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| 01:34.59 | jcims | anyone from Teliax in here? Just did a pay as you go account, but can't access my control panel because i can't pass the authorization question until tomorrow (how much was posted to my cc) |
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| 02:40.04 | kaushal | Hi |
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| 03:40.01 | gurur | i am very new to voip world, want to ask a very smiple question. i wont be sad if noone replies it :) companies are using voip to reduce their costs. i know they can speak through internet. But both sides should have voip to talk each other. Unless, both sides are using voip, how companies are making money and reducing their costs ? |
| 03:40.24 | WIMPy | ITSPs |
| 03:40.31 | WIMPy | ~itsp |
| 03:40.32 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 03:41.05 | WIMPy | And I personally call it a myth. Using VOIP can also increase costs. |
| 03:41.29 | WIMPy | Depends heavily on what you do. |
| 03:41.46 | ChannelZ | And no, both sides don't have to be voip |
| 03:41.52 | gurur | WIMPy: in my countr (turkey) a friend of mine (he is EE like me) told me that, companies are making to much $$$ with using voip, because with voip they dont pay tax |
| 03:42.27 | gurur | ChannelZ: if they the other one doesnt use voip, i need to use a middleman to transfer my call to him called ... ? |
| 03:42.29 | WIMPy | That must be a turkish special case then, I guess. |
| 03:42.32 | ChannelZ | That probably has more to do with your tax system |
| 03:42.51 | gurur | ptsn is the normal landline right? |
| 03:42.52 | ChannelZ | You call regular phones via your ITSP |
| 03:43.15 | WIMPy | It is |
| 03:43.17 | ChannelZ | Basically if YOU did VoIP you're "offloading" the hardware interface to the PSTN to a service provider |
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| 03:43.26 | gurur | i see. i am alittle shamed about asking those stupid questions as an EE |
| 03:43.44 | gurur | so please tell me, how i can be a itsn ? |
| 03:43.49 | ChannelZ | I'd say it's actually fairly rare yet that people do VoIP-to-VoIP without it hitting the telephone network |
| 03:44.59 | WIMPy | By getting both enough PSTN and internet connectivity and some gateways. And quite a bit of logic for handling customers. |
| 03:45.30 | gurur | WIMPy: but how i would handle a phone call from a voip to gsm for ex. ? |
| 03:45.36 | gurur | *forward |
| 03:46.07 | WIMPy | Probably by fording it to the PSTN and let someone else do it. Or you use a GSM gateway. |
| 03:46.16 | gurur | hmm. |
| 03:46.19 | *** join/#asterisk cerberus_za (~coert@196-215-29-178.dynamic.isadsl.co.za) |
| 03:46.31 | gurur | just thinking how i can make money for some small business |
| 03:46.36 | WIMPy | But that will use normal phone avcounts so no passing of caller ID. |
| 03:46.38 | gurur | like setupping their voip network with asterisk |
| 03:47.08 | WIMPy | You can offer it as replacement for classic PBXs. |
| 03:48.16 | gurur | also, i am lookign for a solution, that can auto call m customers and give a recoded message to them, and after tell the customer "to connect customer relationsihp department press 1 (and they can talk with me), to have listed your name press 1( http://mysite.com/script.php?callerid=111 would be called) |
| 03:48.47 | WIMPy | You can do that with Asterisk. |
| 03:48.53 | gurur | WIMPy: i know thati think. |
| 03:49.00 | WIMPy | No matter if via PSTN or VOIP. |
| 03:49.06 | gurur | i have seen some codes with gnudialer, and i really got scared |
| 03:49.24 | gurur | they even wrote codes for AMD architecture of the DSP codes |
| 03:49.31 | gurur | they even wrote codes for AMD architecture of the DSP. |
| 03:49.45 | gurur | is there a tutorial to do that ? |
| 03:49.56 | gurur | or some plugins that i can use ? |
| 03:49.58 | WIMPy | Asterisk has an AMD dialplan app. |
| 03:50.05 | gurur | i use intel. |
| 03:50.30 | gurur | WIMPy: skype is spuuported also ? |
| 03:50.51 | WIMPy | I think it's still supported. |
| 03:50.52 | *** join/#asterisk SaidKLE (~stupid@174-126-42-224.cpe.cableone.net) |
| 03:51.05 | WIMPy | No idea if te M$ deal might change their policy. |
| 03:51.06 | gurur | i wish i can connect my gsm phone with them. because gsm providers provide about 5000 minutes for about $50 |
| 03:51.34 | WIMPy | So we're back to GSM gateways? |
| 03:51.52 | gurur | WIMPy: switchvox is some modules on top of asteriks right ? |
| 03:52.04 | gurur | WIMPy: i dont have any software/hardware gateway |
| 03:52.12 | SaidKLE | Question: no matter what port I specify in sip.conf, asterisk still looks for my softphone on 5060 |
| 03:52.32 | WIMPy | I have heard about switchvox, but no idea what it is. |
| 03:52.38 | SaidKLE | How do I debug that? |
| 03:52.43 | gurur | WIMPy: i just need the whole system for my own case. and if i can do it and like it, i would be thinking aobut making some money for configuring it for oher small biz :) |
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| 03:53.46 | WIMPy | I'd suggest you get the book, get the code, and start experimenting. |
| 03:53.48 | WIMPy | ~thebook |
| 03:53.49 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
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| 03:54.51 | kaldemar | SaidKLE: how did you specify the port? |
| 03:55.14 | WIMPy | SaidKLE: And does the phone register with Asterisk? |
| 03:55.37 | gurur | WIMPy: thanks for book info. |
| 03:55.59 | gurur | WIMPy: what do you guys can recommend me to make money with consulting services for asterisks to small bizs. |
| 03:56.16 | WIMPy | Get a lot of experience first. |
| 03:56.38 | gurur | yep i know that. but i dont want to get experienced with all of them :) dont have too much time. just the basics for now. |
| 03:56.40 | WIMPy | I think you will get the idea if you use and understand it. |
| 03:57.22 | gurur | WIMPy: ok i will sudy http://ofps.oreilly.com/titles/9780596517342/, |
| 03:57.24 | WIMPy | I don't think you will be able to offer any value with basics. |
| 03:57.37 | gurur | WIMPy: considering small bizs, i might be. |
| 03:58.14 | WIMPy | If you don't want any fancy stuff, you might be better off with a classic PBX. |
| 03:58.41 | gurur | WIMPy: pbx what is that exactly ? a special analog line ? |
| 03:58.45 | gurur | other than ptsn i mean |
| 03:59.12 | WIMPy | You in-house telephone exchange. |
| 03:59.17 | WIMPy | Do we have |
| 03:59.22 | WIMPy | ~pbx |
| 03:59.22 | infobot | well, pbx is a Private Branch eXchange |
| 03:59.58 | WIMPy | Good bot :-) |
| 04:01.53 | gurur | so skype is itsn |
| 04:02.07 | gurur | and google provides free sip right ? |
| 04:02.28 | WIMPy | itsp, yes |
| 04:02.39 | WIMPy | Not sure, what google offers ATM. |
| 04:02.48 | gurur | so what do you use for sip ? |
| 04:02.59 | WIMPy | What? |
| 04:04.19 | gurur | where do you get your sip ? |
| 04:04.24 | gurur | service |
| 04:04.33 | gurur | or you create one with asterisks yourself |
| 04:04.42 | WIMPy | That will obviousely depend on what countries you're interested in. |
| 04:04.54 | WIMPy | You can do that, yes. |
| 04:06.11 | WIMPy | Asterisk will interface with mist things, both real and virtual. |
| 04:06.26 | WIMPy | most |
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| 04:10.29 | gurur | hmm i installed |
| 04:10.35 | gurur | voicent |
| 04:10.38 | gurur | and trying it . |
| 04:10.49 | gurur | it is cool. |
| 04:11.04 | gurur | please tell me, i got only my laptop , skype acct with me. |
| 04:11.07 | gurur | can i use asterisks ? |
| 04:11.17 | gurur | i dont have any hardware |
| 04:11.31 | WIMPy | Skype support for Asterisk is commercial. |
| 04:11.39 | WIMPy | ~sfa |
| 04:12.03 | WIMPy | ~skype |
| 04:12.03 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
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| 04:19.50 | gurur | i cant install asterisk for windows |
| 04:19.56 | gurur | it is only for linux i think |
| 04:21.07 | WIMPy | There have been Windows builds, but it's been a while since I heard of anybody trying that. |
| 04:21.47 | WIMPy | But other unixoids operatings systems will also do. |
| 04:22.22 | WIMPy | But hardware support is best on Linux. |
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| 04:33.19 | carrar | or are they ketp on gox |
| 04:33.22 | carrar | err |
| 04:36.43 | Kobaz | fixed another bug in 1.8 |
| 04:44.32 | Kobaz | https://issues.asterisk.org/view.php?id=19346 |
| 04:44.35 | Kobaz | if anyone wants a patch |
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| 05:23.42 | atan | What short term would you use to describe software that handles inbound calls with reps? |
| 05:23.53 | atan | Other than 'call centre software' that is =\ Is there a term for this stuff? |
| 05:24.12 | WIMPy | ACD |
| 05:24.38 | atan | Hmm. Nice. Thanks! |
| 05:25.24 | ChannelZ | India |
| 05:25.35 | WIMPy | LOL |
| 05:25.46 | atan | WIMPy, would there be another term used to describe website-based ACD? |
| 05:27.13 | atan | Oh nice, ACD is also Association of Canadian Distillers :D |
| 05:28.21 | WIMPy | Web? Like frontend? |
| 05:29.57 | WIMPy | A web is something that should never be seen in an orderly house. |
| 05:33.43 | atan | WIMPy, like a hosted call centre. |
| 05:34.50 | WIMPy | Are you more like looking for some sort of CRM with telephony integration? |
| 05:36.07 | atan | Actually I'm working on a website-based portal for a client's reps to take phone calls but I can't think of a neat name for it. |
| 05:36.38 | atan | Reps log in, wait for calls. They get passed out to everyone logged in and that's about that. |
| 05:37.08 | atan | Maybe CRM is what I'm after. I suppose. Seems about right, and call the phone part the ACD. |
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| 06:30.18 | schmidts | good morning |
| 06:32.32 | WiretapSeven | atan, it can also be 'crm' |
| 06:33.13 | WiretapSeven | but I saw you realised that :P |
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| 06:54.06 | wdoekes2 | good morning |
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| 07:12.07 | jacc0 | hi all |
| 07:12.11 | jacc0 | good morning |
| 07:12.29 | jacc0 | I've got mail: "Your license has been formally accepted by our legal department" |
| 07:12.30 | jacc0 | :) |
| 07:13.30 | schmidts | jacc0 morning and congrats ;) |
| 07:15.20 | WiretapSeven | jacc0, whatchalicensing? |
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| 07:18.01 | jacc0 | you need it to be able to post patches |
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| 07:18.39 | jacc0 | I just returned from a week volunteering at Hitb :) |
| 07:19.40 | jacc0 | it ended with a big party on 3 boots payed form bij Microsoft; don't you just love them? |
| 07:19.43 | jacc0 | :p |
| 07:19.53 | jacc0 | *for by |
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| 07:25.25 | WiretapSeven | ah |
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| 07:51.11 | wdoekes2 | *boats, *paid ;) |
| 07:51.15 | jacc0 | yews |
| 07:51.18 | jacc0 | *yes |
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| 08:35.51 | kwk | Hi there. Where can I find the MySQL schema for the CEL table? |
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| 09:27.06 | Nunners | Quick one folks.... I'm trying to remove the queue/onhold music, so that it just rings (or plays a ring). It's so long ago since I did this, I can't remember how I got the music on in the first place, therefore how you remove it ~) Any clues... main config: http://pastebin.com/r6dNDRJB |
| 09:29.32 | wdoekes2 | Nunners: how about pointing that musiconhold directory= to a dir with only a beep-file? |
| 09:30.36 | kaldemar | Nunners: option r or R for app Queue |
| 09:31.02 | kaldemar | Nunners: core show application Queue will tell you more |
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| 09:33.01 | Nunners | kaldemar-> thanks - exactly what I needed.... |
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| 09:51.00 | Drgb_ | hello, I'm trying to send a facsimile using hylafax->iaxmodem->asterisk through an ISDN trunk. I can make/receive calls through the trunk, but calls initiated by the virtual fax seems not to get forwarded to their right destination. There's only a single outbound route configured, and it works. Every call sticks at "DAHDI/i2/somenumber-19 is proceeding passing it to IAX2/765-2199" some seconds then fails reporting either error |
| 09:51.00 | Drgb_ | 3 or error 38. I looked up the meanings of those error codes, they're respectively "no route to destination" and "network out of order". Maybe this is not the right place to ask such a question but I can assume that hylafax and iaxmodem are configured correctly and I'm doing something wrong with the link between iaxmodem and asterisk. |
| 09:55.11 | cneb3000 | Hey Drgb_ |
| 09:55.22 | cneb3000 | while I'm not familiar with your setup exactly... |
| 09:55.54 | cneb3000 | 'no route at destination' normally means what you're sending the invite to (in this case, the 'iax2/765-2199) |
| 09:55.59 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:56.08 | cneb3000 | does not have a route set up to forward it on, and thus ignores it and cancels the call |
| 09:56.36 | cneb3000 | failing that, make a call and copy the logs to pastebin :) |
| 09:56.51 | Drgb_ | ok, just a second |
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| 10:01.16 | Drgb_ | http://pastebin.us/4777 |
| 10:02.30 | Drgb_ | 079957WXYZ is the destination and 079994WXYZ is the source |
| 10:02.54 | cneb3000 | Again, not massively helpful because it's not a setup i'm familar with.. but |
| 10:03.01 | cneb3000 | <PROTECTED> |
| 10:03.01 | cneb3000 | <PROTECTED> |
| 10:03.19 | cneb3000 | normally means the route isn't correct on dest.. in this case the IAX |
| 10:03.28 | WiretapMac | Drgb_: go ask in #freepbx, your problem is going to be freepbx related |
| 10:04.23 | Drgb_ | WiretapMac, could you explain me how freepbx should affect my situation please? Maybe it could be the reason of the failure |
| 10:04.40 | WiretapMac | Drgb_: well, your config looks like it was generated by freepbx |
| 10:04.46 | WiretapMac | so your problem may likely be related to it |
| 10:04.57 | WiretapMac | and I can guarantee that it will just anger those not familiar with freepbx in here |
| 10:04.59 | Drgb_ | yes WiretapMac but I tried manually, obtaining the same result |
| 10:05.16 | WiretapMac | Drgb_: you never edit config files manually with freepbx installed |
| 10:05.27 | WiretapMac | not unless you have a situation freepbx doesn't cover |
| 10:05.28 | Drgb_ | actually I did it before installing it |
| 10:05.42 | Drgb_ | and some config files are left to be edited manually |
| 10:06.36 | Drgb_ | cneb3000, I don't understand exactly what you mean |
| 10:06.51 | Drgb_ | who's the responsible for "routing" the call at that point? |
| 10:07.02 | cneb3000 | Drgb |
| 10:07.23 | cneb3000 | if I rotue a call from asterisk to a voip switch, and the voip switch has no rule to forward the call to a carrier, it will pass that same error |
| 10:07.40 | cneb3000 | not saying it's the problem. but may point you in a direction :) |
| 10:07.57 | cneb3000 | WiretapMac is also correct though. Might be worth talking to #freepbx |
| 10:07.59 | Drgb_ | so it's iaxmodem saying to asterisk "hey, you passed me this call but I don't know what to do with it"? |
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| 10:08.06 | cneb3000 | ^^^ correct.. possibly :) |
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| 10:08.16 | Drgb_ | okay, thank you very much |
| 10:08.21 | cneb3000 | do you have IAXmodem logs? |
| 10:08.25 | cneb3000 | take a look at them ;) |
| 10:09.00 | Drgb_ | hmm, that's something I haven't tried, yet. dumb choice, Drgb_, dumb choice. |
| 10:09.10 | cneb3000 | :D |
| 10:10.51 | kaldemar | Drgb_: whatever is behind DAHDI/g12, responded with cause code 3 to 079957WXYZ. is the dialed number valid? |
| 10:12.04 | Drgb_ | yes, it is, it's my analog fax, and I tried with other destinations as well, such as my cellphone, just in order to make it ring and check if the call was actually going out or not |
| 10:12.14 | Drgb_ | with the same results |
| 10:13.50 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 10:14.15 | *** join/#asterisk luke0512 (~eric@HSI-KBW-091-089-022-230.hsi2.kabelbw.de) |
| 10:14.31 | luke0512 | hello |
| 10:15.33 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:16.27 | luke0512 | i need some help for configuring asterisk 1.6.1.18 on fc12 |
| 10:18.11 | luke0512 | my problem is tried the voicemailbox on fc14 which delivers asterisk 1.6.2.18 and everything works, even german sounds |
| 10:18.25 | luke0512 | the fc14 installation is on virtualbox |
| 10:18.51 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:18.58 | *** join/#asterisk yuvankumar (~quassel@175.139.158.145) |
| 10:19.00 | luke0512 | my server works with fc12 and i made the installation with yum but i always get the rerror |
| 10:19.25 | luke0512 | [May 23 12:06:22] WARNING[2970] pbx.c: No application 'VoiceMailMain' for extension (default, 2999, 1) |
| 10:20.03 | jacc0 | look in extensions.conf under [default] line 2999 |
| 10:20.22 | jacc0 | sorry extension 2999, line 1 |
| 10:21.22 | jacc0 | the application voicemailmain is not available |
| 10:22.02 | cneb3000 | luke0512: Copy and paste the dialplan on pastebin and lets take a look. |
| 10:23.34 | luke0512 | extensions.conf --> http://fpaste.org/E2Ks/ |
| 10:24.06 | luke0512 | i'm an absolut beginner on asterisk and started on working with the book from stefan wintermeyer |
| 10:24.46 | luke0512 | dialin from one to the other client/soffon is working |
| 10:24.53 | luke0512 | +t |
| 10:27.39 | cneb3000 | hmmm |
| 10:28.23 | kaldemar | luke0512: you don't seem to have the VoiceMailMain dialplan application. does 'module show like voice' in CLI list app_voicemail.so? |
| 10:30.02 | luke0512 | Module Description Use Count |
| 10:30.03 | luke0512 | app_voicemail_plain.so Comedian Mail (Voicemail System) 0 |
| 10:30.03 | luke0512 | 1 modules loaded |
| 10:30.16 | luke0512 | only this |
| 10:31.51 | luke0512 | on the vm there's also only that one loaded and the mailbox works |
| 10:33.20 | kaldemar | if not, what does 'module load app_voicemail.so' tell you? |
| 10:34.41 | luke0512 | kaldemar, nothing |
| 10:34.59 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 10:35.26 | luke0512 | that looks strange |
| 10:36.47 | WiretapMac | luke0512: out of interest, what does your modules.conf look like? |
| 10:36.58 | luke0512 | mom |
| 10:37.30 | kaldemar | if there is no module, you should get a warning |
| 10:38.33 | luke0512 | http://fpaste.org/7nRK/ <-- module.conf |
| 10:39.44 | WiretapMac | okay, that looks healthy |
| 10:40.03 | kaldemar | luke0512: do you have app_voicemail.so in /usr/lib/asterisk/modules ? |
| 10:40.46 | kaldemar | or whatever is configured as astmoddir in asterisk.conf |
| 10:44.59 | luke0512 | http://fpaste.org/ndME/ , both systems have same app installed |
| 10:46.37 | luke0512 | both starisk.conf are the same, even differ on lib / lib64, because the server has 64bit installation |
| 10:47.06 | luke0512 | ups asterisk.conf |
| 10:47.27 | skrusty | morning |
| 10:47.40 | luke0512 | hi skrusty |
| 10:47.56 | cneb3000 | luke0512: Looks like your problem is your dont have app_voicemail.so in the /modules/ directory. |
| 10:48.35 | luke0512 | hmm...but the the vm installation the file is missing too but the mailbox is working |
| 10:51.51 | luke0512 | http://fpaste.org/rJSj/ |
| 10:53.04 | cneb3000 | wont hurt to put app_voicemail.so /modules/ and see if it helps ;) |
| 10:54.13 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 10:57.47 | luke0512 | hmmm...therefore i have to look out for the module, cause i installed asterisk via yum and not from source |
| 11:02.46 | luke0512 | hmmm very very strange |
| 11:10.31 | luke0512 | hmm...looks like theres no app_voicemail.so sinlge file/module available |
| 11:10.35 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 11:12.29 | luke0512 | and cause its working on the 32bit version without the app i think it have to work on 64bit system too |
| 11:12.51 | cneb3000 | ahh sorry.. missed that. |
| 11:13.09 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 11:13.41 | luke0512 | but sometimes the ways are unbelivable/unexplainable |
| 11:13.55 | k-man | how does one quit from the console? |
| 11:13.56 | cneb3000 | hmmmm... |
| 11:14.03 | *** part/#asterisk Aas (~aas@static-dsl-88.87-197-156.telecom.sk) |
| 11:14.21 | cneb3000 | So just to be clear... You have EXACTLY the same VM snapshot running on a 32 and 64 bit machine? |
| 11:14.24 | cneb3000 | or something like that |
| 11:14.36 | cneb3000 | the main thing, is both systems have exactly the same config and files one the 32 and 64 bit machine |
| 11:15.19 | k-man | quit and exit both give me No such command |
| 11:15.28 | luke0512 | no the 32 bit version is on my client here, only for testing...the 64bit is my server activ system |
| 11:15.51 | cneb3000 | but they both have the same files on? |
| 11:16.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 11:16.59 | luke0512 | the 32bit is fc14 and the server is running 64bit fc12, the directories look identical |
| 11:17.29 | k-man | oh, nm, worked it out |
| 11:17.37 | luke0512 | therefore the version of asterisk is different |
| 11:18.10 | luke0512 | but both are main version 1.6 |
| 11:19.01 | cneb3000 | Hmm sorry buddy... VoiceMailMain uses the app_voicemail.so. If it's not there it wont work. How you've got other modules running I'm not sure. |
| 11:19.05 | cneb3000 | Sorry I cant be of any more help./ |
| 11:19.11 | kaldemar | k-man: core stop now |
| 11:19.26 | kaldemar | k-man: quit and exit only work when you attach with -r |
| 11:20.04 | kaldemar | actually app_voicemail.so provides VoiceMailMain. |
| 11:20.49 | luke0512 | kaldemar, but on both systems the file is missing and on the 32bit system the mailbox is working |
| 11:20.50 | k-man | kaldemar: yeah, i realised that |
| 11:20.53 | k-man | thanks kaldemar |
| 11:21.15 | cneb3000 | kaldemar: that's what I said :) |
| 11:36.00 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 11:37.35 | k-man | i'm trying to add a new peer so i can connect from a softphone on my computer, but i get this error: Registration from '"woodhouse" <sip:192.168.1.2>' failed for '192.168.1.144' - No matching peer found |
| 11:38.50 | kaldemar | k-man: see "Naming devices" in http://svn.digium.com/svn/asterisk/tags/1.8.4/configs/sip.conf.sample |
| 11:45.41 | k-man | hmm |
| 11:46.39 | k-man | this is the section in question: http://pastebin.ca/2067541 |
| 11:47.01 | k-man | i set my username to woodhouse in the sip client, but still get that error |
| 11:48.14 | k-man | Registration from '<sip:192.168.1.2>' failed for '192.168.1.144' - No matching peer found |
| 11:48.25 | k-man | clearly i'm not understanding something |
| 11:50.15 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:51.55 | *** join/#asterisk coppice (~chatzilla@m121-202-50-32.smartone-vodafone.com) |
| 11:52.20 | luke0512 | the VoiceMailMain on the 32bit system works and on 64bit system not core show applications tells me |
| 11:53.40 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 11:53.44 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:55.31 | cneb3000 | k-man: change type=user to type=friend |
| 11:55.41 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 11:55.45 | cneb3000 | k-main: reload.. then try again |
| 11:58.40 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 12:00.18 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
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| 12:03.30 | kaldemar | k-man: Registration from '<sip:192.168.1.2>' <-- this looks strange, it should be <sip:woodhouse@192.168.1.2>. take a look at the client configuration. |
| 12:10.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:10.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:13.57 | luke0512 | hmmm...i looked out for applications installed on both system...32bit=175 /64bit=153 so here is the different |
| 12:14.12 | luke0512 | this means appps for asterisk |
| 12:14.55 | luke0512 | this maybe an issue with an old version of fedora...so i have to upgrade the server |
| 12:15.15 | cneb3000 | if it's a production test box - should be a duplicate of the other once really. |
| 12:15.24 | cneb3000 | is the production box a vm box? |
| 12:15.43 | cneb3000 | oh wait.. the 64/32bit thing |
| 12:15.45 | cneb3000 | doh |
| 12:16.16 | leifmadsen | if you're using yum, it's possible when installing things that it installs both the 32 bit and 64-bit because yum isn't great at not installing both. You could try, "yum remove *.i386" and see what it returns (make sure you press N to not remove the i386 files, especially if you're not a 64-bit OS) |
| 12:16.23 | luke0512 | the 64bit system is the production system |
| 12:16.42 | leifmadsen | although I wasn't here when you stated the actual problem |
| 12:17.03 | cneb3000 | all hail LeifMadsen |
| 12:19.35 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 12:19.35 | luke0512 | hmmm....i think i will go on with installing/configuring and give him a try after i installed centos 6 on my server |
| 12:20.26 | luke0512 | centos 5 doesn't work for me because it's kernel is to old my dvb-c card |
| 12:21.02 | kaldemar | leifmadsen: missing app_voicemail.so => missing VoiceMailMain |
| 12:21.05 | luke0512 | and an upgrade from f12 to f14 looks dangerous to me because i have an software raid installed |
| 12:21.08 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 12:21.48 | luke0512 | leifmadsen, on 32bit VoiceMail is working without app_voicemail.so |
| 12:22.00 | luke0512 | <PROTECTED> |
| 12:24.52 | k-man | kaldemar: thanks, tried that and still get the same error - i'm using this sip client called Telephone for osx - maybe its dodgey |
| 12:25.44 | kaldemar | k-man: try blink |
| 12:28.04 | k-man | kaldemar: ok, will do |
| 12:28.06 | kaldemar | k-man: http://icanblink.com/ |
| 12:32.14 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:38.10 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 12:38.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:40.14 | *** join/#asterisk wonderworld (~ww@port-92-201-214-69.dynamic.qsc.de) |
| 12:42.08 | k-man | it won't let me have no password for my sip account |
| 12:44.11 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 12:44.29 | *** join/#asterisk bazhang (~bazhang@unaffiliated/bazhang) |
| 12:45.06 | *** part/#asterisk bazhang (~bazhang@unaffiliated/bazhang) |
| 12:46.21 | Cadey | Hi guys, can anyone assist me in the seemingly long story of Xfer and CallerIDs :) |
| 12:46.45 | Cadey | We would like the phones to see the callerID info once the tranfer has taken place |
| 12:49.09 | *** join/#asterisk morningcoffee (~chris@dsl-216-227-10-124.pivot.net) |
| 12:49.19 | leifmadsen | Cadey: what version of asterisk? |
| 12:49.26 | Cadey | 1.8 |
| 12:49.36 | leifmadsen | because I have a 1.8 deployment and it's just works automatically |
| 12:49.38 | leifmadsen | I didn't do anything |
| 12:50.07 | leifmadsen | although I always enable trustrpid=yes and sendrpid=yes (which is not enabled by default) |
| 12:50.07 | leifmadsen | I'm using polycom devices |
| 12:50.15 | Cadey | hum ill try those settings |
| 12:50.22 | Cadey | in sip.conf I take it? |
| 12:50.24 | leifmadsen | after a transfer, the CID just updatees |
| 12:50.27 | leifmadsen | yes sip.conf |
| 12:50.47 | leifmadsen | beyond that there is nothing fancy going on in my sip.conf file or dialplan to specifically enable post-transfer CID updates |
| 12:52.10 | Cadey | kk thanks :D |
| 12:52.16 | Cadey | im going to test those settings |
| 12:52.40 | Cadey | I have another question but ill ask that in a second :) |
| 12:53.12 | Cadey | hum |
| 12:53.38 | Cadey | that didnt seem to work. leifmadsen, are you using the asterisk Xfer or a Xfer started from the polycom? |
| 12:53.46 | leifmadsen | transfer on the polycom device |
| 12:53.50 | Cadey | ok |
| 12:53.59 | Cadey | did you put those settings under general or on the phone? |
| 12:54.09 | leifmadsen | those are general settings |
| 12:54.16 | leifmadsen | they are not peer level settings |
| 12:54.30 | Cadey | :( |
| 12:54.40 | Cadey | does not seem to work on the 57i phones |
| 12:54.44 | Cadey | Aastra |
| 12:54.48 | *** join/#asterisk morningcoffee (~chris@dsl-216-227-10-124.pivot.net) |
| 12:54.50 | leifmadsen | ya I don't use aastra phones |
| 12:55.25 | morningcoffee | Good morning.. I have a noob question...is this the correct place to ask? |
| 12:57.10 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:57.52 | leifmadsen | morningcoffee: yes |
| 12:58.08 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 12:58.48 | morningcoffee | ty. I would like to play around w/astersik in my home. I have 3 lines, 2 voip, and 1 pots. I know I would need an ATA type device to energize my home pots lines, but Can I use a modem for the pots line in on the asterisk box? |
| 12:59.17 | leifmadsen | morningcoffee: you cannot |
| 12:59.35 | leifmadsen | unless you have the skills to write low level drivers for the device |
| 12:59.49 | morningcoffee | yah..no i don't ;) what would i need to get? |
| 12:59.50 | leifmadsen | spend the $50 on the ATA and be done with it |
| 13:00.14 | leifmadsen | get a SIP ATA from someone like Linksys |
| 13:00.17 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 13:00.33 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 13:00.50 | morningcoffee | i was looking into getting one of those for the ability to use my existing phones in my home...but that will do both? |
| 13:00.51 | leifmadsen | morningcoffee: and you don't "energize" your home POTS line, as that line will already be providing voltage, so it would accept the "energization" |
| 13:01.04 | morningcoffee | ok.. |
| 13:01.09 | leifmadsen | morningcoffee: if you get something like a SPA3102 then yes, you have both FXS and FXO |
| 13:01.24 | leifmadsen | FXS == station (phones) and FXO == office (phone line) |
| 13:01.34 | morningcoffee | oh..nice..what i was looking at getting... |
| 13:01.36 | morningcoffee | perfect.. |
| 13:01.49 | leifmadsen | FXS provides voltage for the phones, and FXO accepts voltage from the CO (central office) |
| 13:02.14 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:02.16 | morningcoffee | thank you.. told you stupid easy..dumb question... |
| 13:02.29 | morningcoffee | thank you for your help!! you all rock for the support you provide |
| 13:02.38 | MariusAgon | how to get as smart as leifmadsen is? O_o |
| 13:03.46 | leifmadsen | MariusAgon: read lots, and spend over 7 years with asterisk full time :) |
| 13:03.56 | leifmadsen | I've been doing this since about 2002 |
| 13:04.04 | jacc0 | ~thebook |
| 13:04.04 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:04.06 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:04.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:04.08 | jacc0 | :P |
| 13:04.13 | *** join/#asterisk serafie (~erin@nat/digium/x-jqrbhxsjwmqvdidh) |
| 13:04.16 | leifmadsen | and there is probably more I don't know, than I do know |
| 13:04.19 | morningcoffee | thank you again.. back to waiting for amazon to deliver. |
| 13:04.34 | leifmadsen | you can read it online for free right now though |
| 13:04.39 | leifmadsen | burn |
| 13:04.47 | MariusAgon | you won't know everything, but u have to try anyways |
| 13:05.09 | leifmadsen | I don't *have* to do anything :) |
| 13:05.35 | MariusAgon | I was just thinking loud :) |
| 13:05.46 | leifmadsen | I'm just kidding around |
| 13:06.11 | MariusAgon | yeah, I know :) |
| 13:07.18 | luke0512 | where can i get a howto to configure a system : connect asterisk to an internal S0 ISDN behind a GTB Allegro? |
| 13:07.40 | luke0512 | i have an internal ISDn Card with HFC-S chip |
| 13:07.58 | luke0512 | Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
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| 13:11.58 | *** join/#asterisk irroot (~gregory@dsl-185-161-93.dynamic.wa.co.za) |
| 13:12.15 | Cadey | @leifmadsen : I cant seem to get this working :( |
| 13:12.31 | irroot | afternoon ladies |
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| 13:14.14 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:14.32 | leifmadsen | Cadey: I have no further suggestions -- I'd look at the aastra manual to determine if there is something it needs enabled |
| 13:17.02 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
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| 13:22.49 | Superstar | Is it possible to manually edit the music on hold order? I've tried renaming the files to s00001, s00002 etc but it still seems to have it's own order figured out |
| 13:24.10 | Superstar | I've checked the MySQL databases for filenames with no luck either |
| 13:25.20 | JonathanRose | sort=alpha |
| 13:25.26 | JonathanRose | in musiconhold.conf |
| 13:25.34 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:25.47 | JonathanRose | There's in an example with the included config files. |
| 13:25.57 | JonathanRose | *there is an example |
| 13:26.22 | leifmadsen | JonathanRose: if I understand correctly, he is using MySQL realtime MOH |
| 13:26.39 | leifmadsen | in which case, my guess is the order in which the rows are returned makes a diff |
| 13:26.55 | Superstar | sort=alpha doesn't work |
| 13:27.00 | mazpe | anyone running asterisk on linode? preferably Debian? |
| 13:27.13 | leifmadsen | mazpe: many people are |
| 13:27.16 | JonathanRose | Oh, I guess that changes things. |
| 13:27.21 | Superstar | I checked mysql database to see if it stores music on hold data in there, but it doesn't |
| 13:27.32 | JonathanRose | Oh, then maybe it doesn't. |
| 13:27.42 | mazpe | seems that CONFIG_HZ at 250 needs to be change to 1000 and Dahdi needs to be installed |
| 13:28.27 | JonathanRose | Honestly, I didn't even know you could do MOH with db. |
| 13:29.07 | wdoekes2 | Superstar: use a view with order-by? |
| 13:29.09 | JonathanRose | Actually, are you using a custom player? |
| 13:29.19 | JonathanRose | If you are, that might dictate the order I guess. |
| 13:29.28 | leifmadsen | mazpe: then you need to recompile the kernel |
| 13:29.45 | mazpe | leifmadsen: i thought that linode didnt give you access to the kernel |
| 13:30.15 | leifmadsen | mazpe: that is entirely possible in which case you can't use DAHDI |
| 13:32.53 | mazpe | I tried using ConfBridge, but for a particular application i need all callers to be muted all the time, exepct the Admin. |
| 13:33.25 | mazpe | ConfBridge mutes everyone initially and when the Admin/Marked user logs in, it unmutes everyone |
| 13:35.38 | *** join/#asterisk jmpiloq (~jmpiloq@static-87-79-94-28.netcologne.de) |
| 13:40.40 | Superstar | I'm receiving many lines of errors "WARNING[5351] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line xxx" (xxx = increments of lines) |
| 13:41.07 | Superstar | Any ideas how to stop this without changing the log level? |
| 13:41.30 | *** join/#asterisk guilhermebr (~Guilherme@187.58.192.161) |
| 13:41.43 | pabelanger | Superstar: change your dialplan? |
| 13:41.52 | pabelanger | s/_./_X/ |
| 13:41.56 | Superstar | pabelanger: this is off a fresh install of Asterisk |
| 13:42.07 | jacc0 | _X. can also have unexpected behavior |
| 13:42.12 | Superstar | with no trunks, extensions, queues |
| 13:42.29 | pabelanger | Superstar: which version of Asterisk? |
| 13:42.34 | jacc0 | I would also strongly discourage using it |
| 13:43.03 | jacc0 | consider a phone calling to extension 100&sip/200 |
| 13:43.09 | Superstar | pabelanger: I'm using Asterisk 1.4.21.2 |
| 13:43.11 | pabelanger | I'd be very surprised if extensions.conf.samples contain that |
| 13:43.13 | irroot | got a nasty deadlock on a busy site in ast_channel_cmp_cb see r320499 |
| 13:43.30 | Superstar | pabelanger: actually yes there may be samples causing this. thanks i'll have a look |
| 13:44.41 | luke0512 | no solutions or tips? |
| 13:45.56 | pabelanger | Superstar: Fresh install of Asterisk 1.4.21.2 or an upgrade? Are you using only Asterisk? Just checked our sample configs, '_.' does not exists |
| 13:48.40 | k-man | what do you have to reload after modifying extensions.conf? |
| 13:49.04 | WIMPy | k-man: 'dialplan reload' |
| 13:49.16 | k-man | kaldemar: blink works like a charm thanks - the problem i was having must have been the soft phone i was using |
| 13:49.25 | k-man | thanks WIMPy |
| 13:52.51 | irroot | #19349 with patch add deadlock avoidance to ast_channel_cmp_cb not sure this is a good idea here but it has been locking from time to time |
| 13:53.24 | irroot | oops patch got problem |
| 13:54.43 | jaytee | I've got voicemail attachments working with my Outlook account but when I get push notifications to my LG android phone I try to open them and it never opens, just keeps trying to "fetch" the attachment. Anyone ever run into this? |
| 13:56.08 | Superstar | pabelanger: i used the tarball off asterisks website |
| 13:58.33 | cneb3000 | jaytee: the call is literally attatched to the email. so maybe it's the phone? |
| 13:58.43 | cneb3000 | jaytee: can you open the call on the PC? |
| 13:58.46 | Superstar | I've added a few trunks and an extension, warnings are still there but calls are working. I'm getting warnings for my trunks "WARNING[5351] chan_sip.c: Section 'ArosaSwitzerland' lacks type" but I have specified type=friend? |
| 14:01.53 | jaytee | cneb3000, yep opens fine on the PC |
| 14:02.04 | jaytee | gotta be the phone |
| 14:03.25 | cneb3000 | .... get an iPhone :D:D:D |
| 14:03.58 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:04.54 | k-man | is there any sip clients for the iphone that are free? |
| 14:05.22 | *** join/#asterisk lcat (~lcat@187.45.253.214) |
| 14:06.20 | kaldemar | Superstar: feel free to pastebin your configs. |
| 14:06.42 | cneb3000 | k-man: I dont use it regularly... but I've used '3CX' before. Does the job well. |
| 14:06.51 | cneb3000 | did the job well* |
| 14:07.24 | k-man | thanks cneb3000 |
| 14:08.16 | cneb3000 | so.. who administers Asterisk boxes in a live environment :) |
| 14:08.58 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:08.58 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:09.00 | cneb3000 | what kind of set ups do you have? I have a load balanced cluster of 3 boxes polling static config via ARA on active passive MySQL setup. |
| 14:09.06 | cneb3000 | gets excited. |
| 14:10.32 | puzzled | cneb3000: you seem to be the only one :) |
| 14:10.40 | cneb3000 | I thought as much ¬_¬ |
| 14:11.37 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 14:11.52 | leifmadsen | I have several different setups, some use astdb so they don't rely on anything external, and others use dynamic realtime using MySQL master-master replication |
| 14:11.55 | luke0512 | anyone can help me with my configuration...i have isdn with telephone system that has 1 internal S0 and 4 analog ports |
| 14:12.36 | luke0512 | the internal S0 is available with internal number 46 and i want this for use of asterisk |
| 14:12.43 | WIMPy | luke0512: Where is the Lin to Asterisk? |
| 14:12.53 | WIMPy | ah |
| 14:13.02 | irroot | cneb3000 got over 100 active systems out there |
| 14:13.16 | cneb3000 | leifmadsen, irroot: I feel a little small all of a sudden. |
| 14:13.32 | luke0512 | for connection i have an internal isdn card with cologne chip (HFC-S) |
| 14:13.49 | cneb3000 | irroot: are they all indipendant boxes? |
| 14:14.42 | irroot | have a call gateweay running at a datacentre that has 3servers involved |
| 14:14.58 | irroot | i use PgSQL not MySQL |
| 14:15.08 | luke0512 | as i read i don't need an hfc chip but now i have one and don't know how to connect it with the telephone system |
| 14:15.11 | cneb3000 | Why PgSQL?.. |
| 14:15.17 | cneb3000 | dont really know much about it |
| 14:15.55 | WIMPy | luke0512: If you don't know where to start, here are you options: http://voice.yeti.dk/Asterisk_vs_ISDN |
| 14:16.02 | WIMPy | +r |
| 14:16.04 | irroot | started using it in early 00's had some nice features not in My yet |
| 14:16.56 | k-man | cneb3000: yeah 3ch seems pretty good |
| 14:20.46 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:23.09 | luke0512 | hmm...not even easier |
| 14:25.18 | luke0512 | ok as i can see i want an TE mode because the isdn access is delivered by the internal S0 bus |
| 14:25.51 | WIMPy | correct |
| 14:28.07 | luke0512 | ok...next step i think i should use dahdi because capi may not work |
| 14:28.31 | WIMPy | capi? Did you read the link I gave you? |
| 14:28.35 | luke0512 | with my card |
| 14:29.07 | WIMPy | You can use LCR, dahdi or misn. |
| 14:29.22 | irroot | luke0512 using mISDN is a option |
| 14:29.44 | irroot | LCR is not quite upto spec |
| 14:29.58 | WIMPy | What spec? |
| 14:30.13 | irroot | its not as stable as mISDN |
| 14:30.27 | irroot | everytime i try it in production it breaks |
| 14:30.30 | luke0512 | i think i will use dahdi |
| 14:30.31 | WIMPy | I found it much more stable. |
| 14:30.49 | irroot | maybe we should chat sometime WIMpy ;) |
| 14:31.03 | WIMPy | Feel free to do so :-) |
| 14:31.24 | Superstar | Can anyone help me suppress these errors please? http://codepad.org/RNaEAHcG |
| 14:31.29 | luke0512 | this maybe the reason why it is in the rpm of fedora |
| 14:31.55 | WIMPy | luke0512: Do those RPMs also include HFC-S support? |
| 14:32.09 | Superstar | The MySQL error is there because I've disabled networking but it re establishes a connection through sockets |
| 14:32.28 | irroot | WIMPy my favorite is using USB ISDN |
| 14:32.38 | puzzled | Superstar: replace _. with _X as it suggests? |
| 14:33.09 | puzzled | irroot: did you get those to work? I have 2 billion HFC-S based usb ones. haven't tried them yet |
| 14:33.19 | luke0512 | i don't know WIMPy |
| 14:33.44 | WIMPy | irroot: I do so for testing, but I've been told the USB drivers might indeed not be the most stable. |
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| 14:33.52 | Superstar | puzzled: Sorry, after that. The part of "tries to include non existent context" |
| 14:34.02 | irroot | use the billion lots with mISDN 1 |
| 14:34.10 | WIMPy | puzzled: They will do. |
| 14:34.29 | irroot | WIMpy indeed but for small sites with 2 lines cost effective and most acceptable |
| 14:34.50 | puzzled | wonder if adding the PCI ID of my HFC-S single BRI card to the wcb4xxp will make the card work with latest dahdi |
| 14:35.15 | WIMPy | irroot: Do you use things without PCI slot? |
| 14:35.24 | puzzled | Superstar: well, some context does not exist. typo? or maybe you should add it |
| 14:35.39 | WIMPy | puzzled: I don't think so. |
| 14:35.43 | irroot | yeah straigt into a atom server |
| 14:35.51 | WIMPy | You need that extra module. |
| 14:35.56 | irroot | no PCI sllots at all |
| 14:36.30 | puzzled | WIMPy: hmm that's too bad. a â¬20 card in a small atom box sure would make a cheap small biz pbx |
| 14:37.15 | WIMPy | puzzled: You can either add that 3rd party module to dahdi or use misdn or LCR. Should be enough options. |
| 14:38.13 | puzzled | WIMPy: or just keep on using my Diva Server cards with chan_capi :) It's proven to be a very stable solution over the past years |
| 14:39.00 | WIMPy | puzzled: But I couldn't find a chan_capi for 1.8. |
| 14:39.15 | puzzled | but yes for a HFC-S based card there are enough options. I'll try Digium's version of mISDN and see if I can get this to work |
| 14:39.28 | puzzled | WIMPy: iirc the latest version has 1.8 fixes. lemme check |
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| 14:41.27 | luke0512 | i just tried to start dahdi but get those errors |
| 14:41.28 | luke0512 | http://fpaste.org/E3SW/ |
| 14:42.04 | puzzled | luke0512: iirc the fedora repo does not package the kernel modules |
| 14:43.34 | WIMPy | And there's no mention of zaphfc, either. |
| 14:44.00 | puzzled | WIMPy: I see no mentioning of 1.8 in chan_capi 1.1.5 or trunk. time to email Armin |
| 14:44.10 | irroot | lol @ WIMpy trip down memory lane |
| 14:45.01 | WIMPy | puzzled: That's why I couldn't consider CAPI an option any more. |
| 14:45.10 | puzzled | WIMPy: trunk should work with 1.8: http://lists.melware.net/pipermail/chan-capi-users/2010-December/001162.html |
| 14:45.19 | malcolmd | irroot: M18811. breaks timing somehow? what were the symptoms in your testing? |
| 14:45.51 | WIMPy | I've use chan_capi before, but I think I prefer the current options any way. But depending on your hardware, it might be the only way. |
| 14:46.28 | puzzled | yup |
| 14:47.02 | irroot | malcom think its to do with the messages been written to file it floods the console / logs and does not stop with the patch |
| 14:47.05 | WIMPy | Ok, so capi should be an option again. |
| 14:47.27 | puzzled | yeah that's cool |
| 14:47.44 | irroot | WIMPy capi has the benifit off PPP plugin use one line as a modem line outside * |
| 14:48.10 | WIMPy | Now the interesting question is: Do I have anything that could be used with CAPI in order to test it? |
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| 14:48.23 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:48.29 | WIMPy | irroot: And you can hook hylafax directly to it. |
| 14:48.53 | WIMPy | But otherwise I think chan_capi is a little restricted on the feature list. |
| 14:49.32 | luke0512 | puzzled, do you know where i can get them |
| 14:49.34 | luke0512 | ??? |
| 14:49.35 | puzzled | WIMPy: direct chan_capi -> Hylafax? Wow. I have faxes coming in via chan_capi to iaxmodem and then onto Hylafax+ |
| 14:49.49 | puzzled | luke0512: try rpmfusion.org |
| 14:49.52 | irroot | malcolmd i have gone back to dahdi the one tester of T38 gateway has tested the timmers and dahdi comes out tops pthread fails |
| 14:49.58 | puzzled | luke0512: or maybe elrepo.org |
| 14:50.22 | WIMPy | puzzled: The only way to get G4 fax under Linux AFAIK. |
| 14:50.28 | irroot | hehe so the T38 gateway i got working would not have been needed to work with T38modem/hyla |
| 14:51.04 | puzzled | WIMPy: didn't know that. thanks for the tip |
| 14:51.20 | WIMPy | That's only from what I collected from the web, I haven't tried fax. |
| 14:51.48 | puzzled | the chan_capi -> iaxmodem -> hylafax and vice versa has worked flawlessly for me for many years |
| 14:54.03 | WIMPy | Ha, I have an old AVM BRI/ADSL combo card. That should support CAPI, but probably only on a fairly ancient Linux installation. |
| 14:54.17 | irroot | puzzled i struggled with iaxmodem used ALAW linksys |
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| 14:59.08 | puzzled | WIMPy: any suggestions where I can find an up-to-date zaphfc patch for latest dahdi? This one any good? http://bugs.gentoo.org/attachment.cgi?id=255315 |
| 14:59.45 | puzzled | irroot: I never had any issue with alaw but I don't use any linksys stuff |
| 14:59.58 | WIMPy | Oh, tough one. There seem to be several versions out there. I last tried the one on google code. |
| 15:00.12 | puzzled | seen that one. svn is empty |
| 15:00.37 | WIMPy | It looks empy. There's a sub dir. |
| 15:00.50 | puzzled | doh |
| 15:01.01 | WIMPy | yes |
| 15:01.11 | luke0512 | i found that mISDN is loaded |
| 15:01.14 | luke0512 | mISDN_core 83248 1 hfcpci |
| 15:01.32 | WIMPy | luke0512: What Linux version? |
| 15:01.56 | WIMPy | kernel that is |
| 15:02.53 | irroot | got to love sip notify rebooting 128 polycom phones to get them to use SRTP |
| 15:03.03 | luke0512 | fedora 12 kernel 2.6.31.12-174.2.3.fc12.x86_64 |
| 15:03.15 | WIMPy | And if you want to use dahdi, you have to make sure misdn won't get loaded. |
| 15:03.34 | WIMPy | Ok, that would be misdn2 then. |
| 15:03.36 | puzzled | irroot: heh nice |
| 15:04.23 | luke0512 | dahdi is not available...i havent't found yet |
| 15:04.41 | irroot | use the source LUKE(0512) :P |
| 15:04.42 | luke0512 | so i want to give mISDN a try |
| 15:04.53 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:05.15 | WIMPy | luke0512: That would mean LCR then. |
| 15:05.34 | luke0512 | on produktiv system i don't like using source |
| 15:06.13 | WIMPy | luke0512: that might make it quite hard then. |
| 15:06.14 | irroot | luke have mISDN 1 with 2.6.38 can put it onto SVN |
| 15:07.13 | WIMPy | irroot: You have tried to use misdn1 on a kernel that has misdn2? |
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| 15:07.35 | irroot | yeah use it in production WIMPy |
| 15:08.18 | WIMPy | irroot: Do you need to rebuild the kernel without misdn support first? Or is it possible without comiling your own? |
| 15:08.57 | luke0512 | hmmm...my time for now is empty...maybe back later...thanks so far |
| 15:08.59 | irroot | i compile all me self but supose you could by removing the V2 modules |
| 15:09.13 | WIMPy | So far I have a 2nd ancient kernel on my test system. |
| 15:09.51 | WIMPy | puts that somewhere lower on his to try list. |
| 15:10.35 | Cadey | Hi Guys, anyone here use Aastra 57i phones and got the Xfer to update the caller ID after transfer? |
| 15:21.46 | Cadey | stupid question but doing a sip reload does reload the sop.conf right? |
| 15:21.52 | leifmadsen | yes |
| 15:21.58 | Cadey | kk |
| 15:23.52 | Cadey | lol im having no joy Leif :) |
| 15:24.07 | Cadey | I found a couple of aastra settings that needed setting but still no joy |
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| 15:26.56 | cneb3000 | Cadey: I think the xfer issue is an Asterisk config issue.. nothing to do with the Aastra phone.. |
| 15:28.07 | Cadey | cneb3000 : Possibly mate, I cant find anyone on Google that has it working :( |
| 15:28.10 | cneb3000 | oops.. tell a lie. just googled it - aastra phones seem to make it a little more complicated :) |
| 15:28.21 | Jcook_5xData | any know what this may mean http://pastebin.com/Pg8F414R I try googling it but found no results |
| 15:30.23 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 15:30.56 | asilva | Hello, does anyone know a patch for 1.6.2.xxx to enable more pickgroup groups? |
| 15:31.35 | Superstar | Is it possible to test asterisk against stress by simulating 100 calls (for example)? I only have 2 landlines and my mobile |
| 15:31.46 | leifmadsen | Superstar: use SIPp |
| 15:31.51 | leifmadsen | or another asterisk |
| 15:32.06 | pabelanger | +1 SIPp |
| 15:32.12 | Superstar | thanks guys |
| 15:32.15 | Cadey | <cneb3000> : what did you google by teh way> |
| 15:32.39 | cneb3000 | cadey: i'm googling a few things.. but currently on "original clid in forwarded calls" |
| 15:35.51 | Superstar | Is there less load using WAV instead of MP3 for Music on Hold? |
| 15:35.59 | leifmadsen | asilva: what is the max now.... ? |
| 15:36.15 | leifmadsen | asilva: and what error are you finding? I can't find where a max pickupgroup is defined at all |
| 15:37.05 | leifmadsen | asilva: seems to be 63 I guess |
| 15:37.39 | leifmadsen | I did a search for "pickupgroup" on mantis, and exactly what you want popped up as the 6th issue on the list |
| 15:39.43 | cneb3000 | Cadey: What are you using exactly? Just Asterisk on it's own (which version), FreePBX, Thirdlane?.. |
| 15:40.02 | Cadey | Hi cneb3000, Vanilar 1.8 dude |
| 15:40.35 | cneb3000 | googled 'Vanilar Asterisk' then realised what he was doing |
| 15:40.38 | cneb3000 | >_> |
| 15:40.40 | cneb3000 | <_< |
| 15:40.49 | *** join/#asterisk Hail2theKingBaby (~hail2thek@webmail.mackellar.co.uk) |
| 15:41.18 | Cadey | lol |
| 15:43.00 | cneb3000 | Cadey: Can you copypaste your sip.conf to pastebin please? make sure you remove your passwords ;) |
| 15:43.12 | Cadey | can we PM a second? |
| 15:43.15 | cneb3000 | sure! |
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| 15:55.34 | m_tadeu | hi...when I send the ami action:Events with an actionID, asterisk doesn't reply with the actionID |
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| 16:06.39 | *** join/#asterisk paule32 (~paule22@p5B2B922A.dip0.t-ipconnect.de) |
| 16:06.42 | paule32 | hello |
| 16:07.54 | paule32 | i have register successful the asterrisk to my router ip |
| 16:08.08 | paule32 | calls will be ring on the phones |
| 16:08.46 | paule32 | but nothing voice are on the phone |
| 16:08.58 | paule32 | what can i do? |
| 16:09.33 | Aut0ExeC | paule32: kinda phone? |
| 16:11.04 | paule32 | softphone to asterisk -> router internet phone number -> analog phone |
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| 16:15.07 | paule32 | all ip's that i give to asterisk are private adresses |
| 16:15.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 16:15.46 | paule32 | register => user:password@192.168.1.1 |
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| 16:20.46 | CrashSys | Is there a way to make Asterisk advertise a different Media IP whenever a call is sent out a certain extension? |
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| 16:47.14 | drift- | i'm getting busy signal right now |
| 16:47.17 | drift- | when i call out... |
| 16:47.21 | drift- | any suggestions on how to fix this? |
| 16:47.31 | drift- | was working perfect last week came back from weekend now phone goes busy :( |
| 16:47.35 | drift- | we can recieve phone calls tho |
| 16:47.54 | WIMPy | Have you paid your bill? |
| 16:48.06 | drift- | heh |
| 16:48.09 | drift- | ofcourse |
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| 16:51.55 | asilva | leifmadsen, sorry , i had to leave... the max is 63. i have more than that! that's why i ask that! |
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| 17:23.43 | Jcook_5xData | any one here heard of a SL1 profile or L1 flag in reference to ATT PRI? |
| 17:25.33 | Jcook_5xData | I found no explanation on google |
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| 17:41.16 | bmoraca | has anyone used an Allworx phone with Asterisk? |
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| 18:11.41 | Cadey | Hi guys, ill ask again to see if anyone new has :) So, has anyone got Aastra phones Xfer working so it updates the CallerID post transfer |
| 18:12.08 | Cadey | im interested in the steps they took in order to get it to work |
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| 18:38.29 | _Corey_ | Cadey: I'm not an Aastra guy, but you may want to just check your sip.conf for flags like: sendrpid=yes, rpid_update=yes, etc. |
| 18:38.53 | Cadey | rpid_update... ive not see than one :) |
| 18:39.05 | Cadey | I have the trustrpid and sendrpid |
| 18:39.52 | _Corey_ | look for anything "rpid" and play around with it. I've not had to do anything with the phones themselves |
| 18:40.26 | _Corey_ | works OK on Polycom, Cisco, and Linksys (SPA) stuff |
| 18:40.29 | Cadey | The aastra phones have a setting to allow callerID renames |
| 18:40.33 | Cadey | that is now set |
| 18:40.49 | Cadey | but ill try that sip setting you suggested and look for any more with rpid |
| 18:41.37 | Cadey | ive never seen the rpid_update one tbh |
| 18:43.33 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-098.mobile.uci.edu) |
| 18:47.40 | *** join/#asterisk XXedgeXX (~root@dns1.troubletext.com) |
| 18:54.10 | mazpe | anyone take a look at this: http://paste.scsys.co.uk/106736 i'm trying to install appKonference and it errors out on the make |
| 18:54.28 | mazpe | primarily with this: /usr/src/asterisk-1.8.4/include/asterisk.h:21:33: error: asterisk/autoconfig.h: No such file or director |
| 18:55.01 | Aut0ExeC | anyone here have linux+ ? |
| 18:55.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:55.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:56.12 | Chainsaw | Aut0ExeC: What is that, some sort of kernel fork? |
| 18:56.23 | Aut0ExeC | Chainsaw: lol ... ur funny man |
| 18:57.14 | mazpe | interesting include/asterisk.h refers to asterisk/autoconfig.h |
| 18:57.18 | pabelanger | mazpe: $ sudo apt-get install asterisk-dev |
| 18:57.50 | mazpe | pabelanger: i have the source and i see autoconfig.h.in but no .h |
| 18:57.52 | Jcook_5xData | Aut0ExeC, is that the patch to speed up the kernel |
| 18:58.02 | pabelanger | actually, how did you install asterisk? |
| 18:58.05 | _Corey_ | presses his turbo button |
| 18:58.08 | Aut0ExeC | Jcook_5xData: yes... i guess you can say that |
| 18:58.21 | pabelanger | mazpe: because you need to do ./configure |
| 18:58.28 | irroot | pabelanger SVN :P |
| 18:58.37 | mazpe | pabelanger: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 18:58.56 | pabelanger | mazpe: did you install asterisk-dev ? |
| 18:59.07 | Jcook_5xData | I thought they added that to the next release or hybrid anyway |
| 18:59.26 | mazpe | pabelanger: no i did apt-get install asterisk-dahdi |
| 18:59.47 | pabelanger | mazpe: then you are missing the header files, asterisk-dev should have them |
| 18:59.48 | mazpe | pabelanger: I'm on Debian.. and the apt-get install asterisk installs 1.6 |
| 19:00.09 | mazpe | thats why im installing the from digium packages |
| 19:00.18 | mazpe | hopefully they have it. |
| 19:01.04 | pabelanger | mazpe: Hmm, apt-get install asterisk should get 1.8.4 |
| 19:01.16 | pabelanger | if you have added the Digium repository |
| 19:02.13 | mazpe | which i did.. lets try it |
| 19:02.32 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:02.32 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:02.44 | mazpe | pabelanger: that did it. |
| 19:02.47 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 19:02.50 | Jcook_5xData | Aut0ExeC, my bad you are talking about the cert lol |
| 19:03.08 | malcolmd | irroot: thanks for the _32 change :) |
| 19:03.33 | Chainsaw | Jcook_5xData: Just goes to show one should explain abbrevations/codes before using them. |
| 19:03.37 | irroot | pleasure driving me crazy |
| 19:04.00 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 19:04.38 | irroot | did not want to add a int when few flags will do the job |
| 19:04.59 | malcolmd | i know. :( |
| 19:05.07 | Aut0ExeC | Jcook_5xData: yes |
| 19:05.14 | malcolmd | but we're doing "the right thing" for any future madness that comes up |
| 19:05.18 | Aut0ExeC | Jcook_5xData: cert |
| 19:06.24 | Jcook_5xData | sorry, he said some thing about a fork. and I remember that big patch everone was talking about a couple of month ago. I should shut my fingers :) |
| 19:07.40 | Aut0ExeC | k |
| 19:07.43 | Kobaz | anyone have experience with directed call pickup with polycom phones |
| 19:07.53 | Aut0ExeC | Kobaz: no |
| 19:07.58 | Chainsaw | Kobaz: We just use a feature code here, sorry. |
| 19:08.03 | Khratos | So, this is how hell looks like. I started today at a new job, and all machines have Ubuntu... |
| 19:08.10 | Kobaz | trying to get native call pickup going |
| 19:08.16 | Kobaz | via sip replaces |
| 19:08.20 | Chainsaw | Khratos: And all the servers are running RedHat Expensive Linux? |
| 19:08.32 | Aut0ExeC | Khratos: you shoudl reinstall your fav distro on all |
| 19:08.40 | Aut0ExeC | Khratos: tell the boss u had to do it |
| 19:08.44 | Khratos | I haven't dare to ask |
| 19:08.46 | Khratos | haha |
| 19:08.48 | Aut0ExeC | Khratos: its was best for the company |
| 19:09.14 | Aut0ExeC | Khratos: show some initiative brother |
| 19:09.17 | Aut0ExeC | Khratos: u will get promoted |
| 19:09.28 | Khratos | sure! :P |
| 19:09.32 | Aut0ExeC | :) |
| 19:09.51 | WIMPy | Just produce a power failure and say that most machines didn't survive it. |
| 19:09.57 | Aut0ExeC | lulz |
| 19:10.01 | Aut0ExeC | WIMPy: nice one |
| 19:10.27 | WIMPy | knows someone who used that stategy. [cough] |
| 19:10.45 | Aut0ExeC | lol |
| 19:10.45 | irroot | got some slackware v3 stiffys here somewhere Khratos :P |
| 19:10.45 | Aut0ExeC | WIMPy: nice man |
| 19:11.08 | Aut0ExeC | slackware = no dependencies right? |
| 19:11.10 | Aut0ExeC | ewww |
| 19:11.15 | WIMPy | He didn't convice the boss about that. |
| 19:11.27 | WIMPy | Aut0ExeC: Correct |
| 19:12.18 | Aut0ExeC | WIMPy: gross man |
| 19:12.25 | Aut0ExeC | WIMPy: dependency checks rule |
| 19:12.48 | irroot | hehe heard some it damagers going on about "change control" i pointed out that change begins within they wrere annoyed not sure Y |
| 19:12.51 | malcolmd | Khratos: we'll do some testing of a 3.5x confluence internally within the next couple of weeks; if all's well, it'll roll out to the public wiki.asterisk.org soon after |
| 19:13.40 | luke0512 | i'm out now...cu |
| 19:13.56 | Superstar | Do any corporate companies use Asterisks? |
| 19:14.05 | Khratos | malcolmd: great. Is the official wiki a hosted service or is it installed in a server of yours? |
| 19:14.26 | malcolmd | installed server inside digium - though it should probably be at the co-lo |
| 19:14.36 | irroot | here in south africa 2 of the big 4 banks use it and the largest medical aid company |
| 19:14.49 | Chainsaw | Superstar: corporate companies? Please elaborate. |
| 19:14.54 | _Corey_ | Superstar: http://www.easternmanagement.com/news_1_21_09.htm |
| 19:14.59 | *** join/#asterisk ketema (~ketema@turtle.ketema.net) |
| 19:15.39 | Superstar | chainsaw: are you asking me to define what a corporate company is? |
| 19:17.53 | Khratos | irroot: did you find someone to test t.38 with ooh323? I think I need to buy some hardware in order to help testing |
| 19:18.43 | Superstar | I've found a company called Aspect but the article was from 2006 |
| 19:19.03 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:20.02 | irroot | i have been using t38modem with hylafax got some customers using asterisk as a fax server more the merrier ... see RB 1117 for some changes to ooh323 |
| 19:20.08 | _Corey_ | Superstar: there are literally thousands of companies using Asterisk. Digium has tons of case studies on their website |
| 19:21.09 | Chainsaw | Superstar: No, it just seems a strange thing to ask. You're not interested in a specific field? |
| 19:21.20 | Chainsaw | Superstar: Territory? Continent? |
| 19:21.24 | Superstar | _corey_ I know I read them but none were corporations |
| 19:21.30 | *** join/#asterisk mameluk (~Adium@188.134.33.24) |
| 19:21.55 | Superstar | I just want to know the history of Asterisk |
| 19:23.02 | mameluk | Hi all. Anyone can help me with video in asterisk. I write option videosupport in sip.conf on peers, but when i make video call - video don't up/ |
| 19:23.05 | mameluk | i use http://www.futurewei.com/itemsdetail.asp?dt=products&ID=122 |
| 19:23.06 | irroot | Superstar look at http://www.zapatatelephony.org/ for some history ;) |
| 19:23.36 | Superstar | irroot thank you, appreciate it |
| 19:24.02 | mameluk | in console asterisk i see warning "Unable to parse INFO message from⦠" |
| 19:24.18 | XXedgeXX | Does anyone have any experience setting up Asterisk for a call in radio show using a SIP trunk? Also, is there a directory of Asterisk consultants available? |
| 19:31.15 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-185-miami.txt.hfc.comcastbusiness.net) |
| 19:37.17 | ChannelZ | I suppose it's a matter of how to get the audio out into wherever you need it to go. |
| 19:37.44 | ChannelZ | Could be as easy as a softphone with the audio patched into your mixer |
| 19:40.21 | _Corey_ | We've used this stuff before, works well enough: http://www.bswusa.com/searchresult.asp?searchType=category&subcat=telco-coup |
| 19:49.25 | keith4 | XXedgeXX: I've done PA systems using linphone set to auto-answer |
| 19:49.58 | WIMPy | XXedgeXX: JACK? |
| 19:50.01 | keith4 | it's stupid-easy. all you need is a computer with a sound card and a network connection |
| 19:51.22 | WIMPy | Or if you want analog, chan_alsa or any of the other console channels. |
| 19:55.57 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:59.33 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:09.58 | *** join/#asterisk stegbth (~stegbth@pD9F687E1.dip.t-dialin.net) |
| 20:10.03 | stegbth | hello everybody |
| 20:10.07 | *** join/#asterisk buzzyd (~buzzyd@server.asyouneed.net) |
| 20:10.30 | stegbth | i have two mystery problems with trixbox and asterisknow |
| 20:10.53 | stegbth | i connect via patton smartnode an isdn line to the pbx |
| 20:10.54 | buzzyd | Hi all, When installing 1.8 do I need to use the addons from 1.6? |
| 20:11.19 | stegbth | when creating a inbound route to an extension it works |
| 20:11.38 | stegbth | when i modify the same inbound-route to an ringgroup, all members are ringing |
| 20:12.07 | Chainsaw | buzzyd: 1.8 no longer has an "addons" package. |
| 20:12.25 | stegbth | but when answer the call, the caller get i continue ringtone, the answerer does not hear anything and the remaining phones are still ringing |
| 20:12.26 | Chainsaw | buzzyd: So no, the 1.6.2 add-ons should be gone before 1.8 hits the disk. It will not use them and it most definitely will not like them. |
| 20:12.41 | buzzyd | Chainsaw, what about the mysql cdr aspect is that part of the main 1.8 package now? |
| 20:12.48 | Chainsaw | buzzyd: It is, yes. |
| 20:13.06 | buzzyd | Chainsaw: cool makes it even easier :) |
| 20:13.11 | Chainsaw | buzzyd: Indeed :) |
| 20:13.20 | buzzyd | thank you. |
| 20:19.11 | Kobaz | all you need is sip... |
| 20:19.13 | Kobaz | do de do do do |
| 20:19.22 | Kobaz | sip is all you need |
| 20:20.31 | *** join/#asterisk b0ot (~tom@147.177.42.173) |
| 20:20.53 | *** join/#asterisk caraconan (~jlegido@189.5.20.95.dynamic.jazztel.es) |
| 20:21.18 | acidjazz | can any1 in here name some competitors to twilio? i need to explore other options |
| 20:21.31 | b0ot | If I'm going to have a bunch of mobile stations moving around my network and I want to have voip capabilities in my network... should each station run their own version of asterisk and trunk between them... or would it be better to have few versions of asterisk |
| 20:23.39 | caraconan | Good night. I found errors in the logs regarding my FXO card. The output here: http://pastebin.com/y2PLr9Bh |
| 20:24.12 | caraconan | Any idea if the issue is regarding BIOS? I changed IRQ but no way |
| 20:25.49 | caraconan | Its an X100P FXO card (http://www.x100p.com/products/FXO.php), but I hate to waste my time with this kind of issues. Anybody suggest to me an FXO card supported by linux for testing purposese? |
| 20:26.10 | Chainsaw | caraconan: TDM400 or TDM410. |
| 20:26.18 | Chainsaw | caraconan: Don't waste your time with the poorly converted modems. |
| 20:27.20 | caraconan | Ok, I will take a look. The idea is to buy a low-profile FXO PCI card and start an VMware VM to start learning asterisk |
| 20:27.25 | caraconan | Thanks for the advice |
| 20:27.51 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 20:28.10 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:28.27 | _Corey_ | caraconan: You're planning on using an analog board w/a vmware asterisk guest machine? |
| 20:28.44 | *** join/#asterisk ketema_ (~ketema@kjhmacpro.ketema.net) |
| 20:29.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:29.33 | caraconan | I've got an ESXi, and over that I've got several VM's |
| 20:30.13 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:30.55 | bmoraca | caraconan: you will not be able to use telephony hardware with that |
| 20:31.19 | caraconan | Why? |
| 20:31.28 | bmoraca | because VMWare has no drivers for those cards |
| 20:31.43 | bmoraca | there is no way for your virtual machine to address them |
| 20:33.17 | caraconan | Ok, I will try FXO card in the same PC with a distro like trixbox, and see what happens |
| 20:33.20 | caraconan | Thanks guys! |
| 20:33.45 | _Corey_ | caraconan: Try AsteriskNOW from Digium's website |
| 20:33.56 | caraconan | Ok, thanks a lot |
| 20:34.11 | _Corey_ | get an SIP/analog gateway if you want to stick with VMWare |
| 20:35.16 | caraconan | http://communities.vmware.com/thread/89495 -> ESXi seems not supporting FXO |
| 20:35.54 | _Corey_ | now you have independent confirmation :) |
| 20:37.28 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 20:37.31 | caraconan | Thanks guys |
| 20:38.04 | bmoraca | so, one more ask...has anyone here successfully got an Allworx phone registered with vanila asterisk? if so, any hints on the config file options? :) |
| 20:38.05 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:45.30 | buzzyd | ok what modules do I need to load in 1.8 to get mysql cdr working as cdr_addon_mysql.so doesn't exist |
| 20:46.34 | buzzyd | is there a wiki for 1.8 as the wiki seems quiet out of date |
| 20:58.26 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:04.16 | *** join/#asterisk SkyeBen (~ben@skyefinancialservice.com) |
| 21:05.25 | SkyeBen | Anyone having issues with 1.6.2.18 along the lines of a SIP phone not being able to send audio? |
| 21:06.23 | *** join/#asterisk dinesh___ (~dinesh@46-126-192-144.dynamic.hispeed.ch) |
| 21:06.49 | dinesh___ | hi all, can i get asterisk to play a fake ring tone when i receive an incoming call (in my extension.conf) ? |
| 21:07.19 | dinesh___ | basically a way to silently ignore the incoming call |
| 21:07.23 | SkyeBen | Background(ringtoneFile) |
| 21:07.50 | SkyeBen | Then you'll have to have a ring file (i'm pretty sure * comes with one) |
| 21:08.13 | WIMPy | But ignoring is something different. |
| 21:08.33 | SkyeBen | You could call Playback(), then Voicemai... |
| 21:09.05 | _Corey_ | um... Ringing(), Wait(x), Voicemail |
| 21:11.40 | SkyeBen | Anyone run Cisco phones with SIP firmware? I think I've just found a bug with 1.6.2.18 and I'd love to converse |
| 21:13.41 | dinesh___ | oh that looks great, thanks |
| 21:17.48 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 21:18.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:18.42 | pdtpatrick | Question for u smart folks .. what does this mean ..from 'number' to extension 'blah' rejected because extension not found in context 'default' |
| 21:18.51 | pdtpatrick | what needs to be done to resolve this issue? |
| 21:19.03 | WIMPy | Exactely what it says? |
| 21:19.39 | pdtpatrick | im new to asterisk .. so if u can elaborate further on where to look or what to fix/change -- that'd be awesome. If you don't mind |
| 21:20.03 | WIMPy | You called an extension that does not exist. |
| 21:20.18 | WIMPy | Not in the used context at least. |
| 21:20.45 | pdtpatrick | to fix .. i should then change something in? extensions.conf? sip.conf ? |
| 21:21.20 | WIMPy | Hopefully you don't want to use the default context, so maybe you put a phone in the wrong (or no) context? |
| 21:21.49 | WIMPy | That't the two files involved here, yes. |
| 21:21.54 | pdtpatrick | okay thanks |
| 21:22.48 | Kobaz | does asterisk support polycom BLA ? |
| 21:22.52 | dinesh___ | thanks _Corey_ , Ringing() + Wait() did the trick |
| 21:27.16 | XXedgeXX | Hey, thanks for the answers regarding the radio call in stuff. So I have Asterisk running and I'm using Bria as a softphone. I have Bria configured to use my audio interface for input/output, which runs directly into my broadcast console. What I'm experiencing is an issue with the audio for the person recording sound from the console. I wasn't sure if this was because of the frequency range being produced by my audio interface because things soun |
| 21:27.38 | XXedgeXX | The remote person sounds good in our receive, but our transmit sounds like crap on their end. |
| 21:27.54 | XXedgeXX | Checking out the link to BSW right now. |
| 21:31.06 | _Corey_ | XXedgeXX: fyi, If you buy one of those couplers you'll need to plug it into an FXS port |
| 21:31.38 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 21:32.02 | XXedgeXX | I'm not familiar with an FXS port. Seems like all the couplers assume some kind of analog phone line. Is this why I need this FXS port? |
| 21:32.14 | _Corey_ | yeah, exactly |
| 21:32.38 | pabelanger | Kobaz: BLA? |
| 21:32.46 | XXedgeXX | Ok, where do I find FXS ports? |
| 21:32.50 | Kobaz | bridged line appearance |
| 21:33.18 | _Corey_ | XXedgeXX: Linksys PAP2 would work or an analog card |
| 21:34.02 | _Corey_ | you basically dial the extension assigned to the analog port and it rings the coupler, which answers and bridges you into your board |
| 21:34.51 | XXedgeXX | Thanks for your help. Just so I understand this, basically these FXS ports take ethernet and an RJ45 on the other end so I can plug it into the coupler. Will this single FXS port allow me to support multiple SIP lines? My goal is to be able to support multiple simultaneous callers, conferencing, etc. |
| 21:35.23 | _Corey_ | well, sort-of... you'll get two analog ports with a PAP2 |
| 21:36.04 | _Corey_ | if you want to bridge more callers, doing it with MeetMe() or something on Asterisk is probably better than buying a bunch of ports though |
| 21:36.57 | XXedgeXX | The end goal is to have an 800 number that people can call into and I was hoping to queue up to 10 callers. |
| 21:37.05 | _Corey_ | one on air at a time? |
| 21:37.46 | XXedgeXX | Up to three on air |
| 21:37.59 | *** join/#asterisk robbie` (~bob@74.207.245.153) |
| 21:38.09 | XXedgeXX | That would be ideal, but if I can only do two, that would be acceptable. |
| 21:38.38 | _Corey_ | You going to have someone managing this with a phone transfer or just want to dump them into your sound console? |
| 21:39.18 | robbie` | anyone know where i can get some help with avantfax/hylafax? i have asterisk/freepbx up and running on a VPS, i am running iaxmodem and my device is /dev/ttyIAX, i can see the peer is online, but in avantfax it sais ttIAX [Please Wait] and i get an email when i try to send a fax stating no local dialtone. faxstat says hylafax is running, and my modem ttyIAX is running and idle. faxgetty, hfaxd, and faxq are all running in the background.. any ideas?? |
| 21:39.33 | XXedgeXX | Someone would manage the phone...I was thinking either creating a queue or having a "call screening" extension that would than be passed to the "on air" extension. |
| 21:39.59 | _Corey_ | yeah, I think you really want to use meetme and install the web meetme gui |
| 21:40.18 | _Corey_ | the "screener" can mute/unmute, kick people, etc |
| 21:40.54 | _Corey_ | you can have as many in there as you want and just have the FXS port in the conference all the time |
| 21:41.11 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 21:41.13 | XXedgeXX | Ok, I'll look at that. Is this something that people from anywhere could call? I don't want to force callers to use some kind of interface. I also need to support 800 numbers, preferrably transfer the one I already have. |
| 21:41.30 | XXedgeXX | Never heard of MeetMe, so I'll check that out. |
| 21:41.42 | Kobaz | i'm trying to get native directed pickup working on a polycom |
| 21:41.52 | Kobaz | it looks like the polycom refuses to use it, and it sends a star code instead |
| 21:41.53 | XXedgeXX | MeetMe works in conjunction with Asterisk? |
| 21:42.00 | Kobaz | even though there's an option to turn it on |
| 21:42.08 | _Corey_ | well, that aspect of it is more of a basic asterisk question so I'd refer you to the book to begin |
| 21:42.10 | _Corey_ | ~thebook |
| 21:42.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 21:42.12 | paulc | XXedgeXX: MeetMe is the conferencing app within Asterisk |
| 21:43.00 | _Corey_ | but to actually answer your question about the line... sure you could point your 800# at a line coming into Asterisk |
| 21:43.06 | XXedgeXX | Thanks. THe O'Reilly book is in the mail already :-) |
| 21:43.20 | _Corey_ | :) |
| 21:43.48 | XXedgeXX | THe name kind of threw me off. MeetMe just sounded like something that would be similar to "Webex" or something. If it's built into Asterisk, that makes more sense. |
| 21:44.06 | _Corey_ | type "core show application meetme" on your asterisk CLI |
| 21:44.59 | XXedgeXX | Unfortunately I don't have access from my current location, but I will certainly check it out as soon as I get to the studio. |
| 21:47.02 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 21:49.19 | XXedgeXX | Just searching around, would something like this make any sense for what I'm trying to do? |
| 21:49.21 | XXedgeXX | http://www.voipsupply.com/mediatrix-4116-sip |
| 21:50.11 | paulc | XXedgeXX: I'm late to the party, but I heard "studio" and my ears pricked up - what is it you're trying to do? |
| 21:50.49 | _Corey_ | XXedgeXX: that's a little overkill for what you want |
| 21:51.06 | XXedgeXX | For the past few weeks, I've been experimenting with Asterisk and various softphones trying to figure out a reliable way to do multiple callers for a radio talk show. |
| 21:51.19 | WIMPy | XXedgeXX: If you want ppl to talk to each other: no. If you want to feed them seperately into your mixer: yes. |
| 21:52.03 | paulc | XXedgeXX: Have you seen www.callinstudio.com - it's a hosted offering but sounds kinda right-ish |
| 21:52.08 | WIMPy | thinks the recommendation to use meetme and the web admin tool was a good one. |
| 21:52.26 | paulc | XXedgeXX: I was talking to a guy the other week about callers on air for radio but I don't think it was you was it? (caller screening, a producer, a host, etc?) |
| 21:52.35 | WIMPy | XXedgeXX: Do you also want to do other stuff with it? |
| 21:52.40 | _Corey_ | I'd stick with a single line to your studio equipment and use Asterisk to bridge the calls |
| 21:53.18 | XXedgeXX | I have two problems (that I know of), one, my bandwidth is definitely limited and unless I want to pay a lot of money for a high bandwidth connection, there isn't much I can do about bandwidth. 2, since I'm going straight from the softphone, directly into the radio console, I believe the frequency range being produced by the softphone exceeds what the remote caller can handle, so things are coming out very garbled, yet they sound perfectly fine on o |
| 21:53.37 | XXedgeXX | WIMPy, I have seen that, I will check it out. |
| 21:54.07 | XXedgeXX | paulc, that wasn't me. This is the first time I got on the IRC channel. |
| 21:54.11 | _Corey_ | where is the bandwidth limited? |
| 21:54.27 | paulc | XXedgeXX: how are things sounding garbled? I've done stuff with softphones and DIDs into a mixer and it worked pretty well.. just a case of getting your mix-minuses right :-) |
| 21:54.28 | WIMPy | XXedgeXX: Use a hosted PBX and only the softphone locally. That way you only need bandwidth for one call, or a 2nd for screening. |
| 21:54.40 | XXedgeXX | ADSL - I'm stuck with AT&T in this area. I'm in downtown Los Angeles. |
| 21:56.28 | XXedgeXX | Yes, the console actually has a dedicated caller channel that ensures the mix minus. It sounds garbled in the sense that it's as if there's too much base. If I'm playing background music over the caller channel and I try to talk, you can't hear my voice at all on the remote end, as if the music is just killing the voice. This is why I suspected exceeding the frequency range capable of the receiving call. |
| 21:57.00 | XXedgeXX | base = bass |
| 21:57.23 | paulc | XXedgeXX: Hmm that sounds more like your mic channel's not going to the mix minus - channel assigned to the right bus etc? (what mixer?) |
| 21:57.31 | paulc | WIMPy' |
| 21:58.01 | XXedgeXX | I'm using a broadcast phone hybrid and vonage right now and it's working fine through the hybrid, but of course I'm limited to one call at a time with the analog setup. |
| 21:58.09 | paulc | s got the right idea I think.. colo'd Asterisk server, with a single channel for "on air", which could be shared with screening if needed, or use a channel for each (you could even go low-fi for screening and better quality for on air) |
| 21:59.03 | XXedgeXX | It's not a mixer, it's an actual broadcast console. I'm using a AudioArts Air-3 broadcast console: http://www.bswusa.com/proditem.asp?item=AIR3 |
| 21:59.46 | paulc | Ah yes.. the AIR3 - not a bad board.. |
| 22:00.00 | XXedgeXX | but with a colocated asterisk server, will I be able to queue callers up? and will I be able to have at least two callers on the air simultaneously? |
| 22:00.15 | XXedgeXX | paulc, it's my first console, I'm pretty happy with it so far. |
| 22:00.37 | WIMPy | You can do it all within Asterisk. |
| 22:00.48 | WIMPy | It has queues and conferences. |
| 22:00.50 | _Corey_ | you could have 100 callers on the air simulataneously if you REALLY wanted to... |
| 22:00.54 | paulc | XXedgeXX: Yes.. the conference lives on the Asterisk box, with a single call linking the studio to the conference. Have 1, 2, 5+ etc people in the conference call on the Asterisk box, you're just feeding one channel of audio in/out of your mixer |
| 22:01.05 | XXedgeXX | So any suggestions on a good colocated asterisk host? I'm fully willing to go that route if it means I can make this work. |
| 22:01.07 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:02.00 | paulc | XXedgeXX: I think Link2VoIP offer colo'd Asterisk boxes. Maybe Vitelity too. I've got my own server with someone else, but use both of those for calls etc. |
| 22:05.46 | paulc | XXedgeXX: Where you located? |
| 22:05.54 | XXedgeXX | I'm in downtown Los Angeles |
| 22:06.24 | paulc | Right.. (you said that already, sorry) |
| 22:06.55 | paulc | I was thinking - I did a thing a while back, with Asterisk and a webby front end, for a sort of "moderated" conference call.. |
| 22:07.23 | paulc | it's rough around the edges but had caller screening, Q&A playback, switching listen only callers live to air etc |
| 22:07.41 | paulc | could maybe dig it up when I'm back in the office tomorrow, see if it'd work for you |
| 22:07.57 | XXedgeXX | I would love to try anything that you think might work. |
| 22:08.54 | paulc | Hit me up tomorrow or I'll ping you.. if I put it on a different box I can probably demo the web part of it for you |
| 22:13.14 | mazpe | g729 show licenses: 0/0 encoders/decoders of 0 licensed channels are currently in use |
| 22:13.17 | mazpe | thats not good |
| 22:14.23 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 22:20.45 | Kobaz | ooh, peoples are alive, yay :) |
| 22:20.47 | Kobaz | er |
| 22:26.25 | *** join/#asterisk wonderworld (~ww@port-92-201-214-69.dynamic.qsc.de) |
| 22:33.38 | *** part/#asterisk stegbth (~stegbth@pD9F687E1.dip.t-dialin.net) |
| 22:36.40 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 22:50.22 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 22:53.20 | carrar | staying alive |
| 22:53.26 | carrar | ah ah ah |
| 22:53.29 | carrar | staying alive |
| 23:02.56 | *** join/#asterisk momelod (~smelo@dsl-173-248-230-1.acanac.net) |
| 23:03.03 | momelod | greetings channel. |
| 23:03.05 | ChannelZ | wipes the drool off his lip |
| 23:03.31 | momelod | im running asterisk at home. are there any sip providers that serve residential in the toronto area? |
| 23:03.52 | momelod | *not asterisk@home, asterisk, at my house* |
| 23:04.34 | paulc | momelod: check out voip.ms - a friend of mine does it with an ATA, so it should be fine with Asterisk (ie you're registering to them, rather than them sending calls to a static IP address) |
| 23:05.31 | momelod | perfect, thanks |
| 23:11.16 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 23:12.11 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
| 23:13.07 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 23:17.33 | *** join/#asterisk n0tk (~n0tk@mail.stpaulinternet.net) |
| 23:19.28 | *** join/#asterisk n0tk (debian-tor@gateway/tor-sasl/n0tk) |
| 23:23.34 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
| 23:33.34 | mazpe | did the path for the sounds file for the ivr change in asterisk 1.8 from /var/lib/asterisk/sounds? |
| 23:34.31 | carrar | en |
| 23:34.36 | mazpe | oh |
| 23:35.00 | carrar | h4X0r |
| 23:35.16 | carrar | specify language in your dialplan |
| 23:35.30 | carrar | .. in all your confs |
| 23:36.13 | carrar | and it will look in the right place |
| 23:36.54 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:45.54 | mazpe | so you are saying to "Set(CHANNEL(language)=en)" and then /var/lib/asterisk/sounds/en/myfile.g729? |
| 23:47.07 | mazpe | carrar: not working :( |
| 23:48.12 | carrar | Whats the error? |
| 23:48.32 | carrar | lets see code and output |
| 23:49.58 | mazpe | carrar: http://nopaste.us/5897.html |
| 23:50.12 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 23:50.41 | carrar | "gbpbx/wdo/" |
| 23:50.54 | carrar | that doesn't match /var/lib/asterisk/sounds/en/myfile |
| 23:51.32 | carrar | unless you have /var/lib/asterisk/sounds/en/gbpbx/wdo/business-hours.g729 |
| 23:51.33 | mazpe | well i changed that. |
| 23:52.05 | mazpe | -rw-r--r-- 1 asterisk asterisk 23920 Apr 9 2009 /var/lib/asterisk/sounds/en/gbpbx/wdo/business-hours.g729 |
| 23:52.45 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 23:53.57 | joako | mazpe, Sort of OT, but how's inphonex? Have you used the service for a long while? |
| 23:54.30 | carrar | try jsust putting it in the en directory |
| 23:54.49 | mazpe | joako: i've used them for about two years now. A bit on the expensive side, decent support... they are local to me. |
| 23:55.02 | mazpe | carrar: business-hours.g729? |
| 23:55.13 | carrar | yeah |
| 23:55.34 | ChannelZ | are you sure you don't have languageprefix=no ? (Do other stock system sounds work?) |
| 23:55.54 | joako | mazpe, Yea? Same here... they are literally next door to one of the locations I support, and they get their internet access via an ethernet cable to their datacenter, but I've been hesitatnt to recommend it |
| 23:57.53 | mazpe | ChannelZ: yeah, Authenticate works |