| 00:01.55 | *** join/#asterisk DeltaSonic-Matt (n=DeltaSon@cpe-76-180-254-17.buffalo.res.rr.com) |
| 00:08.37 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
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| 00:25.52 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 00:28.13 | nath0099 | ok i have a minor issue wondered if any one has come acrossed it before. i have tried *1.4.4, trixbox 2.0, trixbox 2.2 and am now using easyvoxbox but the same issue in everyone of them i can get trunks to work but i cant get sipgate trunk to have incoming calls it just comes up engaged when you ring the number. |
| 00:29.01 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 00:32.24 | [TK]D-Fender | nath0099, What do you have installed currently? |
| 00:33.27 | nath0099 | easyvoxbox |
| 00:34.23 | [TK]D-Fender | nath0099, Go install * 1.2 or * 1.4 and then we'll be able to help you |
| 00:34.32 | [TK]D-Fender | nath0099, and I mean FROM SCRATCH |
| 00:35.20 | nath0099 | k |
| 00:35.23 | nath0099 | thx |
| 00:35.40 | *** part/#asterisk nath0099 (i=Jimbean@82-45-112-104.cable.ubr02.maid.blueyonder.co.uk) |
| 00:37.02 | *** join/#asterisk nephfl (i=nephilim@wsip-68-110-130-57.ga.at.cox.net) |
| 00:38.10 | russellb | easyvoxbox? that new? haven't heard of that one |
| 00:38.57 | Sweeper | russellb: it's awsome. basically, it's trixbox with VB support! |
| 00:39.00 | nephfl | hello im new to asterisk and attempting to setup a new system, i am attempting to troublshoot but am not sure where to start, i am using asteriskNOW with a TDM2421e, can someone help me out? |
| 00:39.47 | russellb | VB support? |
| 00:40.06 | Sweeper | russellb: visual basic! 6.0 atm! doesn't that sound AWSOME? |
| 00:40.09 | russellb | nephfl: support@digium.com would be happy to help you |
| 00:40.16 | russellb | Sweeper: sounds absolutely terrible |
| 00:40.23 | Sweeper | lies |
| 00:40.30 | russellb | no, i mean it, it does |
| 00:40.39 | Sweeper | bestest pbx evar mang |
| 00:41.08 | nephfl | i was just looking for some help since the partner i purchased from is closed until tomorrow morning... |
| 00:41.14 | Sweeper | you can even interface with ACCESS |
| 00:42.13 | russellb | OMFG <3 1997 |
| 00:45.41 | [TK]D-Fender | russellb, CrapTASTIC! |
| 00:50.35 | JT | slide projector? |
| 00:50.56 | [TK]D-Fender | JT :Ceiling monted pulldown :) |
| 00:51.12 | [TK]D-Fender | JT : 0 - Space requirement :) |
| 00:52.31 | Sweeper | of course, TK now has to live in total darkness to be able to watch stuff on his awsome VGA projector, but he already does that anyways |
| 00:52.36 | [TK]D-Fender | JT : And I'm selling off my 52" RCA Scenium HDTV and recuperating (25" x 48.5") in floor-space |
| 00:53.10 | [TK]D-Fender | Sweeper, Actually at DAMN nice with my 300w torchere lit up even. 2000 lumens does the job quite well. |
| 00:55.33 | russellb | [TK]D-Fender: you should give me your tv |
| 00:56.04 | [TK]D-Fender | russellb, You can buy it if you like, I'm letting it go for $750 |
| 00:56.12 | Sweeper | [TK]D-Fender: for now, in a month.... :D |
| 00:56.25 | Sweeper | unless it's diy, it's gonna shaft you on bulbs D: |
| 00:57.03 | russellb | [TK]D-Fender: i was hoping for free, but oh well |
| 00:57.15 | russellb | besides, i bet you're nowhere near where I am, anyway |
| 00:57.23 | [TK]D-Fender | Sweeper, Thing is that I have it mounted over my patio doors (well exceeds the width), and leavs only a 1' below it exposed. its a near black-out in daytim all by itself. |
| 00:57.36 | [TK]D-Fender | russellb, Quite correct.... shipping would be a killer. |
| 00:59.34 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.163.72) |
| 01:02.42 | [TK]D-Fender | Tomorrow should receive the projector mount and will begin wiring up the speakers. |
| 01:12.32 | blitzrage | 1984 is a better year |
| 01:12.50 | JT | better than what? |
| 01:13.06 | blitzrage | *.* |
| 01:16.13 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 01:16.25 | [TK]D-Fender | blitzrage, How "Orwellian" |
| 01:18.12 | russellb | it was also the year i was born, which is the real reason that year pwns. |
| 01:18.25 | JT | interesting |
| 01:18.31 | JT | i though russellb was older |
| 01:18.44 | [TK]D-Fender | Was a great yaer for Van Halen too :) |
| 01:19.32 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 01:20.29 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 01:20.54 | lee_is_me | Any care to answer a polycom .cfg file question? |
| 01:22.29 | [TK]D-Fender | lee_is_me, You should try ASKING it :) |
| 01:25.47 | [TK]D-Fender | *crickets* |
| 01:27.51 | *** join/#asterisk Fieldy (i=IPbyox5q@gentoo/contributor/Fieldy) |
| 01:29.36 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 01:33.17 | Corydon76-home | JT: heh. How old am I? |
| 01:33.36 | JT | OLLDD i think |
| 01:33.37 | JT | :P |
| 01:34.11 | Corydon76-home | It's sobering to think that Mark is a year younger than me. |
| 01:34.39 | JT | i have no idea how hold he is... 30? |
| 01:34.47 | Corydon76-home | Correct |
| 01:34.49 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:34.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:35.03 | Corydon76-home | His age is on Wikipedia, btw |
| 01:35.04 | JT | wow what an awesome guess :o |
| 01:35.14 | JT | okay |
| 01:35.31 | [TK]D-Fender | Corydon76-home, I was in the middle of googling it up :) |
| 01:35.37 | Corydon76-home | Maybe one day I'll be worthy of a Wikipedia entry. |
| 01:35.47 | JT | he must look exactly his age |
| 01:35.57 | JT | i was just guessing off seeing him at CeBit Sydney |
| 01:45.25 | cy303 | Any cheap-o FXS recommendations? |
| 01:45.41 | cy303 | My FXO I got for $10 and it rocks, but I'm not finding an ubercheap FXS card.. just need one port :/ |
| 01:46.10 | cy303 | PCI.. |
| 01:46.19 | [TK]D-Fender | cy303, Screw PCI FXS, get an ATA |
| 01:46.39 | cy303 | hm |
| 01:46.46 | [TK]D-Fender | cy303, ATA's also cost a lot less than PCI per port and are more flexible. |
| 01:46.50 | Nugget | You can pick up a sipura spa-3000 for pretty cheap on ebay |
| 01:47.05 | cy303 | Nugget: /me peeps |
| 01:47.06 | [TK]D-Fender | cy303, SPA-2002 is what you should aim for. |
| 01:47.08 | Nugget | it'll be a zillion times better than your clone x100p and it'll have an fxs on it |
| 01:47.11 | cy303 | oh? |
| 01:47.39 | [TK]D-Fender | cy303, dunno about a "zillion" but if you have any complaints, the SPA-3XXX series might be better. |
| 01:48.01 | cy303 | hmm, I'm certainly not familiar with ATA's |
| 01:48.44 | cy303 | so it just grabs a dhcp address .. |
| 01:50.35 | [TK]D-Fender | its a network device, typically programmed through a simple web interface. it talks SIP to your * server and you just plug a borking phone in. |
| 01:50.42 | [TK]D-Fender | boring* |
| 01:51.06 | [TK]D-Fender | gives you all the features you're used to : CW, CID, 3-way call, transfer, hold, etc.... |
| 01:51.27 | Corydon76-home | Basically it's a SIP trunk |
| 01:51.30 | cy303 | that's cool |
| 01:51.55 | cy303 | http://cgi.ebay.com/Linksys-Sipura-SPA-2002-Analog-Telephone-Adapter_W0QQitemZ110124633564QQihZ001QQcategoryZ11908QQrdZ1QQssPageNameZWD1VQQcmdZViewItem |
| 01:52.06 | cy303 | buy now $79.95 |
| 01:52.14 | JT | spa-3102 is worth looking at too |
| 01:52.15 | [TK]D-Fender | cy303, and if speaking G.711 to * incurs extremely little resources and you don't have to muck around with PCI settings and can distance it from your server |
| 01:52.17 | JT | successor to 3000 |
| 01:52.40 | [TK]D-Fender | cy303, SPA-3102 is better than its predecessor and it $75 NEW |
| 01:52.46 | cy303 | haha well then |
| 01:53.36 | [TK]D-Fender | cy303, Again your choice of SPA-2002 VS SPA-3102 depends on your expectations for using it as an FXO as well |
| 01:54.22 | [TK]D-Fender | cy303, If you are indeed happy with your FXO card, then get and SPA-2002 as you'll get 2 FXS on it |
| 01:55.04 | [TK]D-Fender | out for a bit, back later... |
| 01:55.13 | cy303 | yeah, that sounds better to me |
| 01:55.48 | cy303 | so you could basically take that box with you over to say .. another state in the US or something, plug it into a network and register an analog phone to your * box |
| 01:55.53 | cy303 | that's kinda dope |
| 01:56.22 | JT | if you mean "that's awesome!", then yes |
| 01:56.24 | JT | :) |
| 01:56.29 | *** join/#asterisk fbffff (n=fbffff@adsl-66-73-4-221.dsl.chcgil.ameritech.net) |
| 01:56.33 | cy303 | hehe |
| 01:56.36 | cy303 | yea |
| 01:56.52 | cy303 | well screw it, /me orders |
| 01:57.10 | JT | sorry, allergic to gangster slang :P |
| 01:57.26 | JT | but yes it could do that, as long as asterisk is setup correctly |
| 01:57.48 | cy303 | SIP behind NAT sure can be a bitch |
| 01:58.15 | JT | you'd need asterisk to be on a public ip or to be port forwarded |
| 01:58.37 | cy303 | yeah my * boxes are setup correctly for that |
| 01:58.49 | cy303 | just saying I had some struggles with SIP/NAT |
| 01:59.02 | cy303 | actually mostly just shitty routers |
| 02:00.46 | *** part/#asterisk niedobry (n=bbrindle@ip24-254-142-122.rn.hr.cox.net) |
| 02:01.42 | *** join/#asterisk fholmes (n=fholmes@cpe-68-201-251-217.houston.res.rr.com) |
| 02:02.19 | `Sean | guts |
| 02:02.23 | `Sean | anyone know a email |
| 02:02.24 | `Sean | for bkw |
| 02:03.38 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 02:05.50 | flenders | hey, what's the best way to know if a call is incoming or outgoing? on the CLI? AMI? |
| 02:06.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) [NETSPLIT VICTIM] |
| 02:07.50 | Nugget | I fill my dialplan with NoOp() breadcrumbs so that the console tells me a lot about what's going on |
| 02:08.13 | Nugget | ignore the warnings and put some _. extensions with NoOps and it'll do wonders |
| 02:10.27 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:13.12 | *** join/#asterisk mozDown (n=fdsa@c-24-15-226-98.hsd1.il.comcast.net) |
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| 02:54.54 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:07.03 | *** join/#asterisk iote_ (n=iote@mail.lpj.com.br) |
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| 03:15.16 | *** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 03:16.06 | dracosilv | question... why do i need to be 'identified' to be able to join this channel, and what channelflag sets such a thing up? |
| 03:25.43 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 03:38.10 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 03:38.33 | linagee | LOL. what apple wanted you to have 20 years ago. http://video.google.com/videoplay?docid=-5144094928842683632 |
| 03:38.45 | linagee | make asterisk do that! hehehe. :-> |
| 03:39.22 | Corydon76-home | dracosilv: +r, and it's because we've gotten a lot of botnet abuse |
| 03:40.05 | dracosilv | *nods* |
| 03:49.28 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 03:50.19 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:50.28 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 03:51.56 | *** join/#asterisk threat (i=phix@60-240-43-214.static.tpgi.com.au) |
| 03:52.02 | threat | hey |
| 03:52.07 | threat | I have more problems |
| 03:52.21 | threat | fax detect works on some faxes but not on all |
| 03:52.31 | threat | what would cause this and how do I debug / troubleshoot it/ |
| 03:52.32 | threat | ? |
| 03:52.44 | *** join/#asterisk bmg505 (n=leon@196.209.182.194) |
| 03:53.53 | Qwell | hmm, 4.2.2.x are having issues this weekend.. anybody got a good dns server? |
| 03:55.27 | *** part/#asterisk zodell (n=Odell@206.248.3.49) |
| 03:56.22 | Corydon76-home | Qwell: 129.59.1.10 and 129.59.2.10 |
| 03:57.14 | threat | Qwell, G'day |
| 03:57.35 | variable_office | as a matter of opinion, if you were going to setup a new voip network; would you have all the users connecting to asterisk and run openser as a asterisk->asterisk switch || (or) run openser as what the users are connecting to and just use asterisk for voicemail and the like?\ |
| 03:58.00 | JT | what the users connect to |
| 03:58.18 | JT | i see little advantage in putting it as an interface between different servers |
| 03:58.23 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 03:58.37 | variable_office | JT, what would the users connect to? |
| 03:58.41 | Corydon76-home | As a matter of opinion, I wouldn't use openser at all, unless I was trying to load balance servers |
| 03:58.42 | *** join/#asterisk azamba (n=azambuja@189.4.234.74) |
| 03:58.55 | JT | variable_office: openser |
| 03:59.37 | variable_office | Corydon76-home, how would asterisk figure out whois where? |
| 04:00.13 | Corydon76-home | variable_office: let the proxy figure it out |
| 04:00.35 | variable_office | JT, is the NAT support with mediaproxy on openser as good as the support with asterisk? (i havent gotten to the nat in openser yet) |
| 04:01.26 | threat | so any reason why a fax wouldn't be detected? |
| 04:02.01 | JT | variable_office: i'm not sure, but i believe there's a nathelper module |
| 04:02.06 | variable_office | Corydon76-home, so let each asterisk box be capable of figuring out where the call belongs? |
| 04:02.20 | variable_office | JT you running openser w/ mediaproxy? |
| 04:02.27 | JT | threat: flakey code, flakey dialplan, flakey fax tone, pick one :) |
| 04:02.33 | JT | variable_office: nup |
| 04:02.37 | Corydon76-home | variable_office: basically |
| 04:02.45 | JT | i wouldn't use mediaproxy myself, as it's python |
| 04:02.56 | JT | as much as i like python |
| 04:03.05 | JT | it shouldn't be doing something so low level |
| 04:03.11 | threat | JT, ok :) well I can send a fax to this number, so can other people, it is just a few people cannot, it just rings the phone instead of getting picked up as a fax |
| 04:03.17 | threat | JT, how would I troubleshoot this? |
| 04:03.23 | threat | what commands? |
| 04:03.49 | variable_office | JT, so if you had users running nat, you would just use asterisk? |
| 04:04.14 | threat | JT, BTW, I have a tdm400p |
| 04:05.00 | JT | variable_office: if i had a lot of users, i would use openser, see no reason why it can't do the job of proxying |
| 04:05.26 | JT | threat: i don't know... listen to the fax tone, play with timeous? |
| 04:05.37 | variable_office | well openser wants you do use either mediaproxy or rtpproxy |
| 04:06.04 | JT | rtpproxy is C but has less features |
| 04:06.14 | JT | and proxying media is not mandatory in openser |
| 04:06.32 | variable_office | if users are behind nat it is though, correct? |
| 04:06.40 | JT | pretty much |
| 04:06.42 | JT | generally |
| 04:06.58 | threat | JT, interesting, any useful asterisk debug options I Can use? |
| 04:07.26 | JT | threat: why do you keep asking the same question over and over? no, not that i know of. |
| 04:07.51 | variable_office | i guess theres no real reason i cant use some random combination of the two |
| 04:07.52 | threat | JT, ok, well you didn't answer that question before ;) |
| 04:08.09 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:08.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:08.13 | JT | threat: if i don't know the answer, i cannot answer a question |
| 04:08.19 | JT | i don't like feeling harrased |
| 04:08.36 | threat | JT, ok |
| 04:08.45 | threat | JT, you didnt mention that the first time |
| 04:09.00 | threat | the fact that you didn't know the answer, I assumed you didn't read that bit |
| 04:09.18 | JT | it's common irc etiquette to not keep asking the same question over and over :) |
| 04:09.24 | JT | i don't need to specifically state that |
| 04:09.30 | JT | i answered what i could |
| 04:10.29 | threat | ok and I am greatful |
| 04:10.44 | threat | I will just wait here until some one answers my question thrn |
| 04:11.22 | JT | or you could follow the suggestions i gave, your choice :) |
| 04:13.35 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 04:14.11 | variable_office | any idea on whether mediaproxy beats asterisk in simultaneous calls? |
| 04:14.58 | JT | well it doesn't do the same job as asterisk |
| 04:15.18 | JT | but i believe it can handle a hell of a lot more calls than asterisk, especially as a front end to asteriskl |
| 04:15.22 | JT | -l |
| 04:15.52 | variable_office | doesnt mediaproxy actually go through all the rtp data just as asterisk does? |
| 04:18.51 | JT | variable_office: yes but it's a proxy, asterisk isn't |
| 04:21.02 | variable_office | is asterisk considered an application server? |
| 04:21.02 | JT | i guess |
| 04:21.02 | JT | <PROTECTED> |
| 04:21.04 | variable_office | whats b2bua? |
| 04:21.15 | JT | Back To Back User Agent |
| 04:21.19 | JT | in SIP terms |
| 04:26.22 | JT | which is why it's not a Proxy |
| 04:26.22 | variable_office | ah |
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| 05:45.43 | *** join/#asterisk CapriCorn^80 (n=tellu@209.8.41.52) |
| 05:46.32 | CapriCorn^80 | hi ! i need howto configure asterisk between LAN or between two peer to peer softwares |
| 05:46.39 | CapriCorn^80 | hi ! i need howto configure asterisk between LAN or between two peer to peer computers |
| 05:46.58 | CapriCorn^80 | sorry its not softwares . its computer |
| 05:47.00 | JT | ~thebook |
| 05:47.01 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:48.58 | CapriCorn^80 | hi ! i need howto configure asterisk between LAN or between two peer to peer computers |
| 05:49.35 | JT | CapriCorn^80: stop that. |
| 05:49.41 | JT | CapriCorn^80: and read the link i sent |
| 05:51.07 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:51.14 | CapriCorn^80 | JT: the book link . |
| 05:51.19 | CapriCorn^80 | ok |
| 05:51.19 | CapriCorn^80 | but can i get some simple howto on it |
| 05:51.31 | JT | check the section on sip.conf |
| 05:53.12 | CapriCorn^80 | ok |
| 05:59.31 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.163.72) |
| 06:00.02 | CapriCorn^80 | JT: thats fine but i have never configured it in paste n this pdf is too long . i just want a perfect howto which got simple steps to configure asterisk between two computers and properly run it |
| 06:00.29 | JT | you don't need to read the whole pdf |
| 06:00.42 | JT | i know of no such "perfect howto" |
| 06:00.47 | *** join/#asterisk jimboballard (n=zic@adsl-68-88-232-39.dsl.hstntx.swbell.net) |
| 06:00.59 | JT | there is no substitute for learning something properly |
| 06:01.10 | CapriCorn^80 | ok |
| 06:01.22 | CapriCorn^80 | i agree |
| 06:01.23 | mosty | there are lots of asterisk tutorials on the web. there's a simple one on o'reilly's website |
| 06:02.01 | CapriCorn^80 | JT: if i ask u i got two sytems one got linux which will be asterisk server and one window client |
| 06:02.16 | CapriCorn^80 | wat things i need for that ? |
| 06:02.27 | CapriCorn^80 | to configure or up my asterisk voip |
| 06:02.48 | JT | you will need a softphone on the windows box |
| 06:03.48 | CapriCorn^80 | x-lite is softphone i guess |
| 06:04.34 | JT | it is |
| 06:04.52 | jimboballard | Anyone out there have a used 1fxo/1fxs card for sale cheap? |
| 06:05.10 | JT | just buy an SPA-3102? |
| 06:05.24 | mosty | CapriCorn^80, there are a bunch of tutorials linked on this page, pick one: http://www.voip-info.org/wiki-Asterisk |
| 06:05.37 | jimboballard | I'm a poor hobbist... |
| 06:05.42 | CapriCorn^80 | n wat settings i need on linux box ? |
| 06:06.15 | JT | jimboballard: that is the cheapest option. |
| 06:06.22 | JT | except maybe an SPA-3000 |
| 06:06.30 | JT | which the SPA-3102 replaced |
| 06:07.03 | CapriCorn^80 | mosty: ok |
| 06:07.27 | jimboballard | thanks. Can't afford tdm11b. |
| 06:07.28 | CapriCorn^80 | mosty: just looking for perfect setup of it like how to |
| 06:08.37 | mosty | CapriCorn^80, there is no such thing |
| 06:08.53 | JT | ~hafc |
| 06:09.11 | jbot | it has been said that hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 06:09.13 | nynepac | i have an spa2002 hooked up to a few standard phones. i have it configured to talk to my asterisk box. i am having no issues with incoming calls but outgoing calls just don't work.. i have the most "basic" asterisk setup possible whereas i want to recieve calls from one sip provider and make them using another. my provider (for outgoing/termination) gafachi, only supports ulaw.. does anyone have experience getting outgoing calls to wo |
| 06:09.13 | JT | a "perfect" quick setup is to get someone else to do it for you |
| 06:13.24 | jimboballard | Anyone using SPA-3102 with success? In and outbound pots? |
| 06:16.47 | jimboballard | Anyone using 1.4 and FWD (IAX i think). Not working for me. |
| 06:16.58 | n0n4m3 | you think you're using iax? |
| 06:17.17 | n0n4m3 | i'm using sip and iax2 on asterisk 1.2 |
| 06:17.43 | n0n4m3 | why don't you check the sip.conf and iax.conf and decide what are you using |
| 06:18.14 | flenders | jimboballard: I use an SPA3000 |
| 06:18.20 | flenders | it works alright |
| 06:19.14 | jimboballard | cool! Thanks. Looking for cheep alternative! |
| 06:20.24 | jimboballard | n0n4m3: have beentrying to use iax per the fwd asterisk forums. |
| 06:21.55 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 06:21.55 | jimboballard | Am i going down the wrong path? |
| 06:24.32 | jimboballard | Installed OpenSuse 10.2. Recompiled kernel. Compiled 1.4 and zaptel. Installed several x-lites on my lan. |
| 06:25.12 | jimboballard | Works great in house, but no outside connections yet. |
| 06:25.19 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-127-146.w90-36.abo.wanadoo.fr) |
| 06:26.39 | jimboballard | Looking for free sip server. Ideas? |
| 06:27.06 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:27.18 | Juggie | hah |
| 06:27.43 | Juggie | jimboballard, asterisk is a sip server. |
| 06:28.33 | jimboballard | yes, but, i need to connect to outside world. |
| 06:28.57 | JT | connecting to what? |
| 06:30.12 | jimboballard | I just want to be able to call my asterisk box through the internet from out in the field. |
| 06:31.05 | JT | why would you need an outside sip server for that? |
| 06:34.00 | jimboballard | Maybe I'm confused but : my box is behind a firewall with a dynamic ip address. Wouldn't I need to connect to a service such as FWD that provides directory and gateway services? |
| 06:34.36 | Juggie | if the firewall is not under your control then possibly |
| 06:35.11 | JT | get a dynamic dns service |
| 06:35.21 | JT | and make sure the asterisk server is reachable |
| 06:37.18 | *** join/#asterisk friedrich| (n=friedric@e177242181.adsl.alicedsl.de) |
| 06:38.19 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 06:38.42 | jimboballard | The firewall is under my control, but without a dedicated ip address, i won't be able to reach it from outside without knowing that ip. |
| 06:39.04 | jimboballard | It's my understanding that... |
| 06:39.09 | JT | i just mentioned get a dynamic dns service, problem solved. |
| 06:39.55 | jimboballard | Name one, please. |
| 06:40.03 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 06:40.34 | JT | google for it, there's heaps |
| 06:40.50 | friedrich| | http://dyndns.com |
| 06:41.23 | jimboballard | Thanks, all. |
| 06:42.08 | Juggie | i recomend http://www.everydns.net provided you own your own domain |
| 06:42.14 | Juggie | you can setup dynamic dns on your own domain there. |
| 06:43.40 | jimboballard | I don't. Just a lowly dynamic dsl.Thanks, though. |
| 06:46.31 | jimboballard | I'll look into those and educate myself some. Thanks all. Signing off... |
| 06:47.20 | *** part/#asterisk jimboballard (n=zic@adsl-68-88-232-39.dsl.hstntx.swbell.net) |
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| 07:33.11 | CapriCorn^80 | sorry JT i got dc |
| 07:33.25 | CapriCorn^80 | thx for ur information |
| 07:33.35 | CapriCorn^80 | but i asked some simple question |
| 07:33.47 | CapriCorn^80 | i mean wat i required on linux box to configure asterisk ? |
| 07:34.25 | JT | what do you mean? |
| 07:34.26 | mosty | CapriCorn^80, nothing special. just follow any tutorial |
| 07:39.48 | n0n4m3 | CapriCorn^80 you need a text editor |
| 07:40.44 | CapriCorn^80 | ok |
| 07:41.07 | CapriCorn^80 | can u tell me the exact lines from where i should start ? |
| 07:42.30 | n0n4m3 | CapriCorn^80 http://www.voip-info.org/wiki/index.php?page=Asterisk |
| 07:43.03 | n0n4m3 | and check out the http://www.voip-info.org/wiki/index.php?page=Asterisk#Introduction |
| 07:43.14 | n0n4m3 | start with the first link... |
| 07:43.55 | CapriCorn^80 | the first link got so many links |
| 07:44.19 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 07:44.48 | n0n4m3 | Asterisk introduction: An overview for new Asterisk administrators - THE PLACE TO START!! |
| 07:45.08 | JT | you will actually have to read |
| 07:45.11 | n0n4m3 | http://www.asteriskguru.com/ |
| 08:02.28 | nynepac | this is a bit frusterating.. i can call in and out using xlite but when i try to make a call from a regular telephone connected to my spa2002 it just responds with beep beep beep beep after i've dialed a US phone number 1xxxxxxxxxx .. any idea how to debug this.. |
| 08:02.34 | nynepac | incoming works just fine |
| 08:02.53 | JT | does the call come in on the asterisk cli? |
| 08:03.33 | nynepac | no.. i dont see it |
| 08:03.54 | nynepac | but i do see it from xlite :( |
| 08:04.45 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:05.14 | nynepac | any idea how that would happen? |
| 08:05.48 | JT | what is the console verbosity level? |
| 08:06.06 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:06.07 | nynepac | . /usr/sbin/asterisk -vvvvvv -g -dddddd -c |
| 08:06.49 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
| 08:06.55 | *** join/#asterisk saftsack (n=oliver@p54A7D1B2.dip.t-dialin.net) |
| 08:06.56 | Uatec_ | hi |
| 08:07.52 | JT | that would be a verbosity level of 6. |
| 08:08.06 | JT | nynepac: did you hit enter in x-lite after dialling? |
| 08:08.16 | nynepac | yes i do |
| 08:08.36 | nynepac | and in xlite i do see the call status |
| 08:08.45 | JT | do any other numbers work? |
| 08:09.03 | nynepac | i can't call the xlite phone either |
| 08:09.20 | nynepac | and xlite can't call the sipura phone |
| 08:09.32 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:10.14 | JT | have you tried calling anything other than the us phone number? |
| 08:10.59 | nynepac | just tried a friend in sweden.. same behavior.. beep beep beep beep |
| 08:11.37 | nynepac | brasil failed miserably as well |
| 08:12.20 | JT | sounds like a lot of config is incomplete |
| 08:13.32 | nynepac | yeah.. sounds like that to me as well |
| 08:14.19 | nynepac | -- Registered SIP '2001' at 192.168.1.100 port 5061 expires 3600 |
| 08:14.42 | *** join/#asterisk abuyazan (n=khaled@dogbert.palnet.com) |
| 08:14.47 | abuyazan | hello |
| 08:15.26 | abuyazan | how can i delete all messages in my voicemail ? |
| 08:16.12 | nick125_lappy | abuyazan: You should be able to clear out /var/spool/asterisk/voicemail/context/extension/ |
| 08:16.22 | nick125_lappy | That might delete your greetings though |
| 08:16.26 | nynepac | is there a simple startup guide ? |
| 08:16.35 | nick125_lappy | So, try deleting INBOX and Old |
| 08:16.42 | abuyazan | thanks nick125_lappy |
| 08:16.48 | nick125_lappy | abuyazan: No problem |
| 08:18.09 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
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| 08:30.11 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 08:41.05 | nynepac | what could prevent extensions not being able to speak with one another |
| 08:41.51 | drrt | nynepac, you shouldnt include it each other |
| 08:43.57 | nynepac | ah i see whats going on.. if i swap the context from incoming_fonosip to default in my extension |
| 08:44.01 | nynepac | so 2001 for example |
| 08:44.08 | nynepac | either incoming calls or outgoing calls work |
| 08:46.47 | *** join/#asterisk saftsack (n=oliver@p54A7CC3C.dip.t-dialin.net) |
| 08:48.51 | nynepac | why would either one or the other work properly (incoming or outgoing calls?) mind you incoming and outgoing are 2 different providers |
| 08:51.02 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 09:04.37 | Uatec_ | hey, what do people think about this: |
| 09:04.46 | Uatec_ | http://www.kirkhamsystems.com/asttapi |
| 09:04.54 | Uatec_ | for a way of notifying your PC what's going on with your phone? |
| 09:10.02 | cy303 | Uatec_: what are you trying to do? |
| 09:18.36 | Uatec_ | notify asttapi of what is going on with my call |
| 09:20.10 | cy303 | ahh, not familiar with tapi |
| 09:23.02 | snuffy22 | generally AST TAPI is connected via the manager interface of asterisk |
| 09:23.16 | snuffy22 | actually i think that's the only way it works.. |
| 09:23.26 | cy303 | yeah reading about it on voip-info |
| 09:23.37 | cy303 | some windows front end to * manager interface |
| 09:23.38 | snuffy22 | the manager connection to asterisk will report call status |
| 09:24.01 | snuffy22 | and all sorts of other crap :) |
| 09:25.13 | snuffy22 | generally i thought ast tapi only really did stuff for like making outgoing click to call.. |
| 09:25.28 | snuffy22 | not so much handling incoming events |
| 09:25.42 | snuffy22 | but then i've never really played with it |
| 09:26.08 | *** join/#asterisk LH-euhost (n=LH-euhos@87.97.80.170.pool.invitel.hu) |
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| 09:35.24 | tzafrir | snuffy22, the windows client on each user's computer talks directly with the asterisk manager? |
| 09:36.17 | Uatec_ | ok |
| 09:40.17 | Uatec_ | i think, in that case, that using the Asterisk Manager Interface would generally do everything in that link, and do it in a much easier, concise fashion |
| 09:40.30 | Uatec_ | unless that's doing something else that i'm not seeing |
| 09:40.59 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 09:46.29 | tengulre | I developt a asterisk manager , anybody need it ? |
| 09:47.04 | Uatec_ | what exactly did it do? |
| 09:47.21 | tzafrir | Uatec_, if you trust all of your users: yes. But I asked if there's something in the middle that does not force you to blindly trust the good will of your users not to execute arbitrary commands on the asterisk server |
| 09:50.06 | Uatec_ | would you mind sending it to me so i can assess exactly what it does and home? |
| 09:56.57 | *** join/#asterisk fbffff (n=fbffff@c-67-175-209-231.hsd1.co.comcast.net) |
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| 10:54.57 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 10:57.02 | mattfletcher | Hello, I have two offices running asterisk 1.2 machines, with a site-to-site VPN. I've now been asked to look into the possibility of adding video to the mix. What can asterisk offer me? I list my version number as I understand 1.4 can do a lot more, but I'm reluctant to upgrade when it all works! |
| 11:01.50 | *** join/#asterisk mjmarrio (n=mike@210.19.201.38) |
| 11:09.42 | mjmarrio | hello all. Can anyone perhaps answer a couple of questions about mixing Digium TDM and TE cards? |
| 11:09.48 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
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| 11:21.20 | Uatec_ | mjmarrio, not unless you ask them |
| 11:25.19 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:27.20 | Zeeek | yadayada |
| 11:28.19 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 11:28.20 | zeeesh | hi |
| 11:28.27 | Zeeek | hello |
| 11:29.05 | zeeesh | if i want to copy astcc.agi from server A to server then B then .. how to do it ? |
| 11:30.46 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 11:38.26 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 11:38.39 | mjmarrio | ok here goes: I have installed a TDM2400 with 3xFXO quad modules and one TE205P card. I have not yet connected to ISDN line. I am not sure how to set the "signalling" parameter in zapata.conf. setting fx signalling in zapata.conf is ok but when I set pri_cpe I get errors and the "zap restart" command is no longer available from the CLI console |
| 11:38.46 | eeos | hi everybody |
| 11:39.16 | Zeeek | eeos where's pour homework assignment? |
| 11:39.50 | eeos | Zeeek: have not been able to solve the problem :( |
| 11:39.54 | eeos | (yet) |
| 11:39.57 | mjmarrio | do you set the signalling paramater immediately before the channels? |
| 11:40.11 | tzafrir | mjmarrio, you need to give the right signalling to each channel |
| 11:40.16 | tzafrir | right |
| 11:40.52 | mjmarrio | Something like this: signalling=fxo_ks |
| 11:40.53 | mjmarrio | group=2 |
| 11:40.53 | mjmarrio | channel=1-12 |
| 11:40.53 | mjmarrio | ; |
| 11:40.53 | mjmarrio | signalling=pri_cpe |
| 11:40.54 | mjmarrio | group=1 |
| 11:40.56 | mjmarrio | channel=25-39,41-55 |
| 11:40.58 | mjmarrio | channel=56-70,72-86 |
| 11:40.59 | tzafrir | ~pb |
| 11:41.10 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 11:41.12 | tzafrir | but basically yes |
| 11:41.17 | Zeeek | eeos why not? |
| 11:41.44 | tzafrir | assuming that this will be the order in which the modules will load eventually |
| 11:41.50 | eeos | Zeeek: I do not understand how connection between asterisk and the provider works :( |
| 11:41.51 | mjmarrio | tzafrir: Can I private chat you? |
| 11:42.04 | eeos | Zeeek: I am reading some additional documentation now |
| 11:42.41 | Zeeek | eeos what do you not understand? |
| 11:44.09 | eeos | Zeeek: how do I write the extension so that when a user on the local network decides to open an external call the VOIP line to the provider is called |
| 11:45.06 | Zeeek | eeos it is a good idea to use a context dedicated to users that have the right to do this. It will be useful later |
| 11:45.12 | tzanger | morning |
| 11:45.32 | Zeeek | so if that context were called [users-who-can-call-provider] |
| 11:45.48 | mjmarrio | tzafrir: sorry about that |
| 11:46.12 | Zeeek | eeos under that context you would have an extension that allowed the calls |
| 11:46.13 | mjmarrio | is this the correct way to configure zapata.conf? |
| 11:46.36 | tzafrir | probably. if asterisk starts without an error, then it is |
| 11:46.50 | tzafrir | assuming you actually have chan_zap built |
| 11:46.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 11:47.07 | tzafrir | (if you can call through those card: you have it) |
| 11:47.08 | mjmarrio | I am using asteriskNow |
| 11:47.24 | tzafrir | you're using their packages? |
| 11:47.33 | mjmarrio | no that's the trouble I am getting error messages on the console. |
| 11:47.34 | eeos | Zeeek: yes, I did understand that. But how does this extension connect to the sip provider? I want to use SIP protocol not IAX2. |
| 11:48.00 | mjmarrio | I should be able to call out on fxo even though there is a definition for the isdn |
| 11:48.09 | tzafrir | in asterisknow zaptel.conf and zapata.conf are generally generated files. But only the config for the analog part gets generated |
| 11:48.12 | Zeeek | eeos there are a bout 100,000,000 examples of SIP extensions on the INternet |
| 11:48.17 | mjmarrio | my dial plan specifies to use group 2 which is teh fxs |
| 11:48.34 | tzafrir | You need to edit the "template" files to actually add your own stuff |
| 11:48.44 | Zeeek | eeos begin by looking at the dial application and its syntax |
| 11:48.55 | mjmarrio | yes I know. I acutually edit zapata.conf.zapscan and then run zapscan to create config files |
| 11:49.07 | tzafrir | ok |
| 11:49.26 | tzafrir | so what sign do you have of a problem? |
| 11:49.26 | Zeeek | eeos then you could go look at freeworlddialup.com where there are certainly examples of how to connect to a SIP provider |
| 11:49.50 | eeos | Zeeek: thanks! |
| 11:50.43 | mjmarrio | my ztcfg seems ok but I have the following message at the end of it: CAS signalling on span 3 conflicts with Clear channel on channel 64. |
| 11:50.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:50.45 | Zeeek | intellectual curiousity is your most valuable asset. Get and keep it |
| 11:51.11 | mjmarrio | and also when I try to make a call I get: "Unable to specify channel 1: Device or resource busy" on the CLI console |
| 11:53.14 | eeos | Zeeek: it seems to be easy to have them talk through IAX |
| 11:53.25 | eeos | Zeeek: but I do not understand how to do it through SIP |
| 11:53.35 | Zeeek | eeos why waste time conjectruing when you can go find out what you asked |
| 11:54.00 | mjmarrio | tzafrir: so what do you think? |
| 11:55.01 | Zeeek | eeos GOOGLE asterisk SIP extension - 1.3 millions pages talk about it |
| 11:55.20 | mjmarrio | well, /proc/zaptel all seems ok. All three spans recogninsed...... |
| 11:55.23 | mjmarrio | and.... |
| 11:55.43 | mjmarrio | /etc/zaptel.conf seems to cause no problems. |
| 11:56.14 | mjmarrio | when I add the ISDN signalling in zapata.conf is when I get problems. I'm not sure if I am doing it correctly |
| 11:57.12 | mjmarrio | Also I have added the entries in zaptel.conf in the same order as the /proc/zaptel files |
| 11:57.58 | mjmarrio | any thoughts? |
| 11:59.17 | mjmarrio | As I understand the signalling paramater simply preceeds the channel definitions in zapata.conf |
| 11:59.21 | mjmarrio | is that cor5rect? |
| 12:00.05 | eeos | Zeeek: actually I have been reading half a tonn of documentation found through search engines, but not of much help |
| 12:00.14 | *** join/#asterisk ectospasm (n=ectospas@206.166.212.66) |
| 12:00.22 | eeos | Zeeek: I m afraid there is too much available, and most of it of unexceptional quality |
| 12:00.31 | eeos | Zeeek: anyway we will solve it sooner or later |
| 12:00.34 | Zeeek | eeos well if you do what I just said you'd be loking at something like this: http://www.loligo.com/asterisk/current/extensions.conf |
| 12:01.03 | Zeeek | although the above is old it still works and has many commented examples |
| 12:01.12 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 12:01.15 | mjmarrio | tzafrir: any thoughts? |
| 12:01.18 | eeos | Zeeek: that is the article you sent me few days ago |
| 12:01.31 | Zeeek | well if you read it you would have the answer. |
| 12:01.45 | eeos | Zeeek: yes, I am wading my way through, I have not got to that one yet |
| 12:01.47 | Zeeek | it isn't an article it is a config file |
| 12:02.09 | eeos | ye, but was it not connected to the article |
| 12:02.17 | Zeeek | look at the file and search for @fwd |
| 12:02.35 | Zeeek | the whole extension is there written for you |
| 12:02.38 | eeos | Zeeek: also one of our providers uses openSER,that I do not know at all |
| 12:02.43 | eeos | Zeeek: :) |
| 12:03.06 | eeos | Zeeek: the problem with search engines is to understand what is the documentation you want out of 1.3 millions pages |
| 12:03.27 | Zeeek | no it isn't |
| 12:03.37 | Zeeek | the first 5 examples have what you ask for |
| 12:04.14 | Zeeek | now I'm afraid you'll have to be on /ignore until you've actually read some of it |
| 12:04.33 | eeos | Zeeek: some of what? I have read most of the book as well |
| 12:05.04 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 12:07.23 | mjmarrio | Does this config seem ok for zaptel.conf?? span=1,1,2,ccs,hdb3,crc4 |
| 12:07.24 | mjmarrio | bchan=25-39,41-55 |
| 12:07.24 | mjmarrio | dchan=40 |
| 12:07.24 | mjmarrio | span=2,1,2,ccs,hdb3,crc4 |
| 12:07.24 | mjmarrio | bchan=56-70,72-86 |
| 12:07.25 | mjmarrio | dchan=71 |
| 12:07.42 | tzanger | no |
| 12:07.51 | mjmarrio | how come? |
| 12:07.51 | tzanger | you cannot have both span 1 and 2 the primary clock source |
| 12:07.59 | tzanger | span=1,1,2,... is fine |
| 12:08.05 | tzanger | span 2,1,2,... is not |
| 12:08.12 | tzanger | pick which span you want to clock from |
| 12:08.15 | tzanger | make that '1' |
| 12:08.32 | tzafrir | mjmarrio, what do you see on 'zap show channels'? only "pseudo"? |
| 12:08.34 | tzanger | if you want to clock from the other if the primary is down, use clocktype 2 |
| 12:08.43 | tzanger | if you don't wnat to clock from it ever, use '0' |
| 12:09.11 | mjmarrio | ok tq I will try now |
| 12:11.33 | *** join/#asterisk Wing|wrk (n=wing@newoffice-5.tvcom.ru) |
| 12:12.26 | mjmarrio | When I add isdn stuff into zapata.conf then I lose all zap commands |
| 12:12.27 | mjmarrio | No such command |
| 12:13.14 | mjmarrio | I have changed the clocking. I didn't have ISDN line connected yet so that has prbly solved future problem |
| 12:13.22 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.56 | mjmarrio | my /proc/zaptel files seem a bit strange though |
| 12:15.05 | *** join/#asterisk fbffff (n=fbffff@c-67-175-209-231.hsd1.il.comcast.net) |
| 12:15.32 | mjmarrio | span2 is TE205 port 1 ch25-48 with ch 40 as HDLCFCS but then only goes to ch 48. Seems to be missing 7 channels?? |
| 12:16.50 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 12:16.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:19.07 | ectospasm | today I start my job at Digium... |
| 12:19.24 | Uatec_ | Congrats ectospasm |
| 12:19.27 | Uatec_ | what do you do? |
| 12:19.36 | ectospasm | I'll be in the tech support group |
| 12:19.42 | Uatec_ | oh |
| 12:19.43 | Uatec_ | you start |
| 12:19.45 | Uatec_ | not you have started |
| 12:20.10 | ectospasm | I hear my first two tasks will be to install whatever distro I choose, and then install Asterisk |
| 12:20.21 | ectospasm | It's been two years since I did the latter |
| 12:21.37 | ectospasm | But it was fairly straightforward then... |
| 12:21.37 | file | ectospasm: you should also bring muffins for the tech support team and tell them file told you to get them... muahahahaha |
| 12:21.37 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 12:22.06 | Uatec_ | ectospasm, business edition? |
| 12:22.23 | Uatec_ | ectospasm, what's your real name? incase i end up emailing you at some point :) |
| 12:22.34 | ectospasm | I'll tell you later (-: |
| 12:22.37 | Uatec_ | heh |
| 12:23.20 | drrt | ~paste |
| 12:23.30 | jbot | extra, extra, read all about it, paste is http://rafb.net/paste/ |
| 12:28.41 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 12:29.10 | Uatec_ | ectospasm, make sure that you install pound sign linux on a machine with sata raid (mirrored), and use a b410p isdn interface |
| 12:31.48 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 12:36.09 | eeos | Zeeek: I read the file you sent me, but still cannot understand where is the connection opened :( |
| 12:37.16 | Zeeek | eeos I'm trying to understand what you don'tunderstand |
| 12:37.25 | eeos | Zeeek: where are the login (user id) and the password passed to the SIP provider? |
| 12:37.43 | *** join/#asterisk NirS (i=Nir@87.68.75.85.cable.012.net.il) |
| 12:37.56 | eeos | Zeeek: it looks like the server is opening a connection without passing the information for the connection. how is that possible? |
| 12:38.00 | Zeeek | did you see this? exten => _7.,3,Dial(SIP/${EXTEN:1}@fwd) |
| 12:38.05 | eeos | I keep going back to the same point. |
| 12:38.20 | eeos | yes! but where is the login/password passed o? |
| 12:38.25 | eeos | s/o/on |
| 12:38.41 | Zeeek | in sip.conf under the fwd] [peer |
| 12:38.56 | Zeeek | in sip.conf under the [fwd] peer |
| 12:39.33 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 12:40.50 | [TK]D-Fender | Zeeek: exten => _7.,3,Dial(SIP/fwd/${EXTEN:1}) <- more appropriate |
| 12:42.01 | eeos | [TK]D-Fender: is it new syntax? |
| 12:42.04 | Zeeek | why? the other works AFAIK |
| 12:42.32 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136847.home.otenet.gr) |
| 12:42.57 | eeos | Zeeek: I cannot believe, you are right |
| 12:43.02 | eeos | :8 |
| 12:43.04 | Zeeek | eeos did you go look at the sip.conf file? |
| 12:43.26 | [TK]D-Fender | eeos: No, but the latter can have issues like DNS attempting to resolve it because of the naming structure and I've seen some scenarios where it just "doesn't work" for some reason. |
| 12:43.31 | Zeeek | I'm sure this is laid out perfectly well in the book |
| 12:44.15 | Uatec_ | there's a book? |
| 12:44.20 | [TK]D-Fender | ~book |
| 12:44.34 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:45.01 | eeos | Zeeek: yes, probably it is. so it is just treated as a "normal" user |
| 12:45.06 | *** join/#asterisk viperdude (n=jon@195.74.96.120) |
| 12:45.07 | eeos | Zeeek: well, apart registration |
| 12:45.16 | Zeeek | what about registration? |
| 12:45.29 | *** join/#asterisk jrenzema (n=josh@213.180.89.225) |
| 12:45.39 | eeos | Zeeek: thanks a lot, I was going round in circles. |
| 12:45.52 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136847.home.otenet.gr) |
| 12:45.58 | nasls_lsa | aah , tora ok :) |
| 12:46.05 | Zeeek | yeah me too. I kept pointing you to the same documents |
| 12:46.15 | *** part/#asterisk jrenzema (n=josh@213.180.89.225) |
| 12:46.16 | eeos | Zeeek: yes, apart registration. |
| 12:46.17 | Zeeek | http://automated.it/guidetoasterisk.htm |
| 12:46.31 | *** part/#asterisk nasls_lsa (n=chatzill@athedsl-136847.home.otenet.gr) |
| 12:46.51 | eeos | Zeeek: sorry, that was connected with your previous message not with the last |
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| 12:48.25 | *** join/#asterisk dinesh___ (n=dinesha@80-218-249-64.dclient.hispeed.ch) |
| 12:48.32 | dinesh___ | hi all, I've got a little question |
| 12:48.33 | Zeeek | to make a long story short, Albert King had it right all along |
| 12:48.53 | eeos | Zeeek: who is Albert King? |
| 12:49.03 | Zeeek | Albert King : "Everybody wants to go to heaven but nobody wants to die" |
| 12:49.39 | dinesh___ | i would like to use this: GotoIf($["${CALLERIDNUM}" != "mynumber"]?7) , but the problem that it is always being evaluated to "false", how can I display the value of ${CALLERIDNUM} for debugging purpose ? |
| 12:50.31 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 12:50.47 | [TK]D-Fender | dinesh___: NoOp(${CALLERIDNUM}) |
| 12:50.48 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 12:51.05 | [TK]D-Fender | dinesh___: Hoever that variable was deprecated in 1.2 and removed in 1.4 |
| 12:51.47 | dinesh___ | oh |
| 12:51.48 | [TK]D-Fender | dinesh___: You should use the new function for 1.2+ ${CALLERID(num)} |
| 12:51.52 | dinesh___ | i'm actually using 1.4 |
| 12:51.53 | Zeeek | [TK]D-Fender has been deprecated in 1.6, don't listen to him |
| 12:52.50 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:53.06 | dinesh___ | thank you very much |
| 12:53.18 | dinesh___ | <PROTECTED> |
| 12:53.51 | eeos | Zeeek: :p |
| 12:57.56 | dinesh___ | hm, and why doesnt this work, I'm getting some syntax error things when it is being executed: |
| 12:57.58 | dinesh___ | exten => 1000,3,GotoIf($["${CALLERID(num)}" == "0795648910"]?100 |
| 12:58.27 | dinesh___ | [Jun 4 14:53:19] NOTICE[445]: pbx.c:1702 pbx_substitute_variables_helper_full: Error in extension logic (missing ']') |
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| 12:59.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 12:59.08 | viperdude | hi all |
| 12:59.27 | viperdude | anyone get this message when attempting to transfer a call " Supervised transfer requested, but unable to find callid" ? |
| 12:59.29 | dinesh___ | oh nevermind the tailing ) is missing |
| 13:00.12 | tbic | can you make the AGI Call GET DATA play more than one sound file? |
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| 13:02.49 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:03.53 | [TK]D-Fender | dinesh___: And it should be "=", not "==" |
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| 13:06.44 | cy303 | anyone here using callwithus? |
| 13:08.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:10.24 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 13:16.25 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:16.26 | Katty | morning. |
| 13:16.37 | [TK]D-Fender | Katty: *yawn* Mew. |
| 13:18.04 | iote_ | for correct peer matching in realtime sip peers i need "fullcontact" set to "sip:exten@userip"... how do I populate on the fly the fullcontact with information from exten and userip? im using res_conf_ldap.so |
| 13:20.30 | iote_ | of course, i can create an attribute for this on ldap, but since I already got the necessary information in other attributes i want to find a way to build it on realtime request... |
| 13:21.40 | cpm | morn'n |
| 13:22.45 | iote_ | on res_ldap.conf i have "attribute = fullcontact => AstAccountFullContact" it would be nice to have something like: attribute = fullcontact => "sip:".$AstExten."@".$AstIPaddress |
| 13:22.50 | mjmarrio | tzafrir: Now I see only pseudo |
| 13:23.11 | tzafrir | could you pastebin your zapata.conf ? |
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| 13:33.58 | jeremy_g | hi |
| 13:40.30 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
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| 13:41.18 | lee_is_me | [TK]D-Fender: ping |
| 13:43.12 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:43.56 | *** part/#asterisk sventek (n=sventek@c211-30-131-80.carlnfd4.nsw.optusnet.com.au) |
| 13:43.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:45.34 | puzzled | hi |
| 13:45.42 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:45.50 | [TK]D-Fender | lee_is_me: Pong |
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| 13:47.37 | msetim | Hi |
| 13:47.42 | lee_is_me | [TK]D-Fender: Sorry to keep you hanging yesterday. |
| 13:47.53 | msetim | I have been installed the asterisk now for test purposes |
| 13:47.59 | [TK]D-Fender | lee_is_me: Trust me... I didn't lsoe any sleep over it :) |
| 13:48.06 | msetim | What is the password default of front-end? |
| 13:48.24 | [TK]D-Fender | msetim: * doesn't have a "front-end". |
| 13:48.28 | lee_is_me | [TK]D-Fender: I'm sure. But I did have a question on polycom cfg files |
| 13:48.53 | [TK]D-Fender | msetim: If you're talking about AsteriskNOW or FreePBX/Trixbox, please read the channel topic for support channels for them |
| 13:49.07 | [TK]D-Fender | lee_is_me: Ah yes... so ask away and we'll see what we can do. |
| 13:49.14 | msetim | [TK]D-Fender: I'm talking about Asterisk GUI that came with it |
| 13:49.32 | lee_is_me | [TK]D-Fender: reg.x.server.y.address ==> x = line, but what is y stand for? |
| 13:49.37 | [TK]D-Fender | msetim: Feel free to ask in #asterisknow or #asterisk-gui |
| 13:49.54 | msetim | [TK]D-Fender: thanks :-D |
| 13:49.55 | *** join/#asterisk Corydon76-home (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 13:49.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:50.10 | Katty | cory |
| 13:50.13 | [TK]D-Fender | lee_is_me: Fall-back server. if the primary goes down you can specify a failover server for redundency |
| 13:50.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:50.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:50.51 | lee_is_me | [TK]D-Fender: Thanks, but I'm trying to figure out what "y" is a variable for. Am I mistaken? |
| 13:51.04 | [TK]D-Fender | lee_is_me: 1 or 2 |
| 13:51.06 | anthm | heya |
| 13:51.15 | [TK]D-Fender | lee_is_me: as per the Admin Guide |
| 13:51.16 | MindTheGap | in realtime sip peers i need "fullcontact" set to "sip:exten@userip"... how do I populate on the fly the fullcontact with information from exten and userip? im using res_conf_ldap.so |
| 13:51.22 | MindTheGap | on res_ldap.conf i have "attribute = fullcontact => AstAccountFullContact" it would be nice to have something like: attribute = fullcontact => "sip:".$AstExten."@".$AstIPaddress |
| 13:51.36 | lee_is_me | [TK]D-Fender: I'm looking at it, but am still confused. |
| 13:51.51 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 13:52.03 | [TK]D-Fender | lee_is_me: You're not building it from scratch, are you? |
| 13:52.29 | lee_is_me | [TK]D-Fender: I'm thinking of writing a GUI for polycom config |
| 13:53.18 | Katty | ack! |
| 13:53.35 | Katty | heh |
| 13:54.01 | lee_is_me | [TK]D-Fender: I see that x = the line being registered. IE 1 and/or 2 with the 301. I'm not certain what the y variable is for. Is for which server this applies to? |
| 13:56.53 | [TK]D-Fender | lee_is_me: Yes, thats what I jsut said |
| 13:57.19 | lee_is_me | [TK]D-Fender: LOL. Just making sure. |
| 14:00.13 | e-ddie | heh |
| 14:00.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
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| 14:00.37 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:01.31 | tzanger | it's gotta suck to be a female in a geeky-male dominated channel |
| 14:01.43 | anonymouz666 | hehe |
| 14:01.48 | Mercestes | unless your into geeky males |
| 14:01.50 | *** join/#asterisk digus (n=digus@206.222.110.30) |
| 14:02.28 | anonymouz666 | geeky males aka nerds don't care much about woman anyway |
| 14:02.30 | anonymouz666 | hehe |
| 14:04.21 | jkiff | For the ladies, the odds are good, but the goods are odd. :-P |
| 14:04.58 | tzanger | hahahahaha |
| 14:05.07 | *** join/#asterisk fbffff (n=fbffff@adsl-66-73-4-221.dsl.chcgil.ameritech.net) |
| 14:05.24 | Mercestes | Very nice. |
| 14:06.04 | msetim | [TK]D-Fender: asterisk-gui password is set on installation |
| 14:10.45 | [TK]D-Fender | jkiff: definite winner... |
| 14:10.52 | [TK]D-Fender | msetim: I'll take your word for it. |
| 14:15.51 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.59 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:18.54 | Katty | tzanger: it's not that bad ;) |
| 14:19.47 | *** join/#asterisk BSD_Tech (n=BSDTech@ppp-71-128-6-42.dsl.irvnca.pacbell.net) |
| 14:19.47 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net) |
| 14:19.55 | SuPrSluG | hello all |
| 14:21.18 | SuPrSluG | can anyone recommend a decent inexpensive router? I just had a linksys crap out (not forwarding anything). |
| 14:22.04 | SuPrSluG | there are only 4-5 users at any given time. thanks. |
| 14:23.36 | [TK]D-Fender | SuPrSluG: Buy another Linksys router. They generally work jsut fine. |
| 14:24.01 | [TK]D-Fender | SuPrSluG: many D-Links screw up NAT, and most other cheap ones well... ick |
| 14:24.17 | viperdude | anyone get this message when attempting to transfer a call " Supervised transfer requested, but unable to find callid" ? |
| 14:24.24 | BSD_Tech | Superslug go to the linksys site and get the updated firmware for that router |
| 14:24.28 | Mercestes | SuPrSluG, Go get a wrt54gl and put linux on it. |
| 14:24.37 | SuPrSluG | they had a power outage and since pass nothing. i'm resetting to defaults |
| 14:24.47 | Mercestes | iptables ftw |
| 14:25.15 | SuPrSluG | i've done openwrt. maybe i'll do that |
| 14:25.26 | BSD_Tech | Superslug what model linksys do you have ? |
| 14:25.38 | SuPrSluG | wrtG |
| 14:25.50 | *** join/#asterisk _omer (n=omer@lhr-mp-dig-p11-249.brain.net.pk) |
| 14:25.53 | BSD_Tech | put openwrt on it |
| 14:26.17 | tzanger | Katty: :-) |
| 14:26.17 | *** part/#asterisk Wing|wrk (n=wing@newoffice-5.tvcom.ru) |
| 14:26.42 | BSD_Tech | Get a room |
| 14:26.57 | tzanger | I just got a treo 700wx, upgrade from 650 |
| 14:26.57 | BSD_Tech | but make sure to setup the webcams first |
| 14:27.02 | tzanger | not entirely sure it's an upgrade yet |
| 14:27.05 | tzanger | BSD_Tech: ha |
| 14:27.10 | tzanger | I'm engaged, I wouldn't do that |
| 14:28.51 | BSD_Tech | ok who is the lucky guy/woman ? |
| 14:29.15 | _omer | Micheal Jackson |
| 14:29.18 | _omer | :D |
| 14:29.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:31.15 | phearless | hi guys |
| 14:31.15 | phearless | Is this bug ( http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483 ) fixed in the official 1.4.0 release ? |
| 14:31.53 | BSD_Tech | we are upto 1.4.4 |
| 14:32.49 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 14:34.00 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:34.15 | phearless | it does not answer my question |
| 14:34.32 | phearless | asterisk -r |
| 14:34.32 | phearless | Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others. |
| 14:34.35 | phearless | i am using this version |
| 14:36.08 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 14:36.24 | BSD_Tech | then you way behind |
| 14:36.31 | [TK]D-Fender | <PROTECTED> |
| 14:36.32 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:36.40 | [TK]D-Fender | <PROTECTED> |
| 14:36.54 | [TK]D-Fender | Does this SOUND like it was fixed for 1.4.0 release?! *NO* |
| 14:37.18 | sergee | Is it possible to have anonymous calls (insecure=port,reinvite) from multiple ip? (something like host=192.168.1.0/24) or should i create a separate user entry in config for each IP? |
| 14:37.20 | [TK]D-Fender | 1.4.0 = way before the bug was entered |
| 14:37.38 | sergee | insecure=port,invite ofcause... |
| 14:37.58 | phearless | thank you [TK]D-Fender |
| 14:39.44 | [TK]D-Fender | <- Zen master of the blatantly obvious |
| 14:40.18 | [TK]D-Fender | sergee: use "host=dynamic" and set the allow host/mask for that range. |
| 14:41.11 | sergee | [TK]D-Fender: hmmm, let me check... |
| 14:46.10 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-114-25.w90-36.abo.wanadoo.fr) |
| 14:46.34 | *** join/#asterisk fbffff (n=fbffff@adsl-66-73-4-221.dsl.chcgil.ameritech.net) |
| 14:48.11 | MindTheGap | in realtime sip peers i need "fullcontact" set to "sip:exten@userip"... how do I populate on the fly the fullcontact with information from exten and userip? im using res_conf_ldap.so |
| 14:48.14 | MindTheGap | on res_ldap.conf i have "attribute = fullcontact => AstAccountFullContact" it would be nice to have something like: attribute = fullcontact => "sip:".$AstExten."@".$AstIPaddress |
| 14:48.20 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 14:48.52 | QbY | does anyone know of a easy way to convert Cisco 79xx phones to SIP and running the latest firmware? Editing the OS79XX.txt file is getting old.. |
| 14:50.03 | sergee | [TK]D-Fender: doesn't work :) but my question wasn't correct initialy ... |
| 14:51.08 | sergee | [TK]D-Fender: Is it possible to allow calls from non-existent users (insecure=port,reinvite) from multiple ip? (something like host=192.168.1.0/24) or should i create a separate user entry in config for each IP? |
| 14:51.36 | [TK]D-Fender | sergee: Way I mentioned might do it. |
| 14:51.53 | [TK]D-Fender | sergee: and that should be invite, not reinvite |
| 14:51.54 | sergee | DID provider sends calls from multiple IPs, it uses SIP username as CallerID... |
| 14:52.29 | sergee | [TK]D-Fender: unfortunately it doesn't work.. |
| 14:54.04 | [TK]D-Fender | sergee: Ok well you can allow completely un-auth'd calls off [general] |
| 14:54.26 | [TK]D-Fender | sergee: And use a _. to match everything, then check the channel for IP. |
| 14:54.41 | [TK]D-Fender | sergee: Ugly.... but being * that should come as no surprise ;) |
| 14:55.24 | sergee | [TK]D-Fender: :) i'll check source code now, i suppose it shouldn't be hard to add this functionality... |
| 14:58.21 | *** join/#asterisk centrex (i=pputman@nat/digium/x-457233553e9e7e17) |
| 15:00.00 | *** join/#asterisk ctooley (n=ctooley@209.33.108.198) |
| 15:00.05 | *** part/#asterisk ctooley (n=ctooley@209.33.108.198) |
| 15:00.12 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 15:00.14 | *** join/#asterisk bbryant (i=brett@nat/digium/x-1117da6763b4032b) |
| 15:00.31 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:01.42 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 15:01.53 | ifnotwhynot | any fax experts here? |
| 15:02.03 | ifnotwhynot | asterfax that is? |
| 15:04.19 | sergee | ~asterfax |
| 15:04.46 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 15:06.38 | [TK]D-Fender | ~twit |
| 15:06.40 | jbot | Yeah I think ifnotwhynot isn't to bright either... |
| 15:07.21 | [TK]D-Fender | ~botsnack |
| 15:07.21 | jbot | [TK]D-Fender: thanks |
| 15:08.05 | *** join/#asterisk evgeni_71 (n=jaymz_r@207.91.46.139) |
| 15:08.43 | *** join/#asterisk bts3685|work (n=bts3685|@pool-71-244-104-167.phlapa.fios.verizon.net) |
| 15:11.18 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:12.27 | *** join/#asterisk fbffff (n=fbffff@c-67-175-209-231.hsd1.il.comcast.net) |
| 15:13.49 | *** part/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 15:16.30 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 15:16.45 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:16.46 | eeos | hi there |
| 15:18.09 | Mercestes | 'ello |
| 15:25.23 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 15:25.55 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:28.07 | *** join/#asterisk monstertruck (n=monstert@c-75-74-251-82.hsd1.fl.comcast.net) |
| 15:28.32 | *** join/#asterisk angryuser (n=Miranda@213.151.180.176) |
| 15:28.39 | angryuser | good day |
| 15:28.43 | monstertruck | does anyone know of a gateway that can do iax2 and ilbc ? |
| 15:29.54 | angryuser | i need a solution here, i need a contct list, used internally, when i click on phone number, asterisk dials and connect's me (snom phone) how can i do that? |
| 15:30.07 | [TK]D-Fender | monstertruck: LOL |
| 15:30.26 | monstertruck | [TK]D-Fender ? |
| 15:30.36 | [TK]D-Fender | angryuser: Lookup ".call files" and "AMI Originate" on the WIKI |
| 15:30.39 | angryuser | something like sugarcrm but mora basic, just contact list |
| 15:30.40 | [TK]D-Fender | ~wikis |
| 15:30.55 | jbot | wikis is probably http://www.voip-info.org |
| 15:30.56 | angryuser | ok thx |
| 15:32.15 | angryuser | thank you fender ;) |
| 15:33.09 | angryuser | i will develop some application, maybe release it public |
| 15:33.19 | [TK]D-Fender | monstertruck: Very few manufacturers care about IAX2, evern fewer care about ILBC. Decent products are in the "fictitious" category... |
| 15:33.33 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:33.43 | angryuser | i dont need crm machine, just a little "point&call" |
| 15:34.19 | monstertruck | angryuser, web interface with flex, php backend to write a call file |
| 15:34.39 | monstertruck | angryuser, that will make it pretty |
| 15:35.03 | angryuser | monstertruck: no we will integrate it to Windev internal application allready in place |
| 15:35.13 | monstertruck | [TK]D-Fender, so im shit outta luck |
| 15:36.11 | monstertruck | angryuser, here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 15:36.30 | monstertruck | same thing, backend response to user click should be to write a call file |
| 15:36.40 | [TK]D-Fender | monstertruck: what do you actually have to do? |
| 15:37.20 | monstertruck | [TK]D-Fender, i need a bunch of separate fxo in latin america |
| 15:37.27 | angryuser | monstertruck: was on the same page;) |
| 15:37.36 | monstertruck | i could place an asterisk server in each office |
| 15:37.44 | [TK]D-Fender | monstertruck: Mount up an * server then. |
| 15:38.04 | monstertruck | but was lookign for a cleaner solution, all I need is a dumb gateway |
| 15:38.45 | monstertruck | and they are all behind nat, so im trying to avoid the headache with sip |
| 15:40.37 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:41.06 | *** join/#asterisk fbffff (n=fbffff@c-67-175-209-231.hsd1.co.comcast.net) |
| 15:41.18 | monstertruck | [TK]D-Fender, ever used a spa3102 behind nat, accross the net from * ? |
| 15:41.36 | [TK]D-Fender | monstertruck: Yeah, works fine. |
| 15:41.36 | monstertruck | or any other sip gateway for that matter... |
| 15:41.50 | monstertruck | will try one of those then |
| 15:41.53 | monstertruck | thnks |
| 15:42.11 | angryuser | hehe it is allmoust too easy.. * great |
| 15:42.36 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:43.12 | eeos | I keep getting "603 Declined" error |
| 15:43.42 | eeos | :( |
| 15:53.38 | angryuser | bye everybody;) |
| 15:53.39 | *** join/#asterisk salviadud (n=dude@189.156.174.25) |
| 15:53.39 | bts3685|work | asterisk has IPV6 support, yeah? i mean i think it'd matter more on the hardware and kernel, but have there been any issues? |
| 15:53.40 | *** join/#asterisk ecoleman (n=eric@24.75.47.98) |
| 15:55.08 | *** join/#asterisk mightnare (n=mike@p6159-ipad02motosinmat.mie.ocn.ne.jp) |
| 15:56.19 | ecoleman | it's so quiet in here... safe to ask a question? |
| 15:57.08 | mightnare | you're asking one already... ;) |
| 15:57.36 | ecoleman | heh |
| 15:57.48 | ecoleman | we have a dialer macro that checks a few different providers, to see which is available to dial out, and then passes them into a ael context we defined. |
| 15:58.06 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:58.17 | ecoleman | we were passing it to a congesstion to see the response, but for some reason, it continues to pass them into the context after the congestion portion |
| 15:58.44 | [TK]D-Fender | ecoleman: Pastebin what you've made, and the CLI output. |
| 15:58.45 | [TK]D-Fender | ~pb |
| 15:58.50 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:58.51 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 15:59.03 | ecoleman | yeah, i know about PB |
| 16:01.00 | ecoleman | shit i might have figured it out |
| 16:04.32 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:06.42 | *** join/#asterisk ManxPower (n=manxpowe@80.sub-70-220-176.myvzw.com) |
| 16:10.57 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 16:14.36 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 16:15.16 | *** join/#asterisk mfdutra (n=marlon@201.21.222.126) |
| 16:16.01 | mfdutra | hello. I'm receiving this error when asterisk tries to notify a polycom phone of a state change (hints): |
| 16:16.02 | mfdutra | WARNING[8114] chan_sip.c: sip_xmit of 0x83412c8 (len 821) to 192.168.0.239:5060 returned -1: Operation not permitted |
| 16:16.45 | mfdutra | there is no any firewall rules on the output |
| 16:17.01 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 16:17.02 | ManxPower | are you running Asterisk as root |
| 16:17.07 | ManxPower | Is there NAT involved? |
| 16:17.08 | mfdutra | yes |
| 16:17.11 | mfdutra | no |
| 16:17.25 | mfdutra | everything else works ok |
| 16:17.27 | mfdutra | sip, rtp, iax |
| 16:17.39 | mfdutra | that happens only on hints subscriptions |
| 16:17.41 | Hmmhesays | is there something similar to eyebeam for linux? |
| 16:17.49 | ManxPower | other than that does the phone at 192.168.0.239 work correctly. make calls, receiver calls |
| 16:18.00 | mfdutra | yep, perfectly |
| 16:18.05 | errr | anyone know if its possible to get line presence across servers? |
| 16:19.16 | ManxPower | errr: your extensive searching of the mailing lists didn't turn up anything? |
| 16:19.48 | ManxPower | mfdutra: What version of Asterisk? |
| 16:19.54 | mfdutra | 1.4.2 |
| 16:20.04 | ManxPower | ah. |
| 16:20.11 | ManxPower | I can't run 1.4 |
| 16:20.19 | n0n4m3 | how come? |
| 16:20.47 | mfdutra | I can't upgrade to 1.4.4 because of my channel driver. I use a brazilian tdm card. they don't have channel driver for 1.4.4 yet |
| 16:20.47 | ManxPower | It has not been out long enough to have a proven track record. |
| 16:20.47 | mfdutra | although that worked perfecly with 1.2 |
| 16:21.15 | ManxPower | Actually it has a proven track record -- of terrible show stopping bugs. |
| 16:21.33 | Hmmhesays | I don't use 1.4 for anything other than tinkering around |
| 16:23.39 | ManxPower | The LATEST 1.4.x has not been out long enough to know if it has major bugs or not. |
| 16:23.39 | ManxPower | Also, I do not have a lab to test it. |
| 16:23.39 | ManxPower | well, not at the moment at least. |
| 16:23.40 | ManxPower | I do NOT want 200 angry users banging on my door because a shiny new 1.4.x server fails for some reason |
| 16:23.40 | xheliox | Can anyone recommend a loud ringer for a warehouse? (not really an Asterisk question) :) |
| 16:23.40 | lee_is_me | On AMI what is the best way to get list of extensions? SIP SHOW PEERS looks promising, but the output does not seem to provide a way to distinguish between peers that are internal and those that are for ITSP's. |
| 16:23.40 | ManxPower | xheliox: hellodirect.com ? |
| 16:23.40 | mfdutra | it's been working ok. I got only this issue now |
| 16:23.48 | ManxPower | lee_is_me: "show dialplan"? |
| 16:23.54 | xheliox | Hmm, good call. |
| 16:24.02 | ManxPower | mfdutra: what changed between the working and non-working system? |
| 16:24.22 | mfdutra | actually this is a new system, beginning with 1.4.2 |
| 16:24.28 | mfdutra | it's a customer of mine |
| 16:24.46 | mfdutra | in my own office with asterisk 1.2, I have several polycoms working ok with presence watch |
| 16:24.47 | ManxPower | lee_is_me: A SIP DEVICE IS NOT AN EXTENSION!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 16:25.00 | ManxPower | mfdutra: looks like a 1.4.x bug to me. |
| 16:25.06 | lee_is_me | ManxPower: thus my problem. |
| 16:25.08 | mfdutra | maybe |
| 16:25.18 | *** join/#asterisk cayorde (n=flexable@host164-105-dynamic.16-87-r.retail.telecomitalia.it) |
| 16:25.20 | mfdutra | lee_is_me, there is no "list of extensions" |
| 16:25.26 | ManxPower | lee_is_me: A SIP device is a device. You well never ever get a list of extensions with sip show peers. That will give you a list of sip account. |
| 16:25.31 | salviadud | is 1.4 worth the trouble? |
| 16:25.33 | salviadud | i got 1.2 |
| 16:25.35 | salviadud | works fine |
| 16:25.38 | ManxPower | in the CLI, you can do "show dialplan" |
| 16:25.40 | lee_is_me | ManxPower: Got it. |
| 16:25.59 | mfdutra | you can have millions of extensions matching a single rule |
| 16:26.02 | ManxPower | an extension is what maps the logical number to the actual sip.conf entry. |
| 16:26.27 | lee_is_me | ManxPower: Oh my mistake. Not extensions as in the asterisk sense, but extensions as in registered sip devices. |
| 16:26.28 | ManxPower | You can have one extension ring many different devices. |
| 16:26.31 | ManxPower | mfdutra: what is this rule you speak of? |
| 16:26.42 | ManxPower | lee_is_me: then don't use the work "extension". |
| 16:26.48 | mfdutra | sorry? |
| 16:26.56 | lee_is_me | ManxPower: Yeah. My mistake as I said. |
| 16:26.59 | ManxPower | You are asking why, when you shake an apple tree, oranges fall out. |
| 16:27.09 | lee_is_me | ManxPower: LOL. |
| 16:27.11 | ManxPower | it is because you are calling an orange tree and apple tree. |
| 16:27.44 | lee_is_me | ManxPower: so what is the best way to get a list of sip devices that are configured as local phones? |
| 16:27.46 | ManxPower | lee_is_me: a term most people will understand is "sip devices", "sip accounts", "sip.conf entries" (the last one is more correct) |
| 16:27.55 | Hmmhesays | good luck getting chan_sip to handling millions of registrations |
| 16:28.01 | lee_is_me | ManxPower: Thanks. |
| 16:28.13 | ManxPower | lee_is_me: there is no good way. All SIP devices are SIP devices. Asterisk doesn't know or care if they are local or remote. |
| 16:28.32 | ManxPower | The only thing you can really do is look at the IP ADDRESS of the device. |
| 16:28.35 | lee_is_me | So, looks I will have to rely on some kind of naming system. |
| 16:28.48 | ManxPower | or name your local extensions and remote providers differently. What is what we do. |
| 16:29.12 | Hmmhesays | there are many ways you can tackle the problem |
| 16:29.16 | ManxPower | For all phone devices: the SIP IS is the MAC address with a -a -b -c, etc for each line. |
| 16:29.26 | lee_is_me | ManxPower: Well, the norm seems to numerical for internal devices and alph/num for ITSP's |
| 16:29.33 | Hmmhesays | norm? |
| 16:29.36 | Hmmhesays | pffft |
| 16:29.38 | ManxPower | for all service providers: use whatever the hell they tell us to use ad the name. |
| 16:29.46 | lee_is_me | for me |
| 16:29.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 16:30.04 | lee_is_me | from what I have seen in my long tenure of 9 months in the Asterisk biz |
| 16:30.08 | ManxPower | lee_is_me: no, numbering the sip id the same as "the 'extension'" happens because people don't know a thing about bow PBXs work. |
| 16:30.32 | ManxPower | that method is short sighted, limiting, and confusing. |
| 16:30.43 | lee_is_me | ManxPower: I agree. |
| 16:31.06 | Hmmhesays | bingo, your sip id should be something about the device. my sip id's are set to represent where the endpoint is |
| 16:31.15 | lee_is_me | ManxPower: I just need a way to parse out the sip devices in a way that allows me to distinguish internal devices |
| 16:31.36 | ManxPower | The as soon as you want extension "666" to ring 2 or more devices you will start to fully understand how limiting extensions=devices idea really is |
| 16:31.50 | Hmmhesays | <site_id>x<number> |
| 16:32.46 | ManxPower | Extensions 666 is Human Resources, of course, but what if you also want it to ring the VP of marketing. Then you need extension 666 to ring extension 666 and extension 682. |
| 16:33.12 | lee_is_me | ManxPower: Thanks for the heads up. |
| 16:33.27 | ManxPower | or more correctly "Then you need extension 666 to ring device 666 and device 682" |
| 16:33.34 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 16:34.39 | lee_is_me | So, regardless of the syntax used, a naming scheme of some kind seems to be best way to go... |
| 16:34.39 | Hmmhesays | yes |
| 16:34.39 | *** join/#asterisk fbffff (n=fbffff@c-67-175-209-231.hsd1.il.comcast.net) |
| 16:34.39 | lee_is_me | I like Hmmhesays example. I use the same thing in a POS proggie. |
| 16:35.37 | *** join/#asterisk cvspain (n=cvaldess@19.Red-213-98-191.dynamicIP.rima-tde.net) |
| 16:35.38 | Hmmhesays | creating similiar prefix sip id's to group phones based on, well, whatever you want will help you in the long run |
| 16:35.39 | cvspain | Hi |
| 16:35.39 | cvspain | any one have a solution for g729 & astlinux |
| 16:35.46 | lee_is_me | Hmmhesays: Almost like a not notation... |
| 16:35.46 | ManxPower | cvspain: Yes. |
| 16:35.54 | cvspain | MaxPower???? |
| 16:35.57 | Hmmhesays | a "not" notation? |
| 16:36.03 | lee_is_me | Hmmhesays: site.group.phone |
| 16:36.07 | cvspain | have purchaced 4 g729 licenses |
| 16:36.08 | ManxPower | cvspain: Purchase the G729 codec from Digium. |
| 16:36.11 | Hmmhesays | oh yes |
| 16:36.28 | lee_is_me | or assembly referencing if you will... |
| 16:36.51 | ManxPower | cvspain: I really can not help you with astlinux specific issues |
| 16:37.01 | Hmmhesays | I don't know can you use a period as a delimiter in a sip username? |
| 16:37.09 | cvspain | MaxPower> after purchase g729 and try to run register get **: contacting digium ....**: FAILED to contact 'https://register.digium.com/register.php' |
| 16:37.10 | ManxPower | cvspain: Have your tried AsteriskNOW or AsteriskGUI? |
| 16:37.27 | Hmmhesays | the first time around I used an x cause thats what the company wanted |
| 16:37.28 | cvspain | nope, only astlinux |
| 16:37.28 | ManxPower | cvspain: you must contact Digium. |
| 16:37.29 | lee_is_me | Hmmhesays: just an example. or sitexgroupxphone |
| 16:37.40 | Hmmhesays | yep lee_is_me: but you got me curious |
| 16:37.49 | ManxPower | cvspain: I doubt your problem is specific to astlinux. |
| 16:38.05 | lee_is_me | Hmmhesays: I'ts more readable in my opinion for separating values |
| 16:38.12 | ManxPower | Hmmhesays: I'll bet a . will make asterisk think it is an ip address. |
| 16:38.18 | Qwell[] | well, there is also the lack of codec compiled for astlinux (ie; non-x86) |
| 16:38.41 | Hmmhesays | ManxPower: I bet you are right, I'm guessing you would have to escape it |
| 16:38.41 | ManxPower | Qwell[]: so he should contact Digium? |
| 16:38.50 | Hmmhesays | site_group_phone might be better |
| 16:38.58 | ManxPower | Hmmhesays: I don't think asterisk supports escaping in config files. |
| 16:39.04 | ManxPower | or at least sip.conf |
| 16:39.09 | Hmmhesays | extensions.conf does |
| 16:39.41 | Hmmhesays | I have to escape characters for my sql statements |
| 16:40.18 | ManxPower | *nod* |
| 16:40.20 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:40.42 | ManxPower | Qwell[]: aslinux is not x86? |
| 16:41.08 | Qwell[] | isn't it something else? |
| 16:41.15 | Qwell[] | or, maybe there is an x86 version...I don't know |
| 16:41.48 | tzafrir | astlinux? |
| 16:42.05 | *** join/#asterisk JustAGuy1 (n=majortom@adsl-63-195-88-133.dsl.snfc21.pacbell.net) |
| 16:42.14 | Qwell[] | tzafrir: yeah |
| 16:42.30 | tzafrir | astlinux is mainly for x86, though its build system should support some other platform. Not sure if those actually work |
| 16:42.32 | JustAGuy1 | Has anyone successfully connected AOL Phoneline to Asterisk? |
| 16:42.38 | *** part/#asterisk cvspain (n=cvaldess@19.Red-213-98-191.dynamicIP.rima-tde.net) |
| 16:43.23 | JustAGuy1 | (Or if not, anyone know of another free DID provider that gives local US DIDs that can be connected to Asterisk?) :-) |
| 16:43.26 | Daejeo1 | i am looking for a voip provider, can anyone advise? |
| 16:43.34 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:44.10 | Daejeo1 | JustAGuy1: talkdigits,ipkall |
| 16:44.18 | ManxPower | JustAGuy1: you must be a pretty cheap bastard to not want to pay $3/month for a DID. |
| 16:44.24 | Daejeo1 | sipnumbers |
| 16:45.19 | Daejeo1 | voice quality is good |
| 16:45.20 | ManxPower | Daejeo1: Teliax seems to just less that most VoIP providers |
| 16:45.20 | ManxPower | ..e.r.. |
| 16:45.22 | ManxPower | Daejeo1: Teliax seems to suck less that most VoIP providers |
| 16:45.38 | Daejeo1 | ManxPower: let me have a look at Teliax |
| 16:48.18 | JustAGuy1 | IPKall provides local numbers in WA, I need Orlando, Talkdigits doesn't let you pick a city for their free service. |
| 16:48.40 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 16:49.14 | Hmmhesays | heh, hence.. free.. service |
| 16:49.43 | JustAGuy1 | AOL Phoneline is free service as well, but you can pick a city. |
| 16:50.03 | ManxPower | JustAGuy1: I recommend you wait to work with Asterisk until you have some money to spend on it. |
| 16:50.28 | JustAGuy1 | ManxPower Very helpful. |
| 16:50.32 | *** join/#asterisk delphus (n=rodrigog@85.92.130.202) |
| 16:51.02 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 16:51.05 | [TK]D-Fender | JustAGuy1: Ask yourself why some random company is going waste a DID, PSTN channel, and bandwidth to give YOU free service.... |
| 16:51.37 | delphus | question: does anyone know about FreeBSD support for g729a codec ? |
| 16:51.41 | ManxPower | JustAGuy1: It is very good advice. |
| 16:51.41 | JustAGuy1 | IPKall, AOL and others do so already. |
| 16:51.50 | Daejeo1 | JustAGuy1: you cannot get free things whatever you want |
| 16:52.10 | [TK]D-Fender | JustAGuy1: Yeah... AOL I understand since thats how they hook you in in the first place... I'm still not sure on IpKall yet.. |
| 16:52.14 | Qwell[] | delphus: There isn't currently a codec for g729 on freebsd. In the future there will likely be some, but it's unsupported |
| 16:52.17 | ManxPower | delphus: Most of those sorts of questions need to go to Digium, as the G729 is closed source because of patent issues |
| 16:52.24 | JustAGuy1 | IPKall makes it money on termination charges. |
| 16:52.29 | delphus | thanks |
| 16:52.36 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:52.53 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 16:53.09 | JustAGuy1 | They provide numbers in an area where almost all calls to them are long distance and they get paid for the inbound call. |
| 16:53.36 | [TK]D-Fender | JustAGuy1: Sounds like BS to me... SOMEONE is paying for it.... |
| 16:53.47 | delphus | I was afraid I would hear those answers already. I heard that there was a version for freebsd 5.x but the download path from digium main ftp site is not there anymore, anyway thanks guys. |
| 16:53.55 | [TK]D-Fender | Either way, get a real provider, and pay the price for service... |
| 16:53.55 | Zodiacal | anyone know how i can enable call forwarding via the console? i.e. *72 |
| 16:54.06 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 16:54.14 | JustAGuy1 | The IXC pays to terminate calls to IPKall, the local provider. |
| 16:54.16 | [TK]D-Fender | Zodiacal: however YOU set it up in your dialplan |
| 16:54.21 | ManxPower | Zodiacal: Console/1 or Asterisk Command Line Interface. |
| 16:54.33 | Zodiacal | cli |
| 16:54.44 | ManxPower | JustAGuy1: It costs alot more than some .25/cent/mon termination charge to run an ITSP |
| 16:54.52 | Daejeo1 | ManxPower: anyother voip provider except teliax |
| 16:54.53 | ManxPower | Zodiacal: you can't, AFIK |
| 16:55.01 | Zodiacal | okie thanks guys |
| 16:55.07 | ManxPower | Daejeo1: What specifically do you need? |
| 16:55.15 | Daejeo1 | termination for india |
| 16:55.33 | [TK]D-Fender | Daejeo1:termination IN India? |
| 16:55.37 | Daejeo1 | viatalk -- 15cents /min |
| 16:55.39 | *** part/#asterisk delphus (n=rodrigog@85.92.130.202) |
| 16:55.39 | ManxPower | Daejeo1: Teliax's rates won't me much lower than most anyone else's. |
| 16:55.48 | Daejeo1 | I agree |
| 16:55.56 | [TK]D-Fender | Daejeo1: Go look at the ITSP list ont he WIKI |
| 16:56.34 | *** join/#asterisk HaMYaI (i=HaMYaI@125-25-193-158.adsl.totbb.net) |
| 16:56.48 | *** part/#asterisk HaMYaI (i=HaMYaI@125-25-193-158.adsl.totbb.net) |
| 16:57.17 | JustAGuy1 | ManxPower: They get about 1 cent a minute for inbound long distance calls from the IXC. Given they have no customer service at all, they do pretty well. They have been around for many years already. |
| 16:57.17 | ManxPower | Just remmeber: "All ITSPs suck. Some suck less than others."(tm)(c) 2007 ManxPower |
| 16:57.50 | ManxPower | JustAGuy1: How do you know. All termination charges are negotiates on a case by case basis. |
| 16:58.50 | ManxPower | Daejeo1: I think VoIP is illegal in india, but there might be an indian ITSP that has lower prices (if one exists) |
| 17:00.53 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 17:01.00 | Daejeo1 | right."There is one "International VSNL " but these guys only deal with big parties like Verizon etc.. |
| 17:01.32 | JustAGuy1 | ManxPower from doing Due Diligence for our fund. |
| 17:02.43 | ManxPower | Ok. So you are doing Due Dillgence and yet still need free service. |
| 17:02.59 | ManxPower | sorry, this is so sad I have to put you on /ignore |
| 17:03.01 | JustAGuy1 | TK: AOL is certainly has more resources than almost any of those that you call "Real Providers". |
| 17:03.16 | ManxPower | JustAGuy1: best of luck. |
| 17:03.38 | [TK]D-Fender | JustAGuy1: I never said they didn't have real resources. But think of AOL more like fishing... the bait sure looks good till you can't get off the line. |
| 17:03.39 | JustAGuy1 | ManxPower: I don't need anything. I am looking for something to experiment with right now. |
| 17:03.40 | ManxPower | much better |
| 17:04.13 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:04.51 | blitzrage | hey all! I'm using idefisk for an IAX2 softphone on Linux -- anyone have a 2nd recommendation? (I need to test a couple of phones to verify a bug I'm having in an IAX2 gateway I'm building is because of something on the phone, or the gateway itself) |
| 17:05.00 | blitzrage | btw: idefisk is awesome |
| 17:05.09 | blitzrage | but you already knew that :D |
| 17:06.16 | JustAGuy1 | TK: AIM Phoneline is an interesting experimental toy for the moment. They have based their service on PingTel's client. I guess I will see if anyone over there has connected it to SIPXchange. |
| 17:06.20 | JustAGuy1 | Thanks. |
| 17:06.24 | *** part/#asterisk JustAGuy1 (n=majortom@adsl-63-195-88-133.dsl.snfc21.pacbell.net) |
| 17:06.33 | ManxPower | blitzrage: asterisk-to-asterisk would be the best place to test it. |
| 17:07.15 | blitzrage | ya, I guess I could run Dial() on my laptop |
| 17:07.28 | blitzrage | or rather... I mean *CLI> dial :) |
| 17:07.48 | *** join/#asterisk CapRiCoRN^80 (n=cap@203.135.55.37) |
| 17:11.25 | tzafrir | kiax is nice |
| 17:11.44 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 17:11.50 | BSD_Tech | is there going to be 1.4.5 soon ? |
| 17:12.01 | BSD_Tech | its needed |
| 17:12.04 | BSD_Tech | lol |
| 17:12.51 | BSD_Tech | 1.4.4 does not work with the asterisk-gui right unless you use svn. |
| 17:13.32 | blitzrage | BSD_Tech: test chan_sip and make sure we fix the hanging channels bug in 1.4 branch before 1.4.5 |
| 17:13.32 | msetim | Hi... |
| 17:13.38 | BSD_Tech | ok |
| 17:13.44 | BSD_Tech | I will get svn today |
| 17:13.46 | msetim | Someone knows the asterisk http manager? |
| 17:15.12 | msetim | I would like to know why the action command doesn't work by http manager. When I connect by telnet the result of command is printed on screen |
| 17:15.13 | blitzrage | see bug: 9235 |
| 17:15.17 | ManxPower | I don't think the Asterisk HTTP Manager has many friends at the moment |
| 17:15.47 | tzanger | I like it |
| 17:15.58 | tzanger | only because I've used it for my own nefarious plans |
| 17:16.03 | msetim | ManxPower: Yep :( I don't find many users of it |
| 17:16.31 | Katty | hmmmmmmmmmmmmmmmmmmmm |
| 17:16.54 | *** join/#asterisk bhiers (n=chatzill@primary.computerpoint.net) |
| 17:17.08 | msetim | tzanger: You have used the manager from web? |
| 17:17.22 | *** join/#asterisk lyroy (n=lyroy@picachou.csaffluents.qc.ca) |
| 17:17.40 | lyroy | Is there any free oubound only call voip provider? |
| 17:17.51 | tzanger | no |
| 17:18.22 | nick125_lappy | Well, I guess it depends on who you call. I mean, FWD has free tollfree termination.. |
| 17:18.37 | bhiers | using a TDM2400P I get loud click or noise when picking up the line.. is that normal or is there some fix? |
| 17:18.58 | woolbeo | I am planning on using it once I finish fixing the last guy's screwups... Right now my coworkers want a phonesystem that is stable and works like it should more than an update interface for their CRM... |
| 17:19.57 | *** join/#asterisk andydna (n=chatzill@66.246.173.34) |
| 17:19.59 | woolbeo | So by the time I get around to it, HTTP manager should have all the bugs worked out of it.. ;) |
| 17:20.35 | *** join/#asterisk jeffik (n=Valued@206-248-152-65.dsl.teksavvy.com) |
| 17:21.09 | lee_is_me | Still playing around with AMI. Is there a setting that determines the response time for a request sent to AMI? Seems like it takes about 15-20 seconds to broadcast a a response. |
| 17:22.39 | jeffik | anybody familiar with SPA-942? |
| 17:23.12 | lee_is_me | Maybe AMI uses internal polling for processing requests? |
| 17:25.18 | wunderkin | lee_is_me: Async: True |
| 17:25.20 | *** join/#asterisk bhiers (n=chatzill@primary.computerpoint.net) |
| 17:26.12 | woolbeo | bhiers, fxo ->SIP or FXO->FXS? |
| 17:26.31 | MindTheGap | in realtime sip peers i need "fullcontact" set to "sip:exten@userip"... how do I populate on the fly the fullcontact with information from exten and userip? im using res_conf_ldap.so |
| 17:26.33 | MindTheGap | on res_ldap.conf i have "attribute = fullcontact => AstAccountFullContact" it would be nice to have something like: attribute = fullcontact => "sip:".$AstExten."@".$AstIPaddress |
| 17:27.14 | lee_is_me | woolbeo: isn't Async: true for originate only? |
| 17:27.37 | wunderkin | oh.. shrug |
| 17:27.59 | woolbeo | lee_is_me, no idea.. sorry.. |
| 17:28.23 | [TK]D-Fender | MindTheGap: Just a thought. The odds of finding someone here who can help yuo with that are very slim. Perhaps you should try the mailing lists... |
| 17:28.36 | lee_is_me | woolbeo: NP. Just trying to figure out why it takes asterisk to return a result from a simple request... |
| 17:29.35 | bhiers | Anyone know where I can get some help on a TDM2400 issue? |
| 17:29.55 | lee_is_me | <Santity Check/> Just to be sure does anyone else experience such long times (20 seconds +) for asterisk to respond to an AMI request? |
| 17:30.07 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 17:30.23 | [TK]D-Fender | jeffik: What about it? |
| 17:30.24 | woolbeo | lee_is_me, mine is almost instantanious... |
| 17:30.33 | lee_is_me | really? |
| 17:30.43 | woolbeo | bheirs, fxo, fxs, or both? |
| 17:30.46 | lee_is_me | damn, wonder why mine takes so long... |
| 17:31.10 | woolbeo | lee_is_me, 1.4 or 1.2? |
| 17:31.16 | lee_is_me | 1.2 |
| 17:31.52 | slmnhq | Hi all.. do you guys think that there is any value in selling Asterisk boxes running on a real-time environment? |
| 17:32.33 | slmnhq | Or actually, let me rephrase |
| 17:33.04 | slmnhq | What kinds of features could make Asterisk more commercially appealing? |
| 17:33.14 | MindTheGap | [TK]D-Fender, thanks mate... |
| 17:33.35 | *** join/#asterisk ToyMan (n=Stuart@dpc6714368169.direcpc.com) |
| 17:33.52 | bhiers | Woolbeo FXO |
| 17:34.20 | woolbeo | bhiers, is the call fine after it picks up? |
| 17:34.25 | bhiers | Woolbeo , lines are coming from a channel bank to the TDM2400 |
| 17:34.41 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 17:34.48 | bhiers | Yup yup sounds great still got to work on the RX gain but all in all sounds great.. |
| 17:35.02 | bhiers | just get this loud crackle or pop noise on pickup |
| 17:35.06 | woolbeo | bhiers, why not ditch the channel bank completely and just use a T1 card? |
| 17:35.29 | bhiers | what I have right now.. |
| 17:35.44 | msetim | woolbeo: It have many bugs? Asterisk developers appears don't give many importance to it. |
| 17:35.45 | woolbeo | bhiers, what signalling is the CB set to and zaptel.conf/zapata.conf? |
| 17:36.31 | *** part/#asterisk andydna (n=chatzill@66.246.173.34) |
| 17:36.55 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 17:37.12 | woolbeo | msetim, no idea. Its just a personal opinion of mine to not use a brand new feature in a production enviroment. |
| 17:37.37 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 17:37.40 | n0n4m3 | what the hell does 'module embedding means in 1.4's menuconfig? |
| 17:38.29 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 17:38.44 | n0n4m3 | umm |
| 17:38.48 | n0n4m3 | never mind :D |
| 17:38.51 | n0n4m3 | The Module Embedding page is for statically compiling modules, instead of the usual dynamic linking. There aren't a lot of reasons to do this; if you're debugging multiple Asterisk versions it helps keep things sorted out. Or you want a single static binary to install on multiple machines. |
| 17:38.53 | pingwin | i'm trying to find information to allow a sip client to monitor a another sip phone (preferably even while the phone is not connected on a call). is this possible? the only thing I can find thus far is zapbridge (which if disconnected the zap connection is no longer connected) and monitor, which records a file |
| 17:38.57 | pingwin | any tips? |
| 17:39.15 | n0n4m3 | i will check google before asking dumb questions :) |
| 17:39.19 | bhiers | what dir is the zapata.conf normally in |
| 17:39.22 | poppo | Can somebody help me with my php script using fopen to pass variable but not showing up on the logs. Look like its not passing the values. Can somebody help out |
| 17:39.37 | bhiers | never mind |
| 17:40.17 | woolbeo | bheirs, you'll get better sound quality if you ditch the CB, and TDM2400p and use a T1 card. The extra DAC-ADC conversions are going to reduce quality... |
| 17:40.23 | n0n4m3 | too bad 1.4 doesn't supporr mysql by default :D |
| 17:40.38 | Qwell[] | n0n4m3: install asterisk-addons |
| 17:40.43 | woolbeo | bhiers, zapata.conf is in /etc/asterisk/ and zaptel.conf is in /etc/ |
| 17:40.50 | n0n4m3 | Qwell i will |
| 17:41.06 | bhiers | The Quality is rock solid except for this one pop on pickup |
| 17:41.09 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:41.25 | bhiers | what is CB? |
| 17:41.33 | woolbeo | bhiers, if it works, stick with it.. CB=Channel Bank |
| 17:42.44 | *** join/#asterisk Chuji (n=brian@mail.point3media.com) |
| 17:43.10 | woolbeo | bhiers, the pop at the begining makes me think that you have a the CB and the tdm2400p using different signalling, i.e. one set to loop start and one set to ground start.. |
| 17:44.16 | bhiers | where is the setting for signaling in the zap conf? |
| 17:44.20 | [TK]D-Fender | pingwin: This is basic Presence. Go look it up on the WIKI. |
| 17:45.04 | pingwin | [TK]D-Fender k, thank you, i wasn't sure if that's what I wanted or if there was something else because I know some phones won't support it. but thank you for the pointer :) |
| 17:45.21 | [TK]D-Fender | pingwin: What phones do you have? |
| 17:46.33 | pingwin | polycom |
| 17:46.38 | pingwin | not sure the model atm |
| 17:46.42 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 17:47.17 | woolbeo | bheirs, in zaptel.conf it should be something like fxsls=1-24, fxsgs=1-24, or fxsks=1-24. |
| 17:48.06 | [TK]D-Fender | pingwin: All polycoms support it rather well |
| 17:49.03 | ManxPower | [TK]D-Fender: someone had permission denied problems with presence in 1.4.2 |
| 17:49.44 | [TK]D-Fender | ManxPower: All morons are people. Some people are morons. |
| 17:49.59 | [TK]D-Fender | ManxPower: Start drawing logic circles :) |
| 17:50.15 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 17:50.17 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 17:50.45 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 17:51.00 | *** part/#asterisk ecoleman (n=eric@24.75.47.98) |
| 17:51.11 | woolbeo | bhiers, what signalling is your T1 provisioned for? What signalling do you have the fxs ports on your cd set to? |
| 17:52.14 | variable_office | i am using ENUMLOOKUP and when I have a #(%23) or * it always fails, is asterisk's enumlookup not able to handle these chars? |
| 17:54.10 | bhiers | Contacting provider to find the signalling for the T1 |
| 17:55.23 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:56.12 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-148.dsl.irvnca.pacbell.net) |
| 17:56.15 | *** part/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 17:56.19 | [TK]D-Fender | variable_office: set "pedantic=yes" under [general] in sip.conf |
| 17:56.50 | sevard | Does anyone know of what tools in linux would comply with DoD 5220.22-M? |
| 17:57.01 | bkw_ | zero |
| 17:57.20 | bkw_ | SELinux maybe.. |
| 17:58.12 | variable_office | [TK]D-Fender, that didnt help unfortunatly? |
| 17:58.27 | sevard | bkw_: I have a pile of HDDs marked for resale, do they have a livecd? |
| 17:58.59 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 17:59.00 | woolbeo | sevard, scrub? |
| 17:59.02 | [TK]D-Fender | variable_office: it should. |
| 17:59.17 | woolbeo | sevard, I should say scrub with the dod options. |
| 17:59.34 | bkw_ | just scrube them |
| 17:59.40 | variable_office | [TK]D-Fender, are you using # and * in enum? |
| 17:59.44 | bkw_ | http://www.linux-sec.net/Txt/erase.txt |
| 17:59.51 | bhiers | woolbeo only try of signalling I have in the zap confs is signalling=fxs_ks |
| 18:00.05 | [TK]D-Fender | variable_office: Nope |
| 18:00.12 | woolbeo | bhiers, ok, now what are teh fxs ports on your cb set to? |
| 18:00.32 | bkw_ | sevard, its easier to destroy them than it is to wipe and sell them |
| 18:00.34 | variable_office | using enum at all? |
| 18:00.44 | jeffik | all: anybody familiar4y with spa-942? |
| 18:01.10 | sevard | bkw_: it's easier to destroy anything than sell it |
| 18:02.28 | woolbeo | sevard, scrub -p dod, is DoD 5220.22-M compliant. |
| 18:03.17 | woolbeo | To me It seems easier to hook up a HD and run scrub on it than to destroy a HD... |
| 18:03.19 | [TK]D-Fender | jeffik: I already asked what you wanted to know about it. |
| 18:04.15 | [TK]D-Fender | jeffik: Thats like asking if I know what the chemical name for water is and automatically think that I can start discussing quantum theory because its "related" |
| 18:04.40 | *** join/#asterisk BSD_Tech (n=BSDTech@ppp-69-239-114-108.dsl.irvnca.pacbell.net) |
| 18:04.54 | sevard | Would this comply enough with DoD standards for i in `seq 1 3`; do dd if=/dev/urandom of=/dev/drive; dd if=/dev/zero of=/dev/drive; done |
| 18:04.55 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 18:05.29 | bhiers | Woolbeo thanks for info done some research I completely understand now ;-) |
| 18:06.16 | jkiff | Hmm, _[02-19] won't match 02, 03, 04, ..., 17, 18, 19; will it? |
| 18:06.35 | woolbeo | bhiers, so you fixed it? |
| 18:06.48 | [TK]D-Fender | jkiff: well thats an * quesiton, not an SPA on, and NO, it won't. |
| 18:07.01 | bhiers | On hold with ISP |
| 18:07.59 | [TK]D-Fender | jkiff: You'll need 2 matches : _[02-9] and another _1X |
| 18:08.04 | woolbeo | bhiers, good luck, glad I could help. I remeber when I was thrown into the telephony and * world... |
| 18:08.42 | jkiff | [TK]D-Fender: Yeah, that's w.r.t. asterisk. |
| 18:08.44 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 18:08.51 | bhiers | yea I'm a computer guy doing phone stuff |
| 18:09.08 | jkiff | [TK]D-Fender: I see. I don't suppose [(02)-(19)] works then. |
| 18:09.25 | [TK]D-Fender | jkiff: No. go read up on pattern matching in THE BOOK, or on the WIKI |
| 18:09.27 | [TK]D-Fender | ~book |
| 18:09.40 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:09.40 | [TK]D-Fender | ~wikis |
| 18:09.43 | jbot | well, wikis is http://www.voip-info.org |
| 18:09.43 | *** join/#asterisk dalfry (n=dalfry@70.89.177.109) |
| 18:09.45 | nick125_lappy | Just an idea, I think you could do [0-1][2-9] |
| 18:09.47 | [TK]D-Fender | jkiff: the way I showed you is the way to do it. |
| 18:09.58 | [TK]D-Fender | nick125_lappy: NO. |
| 18:10.29 | nick125_lappy | [TK]D-Fender: It's been a while since I've done patterns :p |
| 18:10.34 | [TK]D-Fender | nick125_lappy: 10 & 11 fall throught the gaps. |
| 18:10.42 | woolbeo | bhiers, thats, what happened to me. I was a linux admin and my old boss wanted an ivr, so we setup Bayonne, hated it, and moved over to *. That was back around asterisk 0.4... |
| 18:10.52 | [TK]D-Fender | nick125_lappy: You clearly didn't think more than a second or two on taht one :) |
| 18:11.02 | nick125_lappy | [TK]D-Fender: I was saying for his example [(02)-(19)] |
| 18:12.06 | [TK]D-Fender | nick125_lappy: jkiff>Hmm, _[02-19] won't match 02, 03, 04, ..., 17, 18, 19; will it? |
| 18:15.26 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
| 18:15.26 | jkiff | [TK]D-Fender: Cool, thanks! |
| 18:16.10 | Katty | so let's say i want someone to give me a number, and then i'm going to take that number and turn it into a Record(number:gms) |
| 18:16.13 | Katty | er, gsm |
| 18:16.33 | [TK]D-Fender | Katty: "show application read" |
| 18:16.38 | Katty | thanks |
| 18:16.59 | CapRiCoRN^80 | hi ! is there any how to setup asterisk between linux system and a window system ? |
| 18:17.20 | Qwell[] | CapRiCoRN^80: ssh |
| 18:17.50 | CapRiCoRN^80 | Qwell[]: didnt get u |
| 18:17.53 | bhiers | we were using level3 hosted PBX hated it.. missing all sorts of features so are * is rock solid.. even have animated logos on my phones (little stuff like that makes me happy) LOL |
| 18:18.23 | Qwell[] | ssh from the windows machine to the linux machine, and configure it like normal |
| 18:18.27 | Qwell[] | get something like putty |
| 18:18.31 | Hmmhesays | i'm writing a perl script to control a phone lcd right now |
| 18:19.04 | Hmmhesays | writing user interfaces is just a bitch |
| 18:19.45 | CapRiCoRN^80 | Qwell[]: i mean to say that i need howto site that contain simple way of configuring asterisk on linux box n they using it from window box |
| 18:20.01 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 18:20.04 | bhiers | whats the command to reload zapata.conf? |
| 18:20.22 | Katty | Hmmhesays: mew. |
| 18:20.45 | woolbeo | bheirs, reload chan_zap.so |
| 18:20.55 | bhiers | thanks :-) |
| 18:21.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-5-67.red.bezeqint.net) |
| 18:21.31 | Hmmhesays | Hey Katty |
| 18:22.19 | Katty | [TK]D-Fender: application read seems to be just what i needed. too bad the wiki page with examples is in french :< |
| 18:22.19 | woolbeo | bhiers, you'll need to change the signalling in zaptel.conf and reload your zaptel drivers |
| 18:22.19 | [TK]D-Fender | CapRiCoRN^80: Clarify what you mean by "using it from window box: |
| 18:22.19 | Hmmhesays | Ok I think I finally got this menu to behave the way I want it to |
| 18:22.38 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:22.53 | crimethinker | does it lay flat on the table? does it reflect the cook's current offerings? |
| 18:22.57 | bhiers | woolbeo changed to loop start.. and reload chan_zap.so .. doesn't that reload the zaptel drivers? |
| 18:23.01 | [TK]D-Fender | Katty: exten => 1,1,Read(myvar) exten => 1,2,Record(${myvar}.gsm) |
| 18:23.17 | Hmmhesays | crimethinker: yes |
| 18:24.12 | CapRiCoRN^80 | i mean asterisk server on linux and then window client can use that |
| 18:24.15 | Katty | [TK]D-Fender: and SayDigits(${myvar}) too? |
| 18:24.32 | Qwell[] | CapRiCoRN^80: any windows softphone should be fine |
| 18:25.01 | CapRiCoRN^80 | ok fine |
| 18:25.12 | [TK]D-Fender | Katty: Sure, why not... |
| 18:25.20 | CapRiCoRN^80 | but tell me wat settings will be required by linux |
| 18:25.21 | ManxPower | woolbeo and bhiers you CANNOT change the signalling with a simple reload. You must stop/start asterisk or load/unload chan_zap.so |
| 18:25.32 | CapRiCoRN^80 | i mean any how to on it |
| 18:26.09 | ManxPower | CapRiCoRN^80: too bad you are not trying to just configure a softphone to work with Asterisk. |
| 18:26.13 | CapRiCoRN^80 | some simple site that contain perfect how to setup all this with pics etc |
| 18:26.44 | ManxPower | CapRiCoRN^80: PBXs are complicated systems. It is not possible to set one up "simple". |
| 18:27.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:27.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:27.17 | Hmmhesays | thats why people get paid to set things up |
| 18:27.23 | CapRiCoRN^80 | thats y i need some how to |
| 18:27.37 | CapRiCoRN^80 | i mean step by step configuration |
| 18:27.46 | [TK]D-Fender | CapRiCoRN^80: go read... THE BOOK |
| 18:27.48 | [TK]D-Fender | ~book |
| 18:27.59 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:27.59 | ManxPower | CapRiCoRN^80: such a thing does not exist. |
| 18:28.09 | [TK]D-Fender | <PROTECTED> |
| 18:28.43 | [TK]D-Fender | CapRiCoRN^80: Go download it. Spend a day or few reading. then install a linux distro on your server and GET STARTED |
| 18:29.12 | ManxPower | [TK]D-Fender: that would be a book, not a short howto with pics |
| 18:29.27 | [TK]D-Fender | CapRiCoRN^80: Here's a quick 1.4 current quickie guide |
| 18:29.50 | ManxPower | What wants is Fisher Price(tm) My First PBX. I think you can get it when you buy the Easy Bake Oven and Malibu Barbie. |
| 18:29.55 | [TK]D-Fender | ManxPower: Guess its based on your idea of "quick".... DL'ing the book hardly takes a minute! |
| 18:30.17 | Hmmhesays | reading and understanding is a different thing though |
| 18:30.22 | ManxPower | [TK]D-Fender: not on my connection it doesn't 8-) |
| 18:30.33 | [TK]D-Fender | Hmmhesays: "Not. My. Problem" |
| 18:31.34 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 18:32.20 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 18:32.23 | ManxPower | CapRiCoRN^80: I spent 6 months of working with Asterisk every day before I deployed my first production system. |
| 18:32.31 | ManxPower | And you should too. |
| 18:32.46 | Hmmhesays | that is good advice |
| 18:32.59 | woolbeo | ManxPower, That is what I was trying to tell him.. sorry... |
| 18:33.00 | CapRiCoRN^80 | really |
| 18:33.18 | Hmmhesays | if you think you are just going to step into it and put a system into production, think again |
| 18:33.27 | ManxPower | CapRiCoRN^80: PBXs are complicated things. VoIP PBXs are doubly so. |
| 18:33.43 | ManxPower | With a regular PBX you just need to know the PBX and know telecom (and have lots of money) |
| 18:33.47 | Hmmhesays | theres enough flexibility rope to hang y ourself with |
| 18:34.08 | ManxPower | With a VoIP PBX you need to know the PBX, telecom, Linux, networking, QoS (if using WAN) |
| 18:35.02 | CapRiCoRN^80 | ManxPower: in first phase i just want to configure Asterisk between two systems |
| 18:35.22 | CapRiCoRN^80 | one linux n windown as client |
| 18:35.27 | CapRiCoRN^80 | i will read its doc in future |
| 18:35.38 | CapRiCoRN^80 | but first i m looking to setup between two sytems |
| 18:35.39 | ManxPower | CapRiCoRN^80: you cannot configure Windows as a client. |
| 18:35.50 | ManxPower | You can set up a specific softphone that runs under windows as a client, of course. |
| 18:36.00 | Daejeo1 | i want to play music instead of ring tone on the local extensions |
| 18:36.08 | ManxPower | but Windows itself really has nothing to do with it, except to limit your choices in softphones. |
| 18:36.17 | Daejeo1 | ManxPower? |
| 18:36.19 | ManxPower | Daejeo1: It is called Music On Hold. Read up on it. |
| 18:36.24 | CapRiCoRN^80 | ManxPower: yea |
| 18:36.32 | ManxPower | CapRiCoRN^80: so pick your softphone. |
| 18:36.58 | CapRiCoRN^80 | x-lite |
| 18:37.22 | [TK]D-Fender | CapRiCoRN^80: I linked you to the book, and a quick start guide. Get to work. |
| 18:37.41 | ManxPower | CapRiCoRN^80: nobody can help you until you read the book. |
| 18:38.19 | CapRiCoRN^80 | jbot> hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 ............ this one |
| 18:38.29 | jbot | CapRiCoRN^80: okay |
| 18:38.33 | CapRiCoRN^80 | ? |
| 18:38.43 | Qwell[] | jbot: forget hmm... book |
| 18:38.43 | jbot | i forgot hmm... book, Qwell[] |
| 18:39.00 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:39.24 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 18:39.26 | CapRiCoRN^80 | [TK]D-Fender : u were talking about this book link ..... http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:39.40 | [TK]D-Fender | CapRiCoRN^80: YES |
| 18:40.10 | CapRiCoRN^80 | ok . i got this pdf |
| 18:42.08 | Daejeo1 | ManxPower: |
| 18:43.30 | ManxPower | Daejeo1: |
| 18:44.03 | [TK]D-Fender | *crickets* |
| 18:44.19 | Daejeo1 | music on hold is somewhat different thing |
| 18:44.43 | ManxPower | Daejeo1: not from the standpoint of Asterisk |
| 18:44.45 | ManxPower | and the "m" option of Dial |
| 18:44.56 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 18:45.30 | Daejeo1 | caller should listen the music before i answer the call |
| 18:45.45 | Daejeo1 | not after answering the call |
| 18:46.02 | woolbeo | Daejeo1, use m instead of r in your dial string |
| 18:46.06 | ManxPower | Daejeo1: Asterisk considers all music to be "music on hold" |
| 18:46.17 | ManxPower | woolbeo: please stop talking. |
| 18:46.21 | *** join/#asterisk sysreq (n=sysreq@72.0.197.4) |
| 18:46.32 | ManxPower | You have twice now given information that is really going to screw someone up eventualyl |
| 18:46.56 | ManxPower | you should never ever ever EVER use "r" option to Dial unless you totally and completly understand what it does, why it does it, and why you might need it. |
| 18:47.02 | woolbeo | ManxPower, the first time I told him exactly what wanted, then i corrected myself. |
| 18:47.40 | GaVak | If I don't tunnel SIP, is it unencrypted? |
| 18:47.46 | ManxPower | "r" means override any sounds that should be played (including ringing sounds) and force a fake ringing sound. |
| 18:47.56 | ManxPower | GaVak: correct |
| 18:48.05 | *** join/#asterisk HyPnoLORD (n=hello@189.173.40.97) |
| 18:48.32 | HyPnoLORD | Hello |
| 18:48.36 | ManxPower | It's interensting when you call a busy number using an analog fxo port when using "r". You will hear ringing and then a busy. |
| 18:48.54 | [TK]D-Fender | GaVak: Actually the SIP is ALWAYS unencryped..... the path you SEND it through is another matter. |
| 18:49.12 | woolbeo | Manxpower, it would be nice if the docs said that... |
| 18:49.15 | [TK]D-Fender | GaVak: Anyone en-route of that path can still spy with reckless abandon :) |
| 18:49.20 | GaVak | Murf. So an asterisk port to my outside zone to let in two teleworkers would be a bad idea. |
| 18:49.25 | ManxPower | When using PRIs, and call a cell phone that is not in range will give you a ringing sound instead of "The nexttel subscriber you are calling is not available" |
| 18:49.43 | [TK]D-Fender | GaVak: Depends on your concept of "security". |
| 18:49.46 | ManxPower | GaVak: is your telephone closet locked? |
| 18:49.56 | ManxPower | Is the telco box on the street locked? |
| 18:49.58 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 18:49.59 | [TK]D-Fender | GaVak: use non-standard ports, etc..... iptables lockout, and so on. |
| 18:50.14 | GaVak | I'm trying to QoS the packets, but I can't really do it through the persistant tunnel from the remote workersw. |
| 18:50.21 | [TK]D-Fender | GaVak: Security comes in levels, not absolutes. |
| 18:50.23 | ManxPower | If I was trying to back someone's telecoms I'll just walk into their phone closed and put in a tap. |
| 18:50.24 | GaVak | Man: Yeah, the phone switch is actually in my office. |
| 18:50.31 | ManxPower | no sillyness with the network |
| 18:50.57 | HyPnoLORD | HI.. Have any of you encountered a Voltage Problem or Know Why would a TD400P Work in one Analog Line (Location 1) and Not work at all in a different Analog Line (Location 2) ??? |
| 18:53.11 | HyPnoLORD | Its a funny behavior, but I haven´t figured out the problem already. My guess is that the line has some kind of drop down voltage or not enough to power the card. However, a regular FXO device (my old telephone) works fine in Analog Line (location 2) |
| 18:53.29 | HyPnoLORD | any one? |
| 18:54.30 | yannj_fr | hi all |
| 18:54.47 | HyPnoLORD | hello yannj_fr |
| 18:54.53 | [TK]D-Fender | HyPnoLORD: You mention FXO and "your regular telephone" in the same sentence. Phones are FXS. Get yourself straight |
| 18:55.03 | HyPnoLORD | oops |
| 18:55.09 | HyPnoLORD | sorry I ment FXS |
| 18:55.14 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:55.28 | [TK]D-Fender | HyPnoLORD: Start over again and try to be clearer. |
| 18:55.28 | woolbeo | ManxPower, bheir asked how to reload zapate, so I answered, then I realized he was trying to change his signalling, so I told him he had to reload his zaptel drivers, so I thought that would be clear that you have to stop asterisk, unload zaptel modules, load zaptel modules, start asterisk. I didn't give him bad information, I just didn't give him step by step information. I will shut up though anyways.. |
| 18:55.49 | woolbeo | zapate=zapata |
| 18:56.02 | Daejeo1 | ManxPower: where can i find music on hold in asterisk? |
| 18:56.05 | Daejeo1 | etc? |
| 18:56.05 | [TK]D-Fender | ManxPower: And do be so kind as to spit his head out so he may hope to have it surgically reattached :) |
| 18:56.13 | ManxPower | Daejeo1: there is some incliuded |
| 18:56.22 | Daejeo1 | dir? |
| 18:56.28 | ManxPower | [TK]D-Fender: I try to do that otherwise I choke. |
| 18:56.57 | Daejeo1 | can you tell me the path? |
| 18:56.58 | ManxPower | Daejeo1: I would have to ssh across a 3000ms satellite link to tell you. It is usually listed in /etc/asterisk.conf I think |
| 18:57.10 | Daejeo1 | ok |
| 18:57.26 | [TK]D-Fender | ManxPower: Or chew before you swallow :) |
| 18:58.32 | echo--- | I'm shopping around for single port PRI cards for use in the US. Last card I bought was a T100P, so it's been awhile. How are the Sangoma A101D series cards? Do they suck? |
| 18:58.52 | *** join/#asterisk EduHard (n=user@proxy.donto.com.ar) |
| 18:59.02 | EduHard | Hello everybody |
| 18:59.19 | ManxPower | echo---: Sangoma config sucks, but they are good and reliable cards |
| 18:59.36 | EduHard | Need help with an IVR default action (is that the correct name?). |
| 18:59.44 | HyPnoLORD | [TK]D-Fender ok.. I have a TDM400P with a FXS and FXO module. Im barely expermienting with asterisk. I had it configured, and working for 2,3 days in Location 1. However I just moved the Server to Location 2 and connected the Telephone Line in the right Port. However If I try to make a phone call I get nothing (one a high pitch tone) If I call the Number of the phone line I get a bussy tone due to line problems. |
| 19:00.12 | echo--- | thanks, Manx. |
| 19:00.12 | HyPnoLORD | But if I disconect the Line and connect it to a regular phone it gives me a tone |
| 19:00.15 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 19:00.42 | HyPnoLORD | and no problem |
| 19:00.45 | EduHard | When it times out it loops the welcome recording over and over and over again. |
| 19:01.27 | ManxPower | EduHard: it won't do that unless you set up the IVR to do that |
| 19:01.51 | jcaceres | hello i am using a grandstream gateway, and it take too long to release the FXO line when the caller hangs up |
| 19:01.57 | woolbeo | Manxpower, now the now the misinformation about r I admit that I screwed up.. However I did not tell him to use r, aI told him to use m... So does m overrivde anything that would be played and play music like r would? |
| 19:02.00 | [TK]D-Fender | HyPnoLORD: Double-check your ports. I'm still betting they're wrong. |
| 19:02.02 | ManxPower | HyPnoLORD: I suspect you are confused as to the port number |
| 19:02.05 | jcaceres | any idea of what can i be doing wrong? |
| 19:02.33 | [TK]D-Fender | jcaceres: Go read your GS's manual and see what disconnect supervision options it has. |
| 19:02.45 | *** join/#asterisk rob0 (n=rob0@208.62.162.112) |
| 19:03.31 | EduHard | Any tip on how to set it up? I don't see any setting for that on the freepbx gui, I've done it changing t and i values in extensions_additional.conf file |
| 19:03.47 | [TK]D-Fender | EduHard: there is no "default action". It does whatever you tell it to. Go read up on "asterisk standard extensions" on the WIKI |
| 19:03.51 | [TK]D-Fender | ~freepbx |
| 19:03.52 | jbot | hmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:04.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 19:04.36 | HyPnoLORD | ManxPower (Well I thought so but I had my FXS MOdule Removed) and Im sure Im using the same port as before |
| 19:04.59 | [TK]D-Fender | HyPnoLORD: Regardless, check to see if your FXS is working fine first. |
| 19:05.25 | EduHard | thanks for the data, found info on it. |
| 19:06.29 | ManxPower | EduHard: we do not support FreePBX here |
| 19:06.32 | ManxPower | ~freepbx |
| 19:06.35 | jbot | i heard freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:06.44 | HyPnoLORD | Well I dont have it installed in the card. I removed it because I dont need the FXS card I just have the FXO (RED) module |
| 19:08.15 | [TK]D-Fender | HyPnoLORD: go double-check your card and its jacks individually. do a dialout test and listen in parallel with an analog phone. |
| 19:08.28 | EduHard | Here is command-line support only? |
| 19:09.17 | [TK]D-Fender | EduHard: Non-GUI, yes. |
| 19:09.39 | [TK]D-Fender | EduHard: See the topic and the bot-scripts for help links if you still want to continue using them. |
| 19:11.06 | EduHard | well, there's a lot to read... i'll come back any time soon. |
| 19:11.25 | EduHard | Thanks for your time men. |
| 19:12.37 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 19:12.47 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 19:14.46 | HyPnoLORD | OK |
| 19:14.48 | HyPnoLORD | ;D |
| 19:19.01 | bapril | anyone know a way to detect a hook-flash coming from the called party on a PRI? Trying to detect if called-party is trying to 3-way call etc.? |
| 19:19.19 | ManxPower | bapril: you can't |
| 19:19.42 | ManxPower | It is the caller's PBX that needs to do that |
| 19:20.00 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 19:20.15 | bapril | the callers pbx can do it, I just need to detect it. |
| 19:21.05 | bapril | I see the flash in the sound-file if I record the call. |
| 19:21.43 | *** part/#asterisk EduHard (n=user@proxy.donto.com.ar) |
| 19:22.48 | ManxPower | bapril: no, you do not see the flash in the sound file. you hear the electrical noise of the flash. noise is not a flash. |
| 19:23.14 | ManxPower | Asterisk does not support any kind of FLASH on any kind of PRI in any way shape or form. |
| 19:23.29 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:25.22 | bapril | I see a pretty distinct waveform associated with the flash. I understand that asterisk does not support it, trying to find alternate means to perform the function. |
| 19:27.46 | ManxPower | bapril: bes of luck |
| 19:29.00 | bapril | thx |
| 19:29.48 | dalfry | ping |
| 19:30.15 | *** join/#asterisk AJaymn (n=mypocket@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 19:30.19 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 19:31.08 | AJaymn | Ive found the Click-2-Call script for webpage.. but it calls you then the # that was entered,, Im looking for 1 that will call the # then the programed # |
| 19:31.39 | *** join/#asterisk noco (n=casey@64.81.142.112) |
| 19:32.09 | noco | what would be the easiest way to grab the number of voicemail messages in a user's inbox and display it on a site using php? |
| 19:32.50 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 19:32.52 | noco | can i script it using the console and 'show voicemail users' and then parse out the number of messages displayed there? |
| 19:34.01 | sysreq | noco: I guess so, using the Asterisk Manager Interface (AMI). |
| 19:34.13 | Katty | http://pastebin.ca/537559 <- yay, almost done. |
| 19:34.46 | HyPnoLORD | [TK]D-Fender .. Just tested the ports, and I cant find the problem :S |
| 19:35.07 | HyPnoLORD | I bet that if I return to Location 1 It is going to work just fine |
| 19:35.42 | ManxPower | HyPnoLORD: the line at location 2 is direct from the telephone company and not into a PBX of some sort? |
| 19:35.57 | ixela | HyPnoLORD: You only have an fxo module on the tdm400p correct? |
| 19:35.58 | HyPnoLORD | :S PSTN Line Voltage is 51 Volts is this correct |
| 19:36.10 | HyPnoLORD | Yes I only have a FXO module |
| 19:36.16 | ManxPower | HyPnoLORD: Maybe. |
| 19:36.17 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 19:36.34 | ManxPower | It is supposed to be -48V, but it can vary |
| 19:36.44 | ixela | HyPnoLORD: have you made sure to configure zapata.conf and zaptel.conf correctly for the location change? |
| 19:36.55 | ManxPower | HyPnoLORD: it sounds like your line is off a PBX |
| 19:37.26 | HyPnoLORD | well I didnt reconfigure. The line comes directly from the Company |
| 19:37.51 | ManxPower | HyPnoLORD: The Phone Company or The Employer Company |
| 19:38.27 | HyPnoLORD | But if it worked fine before at Loc 1 shouldnt it work at Loc 2 if both are Lines that comes directly from the Phone Company |
| 19:38.40 | ManxPower | HyPnoLORD: yes, it should |
| 19:38.48 | ManxPower | unless they are different type of line |
| 19:38.54 | ixela | HyPnoLORD: and you are using the same module with your configuration modified to use port 2 instead of port 1? |
| 19:38.59 | HyPnoLORD | that is what Im trying to find out |
| 19:39.20 | ManxPower | ixela: all he did was move it to a new place and plug a phone line into it |
| 19:39.30 | ManxPower | HyPnoLORD: we can't tell you the type of line you have |
| 19:39.31 | HyPnoLORD | ManxPower Thats right |
| 19:39.33 | ixela | i thought he changed ports on the tdm card |
| 19:40.11 | jcaceres | is it posible to sedn a fax throught a sip client? |
| 19:40.15 | ixela | sorry then, i misread what he said earlier. |
| 19:40.22 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 19:40.28 | jcaceres | does any body know a client that can do that? |
| 19:40.36 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 19:41.00 | HyPnoLORD | ManxPower: Well I know that the Analog Line (loc 1 - My house :P ) Works fine.. but Analog Line (Loc 2 My office) is not working and Im sure that is not going trough an another PBX |
| 19:41.40 | robin_sz | OK, so what I want to do is: incoming call .. phone 1 rings ... then phone 1 and 2 .... then phone 1 and 2 and 3 anf 4 and 5 ... hopefully eventually someone answers |
| 19:41.57 | robin_sz | this must be a very VERY common scenario |
| 19:42.56 | robin_sz | the xample I found on the Wiki works very badly indeed |
| 19:43.08 | HyPnoLORD | [TK]D-Fender ManxPower ixela : Im going to tray a 3rd Line to see What Happens.. Be right back |
| 19:43.43 | robin_sz | [TK]D-Fender, you know that crazy delayed dial thing I was playing with and you were pointing me in the right direction regarding DND etc? |
| 19:44.12 | [TK]D-Fender | robin_sz: You mean that macro-bungled mess? ... uuhh yeah, ;) |
| 19:44.25 | robin_sz | well, I think that approach is doomed |
| 19:44.37 | [TK]D-Fender | robin_sz: I did tell you you could abstract the overrides with about 3 lines of dialplan.... |
| 19:44.49 | Chuji | robin_sz: How many phones are we talking about? |
| 19:44.50 | robin_sz | yes |
| 19:44.52 | [TK]D-Fender | robin_sz: Its not doomed... only your sanity :) |
| 19:45.03 | [TK]D-Fender | robin_sz: My rates are very accessable ;) |
| 19:45.24 | robin_sz | Chuji, 5 max |
| 19:46.26 | robin_sz | [TK]D-Fender, one of the extensions it eventually will dial is a bell in the factory ... we've noticed this sometimes gets rung even though one of the extensions has picked up the call .. it then rings for a whole minute, sometimes even after the call has finished |
| 19:47.05 | [TK]D-Fender | robin_sz: Like I said... you have some OVERRIDING of your macros to do. |
| 19:47.23 | Chuji | robin_sz: creating the dialplan stepping through each would be easy, but skipping ones that were busy would be a little funky |
| 19:47.23 | [TK]D-Fender | robin_sz: This is not Raw Cat science ;) |
| 19:47.39 | [TK]D-Fender | Chuji: Not really... |
| 19:47.46 | [TK]D-Fender | Chuji: I've seen the setup. |
| 19:48.26 | Chuji | Wouldn't it just be Dial(SIP/1) Dial(SIP/1&SIP/2) etc? |
| 19:48.33 | robin_sz | the original setup was easier ... dial a,10,t, dial a&b, 20,t, dial a&b&c&d&e,20,t |
| 19:48.49 | robin_sz | Chuji, thats what it originally did do |
| 19:49.19 | robin_sz | but there are problems with that approach |
| 19:50.08 | robin_sz | eg sip1 gets a call for 10 seconds, then its dropped and then gets the call again (along with sip/2) ... so it registers as 2 missed calls |
| 19:50.19 | *** part/#asterisk AJaymn (n=mypocket@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 19:50.24 | [TK]D-Fender | robin_sz: Its only your silly marco, get over it! * is fine. Dial is fine. Your macro..... needs a ltitle work ;) |
| 19:50.49 | robin_sz | and if the phoen can't drop the first oen and setup for the second one .l. then all hell breaks loose |
| 19:51.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:51.13 | robin_sz | [TK]D-Fender, its nto my macro, its the as-supplied [std-extension] macro |
| 19:51.41 | [TK]D-Fender | robin_sz: The fact you cut & pasted it just means your IGNORANT *AND* GUILTY ;) |
| 19:51.56 | robin_sz | s/your/you're/ |
| 19:52.15 | [TK]D-Fender | s/your/you're/ |
| 19:52.22 | [TK]D-Fender | THERE.... happy? |
| 19:52.26 | [TK]D-Fender | :D |
| 19:52.37 | robin_sz | always check your spelling when accusing people of being ignorant ;) ;) ;) |
| 19:52.52 | robin_sz | happy, no .. but .. I'm working on it |
| 19:53.09 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 19:53.11 | [TK]D-Fender | I'm multi-tasking far beyond my means right now ... |
| 19:53.26 | shido6 | hehe |
| 19:53.43 | robin_sz | OK, I'll go work on thr macro again |
| 19:54.02 | [TK]D-Fender | robin_sz: *yay* |
| 19:54.59 | jcaceres | is it posible to sedn a fax throught a sip client? |
| 19:55.02 | jcaceres | does any body know a client that can do that? |
| 19:55.11 | *** join/#asterisk thansen|laptop (n=thansen@151.155.248.199) |
| 19:55.28 | *** join/#asterisk BSD_Tech (n=BSDTech@ppp-69-239-114-108.dsl.irvnca.pacbell.net) |
| 19:55.47 | robin_sz | sip soft client or hardware client? hardware clients with u/a law seem to work fine |
| 19:55.52 | [TK]D-Fender | jcaceres: Fax over VoIP = serious pain with *. Do NOT go there.... |
| 19:56.30 | robin_sz | that true .. we terminate directly to ISDN, that works |
| 19:56.57 | robin_sz | hylafax and an old modem is a better plan |
| 19:57.10 | robin_sz | plenty of hylafax clients out there |
| 19:58.43 | jcaceres | i meant a soft client |
| 19:58.52 | [TK]D-Fender | indeed. Get a separate analog line that * has NOTHING to do with and run Hylafax on it. |
| 19:59.10 | [TK]D-Fender | robin_sz: Oh, and BTW there was nothing wrong with my spelling ;) |
| 19:59.30 | jcaceres | and a regulam modem |
| 19:59.31 | jcaceres | ? |
| 19:59.39 | [TK]D-Fender | Grammar Rangers.... ATTACK!!!!! |
| 19:59.44 | [TK]D-Fender | jcaceres: Yes. |
| 19:59.56 | [TK]D-Fender | jcaceres: go to www.hylafax.org and start reading |
| 20:00.06 | jcaceres | thnks |
| 20:00.49 | robin_sz | [TK]D-Fender, btw at the risk of being a pedant, 'your' was incorrect. |
| 20:02.32 | [TK]D-Fender | robin_sz: It's selection, not its spelling :) That would be a grammar error, not a spelling one. If I'm to be hung for my crimes... at least get the crime right ;) |
| 20:02.49 | thansen|laptop | do I need a sound card in my server to make outbound calls "talk"? |
| 20:02.57 | [TK]D-Fender | thansen|laptop: No |
| 20:03.34 | thansen|laptop | [TK]D-Fender: thanks...I've dialed from the console..is there anyway I can test it out..like make it say a number or something |
| 20:03.49 | [TK]D-Fender | thansen|laptop: What have you configured so far? |
| 20:04.08 | robin_sz | [TK]D-Fender, you're loosing the argument ;) |
| 20:04.09 | thansen|laptop | [TK]D-Fender: I have my incoming sip account..a few extensions |
| 20:05.14 | HyPnoLORD | BRB |
| 20:05.16 | HyPnoLORD | quit |
| 20:05.26 | [TK]D-Fender | robin_sz: Nope, a comfortable stand-still :) |
| 20:06.22 | [TK]D-Fender | thansen|laptop: go try and use your account to dial out, etc. |
| 20:06.23 | *** join/#asterisk notoriousrab1982 (n=root@76.195.14.206) |
| 20:06.40 | thansen|laptop | [TK]D-Fender: I already dialed out successfully |
| 20:06.57 | [TK]D-Fender | thansen|laptop: So inbound is not working so great? |
| 20:06.57 | thansen|laptop | just using the console...and one of my extensions as well |
| 20:07.10 | thansen|laptop | inbound is fine as well |
| 20:07.41 | [TK]D-Fender | thansen|laptop: So.....the problem is what exactly? |
| 20:08.16 | thansen|laptop | [TK]D-Fender: from the console/server...I want to dial and send out some "sound" |
| 20:08.33 | thansen|laptop | like make it say a number or a greeting or something |
| 20:08.46 | [TK]D-Fender | thansen|laptop: I somehow thought you'd have set up a phone of some kind. You have NOT done this? |
| 20:09.02 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:09.27 | thansen|laptop | [TK]D-Fender: I've got a couple extensions connected which work fine..but I want to make the server talk |
| 20:11.00 | [TK]D-Fender | thansen|laptop: When, how, to "say" what? |
| 20:11.00 | Katty | right. |
| 20:11.00 | thansen|laptop | [TK]D-Fender: well, I'm toying with an automated voice system, I just am trying to find out how to make it playback a number or greeting or something |
| 20:11.00 | Katty | so i was thinking... |
| 20:11.03 | Katty | most of our clients don't know our last name. and the directory program asks for the first three letters of the person's last name. is there a way to switch those around. |
| 20:11.09 | [TK]D-Fender | thansen|laptop: Ah, you want to make an IVR. Well.... time to crack opten the BOOK, and check out the WIKI |
| 20:11.10 | [TK]D-Fender | ~book |
| 20:11.21 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:11.23 | [TK]D-Fender | ~wikis |
| 20:11.24 | jbot | [wikis] http://www.voip-info.org |
| 20:11.26 | Katty | or, does that simply mean you have to swap the names around in voicemail.conf and simply overwrite the audio files? |
| 20:11.38 | [TK]D-Fender | thansen|laptop: on the WIKI go lookup "asterisk IVR tips" and keep in mind its quite deprecated, but a place to start. |
| 20:11.52 | thansen|laptop | [TK]D-Fender: rock on! thanks |
| 20:12.09 | [TK]D-Fender | Katty: "show application directory" <- you can tell it to use the FIRST name. |
| 20:12.35 | Katty | neat |
| 20:12.36 | [TK]D-Fender | thansen|laptop: No probelm. This will take some work on your part, but its not that much really. |
| 20:13.05 | thansen|laptop | [TK]D-Fender: I'll go have a look..I might be back :) |
| 20:17.52 | *** join/#asterisk Netgeeks_ (n=root@pbx5.netgeeks.net) |
| 20:19.48 | [TK]D-Fender | Katty: you have a context in voicemail.conf. this is used to see which list of people it will look through. the OTHER is the DIALPLAN context it will take the box # as an exten in and dial out to. |
| 20:21.04 | Hmmhesays | I guess i'm going to a taylor guitar demonstration tonight |
| 20:21.10 | Katty | [TK]D-Fender: oooh, k'then |
| 20:22.10 | Katty | [TK]D-Fender: so Directory(downstairs[|downstairs[|b]])? |
| 20:22.33 | Katty | [TK]D-Fender: i have a hard time following syntax ^_- |
| 20:23.10 | *** join/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 20:25.00 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 20:25.36 | [TK]D-Fender | Katty: .... |
| 20:25.50 | Katty | lol, fine, i'll ask someone else (= |
| 20:25.56 | *** join/#asterisk coolfreecode (i=root@190.41.82.6) |
| 20:26.03 | Katty | [TK]D-Fender: don't bother stooping to my level ;) |
| 20:26.18 | aptura | I am trying to lookup 24 pair pinout for t1 but I have not seen site with the color coding of the wires included. If anyone knows of a url that would be helpfull. |
| 20:26.33 | [TK]D-Fender | Katty: I mean its 3 parms and they show you the order in the sample! |
| 20:26.37 | drewr | Is there any way to use virtual channels with a conference bridge so that my PRI channels don't get eaten up when internal phones are connected? |
| 20:26.58 | [TK]D-Fender | drewr: What is a "vitual channel"? |
| 20:27.16 | [TK]D-Fender | drewr: Just have your local phones call MeetMe direct |
| 20:27.41 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 20:27.42 | [TK]D-Fender | aptura: T1 use 2 pair, not 24.... |
| 20:28.12 | coolfreecode | @_@ |
| 20:28.21 | drewr | [TK]D-Fender: How do I figure out what extension that is? For example, we just call x601 to get in a room, but that eats up an inbound channel. |
| 20:29.11 | [TK]D-Fender | drewr: its your dialplan.... this is 101 stuff... |
| 20:30.00 | Mercestes | <PROTECTED> |
| 20:30.20 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 20:30.22 | aptura | TK that part I know but what about the cases of 24 pair cabling comming onto a backbboard? |
| 20:30.36 | drewr | [TK]D-Fender: I'm sure it is. I inherited this system and piecing information together as I go. Thanks. |
| 20:30.54 | [TK]D-Fender | drewr: Starting from scratch huh? |
| 20:30.59 | Mercestes | aptura, are you referring to an ansenol cable coming from a T1 adtran? |
| 20:31.21 | [hC] | anyone here recommend any PoE switches that do CDP? |
| 20:31.34 | drewr | [TK]D-Fender: Well, I've learned a fair amount, but not having set it up I always find a nook or cranny I don't know anything about. |
| 20:31.34 | [hC] | the netgears i have been using dont seem to support it. and hp just removed support. |
| 20:31.49 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 20:32.09 | drewr | [TK]D-Fender: Today we ran out of outbound channels because of a conference call. I didn't realize that internal participants took a channel. |
| 20:32.21 | [TK]D-Fender | [hC]: How abaout a Catalyst ;) |
| 20:32.24 | [hC] | drewr: they dont. |
| 20:32.32 | [hC] | [TK]D-Fender: yeah, there's always that. :) $$ |
| 20:32.38 | [TK]D-Fender | drewr: Depends how you set them up to dial into it. |
| 20:32.42 | drewr | [hC]: "show channels" showed them as all active. |
| 20:32.54 | drewr | [TK]D-Fender: Can you point me to some reading about how to reconfigure that? |
| 20:33.05 | aptura | Mercestes cable came from outside plant "I suspect" and terminated on the punch down block. I am tying to learn everything about T1 specifications. |
| 20:33.07 | [TK]D-Fender | drewr: Um.... EVERY call is a "channel", I was presuming you meant concerning your PRI |
| 20:33.19 | [hC] | drewr: well... sure, they are a channel,but if you're calling into a meetme from a locally connected extension you wont take up an outbound zap/iax/sip channel |
| 20:33.24 | [TK]D-Fender | drewr: A SIP phone accessing its voicemail is a channel. |
| 20:33.37 | [TK]D-Fender | drewr: Doesn't mean its eating up your PRI... |
| 20:33.50 | drewr | Ah. I see. |
| 20:34.22 | drewr | So why would I get an operator while trying to dial outbound? |
| 20:34.48 | drewr | [hC]: OK, then perhaps there was another issue. |
| 20:35.46 | [TK]D-Fender | drewr: Quite likely. |
| 20:36.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 20:39.51 | *** join/#asterisk FF|Elliot (n=elliot@host81-155-72-43.range81-155.btcentralplus.com) |
| 20:40.22 | aptura | Avaya reporedly exploring a sale |
| 20:40.39 | FF|Elliot | hi i keep getting this error Unable to create channel of type 'IAX2' (cause 3 - No route to destination) but my phone only uses SIP |
| 20:40.53 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 20:40.53 | aptura | Seems the company wants to sell part of all of its corporation. |
| 20:40.58 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 20:42.27 | [TK]D-Fender | FF|Elliot: Do you regularly talk in English to people who only speak Greek? |
| 20:42.42 | FF|Elliot | no? |
| 20:43.56 | FF|Elliot | is anybody able to help? |
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| 21:06.36 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 21:07.57 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:08.24 | *** join/#asterisk rocket007 (n=youga@86.99.208.131) |
| 21:09.21 | [[blah]asfd | I am trying to correct something causing my asterisk to drop calls. I have been told to adjust my mtu. I am trying to find where I do that and am not finding it. could anyone guide me? |
| 21:09.48 | rocket007 | Cisco Vlan question- Is there a way I can utilize support in my cisco switches for Voice VLANs, with asterisk and polycom fones ? |
| 21:12.52 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 21:18.24 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:19.01 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
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| 21:23.53 | JoeDeveloper | Hi. I am running Asterisk 1.4 on Fedora Linux. When I call the "Playtones(ring)" command, I get nothing. Everything else seems to work ok. Is the ring tone a file that maybe is missing on my config or something? |
| 21:25.03 | *** join/#asterisk ruied (n=ruied@bl10-124-203.dsl.telepac.pt) |
| 21:26.45 | *** part/#asterisk nasls_lsa (n=chatzill@athedsl-136847.home.otenet.gr) |
| 21:27.45 | ruied | Hi, how can I make a decision with the inbound caller id, I have a TDM400, running on my system. I would like to make an automatic voip call, autenticated by the inbound call number trough ZAP/4 is that possible? |
| 21:28.10 | ruied | the callerID appears in my phone.... |
| 21:29.19 | [TK]D-Fender | ruied, Sure. "show application gotoif" |
| 21:29.32 | [TK]D-Fender | ruied, "show function CALLERID" |
| 21:30.17 | ruied | [TK]D-Fender, thanks! going to check! :) |
| 21:33.27 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
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| 21:46.52 | *** join/#asterisk EricL (n=eric@74.9.83.194) |
| 21:47.14 | EricL | Why do I keep getting: No application 'SetVar' for extension (internal, 9100, 3) |
| 21:47.48 | EricL | Isn't SetVar() a command? |
| 21:49.35 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:50.02 | Nate9939 | funny thing happening, i'm reloading my extensions.conf file with dialplan reload, and it is reloading some extensions.conf i have no idea where from.. I am changing the /etc/asterisk/extensions.conf file, but it is loading something else. |
| 21:50.14 | Nate9939 | where should i be saving my extensions.conf file? |
| 21:50.31 | centrex | Nate9939, check your /etc/asterisk/asterisk.conf file to see where it is set to load your dialplan from. |
| 21:50.42 | Nate9939 | ok doke. |
| 21:50.53 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 21:51.27 | centrex | Nate9939, astetcdir => /etc/asterisk is what it should say if it's from /etc/asterisk |
| 21:51.29 | Nate9939 | what is it listed as under the directories headings? |
| 21:52.11 | Nate9939 | doesn't look like as far as i can tell there is any directory set for it. |
| 21:52.26 | centrex | Did you do a make samples when you compiled it? |
| 21:52.43 | Nate9939 | yes. |
| 21:53.12 | centrex | Hrm. well there should be a section under directories that says astetcdir |
| 21:53.26 | Nate9939 | ok there is, its set at /etc/asterisk |
| 21:53.47 | Nate9939 | and thats where my extensions.conf file is. |
| 21:54.11 | centrex | you might have your extensions.conf misconfigured. |
| 21:54.45 | Nate9939 | hrmm, its just very basic, it only has 4 lines. |
| 21:54.51 | [TK]D-Fender | EricL, Welcome to the wonderful world of DEPRECATION. |
| 21:54.55 | Nate9939 | 1 context and 3 lines. |
| 21:55.01 | [TK]D-Fender | EricL, SetVar was replace by Set in 1.2 |
| 21:55.22 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 21:57.19 | EricL | [TK]D-Fender:Ah, thanks. Wish that one was better documented as the MeetMe stuff on VoIP-info.org still uses SetVar(). |
| 21:57.19 | Nate9939 | this may help, the dialplan it is loading says all contexts where made by pbx_config, or pbx_ael |
| 21:58.22 | ruied | [TK]D-Fender, is something like: ' exten =>s,1,GotoIf(CALLERID(num,918116999)?3:4)' being s,3 the true condition and s,4 the false one? |
| 21:58.28 | *** part/#asterisk rocket007 (n=youga@86.99.208.131) |
| 21:58.31 | [TK]D-Fender | EricL, voip-info is not official documentation. This is included in all the changelogs, in the /docs folder etc. |
| 21:58.47 | [TK]D-Fender | EricL, when in doubt : "show applications" "show functions" |
| 22:00.16 | EricL | [TK]D-Fender:I know, but its the best reference besides the two books out there (which are slowly becoming dated). |
| 22:00.22 | ruied | and the 91........ the inbound callerID ? |
| 22:00.45 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 22:02.04 | [TK]D-Fender | ruied, Nope. Go look on the WIKI for "asterisk expressions" to learn how to do your test. and while you're there, "asterisk functions" |
| 22:02.27 | ruied | ok |
| 22:02.31 | [TK]D-Fender | ruied, getting warmer though |
| 22:03.27 | ruied | :) thanks |
| 22:04.25 | *** join/#asterisk Mad|Cow (n=madcow@63.96.151.145) |
| 22:05.15 | Nate9939 | hey tkd sorry to bother you, can't get my * box to load the correct extensions.conf file when doing a dialplan reload, anything i can try? |
| 22:05.15 | Mad|Cow | Anyone have any experience with the Cisco 7936 with Asterisk. Is a conf. phone... but it only supports skinny... trying to figure out if it will work or not with Asterisk |
| 22:05.38 | Qwell[] | Mad|Cow: There is a patch on our bug tracker for a 7935, but it should work for a 7936 also - please do test it |
| 22:05.46 | Qwell[] | (and report back results, so I can finally commit it...) |
| 22:06.51 | Mad|Cow | @Qwell[]: I'd be happy to. Could you point me in the direction of the bug? |
| 22:07.06 | Qwell[] | bugs.digium.com - do a search for 7935 |
| 22:10.08 | *** join/#asterisk Mad||Cow (n=madcow@63.96.151.145) |
| 22:11.05 | [TK]D-Fender | Nate9939, pastebin the CLI output of the failure, and an "ls -l" dump of the folder its in with a "cat" dump whil you're at it (and w/ the CLI command to back it. |
| 22:12.25 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:14.41 | Nate9939 | tkd no probl. |
| 22:16.27 | Nate9939 | cat dump, how do i cat all the files? |
| 22:18.03 | [TK]D-Fender | ~pb |
| 22:18.20 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 22:18.24 | [TK]D-Fender | "cat /etc/asterisk/extensions.conf" etc..... |
| 22:20.32 | Nate9939 | http://pastebin.ca/537981 |
| 22:22.40 | Nate9939 | tkd i updated that paste with the dial plan that it DOES load... |
| 22:22.42 | Nate9939 | http://pastebin.ca/537988 |
| 22:23.00 | yannj_fr | I started using realtime, odb workfine with postgresql , I had a user but get the error on CLI : device not match ACL |
| 22:23.22 | yannj_fr | any idea? |
| 22:25.54 | [TK]D-Fender | Nate9939, You did NOT provide me the linux CLI command showing your "cat" for your dialplan. |
| 22:25.55 | *** join/#asterisk Mad|Cow (n=madcow@63.96.151.145) |
| 22:26.09 | [TK]D-Fender | Nate9939, I said I wanted to see EVERYTHING. |
| 22:26.56 | Nate9939 | tkd, i did i thought, i did the cat for extensions.conf, then did the show dialplan at the cli for the dialplan it does load, check it again, as i updated it 2 times. |
| 22:28.17 | Nate9939 | the cat for extensions.conf is right after directory listing, its only 4 lines. |
| 22:28.46 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 22:30.36 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 22:31.19 | [TK]D-Fender | Nate9939, well that 91 byte extensions.conf sure SEEMS to be whats loaded.... so whats the problem? |
| 22:31.35 | [TK]D-Fender | Nate9939, (I pasted it MYSELF intoa file. indeed 91 bytes) |
| 22:32.12 | ruied | [TK]D-Fender, it's working! :) thanks... |
| 22:33.10 | [TK]D-Fender | ruied, glad to hear.... probably looks like GotoIf($["${CALLERID(num)}"="918116999"]?3:4) |
| 22:33.51 | ruied | yep |
| 22:33.55 | [TK]D-Fender | :) |
| 22:34.06 | ManxPower | I assume 1.4 doesn't require spaces around the = |
| 22:34.12 | yannj_fr | I started using realtime, odb workfine with postgresql , I had a user but get the error on CLI : device not match ACL , any idea?? |
| 22:34.24 | [TK]D-Fender | ruied, Sure ... I could just GIVE it to you.. but knowledge earned > cut & paste |
| 22:34.38 | ruied | yep ;) |
| 22:34.40 | [TK]D-Fender | yannj_fr, we heard you 10 minutes ago...... |
| 22:34.51 | yannj_fr | ok |
| 22:34.55 | rbd | hi guys, Asterisk 1.2.13... I have an asterisk app that runs a lot like the MP3Player app (i.e. executes an external app, and pipes in the data from stdin. which is piped to the stdout of the external app)...it is controlled by an AGI script that accepts DTMF 1 to end it and recall it with a different agent ID param (it is for silent monitoring agents). however, sometimes a caller will be dropped after advancing through conversations |
| 22:34.55 | ManxPower | [TK]D-Fender: Build a fire for a man and keep him warm for a night. Light a man on fire and keep him warm the rest of his life. |
| 22:35.01 | rbd | the log we get is at: http://cut.and.paste.org/index.php?id=1156 ....any ideas? |
| 22:35.14 | [TK]D-Fender | ManxPower, I try to save that quote for more special occasions ;) |
| 22:35.14 | Nate9939 | tkd sorry for wasting you time, i just saw all that. pbx_ael stuff when doing show dial plan, so didn't see my little 4 lines at the top. |
| 22:35.25 | Nate9939 | what the heck is pbx_ael anyways? |
| 22:35.29 | [TK]D-Fender | Nate9939, Go caffeinate! |
| 22:35.35 | [TK]D-Fender | Nate9939, AEL. |
| 22:35.36 | [TK]D-Fender | ~ael |
| 22:35.48 | jbot | from memory, ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 22:35.49 | [TK]D-Fender | Nate9939, wake up to 2 YEARS AGO :) |
| 22:36.08 | [TK]D-Fender | Nate9939, and that'd be the contents of extensions.ael |
| 22:36.34 | *** join/#asterisk CoolGuy21 (n=77889789@cpe-76-173-56-41.socal.res.rr.com) |
| 22:36.36 | CoolGuy21 | hi |
| 22:36.55 | CoolGuy21 | i have asterisk 1.2, how can i setup a wakup call service? |
| 22:37.16 | *** join/#asterisk daveburr (i=Miranda@66.7.122.92) |
| 22:38.03 | yannj_fr | CoolGuy21 : i was thinking about that |
| 22:38.48 | CoolGuy21 | yannj_fr any good news? |
| 22:38.48 | [hC] | Anyone here using PoE switches w/ CDP support for VLAN ID auto provisioning, (and not cisco switches) |
| 22:38.49 | [TK]D-Fender | CoolGuy21, lookup ".call files" and "AMI Originate" on the WIKI |
| 22:38.52 | [TK]D-Fender | ~wikis |
| 22:38.58 | jbot | i guess wikis is http://www.voip-info.org |
| 22:39.01 | ManxPower | CoolGuy21: your extensive search of the MAILING LIST ARCHIVES and the WIKI was not helpful? |
| 22:39.08 | yannj_fr | thought about a daemon checking subscription to wake and creating .call filles |
| 22:39.15 | ManxPower | ~malinglist |
| 22:39.19 | [TK]D-Fender | [hC], Get a mid-span PoE injector like the PowerDsine series |
| 22:39.19 | ManxPower | ~mailinglist |
| 22:39.21 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 22:39.43 | ManxPower | [TK]D-Fender: I doubt a midspan will support VLAN ID via CDP |
| 22:39.53 | ManxPower | [hC]: why not Cisco? |
| 22:40.11 | [hC] | ManxPower: $$$. |
| 22:40.15 | [TK]D-Fender | ManxPower, because he's CHEAP.... and still stuck with Cisco's ;) |
| 22:40.24 | ManxPower | yannj_fr: Hint: If you create the call file with a FUTURE date it won't run until that date. |
| 22:40.25 | [hC] | I have been using NetGear or HP, but they both recently removed support for CDP |
| 22:40.38 | ManxPower | [hC]: We spend like $1200 on a 48 port Cisco |
| 22:40.46 | [hC] | Err. |
| 22:40.46 | ManxPower | Cisco 5509 with SPECIFIC cards off ebay |
| 22:41.05 | ManxPower | I spent like $300 on my 24 port cisco |
| 22:41.11 | yannj_fr | ManxPower : ok, so everyday you recreate file , true? with a cron job for example? |
| 22:41.14 | ManxPower | I think mine is a 5505 |
| 22:41.16 | [hC] | I have been spending about $140 on HP or Netgear switches, and the cisco ones ive been looking at are like, $6-9000/ea |
| 22:41.22 | [hC] | Which is why i said $$$ |
| 22:41.31 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 22:41.39 | *** part/#asterisk daveburr (i=Miranda@66.7.122.92) |
| 22:41.46 | [hC] | Id prefer 24/48 port PoE capable (IEEE 802.3af) and VLAN capabilities, thats it. |
| 22:41.48 | ManxPower | [hC]: your problem is not that Cisco (used) is so expensive, but that the switches you use are so cheap. |
| 22:42.03 | ManxPower | [hC]: many used cisco switches are cheap |
| 22:42.07 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 22:42.13 | [hC] | I cannot resell used switches to business clients. |
| 22:42.16 | ManxPower | BUT, you will still need a power injector. |
| 22:42.27 | ManxPower | [hC]: even if they are cisco certified. |
| 22:42.28 | [hC] | The switches i am talking about provide PoE |
| 22:42.36 | ManxPower | you'll pay more for a certified used..... |
| 22:42.56 | yannj_fr | try nortel |
| 22:43.02 | yannj_fr | or huawey |
| 22:43.05 | [hC] | It has nothing to do with certified or not, really.. they dont know the difference, but if i buy a used one, if it breaks, i have no warranty on it or anything. |
| 22:43.10 | yannj_fr | or extreme |
| 22:43.12 | ManxPower | [hC]: Well when you find what you need in the price range you need, look around and see if there is a Unicorn nearby. I'm looking for one. |
| 22:43.28 | ManxPower | [hC]: certified allows you to buy a service contract. |
| 22:43.40 | [hC] | ManxPower: so far there is nothing that the HP or Netgear cannot do, aside from CDP. |
| 22:43.55 | ManxPower | You don't really want the 4 week turn around time for warrenty repairs anyway |
| 22:44.11 | ManxPower | [hC]: As I said, when you find what you are looking for...... |
| 22:44.28 | [hC] | ManxPower: I guess the most common thing to look for then would be a Cisco 3550 w/ PoE |
| 22:44.32 | ManxPower | We have like 15 of the Cisco 550x switches in production. No problems. |
| 22:44.45 | [hC] | [TK]D-Fender: whats the cluebat for? what do you use? |
| 22:44.59 | ManxPower | [hC]: find a switch that does CDP, then find a power injector |
| 22:45.38 | [hC] | ManxPower: why wouldnt i just get a switch that does CDP -and- PoE? |
| 22:45.47 | JT | hp can't do cdp? |
| 22:45.51 | ManxPower | [hC]: price |
| 22:45.57 | [hC] | JT: they just recently removed CDP support in favor of LLDP |
| 22:45.58 | ManxPower | built in POE is expensive |
| 22:46.02 | [hC] | (I hadnt heard of LLDP either) |
| 22:46.25 | [hC] | I just found a 3550 with PoE on ebay for $1500 |
| 22:46.29 | [hC] | thats completely reasonable. |
| 22:46.35 | yannj_fr | http://www.huawei.com/products/datacomm/detailitem/view.do?id=963&rid=66 |
| 22:46.52 | JT | cisco gear is a waste of money unless you have support contracts and what not |
| 22:46.54 | [hC] | yannj_fr: thanks for the url, but who the heck is huawei? |
| 22:46.56 | JT | unsupported |
| 22:46.59 | JT | unwarranted |
| 22:47.02 | JT | unfriendly company |
| 22:47.25 | yannj_fr | huawei is the "chinese cisco" |
| 22:47.31 | *** join/#asterisk ertyu (i=left@S010600d0b7928a07.wp.shawcable.net) |
| 22:47.35 | [hC] | JT: here is my only dilemma. I dont care if i use cisco, or hp, or netgear, or whoever. I simply want to be able to auto provision phones without interaction. |
| 22:47.48 | [hC] | JT: so far, the only thing i can tell that allows this to happen (with polycom anyways) is CDP. |
| 22:47.56 | [hC] | And by auto provision i mean VLAN ID discovery |
| 22:47.58 | JT | err |
| 22:48.00 | yannj_fr | they were a subcontracter for cisco IOS coding |
| 22:48.07 | JT | i see |
| 22:48.19 | JT | hp has lifetime warranty, that seals the deal for me |
| 22:48.22 | [hC] | In a shared wiring environment, I prefer to run all the phones on an isolated vlan. |
| 22:48.53 | [hC] | hp's switches are great, i like them. but since they removed support for CDP I cant accomplish what i need, unless i can get all the other phone manufacturers to implement LLDP |
| 22:49.41 | ManxPower | JT: Only when it is new. |
| 22:49.45 | JT | i'm surprised that a proprietary cisco protocol is the only way to provision a polycom |
| 22:49.50 | [hC] | I definitely do not like the idea of a midspan injector though, its just another piece to break. |
| 22:49.50 | JT | ManxPower: what? |
| 22:50.06 | JT | [hC]: midspan injector is probably not a good idea |
| 22:50.11 | ManxPower | JT: it isn't. It's the only way to automatically assign the vlan..... until SIP 2.0 and recent bootroms |
| 22:50.18 | [hC] | JT: well they have this thing called DHCP VLAN ID Discovery, however, it is disabled by default ...... (wtf?) |
| 22:50.19 | JT | cisco do make multiport poe injectors, but still |
| 22:50.28 | JT | ManxPower: that's what i mean |
| 22:50.41 | JT | ManxPower: < ManxPower> JT: Only when it is new. <--- ? |
| 22:50.48 | [hC] | ManxPower: there are other ways now with new bootrom and sip2.0? |
| 22:50.51 | ManxPower | JT: Cisco is great, and if you can use used Ciscos the price is not bad either. |
| 22:51.07 | yannj_fr | [hC] are you sure for CDP and polycom |
| 22:51.10 | JT | i don't like cisco that much |
| 22:51.12 | [hC] | ManxPower: the only other thing i found was DHCP VLAN Id discovery, which seemed to be disabled by default, hence unusable without interction. |
| 22:51.25 | ManxPower | [hC]: there is some sort of device.xxx options that allow you to set the bootrom options and reboot the phone, all from the config file. |
| 22:51.57 | JT | ManxPower: what were you talking about... when some item is new? |
| 22:51.58 | ManxPower | We use CDP switches so I've never had to use it, I read it in the release notes or the admin guide |
| 22:52.05 | [hC] | ManxPower: so, that would then require the phone to provision itself once on the wrong vlan, resave settings, reboot onto the correct vlan, and provision again and go. |
| 22:52.26 | [hC] | ManxPower: because by default the dhcp discovery option is off. |
| 22:52.39 | ManxPower | [hC]: correct |
| 22:53.04 | yannj_fr | isnt it 802.1d that provide vlan id discovery? |
| 22:53.11 | JT | earth to ManxPower |
| 22:54.28 | ManxPower | JT: Ciscos are good equipment, but the price does not make the a good value. Used Cisco equiopment (eBay or used networking vendor) is much cheaper and so a much better value than buying it new. |
| 22:54.42 | ManxPower | Used Cisco equipment is so cheap you might as well just get a spare box. |
| 22:54.47 | ManxPower | JT: Do you understand now? |
| 22:55.05 | JT | i swear you were replying to < JT> hp has lifetime warranty, that seals the deal for me |
| 22:55.18 | ManxPower | If you really must have a service contract, spend a little more and get a certified used cisco box |
| 22:55.37 | CoolGuy21 | all the wakeup scripts dont have an option to put the phone number you want it to call |
| 22:55.55 | ManxPower | JT: no |
| 22:56.28 | [hC] | It appears as though LLDP is the discovery protocol replacing CDP, and its open |
| 22:56.35 | JT | ManxPower: but basically you're deploying unsupported and unwarranted hardware if you deploy a used cisco that isn't covered by an agreement with cisco |
| 22:56.38 | [hC] | so i guess if everyone jumps on the bandwagon there, we'll be set. |
| 22:56.55 | [TK]D-Fender | CoolGuy21, <[TK]D-Fender> CoolGuy21, lookup ".call files" and "AMI Originate" on the WIKI |
| 22:56.55 | [TK]D-Fender | <[TK]D-Fender> ~wikis |
| 22:56.55 | [TK]D-Fender | <jbot> i guess wikis is http://www.voip-info.org |
| 22:58.05 | ManxPower | JT: CERTIFIED Cisco stuff qualifies for support contracts. one of the vendors we use even will give you a refund if cisco refuses your service contract request. |
| 22:58.07 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 22:58.38 | JT | is it warranted? |
| 22:58.38 | ManxPower | We don't use many support contracts on our equipment, we just buy a spare. |
| 22:58.43 | JT | heh |
| 22:58.44 | J4k3 | Aren't there third party Cisco support groups? |
| 22:58.54 | JT | i understand the logic but if it's a customer site... |
| 22:59.02 | [TK]D-Fender | J4k3, Right next to Alcohol Anonymous ;) |
| 22:59.07 | J4k3 | I mean, Vendor support isn't always the best way to get service. |
| 22:59.13 | JT | IOS upgrades are controlled by cisco |
| 22:59.30 | [hC] | personally i dont need support for implementation, its when the thing dies. |
| 22:59.36 | BSD_Tech | these shoes rule those shoes suck |
| 22:59.38 | J4k3 | [hC]: buy two. |
| 22:59.38 | JT | exactly |
| 22:59.47 | JT | buy hp :) |
| 22:59.50 | J4k3 | spares are the best |
| 23:00.04 | J4k3 | warranty = downtime |
| 23:00.08 | yannj_fr | hp arent so good product |
| 23:00.14 | JT | i prefer to support a company that isn't an absolute arse to its customers |
| 23:00.14 | yannj_fr | we stop installing it |
| 23:00.18 | J4k3 | spares = swap the crap and go get a beer. |
| 23:00.38 | JT | spares are useful |
| 23:00.38 | yannj_fr | because of too much die |
| 23:00.38 | ManxPower | JT: if you mean the 90-day warrenty you get, no it is not warrented. |
| 23:00.38 | JT | yannj_fr: what sort of hp? |
| 23:00.49 | ManxPower | If you mean "can I get a next business day support contract" then the answer is "yes" |
| 23:00.53 | JT | ManxPower: hp procurves have lifetime warranties |
| 23:01.05 | J4k3 | JT: and in a week, they'll get you a replacement. |
| 23:01.07 | J4k3 | ;) |
| 23:01.11 | ManxPower | JT: How long from the time it breaks until you get it back. |
| 23:01.17 | yannj_fr | 3xxx |
| 23:01.31 | J4k3 | BUY SPARES, thats the best answer. |
| 23:01.42 | [hC] | there is no debating in my opinion that service contracts on switches, and warranties are pretty much useless when you can just buy another |
| 23:01.45 | [hC] | and learn how to set them up. |
| 23:02.03 | [hC] | The real factor here comes from functionality, reliability, and price. |
| 23:02.19 | JT | yannj_fr: what product.... |
| 23:02.40 | yannj_fr | dont remember, I didnt installed them |
| 23:02.50 | J4k3 | all my cisco crap is ancient. It was such junk when I bought it (for high dollar) that I do everthing possible to avoid buying any more of their stuff. |
| 23:03.07 | yannj_fr | I only handle cisco |
| 23:03.22 | yannj_fr | when we need cheap product, we use 3com |
| 23:03.26 | JT | yannj_fr: what are they, switches or what? stop being so vague |
| 23:03.26 | ManxPower | J4k3: what did you buy? |
| 23:03.44 | yannj_fr | L2 switches |
| 23:03.45 | J4k3 | ManxPower: Cisco 4500 |
| 23:03.55 | ManxPower | J4k3: you poor thing |
| 23:03.58 | JT | yannj_fr: i see |
| 23:04.12 | J4k3 | ManxPower: with the extremely overpriced completely worthless 100mbit ethernet interface! |
| 23:04.17 | ManxPower | I think we spent under $2,000 for our 3604 |
| 23:04.28 | ManxPower | and that included the cards we needed |
| 23:04.31 | J4k3 | my border router is a PC with a sync serial card now. |
| 23:04.37 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:04.44 | J4k3 | and a linuxy OS (Mikrotik RouterOS) |
| 23:04.58 | J4k3 | I paid about $1500 for the 4500... that was like 1998. |
| 23:05.00 | yannj_fr | j4k3, wich serial card? |
| 23:05.04 | J4k3 | our 2501 ran out of horsepower |
| 23:05.07 | [hC] | All I want is a switch with at least 24 port port desnsity, PoE support NATIVELY, and preferrably CDP, LLDP, or some sort of device discovery/vlan provisioning system for under 200$ |
| 23:05.10 | ManxPower | The ONLY Cisco stuff we buy new are 2621 routers and even those we are starting to look at used 2621s |
| 23:05.13 | J4k3 | yannj_fr: etinc pcisync 4-port. |
| 23:05.27 | JT | [hC]: are you dreaming? |
| 23:05.37 | [hC] | JT: Ive found it, twice. just no CDP. |
| 23:05.43 | ManxPower | GOOD LORD! You didn't sell Asterisk on PRICE, did you? |
| 23:05.50 | JT | [hC]: what item? |
| 23:05.58 | yannj_fr | CDP is cisco property |
| 23:06.04 | [hC] | JT: The HP ProCurve 24 port PoE switch (which USED to do CDP) we buy for $1300 |
| 23:06.17 | JT | proprietary |
| 23:06.26 | JT | [hC]: hell of a lot more than $200 |
| 23:06.33 | [hC] | oh shit |
| 23:06.35 | [hC] | I meant $2000 |
| 23:06.37 | [hC] | sorry. |
| 23:06.39 | JT | ah |
| 23:06.47 | [hC] | I was wondering why you thought I was out of my mind. |
| 23:06.52 | [hC] | :) |
| 23:06.56 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 23:07.06 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 23:07.11 | J4k3 | for $2000 I could buy a really badass server and slap in quad ethernet cards til I ran out of slots |
| 23:07.11 | [hC] | hp moved from CDP to LLDP and... nobody else uses LLDP :) |
| 23:07.30 | J4k3 | of course, thats not as cool. |
| 23:07.41 | JT | J4k3: quad ethernet cards that provide PoE? |
| 23:07.45 | [hC] | J4k3: you know how much quad ethernet cards are? and i'd like to see you get 24 port port density in a server. or PoE for that matter. |
| 23:07.50 | J4k3 | JT: they do after I get done with my soldering iron :P |
| 23:07.53 | [hC] | a quad port intel card is around $800 |
| 23:08.08 | JT | for 100base? |
| 23:08.10 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 23:08.13 | J4k3 | [hC]: quad 100 is way cheap on ebay |
| 23:08.16 | [hC] | 10/100/1000 |
| 23:08.17 | JT | maybe for gigabit |
| 23:08.22 | J4k3 | your phones need 1gbit? |
| 23:08.26 | [hC] | I couldnt find 10/100 recently |
| 23:08.26 | J4k3 | you talk a LOT. |
| 23:08.27 | JT | who on earth needs gigabit to their phone? |
| 23:08.32 | [hC] | i dont |
| 23:08.35 | J4k3 | who needs more than 10 to a phone? |
| 23:08.37 | [hC] | I just know thats the price i found last week |
| 23:08.44 | [hC] | you cant buy 10/100 quad cards anymore new |
| 23:09.01 | [hC] | at least my distributor doesnt sell them. only 10/100/1000 |
| 23:09.02 | J4k3 | bah, its not new, its pretested ;) |
| 23:09.07 | [hC] | lol |
| 23:09.12 | J4k3 | in PC parts terms |
| 23:09.16 | J4k3 | thats the truth |
| 23:09.20 | [hC] | burnt in :) |
| 23:09.27 | J4k3 | stuff works til it dies... and you keep spares for that ;) |
| 23:09.50 | JT | it also depends how easy it is to get to the site to replace a part |
| 23:09.50 | [TK]D-Fender | J4k3, NBC : (May repeats) "If you haven't seen it... it's new to you!" |
| 23:09.55 | variable_office | J4k3, working on asterisk now eh? |
| 23:10.13 | JT | you'd choose a more reliable part for a difficult or expensive to access site |
| 23:10.17 | variable_office | got a cool project going? |
| 23:10.22 | J4k3 | variable_office: been using it in the office since december. its nice, and a great test of network performance ;) |
| 23:10.36 | [hC] | well, im going to have to look into how to auto provision polycoms without CDP support at the switch, it will be a pain to have to have to get DHCP for a phone on vlan1, provision, move to vlan20, re-ip, reprovision, |
| 23:10.36 | [hC] | ugh |
| 23:10.38 | J4k3 | but no, so far no cool projects... just getting the wireless in the air has been enough hassle. |
| 23:10.38 | variable_office | ya, it works great as a pbx |
| 23:10.38 | [hC] | ugly. |
| 23:11.05 | J4k3 | considering the price |
| 23:11.08 | JT | sharing eithernet segments with computers and phones is ugly |
| 23:11.08 | J4k3 | of a SD card |
| 23:11.09 | J4k3 | is like $5 |
| 23:11.19 | Mad|Cow | Does anyone out there have any experience with the Cisco 7936 Conf phone? |
| 23:11.20 | J4k3 | why don't these phones use SD cards like GSM cells use SIM cards? |
| 23:11.41 | J4k3 | a 32MB SD card could be had for less than $5/ea in quantity... "network provisioning" starts sounding a lot less cool with this. |
| 23:11.56 | variable_office | J4k3, ya, same here, i am really trying to expand |
| 23:12.20 | Sweeper | network provisioning is awsome |
| 23:12.34 | Sweeper | you guys just don't know how to easily write cool interfaces for it :D |
| 23:12.56 | variable_office | the actual building is cool, the legal stuff of where can i put what is annoying |
| 23:13.20 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:13.24 | J4k3 | variable_office: thats why I stick to rural areas only. |
| 23:13.42 | J4k3 | cities are just too damned noisy anyways. |
| 23:14.17 | variable_office | J4k3, so do I, its just as bad here |
| 23:14.23 | [hC] | JT: sometimes you cant help it. |
| 23:14.37 | J4k3 | variable_office: where are you at these days? |
| 23:14.47 | [hC] | JT: I'm talking to a movie production studio which is hundreds of thousands of square feet large, and has a single drop to places they need phones. |
| 23:15.14 | variable_office | rural IL, same as always |
| 23:15.20 | [hC] | JT: coming in as an underdog competing with other people who can use their existing phone wiring, i have to make it work |
| 23:15.30 | J4k3 | variable_office: you mean the place where you've got a massive WISP attempting to muscle your state government? |
| 23:16.15 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 23:16.31 | Nate9939 | if i have created a sip trunk called test do i use dial (test, xxxxx) xxxx being the number to dial out on it? |
| 23:16.36 | JT | [hC]: you can always use existing phone wiring :) |
| 23:16.40 | variable_office | J4k3, you mean am i big enough to do that? hell no |
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| 23:16.50 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 23:16.52 | [hC] | JT: oh yeah, thats a great idea. :) |
| 23:17.10 | [hC] | 500+ft cat3 runs work FIIIIIIIINE for IP :P |
| 23:17.18 | J4k3 | variable_office: I heard through the grapevine another group (a merger-based ISP) was doing that. |
| 23:17.19 | JT | [hC]: either analogue, or if it's not generally more than 60metres, ethernet |
| 23:17.22 | J4k3 | up there. |
| 23:17.30 | JT | 500ft is quite excessive |
| 23:17.36 | variable_office | J4k3, whats the name? |
| 23:17.49 | J4k3 | 500ft cat3? 2.3mbit SDSL. |
| 23:18.02 | JT | J4k3: expensive as hell |
| 23:18.23 | JT | running any dsl to each workstation or whatever, very uneconomical |
| 23:18.27 | J4k3 | JT: bah, SDSL modems can talk to each other.. just have to have matching chipsets at each end |
| 23:18.30 | J4k3 | yeah |
| 23:18.31 | [TK]D-Fender | [hC], IP != Ethernet. :) |
| 23:18.38 | J4k3 | it'd add up depending on how much you did it. |
| 23:18.51 | JT | i assume he has a lot of handsets |
| 23:18.52 | [hC] | [TK]D-Fender: touche salesman. |
| 23:18.53 | [hC] | :) |
| 23:18.53 | J4k3 | 802.11a offers great performance if you've got perfect LOS |
| 23:19.16 | J4k3 | variable_office: they're whoever prarie-inet merged with? |
| 23:19.20 | yannj_fr | bye all, good night/evening |
| 23:19.32 | variable_office | prarie inet merged with someone? |
| 23:19.38 | J4k3 | I believe so |
| 23:19.39 | variable_office | prarie inet fell apart i thought |
| 23:20.27 | J4k3 | I can't remember the name of the main company... one of their employees was joining #wireless and talking mad quantities of shit |
| 23:20.39 | variable_office | still in there/ |
| 23:20.41 | variable_office | ? |
| 23:20.45 | J4k3 | not that I notice |
| 23:20.48 | J4k3 | I forget the nick |
| 23:20.56 | J4k3 | if so they haven't said anything in a long time. |
| 23:21.22 | variable_office | damn, i wouldve liked to have heard that |
| 23:21.33 | variable_office | pi got ripped to shreds here a few years back |
| 23:23.35 | *** join/#asterisk anthm (n=anthm@m010f36d0.tmodns.net) |
| 23:23.35 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 23:33.40 | michael-i | How does one get ahold of an extension's technology? I'm using a macro for all internal extensions and have both IAX2 and SIP clients. |
| 23:34.59 | [TK]D-Fender | extensions have nothing to do with SIP or IAX2 devices |
| 23:35.16 | [TK]D-Fender | and extension is merely something you can call in your dialplan |
| 23:35.28 | [TK]D-Fender | What it does is irrelevent. |
| 23:37.03 | rob0 | Wow, the zaptel driver has come a long way since I first messed with it. Set up my zaptel.conf, modprobe the driver, and it works! No udev tweaking needed. |
| 23:37.11 | rob0 | <== happy |
| 23:37.26 | michael-i | I completely missed chan_local. I'm probably using the wrong terminology, sorry. I just wanted one macro to dial all internal phones and chan_local does just that I think. |
| 23:40.53 | [TK]D-Fender | michael-i, No more that anything else does. "show application dial" |
| 23:41.24 | [TK]D-Fender | michael-i, Dial(SIP/1@SIP/2@IAX2/3@IAX2/4) |
| 23:41.47 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 23:43.55 | Nate9939 | Dial(${OUTBOUNDTRUNK}/702${EXTEN}) would this statement append the digits 702 before the dialed digits for the area code? |
| 23:44.32 | [TK]D-Fender | Nate9939, it will parse out that line exactly as it looks. |
| 23:45.03 | [TK]D-Fender | Nate9939, putting "702" between "/" and the current Exten |
| 23:45.03 | Nate9939 | what is the syntax to add the digits 702 to the dialed digits? |
| 23:46.12 | Nate9939 | i think that is what i want. |
| 23:46.13 | [TK]D-Fender | Nate9939, You have to rethink your concept of "adding". You are just calling an app. in it you reference a variable, something that I presume is a constant (${OUTBOUNDTRUNK}), and 1 fixed char. |
| 23:46.33 | Nate9939 | the user dials a local number and this will append 702 to the beginning of those digits and send it out. |
| 23:46.59 | [TK]D-Fender | Nate9939, It parses out their value and passes it to the app. Same goes for ANY app. the concept that you are "adding digits" is not a good frame of mind |
| 23:47.27 | [TK]D-Fender | Nate9939, You WILL get the 702 in front of the EXTEN you referenced there |
| 23:48.01 | [TK]D-Fender | Nate9939, so from a functional POV yes, you could say "you are adding digits", but only in the scop of that Dial command. |
| 23:48.58 | CoolGuy21 | hey guys |
| 23:49.29 | ManxPower | variables in the dialplan are evaluated and replaced with their value BEFORE the exten => line is run. |
| 23:50.31 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 23:50.59 | [TK]D-Fender | ManxPower, Given how dumb pbx_config is I'm surprised you can't use vars to substitute EVERYTHING from after the priority incluing the app. |
| 23:51.09 | [TK]D-Fender | ManxPower, or WORSE :) |
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