwell | touche |
| 17:36.54 | Qwell | allow me to rephrase. |
| 17:37.09 | Qwell | Without duct tape, you cannot. |
| 17:37.16 | *** join/#asterisk BitBandit (n=PX2@mail.dutro.com) |
| 17:37.17 | Qwell | All uses would involve duct tape. |
| 17:38.02 | dw | heh. :) there is at least http://www.mhspot.com/mhspot/sippyskype.htm , and skype also sell a library. so your statement could still use some refinement :) |
| 17:38.23 | dw | im guessing your statement means something like "nobody here ever got it working well enough to be useful |
| 17:38.42 | Qwell | "SippySkype Sip to Skype Gateway System Requirements: |
| 17:38.42 | Qwell | <PROTECTED> |
| 17:38.48 | Qwell | ie; duct tape |
| 17:38.58 | *** join/#asterisk aksyn (n=aksyn@78.86.127.229) |
| 17:39.00 | dw | lol |
| 17:39.10 | jaytee | I'm thinking someone should get out their C programming guide and go write their own Skype module for Asterisk if they think's it's such a necessary deal. |
| 17:39.19 | Qwell | Your duct tape + robot idea is a better one. |
| 17:39.25 | Qwell | jaytee: Not possible. |
| 17:39.45 | Qwell | not without using the actual skype client, and an X session per instance |
| 17:39.56 | jaytee | sure it is, just place a function call to "ducttape(void) in there somewhere |
| 17:40.06 | keith4 | there are commercial libraries available... |
| 17:40.17 | Qwell | keith4: see above |
| 17:40.25 | keith4 | http://www.chanskype.com/ |
| 17:40.44 | Qwell | that uses the actual skype client and VNC or something stupid |
| 17:40.48 | dw | hrm, i wonder how they can sell that |
| 17:40.54 | Qwell | dw: no comment |
| 17:41.01 | dw | the skype stuff is commercial, and would need linked into the gpl asterisk |
| 17:41.20 | jaytee | that's liking mixing matter with antimatter |
| 17:41.33 | Qwell | I believe that particular one uses a binary kernel module to...stuff. |
| 17:42.20 | keith4 | ugh.. so, it literally runs skype, and just hooks into the audio stream? |
| 17:42.26 | keith4 | what an ugly, ugly hack |
| 17:42.26 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-42.hsd1.ky.comcast.net) |
| 17:42.38 | Qwell | keith4: not only that, but it runs an instance of skype PER call |
| 17:42.47 | keith4 | wtf |
| 17:42.48 | Qwell | which means it also runs an X session per instance |
| 17:42.58 | jaytee | that ain't ugly, that's fugly! |
| 17:43.05 | Qwell | duct tape + robot would be a better method |
| 17:43.07 | keith4 | sounds like a great way to bring a server crashing to its knees |
| 17:43.27 | jeffspeff | can i use my same config files with 1.2 that i used with 1.1.x |
| 17:43.37 | Qwell | jeffspeff: There was no 1.1.x |
| 17:43.41 | Qwell | so...probably not, no |
| 17:44.03 | jeffspeff | 1.4.19. sorry. :p |
| 17:44.10 | hsv-al | qwell whats up w/ the phone sytem, it does it from my cell |
| 17:44.12 | jeffspeff | here, hold on, i got my software versions mixed up... |
| 17:44.14 | hsv-al | regular phone, and work phone |
| 17:44.24 | Qwell | hsv-al: got me.. |
| 17:44.26 | jaytee | you want to use files from 1.4 on 1.2? why are you going backwards? |
| 17:44.30 | hsv-al | cant burst through menus |
| 17:44.35 | hsv-al | throttled ;/ |
| 17:44.45 | Qwell | we don't get to maintain the Digium PBX anymore :p |
| 17:44.51 | Qwell | s/we/the developers/ |
| 17:44.52 | hsv-al | eh? |
| 17:45.10 | keith4 | ew |
| 17:45.16 | jeffspeff | lets try this again. can i use my same config files with 1.4.20.1 that i used with 1.4.19? |
| 17:45.19 | keith4 | there are other ways to connect to skype, but they're expensive and cumbersome |
| 17:46.02 | jaytee | jeffspeff, yeah, you should be able to to. I don't think much has been "deprectated" between .19 and .20 :-) |
| 17:46.06 | keith4 | or are small-scale http://www.rsdevs.com/psgw.shtml |
| 17:46.17 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 17:46.40 | keith4 | ooh, windows-only! http://www.nch.com.au/skypetosip/ |
| 17:47.42 | keith4 | or maybe even this: http://www.skip2pbx.com/ |
| 17:47.54 | Qwell | keith4: care to guess how it works? |
| 17:48.06 | Qwell | "PSGw supports only a single concurrent connection between Skype and SIP/H.323 network." |
| 17:48.11 | Qwell | wonder why that might be... |
| 17:48.19 | keith4 | doesn't get much smaller-scale than that |
| 17:48.35 | hsv-al | [12:48pm] -ix33- You do not have access... -ScrollZ- |
| 17:48.37 | hsv-al | pike off ix33 :) |
| 17:52.49 | keith4 | I wonder if people would pay for an actual skype-integration module, like officially released by the skype people |
| 17:53.23 | hsv-al | each of the 3 packages i bought, fxo/fxs, and tdm card all say register them |
| 17:53.30 | hsv-al | does the 411P only need the registering? |
| 17:55.31 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 17:56.14 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:56.39 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 18:00.48 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 18:01.15 | e` | Asterisk started to drop all calls when it was being flooded with "Received trunked frame before first full voice frame' warnings. restarting asterisk fixed the issue, but can someone explain the warning message to me? |
| 18:02.18 | *** join/#asterisk s0lid (n=s0lid@61.28.163.132) |
| 18:02.35 | *** part/#asterisk jbeez (i=jbeez@jbeez.net) |
| 18:03.32 | hsv-al | what the hell |
| 18:03.41 | hsv-al | when i just called my * it said weasels have eaten the phone system lol |
| 18:06.02 | talntid | lol |
| 18:07.44 | hsv-al | forgot about that confi entry i put from the book |
| 18:07.46 | hsv-al | tt-weaseals |
| 18:13.54 | styelz | i like to use the "what are you wearing one".. sounds sexy |
| 18:17.52 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-81-170.vif.net) |
| 18:18.55 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:21.38 | hsv-al | heh |
| 18:22.33 | *** join/#asterisk DanyWalker (n=ww@201.230.26.10) |
| 18:22.35 | *** join/#asterisk arctic_import (n=jasonj@mail.uui-alaska.com) |
| 18:22.43 | DanyWalker | hi people, i have a problem |
| 18:23.19 | DanyWalker | i need see more extensions of asterisk (actually i see only 40 in the control panel) |
| 18:23.27 | Qwell | ~freepbx |
| 18:23.28 | jbot | [~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 |
| 18:23.55 | Qwell | DanyWalker: nearly nobody here uses it, and won't be able to help you |
| 18:24.49 | arctic_import | Okay when I run a ztcfg -vv Why doe sit always tell me ""Channel 01: E & M (Default) (Slaves: 01)"" no matter what I set my signalling line to say in my zapata.conf. Shouldn't this change if I set signalling=fxo_ks |
| 18:24.50 | NovceGuru | hmm so I've been on this for a few days know. bandwidth.com or teliax.com for hosted service |
| 18:25.21 | [TK]D-Fender | arctic_import: Probably because ztcfg uses zaptel.con, NOT zapata.conf |
| 18:25.23 | Qwell | arctic_import: ztcfg uses zaptel.conf |
| 18:27.33 | arctic_import | Qwell: so is featb still an option? I guess I'm confused. I'm supposed to connect to another pbx using feature group b. So should I set my zaptel.conf to use e&m, and then set my zapata.conf to signalling=featb ?? Will that accomplish what I'm after. |
| 18:27.59 | Qwell | arctic_import: I don't know much about T1 signaling... |
| 18:28.20 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 18:29.20 | *** join/#asterisk Piedpyper (n=Geoff_La@rrcs-71-41-62-10.se.biz.rr.com) |
| 18:29.31 | *** part/#asterisk dw (i=dmwdmw@unaffiliated/dw) |
| 18:31.31 | *** join/#asterisk kdotmo (n=kyle@nv-69-69-253-142.sta.embarqhsd.net) |
| 18:32.03 | *** join/#asterisk dlynes (n=daniel@d206-116-189-12.bchsia.telus.net) |
| 18:32.56 | Juggie | arctic_import, you just set your span line and your b/d channels |
| 18:33.17 | Juggie | you dont set e&m |
| 18:34.28 | arctic_import | Juggie: I'm not doing ISDN, its a old style T1, that I'm supposed to using MF style signalling so I need to use featb in my /etc/asterisk/zapata.conf. I believe I'll need to set my /etc/zaptel.conf to use e&m. I'll just have to try it. |
| 18:34.47 | Juggie | oh, your doing clear channel |
| 18:35.23 | arctic_import | Juggie: yes. They want Feature Group B. |
| 18:35.27 | jeffspeff | I'm using voip and softphones... how do you set the caller id to be a company name or persons name instead of the phone number? When I call from my softphone to my cell phone, it shows my voip phone number only. also, i'm using sip, and my provider is teliax if that helps any. |
| 18:35.53 | denon | jeffspeff: the name is looked up by the pnone number, you can't set it, the remote side looks it up from the carrier |
| 18:35.55 | Juggie | ya what you are proposing sounds about right then |
| 18:36.19 | Juggie | denon, thats not allways true |
| 18:36.21 | Juggie | depends on your location |
| 18:36.48 | denon | Juggie: well .. he's using teliax, so probably US |
| 18:37.05 | Juggie | it will work, its weird |
| 18:37.21 | Juggie | in canada its much nicer, i can set cidname and it will pass almost everwhere |
| 18:37.36 | jeffspeff | well, i've set the info in my teliax account |
| 18:38.03 | denon | jeffspeff: well, you can try doing it like this: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 18:38.07 | jeffspeff | how do you set cidnames? could i use that in the us? |
| 18:38.10 | Juggie | you would have to check w/ teliax on that. all you can do is do Set(Callerid('whatever')=meh) |
| 18:38.19 | denon | it may work, it may not |
| 18:38.25 | jeffspeff | true |
| 18:38.29 | jeffspeff | ok, thanks. |
| 18:38.38 | Juggie | ya it works in canada, its nice, can set it to whatever you want |
| 18:38.45 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-116-111-122.rgv.res.rr.com) |
| 18:38.56 | denon | Juggie: canada's gotta have some advantages, I guess |
| 18:38.57 | denon | ducks :) |
| 18:39.03 | *** join/#asterisk bcl (n=bcl@neil.brianlane.com) |
| 18:39.13 | hsv-al | whats another way to view the output after -vvvc without piping it into blah.txt |
| 18:39.24 | hsv-al | while * is running, im getting red warning, but i want to check what it is |
| 18:39.28 | hsv-al | my buffer wont let me scroll back far enough |
| 18:39.34 | denon | hsv-al: use -r? |
| 18:39.39 | keith4 | uh, pipe it to less? |
| 18:39.39 | tzafrir_laptop | hsv-al, look at /var/log/asterisk/somelog |
| 18:39.45 | Juggie | denon, its one of many including free healthcare and a leader who isnt mentally retarded :) |
| 18:39.53 | tzafrir_laptop | tail -f can also be handy |
| 18:40.04 | Juggie | ./var/log/asterisk? |
| 18:40.04 | denon | yeah, tail -f is definitely your friend |
| 18:40.20 | Juggie | edit logger.conf to change logging settings |
| 18:40.58 | hsv-al | console => notice,warning,error |
| 18:41.05 | hsv-al | messages = > notice,warning, error |
| 18:41.38 | DanyWalker | somebody has used the Flash Operator Panel ? |
| 18:41.44 | denon | nope, no one has |
| 18:41.50 | denon | even the developer hasn never actually used it |
| 18:41.54 | denon | -n |
| 18:42.04 | hsv-al | I wish nano would support color codes |
| 18:42.24 | [TK]D-Fender | BRB |
| 18:42.33 | spokra | real unix guys use vi.. :> |
| 18:42.44 | denon | or at least say they're using vi, when they're really using vim |
| 18:42.57 | spokra | you have a point there!! |
| 18:43.20 | hsv-al | only warnings i get in asterisk -vvvc are now |
| 18:43.30 | hsv-al | red_smdi.c: no smdi interfaces are available to listen on |
| 18:43.33 | hsv-al | other then that, its all good |
| 18:44.32 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 18:45.17 | tzafrir_laptop | spokra, you mean nvi (a clone of the original berkely vi)? |
| 18:45.50 | tzafrir_laptop | hsv-al, you should just use vim. accept the facts of life |
| 18:46.26 | spokra | http://www.homebrew.net/visign/ |
| 18:46.53 | wwalker | I've got calls that occassionally get to a ":timeout" state. I've verified this with "exten => t,1,NoOp()" and see that we are entering the t portion of the context's dialplan. How do I get asterisk to drop the channel (at this point Hangup() was called before the timeout) |
| 18:46.54 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 18:46.54 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:47.12 | hsv-al | ack |
| 18:47.13 | hsv-al | a modal editor |
| 18:48.29 | *** part/#asterisk kdotmo (n=kyle@nv-69-69-253-142.sta.embarqhsd.net) |
| 18:50.16 | *** join/#asterisk kdotmo (n=kyle@nv-69-69-253-142.sta.embarqhsd.net) |
| 18:50.54 | DanyWalker | somebody know how show more extensions in the flash operator channel ? (fop), please |
| 18:51.17 | TrentCreek | http://www.voip-info.org/wiki/view/Asterisk+PBX+functions |
| 18:51.37 | TrentCreek | How do I install asterisk? |
| 18:51.55 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 18:51.59 | tzafrir_laptop | spokra, vi: the editor for the three-fingered folks? |
| 18:52.09 | Qwell | emacs: the editor for 20 |
| 18:52.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:52.53 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 18:53.40 | jayrod422 | any body have any idea why when i put a extensions context in sip context (ie [xyz] context=from-xyz) the incoming calls always go the default extensions context? |
| 18:53.59 | Nasra | sudo get-apt install Asterisk |
| 18:54.11 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 18:54.33 | [hC] | how is kernel 2.6.25 with asterisk? ive upgraded and now im having some weird call drop issues and what not |
| 18:54.38 | [TK]D-Fender | jayrod422: Go look at the SIP debug of an actual call to see whats happening. PASTEBIN is your friend... |
| 18:54.40 | [TK]D-Fender | ~pb |
| 18:54.41 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:54.43 | [TK]D-Fender | ^^^^^^^^^^ |
| 18:54.54 | [hC] | way more "avoiding initial deadlock, 10 retries" messages |
| 18:55.11 | TrentCreek | [TK]D-Fender: sip debug is deperciated |
| 18:56.00 | [TK]D-Fender | TrentCreek: Care to qualify that? Careful... I'm caffeinated ;) |
| 18:56.15 | lmadsen | ~ainap |
| 18:56.20 | jayrod422 | k |
| 18:56.24 | TrentCreek | [TK]D-Fender: yeah I ran it and got that message |
| 18:56.36 | TrentCreek | let me see |
| 18:57.00 | [TK]D-Fender | TrentCreek: "it"? What exactly prompted you to give me that warning jsut now? |
| 18:57.27 | TrentCreek | [TK]D-Fender: typing in "SIP DEBUG" |
| 18:57.48 | [TK]D-Fender | TrentCreek: that command format is indeed deprecated in 1.4, but still works. |
| 18:58.09 | TrentCreek | okay...groovy |
| 18:58.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:00.40 | *** part/#asterisk fainsys (n=fainsys@c-76-17-121-45.hsd1.ga.comcast.net) |
| 19:01.39 | TrentCreek | The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. |
| 19:02.55 | [TK]D-Fender | TrentCreek: Yes, sometimes the DO hide it in the "big print". |
| 19:03.04 | keith4 | deprecated != "doesn't work" |
| 19:03.15 | keith4 | uh... also != "depreciated" |
| 19:03.43 | TrentCreek | Yes it did not just come out and scream..i just happen to see it |
| 19:05.25 | wwalker | Qwell: do you really believe emacs is usable with only twenty fingers? |
| 19:05.40 | Qwell | wwalker: per hand |
| 19:05.47 | wwalker | oh, my mistake. |
| 19:09.18 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 19:14.00 | *** join/#asterisk Sephen (n=Sephen@proxy5.med-web.com) |
| 19:14.07 | seanbright | Qwell: nub |
| 19:15.16 | Sephen | Does anyone know if there is a way to control the automatic gain control that is happening on ZAP channels? The gain control is so harsh, that if someone whispers, it doesn't get picked up, and it is also concerning, as it makes the impression that the call has been dropped because it mutes the channel if the level isn't high enough. |
| 19:16.13 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 19:16.46 | deeperror | anyone familiar with the following warnings http://pastebin.ca/1043168 |
| 19:17.27 | deeperror | i seem to get thousands of these on occasions and I believe it is when someone is being put on hold and MOH is playing but i'm unable to duplicate |
| 19:22.35 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 19:22.54 | luke-jr | FYI, if you get iConnectHere service (bad idea), do be sure to tell them you are using it from outside the US |
| 19:23.09 | luke-jr | or else they will just repeatedly disable your account while they verify E911 support every few months |
| 19:23.39 | Qwell | luke-jr: sounds like a perfectly reasonable thing for them to do.. |
| 19:24.54 | luke-jr | Qwell: to shut off service unannounced every few months? |
| 19:25.26 | luke-jr | with no changes to account (service address the same as it was before) |
| 19:25.52 | Qwell | luke-jr: well, if they called/emailed before shutting it off.. |
| 19:26.08 | luke-jr | Qwell: they emailed after shutting it off |
| 19:26.17 | Qwell | luke-jr: oh, well then |
| 19:26.23 | luke-jr | and again, for no real reason |
| 19:29.05 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:30.09 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 19:31.50 | [hC] | Is there a problem with asterisk and kernel 2.6.25? Im getting tons of dropped calls now on this kernel |
| 19:32.43 | luke-jr | [hC]: well, Asterisk > 1.4.18.1 has never worked for me, so I'd try downgrading it |
| 19:33.09 | [hC] | I'm using asterisk 1.2 on this box. |
| 19:33.43 | luke-jr | that might be a good idea ⺠|
| 19:34.17 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 19:34.46 | iratik | Dumb Question: I don't know anything about sms, but how can I setup a number on my asterisk box to receive an sms text message ? |
| 19:35.28 | nny_1 | any 962 hackers here? |
| 19:35.29 | TrentCreek | is it in the book? |
| 19:35.37 | nny_1 | spa 962* |
| 19:35.47 | nny_1 | trying to figure out if we can move the soft buttons |
| 19:37.12 | TrentCreek | i have the latest kernel and no dropped calls |
| 19:38.13 | [hC] | TrentCreek: which kernel version/asterisk version? |
| 19:38.23 | TrentCreek | i am using 1.4.11 without problems |
| 19:38.50 | [hC] | then again i think im starting with a flawed base. I only upgraded my kernel to fix the /dev/rtc issue on this dell |
| 19:38.54 | TrentCreek | i have to look up the kerel..i alwas upgrade when an update comes out |
| 19:38.57 | [hC] | and im using ztdummy for iax2 trunking |
| 19:39.01 | [hC] | clearly something is wrong with timing. |
| 19:39.17 | TrentCreek | maybe you should upgrade |
| 19:39.20 | *** join/#asterisk dom_aheeva (n=chatzill@atelka.info) |
| 19:39.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:39.37 | [hC] | yes i should, but this is a large scale production box that is not onsite, so its kind of challenging :) |
| 19:39.52 | TrentCreek | what version of Linux? |
| 19:39.58 | [hC] | debian |
| 19:40.10 | TrentCreek | i think that uses Yum, doe sit not? |
| 19:40.11 | [hC] | upgrading linux or asterisk isnt the problem |
| 19:40.17 | [hC] | its making sure that the dial plan is 1.4 compatible :) |
| 19:40.31 | [hC] | no, it uses apt-get. I dont install asterisk from packages anyhow. |
| 19:40.52 | dom_aheeva | anyone know of free softphone that supports URLs popups |
| 19:41.02 | TrentCreek | 1.2 - 1.4 is not that big of a upgrade...just look at the console and it will report anything that will be removed at a later time |
| 19:41.31 | deeperror | in sip.conf what is the consensus on allow and disallow? Can they both be removed and let it negotiate on it's own? Or keep it tight to only allow what you want? |
| 19:42.09 | [hC] | deeperror: disallow=all then allow= the ones you want. |
| 19:42.25 | [hC] | deeperror: makes troubleshooting much much easier. and you also are left with a perfect idea of whats going on. |
| 19:43.30 | nny_1 | is it possible to simulate the transfer feature of a phone through a macro? |
| 19:43.34 | deeperror | [hC], http://pastebin.ca/1043168 this is the warning i'm trying to squash. I get thousands of these a few times an hour I think it has to do with MoH but am unable to duplicate it. What other codec should I allow do you think as currently i'm disallow=all allow=ulaw only |
| 19:44.15 | deeperror | nny_1, check into features.conf for blind and attended transfer |
| 19:44.30 | nny_1 | deeperror: thanks |
| 19:44.41 | [hC] | deeperror: frame type 64 = slin |
| 19:44.49 | [hC] | deeperror: but you should be able to transcode between slin and ulaw |
| 19:45.32 | deeperror | so would that be the format that particular MoH file is and the error occurs when that one gets played? |
| 19:45.49 | [hC] | I suppose it would be yes. |
| 19:45.57 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:46.06 | [hC] | but the thing that is weird, is that you should be able to transcode between slin and ulaw, so you shouldnt see this problem |
| 19:46.19 | TrentCreek | [hC]: When I put 1.4 onhere..I used the dial plans from the 1.2 book because the 1.4 book was not yet. I get warnings "xxxx has been depreciated and will be removed in the near future, use bla, bla, bla," |
| 19:46.20 | [hC] | if you've somehow disabled the slin codec, turn it back on. things like meetme use it, as well as parts of the DTMF module |
| 19:46.33 | deeperror | ok i'll enable that to see if it stops the warnings |
| 19:46.42 | [hC] | TrentCreek: yeah, Ive done it once before for a 200 seat installation. when you have 8-10 ivr's and a bunch of custom stuff, it takes a while. |
| 19:46.52 | [hC] | deeperror: you dont enable it in sip.conf |
| 19:46.58 | [hC] | deeperror: its just a usable module in asterisk |
| 19:47.11 | deeperror | i wouldn't put allow=slin? |
| 19:47.52 | [hC] | no. |
| 19:47.57 | [hC] | asterisk itself is what transcodes the audio, not your phone |
| 19:48.15 | [hC] | if you're getting this error, its because asterisk is unable to transcode slin to ulaw. find out why the slin codec isnt enabled in asterisk itself |
| 19:48.18 | [hC] | start with show codecs |
| 19:48.56 | deeperror | shows up in the list |
| 19:49.03 | [hC] | show modules like sln |
| 19:49.09 | [hC] | you should see format_sln.so |
| 19:49.28 | deeperror | format_sln.so Use Count = 0? |
| 19:49.33 | iratik | Any ideas on receiving sms texts with asterisk ? |
| 19:49.59 | [hC] | hm... looking at this post some more, asterisk is infact determined to think you are capable of receiving slin audio |
| 19:50.07 | [hC] | in your sip.conf make sure disallow=all comes first, then allow=ulaw comes next. |
| 19:50.25 | [hC] | next if this is pertaining to MoH you may as well go look for slinear files in your moh directory and remove them. |
| 19:50.29 | deeperror | that is correct but they are not on the direct above line would that matter? |
| 19:50.42 | [hC] | deeperror: order matters. disallow first, allow second. |
| 19:50.43 | deeperror | MoH files are all default asterisk files |
| 19:50.51 | [hC] | deeperror: if disallow did not come first, that is your problem |
| 19:51.06 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 19:51.34 | deeperror | it was first...but had a few other options between it and allow |
| 19:51.34 | [hC] | deeperror: yeah, the more i look at this, whats happening is asterisk thinks your phone will take slin, so its not transcoding, and its trying to send to your phone directly. your codec order is probably the cause here. |
| 19:51.57 | TrentCreek | iratik: did you look in the book? |
| 19:52.04 | deeperror | also the phones are all zap and the sip is between pbx and termination |
| 19:52.15 | _ShrikE | Im having an interesting issue with * 1.6B9, sip tcp, and promiscredir=yes. I am trying to talk to an exchange 2007 um server. By default exchange sends a 302 forwarding you to worker thread on a different port, in this case 5065. It appears that asterisk is not honoring the port 5065 in the contact field, and instead continues sending to port 5060. Any suggestions? http://www.pastebin.ca/1043213 |
| 19:53.04 | iratik | hmmm |
| 19:53.21 | TrentCreek | iratik: .01 seconds via google http://www.the-asterisk-book.com/unstable/applikationen-sms.html |
| 19:53.32 | TrentCreek | ~vampire |
| 19:53.40 | iratik | I saw that already |
| 19:53.58 | deeperror | maybe the provider is sending slin to me and i've got it disallowed? |
| 19:53.58 | iratik | but there is no such directory under /var/spool/asterisk |
| 19:54.14 | iratik | and i don't know what an smsc is |
| 19:54.22 | iratik | i googled smsc asterisk, and it took me to the same page |
| 19:54.31 | TrentCreek | Then /var/spool/asterisk may be in anothe rlocation |
| 19:55.04 | iratik | no... /var/spool/asterisk is where all such files are located on my system |
| 19:55.33 | TrentCreek | iratik: .01 seconds via google An SMS center (SMSC) is responsible for handling the SMS operations of a wireless network. When an SMS message is sent from a mobile phone, it will reach an SMS center first. The SMS center then forwards the SMS message towards the destination. An SMS message may need to pass through more than one network entity (e.g. SMSC and SMS gateway) before reaching the destination. The main duty of an SMSC |
| 19:55.33 | TrentCreek | <PROTECTED> |
| 19:55.37 | iratik | i set the system up... just never thought about nor encountered sms during |
| 19:55.39 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 19:55.53 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 19:56.37 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 19:57.00 | iratik | TrenetCreek: like i said , i searched for "smsc asterisk" ... it led to the exact same page .. in 0.05 seconds |
| 19:57.40 | iratik | lol... you act like i'm a complete idiot who doesn't know anything about solving things for himself ... but i'm not ... maybe i'm just not as proficient at coming up with masterfully conceived google queries ... but i do look |
| 19:57.54 | TrentCreek | lol |
| 19:58.04 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583881.dsl.bell.ca) |
| 19:58.10 | TrentCreek | a lot of people on here want to be led step by step a lot of times |
| 19:58.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:59.56 | TrentCreek | iratik: i would also like to know about SMS too |
| 20:00.21 | iratik | sms asterisk: .01 seconds by google http://www.ozekisms.com/examples-and-solutions/asterisk-pbx-sms/index_p_php_q_owpn_e_319opt.html |
| 20:00.24 | iratik | lol |
| 20:00.25 | TrentCreek | Seems it can be done...as I got some text spam the other day |
| 20:00.33 | TrentCreek | so it has to be possible |
| 20:00.54 | TrentCreek | Let me ask someone if there is anything special thatneed to be done |
| 20:02.33 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 20:02.48 | TrentCreek | iratik: I found out |
| 20:03.07 | iratik | yeah... why do i get a reject from my mobile phone when i try messaging to the DID ? |
| 20:03.50 | TrentCreek | you need to get a termination service to handle the SMS |
| 20:04.25 | TrentCreek | and you use APIs to interface with their termintion |
| 20:04.46 | iratik | can you give me an example of a company that provides sms termination and an API? |
| 20:04.48 | iratik | that would be great |
| 20:05.42 | TrentCreek | voicetrading.com, but require $700 per purchase in credits minimum....though they will give you a $5 credit to try their service...and their SMS are NOT cheap |
| 20:06.38 | TrentCreek | acutally it's 500 Euros, but by the time you convert..it's over $700 |
| 20:06.43 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 20:06.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:07.18 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 20:08.40 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 20:08.52 | iratik | hmmm |
| 20:09.00 | anonymouz666 | putnopvut: is it normal app_queue generate a Hangup event to manager after time cycle? |
| 20:09.45 | putnopvut | what do you mean by "time cycle?" |
| 20:10.17 | anonymouz666 | I am listening the manager, and I see a different behaviour after 1.4. I see Hangup events from each members being called |
| 20:10.22 | anonymouz666 | while ringing |
| 20:10.58 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 20:11.30 | anonymouz666 | I wonder where this come from since I can't see any manager_event() named "Hangup" or something like that |
| 20:11.41 | shtoom | Hi I am trying to install PRI E1 when I treid to place a call I am getting busy congested error |
| 20:11.56 | putnopvut | anonymouz666: yeah, the hangup is not from app_queue. I guess that's coming from elsewhere. I'll take a look. |
| 20:11.59 | shtoom | <PROTECTED> |
| 20:11.59 | shtoom | <PROTECTED> |
| 20:12.33 | shtoom | how ever the call is going through PRI testing equipment |
| 20:12.35 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 20:12.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:13.23 | Sephen | Does anyone know if there is a way to control the automatic gain control that is happening on ZAP channels? The gain control is so harsh, that if someone whispers, it doesn't get picked up, and it is also concerning, as it makes the impression that the call has been dropped because it mutes the channel if the level isn't high enough. |
| 20:13.39 | putnopvut | anonymouz666: are you using local channels? |
| 20:13.49 | anonymouz666 | no |
| 20:14.10 | putnopvut | anonymouz666: okay, there are two places in channel.c that issue a manager event called "Hangup" |
| 20:14.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:14.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:15.31 | anonymouz666 | I see |
| 20:15.37 | putnopvut | anonymouz666: I'm guessing that somehow ast_hangup is being called from somewhere, but I'm not sure where. |
| 20:15.42 | putnopvut | anonymouz666: did you say you are using 1.4? |
| 20:15.48 | anonymouz666 | 1.4.20 |
| 20:18.09 | putnopvut | anonymouz666: the only places I see that ast_hangup is being called directly from app_queue are in failure situations (like if the channel could not be requested, or if the member was busy). |
| 20:20.16 | drako | how can i determinate whos trying (IP) to connect to my server? |
| 20:21.01 | drako | [Jun 9 16:10:09] NOTICE[28675]: chan_sip.c:13815 handle_request_invite: Failed to authenticate user "fox" <sip:fox@90.122.32.23>;tag=as297b65ab |
| 20:21.22 | anonymouz666 | putnopvut: very very strange then. because all members are available. |
| 20:21.40 | hsv-al | drako, use ip acl's |
| 20:21.41 | anonymouz666 | registered/not in use. |
| 20:21.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:21.47 | hsv-al | and logs show originating ip's of connection |
| 20:22.10 | putnopvut | anonymouz666: are the hangups happening for the members whose phones are not ringing? |
| 20:22.11 | drako | hsv-al, where how? |
| 20:22.21 | hsv-al | deny and permit statements in sip.conf i think |
| 20:22.30 | hsv-al | need to know basic networking and cidr/subnetting |
| 20:22.32 | putnopvut | app_queue makes a list of all members of a queue. Then it will call one of those and send a hangup to all the other members. |
| 20:22.58 | putnopvut | That shouldn't be noticeable though because there shouldn't be a channel associated with those members. |
| 20:23.06 | drako | hsv-al, thats no problem, i just want to know whos trying to connect |
| 20:23.15 | drako | i understand networking |
| 20:23.31 | hsv-al | their ip will show in the logs of originating connections when they attempt to connect |
| 20:23.41 | hsv-al | whats wrong? |
| 20:24.30 | anonymouz666 | putnopvut: at the point I got all hangups from each member, there's no bridged call yet |
| 20:25.58 | anonymouz666 | putnopvut: let me ask you a question, this has anything to do with call-limit? each peer has call-limit=99 |
| 20:26.09 | putnopvut | anonymouz666: it shouldn't matter. |
| 20:26.19 | putnopvut | anonymouz666: did this actually have any affect on the calls in the queue? |
| 20:26.19 | hsv-al | drako, page 98, and page 100 of the PDF |
| 20:26.38 | anonymouz666 | putnopvut: no, it works fine. |
| 20:26.50 | putnopvut | anonymouz666: that's even more strange. |
| 20:27.30 | anonymouz666 | this behaviour supposed to happen only if I pickup the call, right? |
| 20:27.52 | drako | ~book |
| 20:27.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:28.09 | anonymouz666 | but I am seeing things before the call got bridged |
| 20:31.44 | *** join/#asterisk mmartinn (n=martins@n128-227-41-215.xlate.ufl.edu) |
| 20:32.05 | hsv-al | this chapter 5 is really explaining alot of things |
| 20:32.11 | hsv-al | dialplan chap would of been nice earlier in the book :) |
| 20:35.03 | maqr | ok, i got a polycom 330 ip phone, but i didn't realize that i don't have a power adapter for it or PoE.... can someone kick me in the right direction for how i'd setup PoE with a common linksys router so i can use the phone? |
| 20:35.44 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 20:36.01 | seanmh | If someone were to look for cisco 7960 and 7940 images.. where might one look? |
| 20:36.36 | hsv-al | www.fbi.gov/warez |
| 20:36.44 | putnopvut | anonymouz666: Sorry if I asked you this before, but do you see a hangup occur on channel that's actually ringing? |
| 20:36.58 | maqr | seanmh: google image search? :p |
| 20:37.07 | seanmh | hahah |
| 20:37.18 | seanmh | the SIP images.. but I think you knew what I meant ;) |
| 20:37.34 | maqr | i'm pretty sure that would count as warez |
| 20:37.41 | mmartinn | Can anyone tell me if the rtp debug at http://pastebin.com/m2f6f1022 is correctly sending rfc2833 dtmf digits "2 -- 2 -- 1" |
| 20:39.21 | anonymouz666 | putnopvut: the setup is like this: I am listening the Manager events, call the queue. It rings all members and each one send a Hangup event to my listener. However, If I pickup the phone everything works fine. |
| 20:40.04 | putnopvut | anonymouz666: Weird. At what point does the hangup get sent? |
| 20:40.37 | anonymouz666 | putnopvut: 10 secs approx |
| 20:40.44 | anonymouz666 | after ringing everyone |
| 20:40.53 | putnopvut | What's the timeout for the queue? |
| 20:40.58 | putnopvut | in queues.conf? |
| 20:40.58 | anonymouz666 | 300 :) |
| 20:41.15 | putnopvut | yikes. |
| 20:41.23 | putnopvut | What type of channels are you using? |
| 20:41.59 | maqr | are most PoE IP phones 48V? |
| 20:42.15 | anonymouz666 | you mean timeout= parameter? |
| 20:43.41 | anonymouz666 | it's 300 on Queue() and timeout=15 in queue.conf. |
| 20:44.37 | putnopvut | okay, I was curious if the hangup is happening 15 seconds after the call is made. |
| 20:44.47 | putnopvut | But you said it happens about 10 seconds after. |
| 20:45.41 | anonymouz666 | so in this condition if we hit the timeout the Hangup event is supposed to happen? |
| 20:46.18 | putnopvut | anonymouz666: yes. |
| 20:48.21 | anonymouz666 | thanks then. that's why |
| 20:48.44 | putnopvut | anonymouz666: ah, okay, then. You had me worried for a minute :) |
| 20:49.03 | anonymouz666 | hehe |
| 20:49.50 | *** join/#asterisk methods (n=methods@c-68-36-237-152.hsd1.nj.comcast.net) |
| 20:51.21 | methods | anyone know what setting on the pap2 causes it to send back a busy signal after like 3 rings ? |
| 20:51.22 | deeperror | [hC], the default sounds have been converted to ulaw so will see if this fixes the warnings...thanks for the pointers |
| 20:51.44 | hsv-al | http://thevoice.digium.com/ |
| 20:51.47 | hsv-al | down? |
| 20:51.55 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 20:51.58 | mmartinn | What is VLDTMF and why would I sometimes not get one correctly sent out on Zap hardware? |
| 20:52.43 | *** join/#asterisk ruied (n=ruied@bl7-209-241.dsl.telepac.pt) |
| 20:53.13 | deeperror | mmartinn, variable length you see warnings once in a while? |
| 20:53.58 | mmartinn | deeperror: Sometimes my users dial a remote PBX and then try to dial an extension, and the dtmf doesn't always make it, and the remote phone system complains to them. |
| 20:54.06 | mmartinn | deeperror: But other times it works... |
| 20:55.05 | mmartinn | deeperror: In the log of an example where it fails, the only thing that seems important is that VLDTMF tones are missing... the rtp parts seem like they are the same in the log. |
| 20:57.07 | *** part/#asterisk kdotmo (n=kyle@nv-69-69-253-142.sta.embarqhsd.net) |
| 20:57.19 | *** join/#asterisk dieno (i=771e645d@gateway/web/ajax/mibbit.com/x-de41e99fb69ea554) |
| 20:57.42 | dieno | does any one have experience on web click to call |
| 20:58.10 | methods | anyone know where i can set the timers on this pap2 ? |
| 20:59.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:59.48 | nny_1 | if i enable #1 blind xfer in asterisk's features.conf is it supposed to "just work"? Do I press #1 during the call? |
| 21:00.25 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 21:02.38 | *** part/#asterisk Piedpyper (n=Geoff_La@rrcs-71-41-62-10.se.biz.rr.com) |
| 21:04.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:07.07 | [hC] | nny_1: you either have to add the t or T argument to Dial() to enable it |
| 21:07.07 | NovceGuru | anybody have a sipura spa-20xx or pap2 serial/mac address I could borrow |
| 21:07.13 | [hC] | nny_1: show application Dial will explain it |
| 21:07.30 | nny_1 | [hC]: thank ya sir |
| 21:12.03 | [hC] | nny_1: my pleasure |
| 21:12.13 | nny_1 | anyone here a linksys 962 user? |
| 21:12.35 | methods | anyone have any idea why my pap2 would stop inbound ringing after 2 rings ? |
| 21:12.40 | nny_1 | noticed when you are on a line, the BLF presses are ignored |
| 21:12.50 | nny_1 | which *sucks* cause I had high hopes for them |
| 21:12.55 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:12.55 | *** part/#asterisk bcl (n=bcl@neil.brianlane.com) |
| 21:13.01 | nny_1 | hoping maybe there is a way to change that |
| 21:13.09 | nny_1 | 962/932 sorry |
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| 21:19.17 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 21:19.19 | shtoom | Hi I am trying to install PRI E1 when I treid to place a call I am getting busy congested error |
| 21:19.30 | shtoom | app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 21:19.37 | shtoom | == Everyone is busy/congested at this time (1:0/0/1) |
| 21:20.01 | nny_1 | lol |
| 21:20.03 | nny_1 | someone shoot me |
| 21:20.11 | nny_1 | trying to implement transfer with blf |
| 21:20.13 | nny_1 | with 962 |
| 21:20.16 | nny_1 | blind xfer |
| 21:20.20 | nny_1 | it works already |
| 21:20.21 | nny_1 | no need to do anything |
| 21:20.34 | nny_1 | lol |
| 21:20.41 | shtoom | On pri intense debug I get to see this message Sending Set Asynchronous Balanced Mode Extended continoually |
| 21:21.08 | shtoom | I am using asterisk 1.4.9 |
| 21:21.36 | shtoom | zaptel 1.4.11 and libpri-1.4.4 |
| 21:22.28 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:23.29 | shtoom | hi [TK]D-Fender can you help with this problem ? |
| 21:23.53 | [TK]D-Fender | shtoom, At this point.... only if Iwere psychic, which I'm not. |
| 21:24.06 | shtoom | I am getting app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) busy congested when I try to dialout on E1 |
| 21:24.41 | shtoom | i tried 1.2 and 1.4 versions of libpri / zaptel /asterisk |
| 21:24.52 | shtoom | but still I am getting the same error |
| 21:25.04 | shtoom | On pri intense debug I get to see this message Sending Set Asynchronous Balanced Mode Extended continoually |
| 21:25.04 | keith4 | [TK]D-Fender: i thought you *were* psychic |
| 21:25.08 | keith4 | is disappointed |
| 21:25.17 | [TK]D-Fender | shtoom, error means nothing. Pastebin yoru configs and the complete CLI output of the failed attempt at verbose 10 |
| 21:27.41 | shtoom | zaptel.conf - http://pastebin.com/d6f14f6d4 |
| 21:28.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:30.04 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 21:30.07 | shtoom | zapata.conf -http://pastebin.com/d70148c06 |
| 21:30.15 | mike8901 | the jesus phone has arrived! |
| 21:30.20 | methods | yea i have no idea waht's goin on but my ata only rings twice... |
| 21:30.21 | mike8901 | (the second jesus phone, that is) |
| 21:31.58 | shtoom | console out put - http://pastebin.com/d20ba9b79 |
| 21:32.34 | shtoom | [TK]D-Fender: do you see any thing causing this error ? |
| 21:33.18 | shtoom | zttool says ok no alarms |
| 21:33.34 | [TK]D-Fender | shtoom, your OSS chanel is clearly failing so you should not be using that to "test" with. Next, why are you targeting channel 20 on your PRI directly? |
| 21:35.18 | shtoom | [TK]D-Fender:even though OSS fails call is supposed to go thru. Infact I tried with grouping the channels also with Zap/g0/number |
| 21:35.31 | shtoom | I am getting the same thing with that as well |
| 21:35.33 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 21:35.50 | shtoom | call is going thru with the PRI test equipment |
| 21:36.38 | [TK]D-Fender | shtoom, enable PRI debug and retest, and DON'T do the test with your busted OSS |
| 21:36.53 | pjz | anyone know why transfers might fail to work? I've got some polycom 330's talking to asterisk, and they can all xfer internally just fine but if an external call comes in, only the inbound side seems to get xfer'd... the outbound voice channel is essentially muted |
| 21:39.04 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-2-192.lns10.syd7.internode.on.net) |
| 21:40.40 | shtoom | [TK]D-Fender:Unfortunately I am not at the location and no one is and the machine is behind nat only ssh opened so i can't test with softphone |
| 21:41.05 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 21:41.07 | pjz | shtoom: tunnel through ssh |
| 21:41.37 | shtoom | Mas-Dataserver*CLI> pri show spans |
| 21:41.38 | shtoom | PRI span 1/0: Provisioned, Down, Active |
| 21:41.38 | shtoom | PRI span 2/0: Provisioned, In Alarm, Down, Active |
| 21:42.12 | shtoom | span 1 seems to be down what might be the reason |
| 21:42.19 | [TK]D-Fender | shtoom, Also, are you SUER you're supposed to be acting as a timing SOURCE? |
| 21:42.31 | [TK]D-Fender | SURE* |
| 21:43.18 | shtoom | [TK]D-Fender: no remote end should be my timing source |
| 21:43.57 | shtoom | in zaptel..conf 0-for timing source means, remote end is master right ? |
| 21:45.34 | [TK]D-Fender | shtoom, then your span line is wrong |
| 21:45.44 | [TK]D-Fender | shtoom, no, the exact oposite |
| 21:45.49 | [TK]D-Fender | opposite* |
| 21:47.50 | shtoom | [TK]D-Fender: then how come this sangoma utility is configuring it with 0 even though I selected colck= normal instead of master |
| 21:48.35 | [TK]D-Fender | shtoom, Don't know, don't care. |
| 21:48.47 | [TK]D-Fender | shtoom, I never use those scripts. |
| 21:48.55 | [TK]D-Fender | shtoom, You should know * for yourself. |
| 21:51.03 | alrs | shtoom: some of the options in zaptel.conf are ignored by the wanpipe stuff. If you're having problems with a Sangoma card, you should give them a call. Their support department should pick up the phone pretty quickly if you call during business hours. |
| 21:51.26 | alrs | shtoom: I've never had to wait more than 30 seconds to talk to someone in support there. |
| 21:51.58 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
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| 22:01.46 | pjz | anyone know why transfers might fail to work? I've got some polycom 330's talking to asterisk, and they can all xfer internally just fine but if an external call comes in, only the inbound side seems to get xfer'd... the outbound voice channel is essentially muted. But it only happens on a transfer-with-consult; blind transfers work just fine |
| 22:02.56 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 22:06.16 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 22:06.30 | shtoom | power went out |
| 22:06.32 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 22:08.57 | *** join/#asterisk telephreak (n=slestak@12.145.241.251) |
| 22:09.26 | telephreak | hello, has anybody ever got an iaxy device to allow pulse diallng? |
| 22:09.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:10.20 | dieno | is anyone with Click to Dial Experience |
| 22:11.41 | [TK]D-Fender | dieno, What about it? |
| 22:11.56 | dieno | like voipjots script |
| 22:12.00 | dieno | need to route on my cell |
| 22:12.25 | [TK]D-Fender | dieno, Go read up on "call files" and "AMI Originate" on the WIKI |
| 22:12.26 | [TK]D-Fender | ~wikis |
| 22:12.27 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:12.47 | dieno | hmm i did |
| 22:13.02 | dieno | i can transfer call to my SIP using SIP/5000 |
| 22:13.28 | [TK]D-Fender | dieno, Ok, none of that made any sense... continue... |
| 22:13.49 | dieno | but when i enter Local/1xxxxxxx@from-internal it returns call to Local/1xxxxxxx@from-internal rather than trasfer to another number |
| 22:14.13 | [TK]D-Fender | dieno, enter what into where? |
| 22:14.38 | [TK]D-Fender | dieno, and ditch this term "transfer". You are not trasferring anything with a call file.... |
| 22:14.50 | dieno | let me send you my code |
| 22:14.54 | [TK]D-Fender | ~pb |
| 22:14.55 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:14.56 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 22:15.31 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 22:20.11 | dieno | [TK]D-Fender http://pastebin.ca/1043313 please take a look at this code |
| 22:21.15 | *** join/#asterisk dlynes (n=daniel@dsl-vlan468-66-18-244-66.nucleus.com) |
| 22:22.41 | dieno | [TK]D-Fender soo are you still reading :) |
| 22:24.10 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 22:24.17 | jaytee | <PROTECTED> |
| 22:25.31 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 22:25.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:28.27 | dieno | hmmmmmmmmmmmm |
| 22:28.44 | dieno | so anyone else with click to dial knowledge |
| 22:28.52 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 22:36.37 | TrentCreek | ~peanutbutter |
| 22:42.09 | outtolunc | .. |
| 22:42.25 | outtolunc | oopsie |
| 22:43.25 | mwalling | clean up, aisle 5 |
| 22:52.22 | [TK]D-Fender | dieno, That is a large script and you aren't doing much to narrow down the point of failure. |
| 22:52.57 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 23:01.37 | *** join/#asterisk d-k-t (n=dt@125.120.129.131) |
| 23:02.37 | [TK]D-Fender | Ok, off to rebuild my system. BBL |
| 23:04.38 | imcdona | I have a question.....how often does anyone see a T-1 ATM circuit from a telco into an Adtran channel bank to break out FXS lines? This is the first time I have seen something other than TDM. Or is ATM more common than I think it is |
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| 23:17.42 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 23:18.14 | ZX81 | hi all, having a problem with background noise + digium hardware echo can killing the remote speaker. Any ideas? |
| 23:18.30 | ZX81 | aex800 card |
| 23:18.41 | *** join/#asterisk dlynes_laptop (n=daniel@dsl-vlan468-66-18-244-66.nucleus.com) |
| 23:18.42 | ZX81 | I've changed it from mute to drop volume |
| 23:18.46 | ZX81 | in modprobe |
| 23:18.53 | ZX81 | and changed it to only drop 5db |
| 23:18.56 | drmessano | Call Digium support |
| 23:18.59 | ZX81 | but now there's echo |
| 23:19.00 | ZX81 | yeah |
| 23:19.01 | ZX81 | ok |
| 23:19.05 | ZX81 | fair enough :) |
| 23:19.27 | alrs | ZX81: are you using the fancy new voicebus zaptel |
| 23:19.27 | alrs | ZX81: 1.4.10 + |
| 23:19.27 | drmessano | Why waste your time on IRC.. you have FREE support |
| 23:19.27 | ZX81 | yeah |
| 23:19.38 | ZX81 | IRC is easier to do lots of things at the same time |
| 23:19.45 | ZX81 | phone requires attention :) |
| 23:19.46 | alrs | drmessano: because FREE digium support is sometimes pretty iffy |
| 23:20.17 | jameswf-home | offers non iffy free support :) |
| 23:20.28 | alrs | drmessano: I only know about the voicebus stuff because digium support told me it would help with interrupt problems on a 4-span t1 card |
| 23:20.39 | drmessano | alrs: They make the card.. if they are offering FREE support, and have the power to actually CORRECT things like hardware and software issues from the backend, it's silly to ask elsewhere |
| 23:20.42 | alrs | drmessano: turns out that voicebus is for their analog cards |
| 23:21.06 | alrs | drmessano: where on Digium's site do they mention hardhdlc being miles better than using dchan in zaptel.conf? |
| 23:21.23 | *** join/#asterisk CaRb0n^ (n=playa@203.81.221.240) |
| 23:21.25 | drmessano | Who said anything about their website? WTF? |
| 23:22.23 | alrs | drmessano: WTF is that the people on this channel often know more about this stuff than Digium support |
| 23:22.37 | jameswf-home | thinks hardhdlc requires a patch but is recomended in certain situations |
| 23:22.43 | alrs | No patch required |
| 23:22.47 | alrs | It's in Zap 1.4 |
| 23:22.58 | drmessano | Point made |
| 23:23.18 | jameswf-home | hardhdlc is good when you have multiple spans and decent volume' |
| 23:23.31 | jameswf-home | volume as in call count |
| 23:23.39 | alrs | hardhdlc is good any time you don't want to see spans dropping and HDLC errors all over the place |
| 23:23.42 | alrs | I've seen no downside |
| 23:23.49 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 23:23.55 | voxter | hum... im on 1.2 with 2 spans almost maxing them out (sangoma though) i wonder if hardhdlc is necessary? |
| 23:24.05 | voxter | Ah, I dont have any dropped spans or hdlc errors. |
| 23:24.15 | alrs | Sangoma's zaptel patches to 1.2 were to put in hardware hdlc |
| 23:24.26 | voxter | well there ya go. nevermind then! :) |
| 23:24.29 | alrs | Sangoma doesn't patch Zaptel anymore in 1.4 because the support is built-in now |
| 23:24.31 | voxter | Totally forgot about that. |
| 23:24.50 | jameswf-home | voxter: only time i have seen it necissary is a user had a quad span 4 pris which he maxed out often |
| 23:25.29 | alrs | it only works on the 2 and 4-port cards |
| 23:25.36 | alrs | but it helps a lot |
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| 23:28.58 | errr | 208.67.222.222 |
| 23:29.05 | errr | oops |
| 23:29.55 | file | that would be an OpenDNS server |
| 23:29.59 | errr | yep |
| 23:30.19 | errr | I was cleaning off my touchpad and it was in my paste buffer when I started |
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| 23:44.03 | mwalling | always detach screen before doing such actions :) |
| 23:48.45 | *** join/#asterisk kdotmo (n=quiksilv@ip68-96-223-253.lv.lv.cox.net) |
| 23:49.37 | voxter | Dont suppose anyone is aware of the interval that qualifys are done at (SIP devices)? or if it is changeable? |
| 23:51.30 | dieno | http://pastebin.ca/1043372 can any please tell me what that is |
| 23:51.49 | voxter | nevermind, found it |
| 23:52.15 | Juggie | voxter, its not changeable i dont think |
| 23:52.22 | voxter | its in chan_sip.c |
| 23:52.33 | voxter | default is 60sec if the peer is okay, and 10sec if its not. |
| 23:52.38 | Juggie | ya, well everything is changeable :) |
| 23:52.42 | Juggie | if you change the source |
| 23:52.47 | voxter | touche :) |
| 23:53.00 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:53.02 | voxter | it is not a conf file change though, no. |
| 23:53.08 | Juggie | all qualify=2000 would do for example is so long as the peer responded in 2 seconds it would be up |
| 23:53.17 | Juggie | it doesnt change the time |
| 23:53.45 | dlynes | Does anyone have a contact at Solunet, or Audiocodes that's helpful for getting access to the latest firmware for the MP-124? |
| 23:54.36 | dlynes | I've been getting nowhere getting firmware for it from Solunet ever since we bought two of their devices |
