| 00:00.19 | jjshoe | I remember gasoline selling for under a dollar in this era |
| 00:00.29 | jjshoe | I remember it dipping down to 98 cents a gallon |
| 00:00.56 | jjshoe | during the summer of the early 2000's |
| 00:01.08 | alrs | jjshoe: not in California, I assume |
| 00:01.27 | EmleyMoor | We're paying over 4.50 a gallon now! |
| 00:01.33 | alrs | I'm hoping for $10/gallon |
| 00:01.43 | alrs | I've been paying $4.35 for diesel |
| 00:01.46 | EmleyMoor | (GBP) |
| 00:02.00 | jjshoe | cost me $50 to fill my mini cooper the last time around |
| 00:02.07 | EmleyMoor | Diesel is nearly 5 a gallon |
| 00:02.27 | Nivex | is so glad he rides the bus |
| 00:02.34 | jjshoe | I'm so glad I don't ride the bus |
| 00:02.36 | alrs | I prefer my bicycle over my van |
| 00:02.38 | alrs | but it gets 27mpg |
| 00:02.42 | gitguy | does asterisk have notification sounds for conferences? i use app_conference and i want to have sounds when people join/quits conferences, etc |
| 00:02.54 | jjshoe | gitguy of course |
| 00:03.44 | gitguy | where can i see that? |
| 00:03.53 | jjshoe | gitguy http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference |
| 00:04.06 | gitguy | cool, thanks |
| 00:07.10 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:09.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:11.04 | *** join/#asterisk Katty (n=Angela@adsl-70-239-217-207.dsl.stlsmo.sbcglobal.net) |
| 00:11.07 | Katty | hai |
| 00:13.43 | *** join/#asterisk jfg (n=jfg@dyn-88-121-11-212.ppp.tiscali.fr) |
| 00:13.47 | jfg | hi |
| 00:14.03 | jfg | does anyone tried pjsip/pjmedia ? |
| 00:14.10 | Katty | hi |
| 00:14.12 | Katty | how're you |
| 00:14.31 | _ShrikE | Katty! |
| 00:14.42 | Katty | herro. |
| 00:14.46 | Katty | hugs _ShrikE |
| 00:17.11 | _ShrikE | after this week... thats exactly what I needed ;) |
| 00:27.31 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:28.33 | RypPn | What exactly does gtkconsole do? I'd assumed it would pop up a little monitoring terminal on-screen when loaded. |
| 00:28.53 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 00:33.21 | Qwell | RypPn: it's a gtk console...go figure :p |
| 00:34.04 | RypPn | Qwell: Can you shed any light on this please? http://rafb.net/p/ImhjQS49.html |
| 00:34.33 | RypPn | If gtk wasn't available it wouldn't have compiled, so I'm kinda stumped |
| 00:34.42 | Qwell | setup your display properly |
| 00:35.10 | Qwell | the terminal running asterisk would need access to your X (gtk) session |
| 00:35.11 | RypPn | Is there a source of info for this I'm overlooking? |
| 00:38.54 | Qwell | not really. I expect that the number of users of that is very limited |
| 00:39.46 | Qwell | it's basically...open a shell. type Asterisk |
| 00:39.52 | Qwell | it'll load the console |
| 00:40.39 | RypPn | ok, I think I get it, I should be running x with the same user that asterisk is running with, or at least grant that user access to the X session |
| 00:40.59 | Qwell | however it's run, the shell needs to know how to access an existing X session |
| 00:43.32 | RypPn | ok, thats given me plenty to think about, thanks for the insight :) |
| 00:43.47 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-222-150.myvzw.com) |
| 00:44.37 | Qwell | it's one of those things that are like "oh, that's cool. ...I'll never use it though" |
| 00:46.30 | RypPn | ok, I'll admit it... it's like everest, "why? cos it's there." |
| 00:46.47 | Qwell | I went through the same thing :p |
| 00:47.03 | RypPn | I've passed it by for the last year and curiosity finally got the better of me |
| 00:47.18 | Qwell | there was kdeconsole too at one point |
| 00:47.41 | JayTee52 | I've been digging through "the book" and I can't find anyplace where the w parameter in the Dial application is documented as a wait. Matthew said it was chan_zap specific. Was this deprecated in 1.4 or is it documented somewhere else? |
| 00:47.50 | RypPn | yeah, I'd noticed it got dropped, so I thought I'd better have a look at the gtk one, just in csae it proved useful |
| 00:47.57 | RypPn | case* |
| 00:48.02 | Qwell | JayTee52: it's just chan_zap |
| 00:48.42 | Qwell | woah, I just got SMS spammed |
| 00:49.00 | jbeez | do you want me to stop sending those to you |
| 00:50.49 | JayTee52 | Qwell, you mean it's only documented in the source code? |
| 00:52.19 | JayTee52 | this was the line that was giving me problems, [TK]D-Fender showed me another way to do what I want. This was from a 1.2 install that someone else setup for FXO to another PBX. |
| 00:52.22 | JayTee52 | exten => _XXXX,1,Dial(Zap/g1/www,15,mD(${EXTEN})) |
| 00:53.18 | JayTee52 | and matthew said the www is a wait statement and the mD sends the digits in EXTEN but I can't find any documentation on those parameters. |
| 00:54.17 | JayTee52 | and [TK]D-Fender showed me the correct syntax for dialing an extension over PRI but I'm still curious where my predecessor got those parameters from. |
| 00:56.36 | *** join/#asterisk charkins (n=casey@pdpc/supporter/active/charkins) |
| 00:58.19 | ManxPower | JayTee52: on analog ports "w" in the dial string cause asterisk to stop dialing for .5 seconds. |
| 00:59.16 | ManxPower | The system must have been on an analog connection, as that dial says Go off hook, wait 1.5 seconds, after answer (which is immediatly after dialing on analog fxo, dial whatever is in ${EXTEN} |
| 00:59.23 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 00:59.55 | ManxPower | A totally moronic way of doing it unless we has doing something like trying to interface with a PBX, and even then I doubt that would be needed |
| 01:05.36 | SomethingISODD | hello all question i am creating a php agi script i was wondering how i can set it so as soon as the call connect i create a variable for the date/time? or is there anyway to do that |
| 01:05.45 | SomethingISODD | just so i can get start time and finished time.. |
| 01:07.49 | charkins | should it be possible to use the one step parking ("parkcall" from features.conf) from a call that was picked up from a parking stall? |
| 01:08.16 | Qwell | charkins: I think there is/was a bug on that, where it wouldn't reenable it |
| 01:08.34 | JayTee52 | ManxPower, it was to interface with a PBX using FXO ports but like I said, I can't find it in the book anywhere and was wondering where it was documented. |
| 01:08.36 | charkins | thanks, i'll dig through the bug tracker (hopefully there's a patch) |
| 01:10.24 | ManxPower | SomethingISODD: like the info in /var/log/asterisk/cdr-csv ? |
| 01:11.00 | SomethingISODD | ya basically |
| 01:12.36 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 01:13.30 | x86 | ok so what's the trick on the idle image on a polycom phone? |
| 01:14.23 | x86 | I've tried png, which the polycom told me "server sent unknown content type image/png", and then I tried jpeg and got the same results... now I've tried BMP and it doesn't give me that error, but it's still not showing the image |
| 01:14.28 | x86 | any ideas? |
| 01:21.15 | Qwell | x86: bmp I think |
| 01:21.25 | Qwell | there are certain dimension/bpp requirements though |
| 01:21.34 | Qwell | (afaik) |
| 01:21.48 | Qwell | it should be in the admin guide |
| 01:25.43 | x86 | ah |
| 01:25.45 | x86 | ok |
| 01:25.47 | x86 | thx |
| 01:41.43 | ManxPower | x86: read the admin guide. front to back. Then do it again. |
| 01:45.56 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:46.33 | *** join/#asterisk runoff (n=ham@c-24-147-160-185.hsd1.ma.comcast.net) |
| 01:48.30 | runoff | no support for SPA841, how do I hack from 2 lines to 4 lines Thanks for any help |
| 01:53.01 | gitguy | what do you recommend? should I try to make asterisk work behind a NAT or I get a server with public IP address? |
| 01:55.49 | Qwell | ~sipnat |
| 01:55.50 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:55.54 | Qwell | read that and you'll be fine |
| 01:57.30 | gitguy | ok |
| 02:07.31 | *** join/#asterisk euphor][a (n=martin@mail.dataproservices.co.uk) |
| 02:07.33 | euphor][a | hello |
| 02:07.58 | euphor][a | my asterisk server is broken, it hangs loading zttranscode module, any help greatly welcomed |
| 02:09.14 | C4away | is it a virtual machine? |
| 02:09.18 | C4away | euphor][a? |
| 02:09.22 | euphor][a | yes |
| 02:09.24 | euphor][a | no |
| 02:09.25 | C4away | xen |
| 02:09.27 | C4away | no? |
| 02:09.39 | euphor][a | its a physical machine, how do you mean virtual? |
| 02:09.46 | euphor][a | it has a digium card in it |
| 02:09.46 | C4away | like a xen virtual machine |
| 02:09.55 | euphor][a | no, just a mandriva box |
| 02:09.58 | C4away | if you don't know what xen is then it probably isn't |
| 02:09.59 | C4away | lol |
| 02:10.05 | euphor][a | i know xen |
| 02:10.22 | C4away | ah, well I can help you compile zaptel for xen, but that doesn't seem to be the problem |
| 02:10.32 | C4away | where did you get your zaptel source? digium's site or a repo? |
| 02:10.59 | euphor][a | i think its built from digium site |
| 02:11.11 | euphor][a | the box has been in production, it stopped working an hour ago |
| 02:11.12 | C4away | is this a distro like Elastix or Trixbox? |
| 02:11.16 | C4away | oh |
| 02:11.16 | euphor][a | I am thinkg hardware fault? |
| 02:11.20 | euphor][a | mandriva |
| 02:11.25 | C4away | yea, sorry |
| 02:11.28 | C4away | a bit distracted |
| 02:11.43 | C4away | could be a hardware fault |
| 02:11.44 | euphor][a | its a mission-critical box :( |
| 02:11.47 | C4away | ack |
| 02:11.56 | euphor][a | can I use a 2 port digium card in place of a 4 port? |
| 02:12.02 | C4away | yes |
| 02:12.08 | C4away | if you only need 2 pors |
| 02:12.10 | C4away | ports |
| 02:12.18 | euphor][a | there's 2 plugged in, not sure how many are used |
| 02:12.21 | euphor][a | *3 |
| 02:12.32 | C4away | and if zapata.conf is using the correct ports |
| 02:12.47 | euphor][a | would I need change configs if I swapped cards? |
| 02:13.03 | C4away | yes, unless you are pulling the last card and not using those ports |
| 02:13.15 | euphor][a | or would it just work, minus the line that isn't plugged in |
| 02:13.22 | C4away | if you have ports 1-4 configured and you pull ports 5-8 then no problem |
| 02:13.42 | euphor][a | I believe 1-3 are configured, and I'd like to put a 2 port in, and just plug in 1 and 2 |
| 02:13.45 | C4away | if you are using ports 5-8 and pull ports 1-4 then ports 5-8 become 1-4 |
| 02:14.00 | euphor][a | it has just 1 x 4-port card |
| 02:14.03 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 02:14.08 | euphor][a | I'd like to replace with 1 x 2-port |
| 02:14.17 | C4away | you will have to change the group settings in zapata.conf |
| 02:14.23 | euphor][a | bugger |
| 02:14.28 | C4away | well |
| 02:14.41 | C4away | is it kernel panicing? |
| 02:14.44 | C4away | panicking |
| 02:14.49 | C4away | however you spell that |
| 02:15.03 | C4away | picknicking |
| 02:15.13 | C4away | pancakeking |
| 02:15.14 | euphor][a | not to tty, but it hangs loading zttranscode, no response |
| 02:15.35 | euphor][a | I guess a kernel oops somewhre |
| 02:15.36 | C4away | can you enter interactive startup and not load zttranscode? |
| 02:15.40 | euphor][a | yes |
| 02:15.45 | C4away | it loads? |
| 02:15.47 | euphor][a | well, I can do single |
| 02:15.53 | euphor][a | I haven't tried interactive |
| 02:16.16 | euphor][a | but I can rename init.d for zaptel and asterisk and it boots normally |
| 02:16.26 | euphor][a | so something there causing it to hang |
| 02:16.41 | C4away | and all of a sudden |
| 02:16.43 | C4away | out of the blue |
| 02:16.45 | C4away | no warning |
| 02:16.46 | C4away | etc |
| 02:16.55 | C4away | have you changed anything, updated anything/ |
| 02:16.59 | euphor][a | yes, whilst booting, it gets to zttranscode and stops |
| 02:17.02 | C4away | not updated something you should have? |
| 02:17.05 | euphor][a | no, nothing changed |
| 02:17.24 | euphor][a | it just stopped working -- I am thinking hardware |
| 02:17.33 | C4away | pull the card and boot it |
| 02:17.45 | euphor][a | ok |
| 02:17.47 | euphor][a | brb |
| 02:17.51 | C4away | if it boots fine without the card then you might be ok |
| 02:17.58 | C4away | if it still hangs you probably want to reload zaptel |
| 02:18.11 | C4away | backup your zap* files in /etc/asterisk ... specifically zapata.conf |
| 02:18.25 | C4away | print it out, email it to your gmail account, etc |
| 02:24.13 | euphor][a | it boots |
| 02:24.34 | euphor][a | I have backups of configs, all stored in cvs |
| 02:26.05 | euphor][a | okay, I'm guessing hardware.. I wish I had a spare card |
| 02:27.47 | euphor][a | so, any suggestions? :) |
| 02:28.56 | Qwell | call Digium |
| 02:29.56 | euphor][a | hrm |
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| 02:52.41 | outtolunc | ~itsp |
| 02:52.42 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 02:53.12 | outtolunc | ~itsplist-us |
| 02:53.13 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 03:06.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 03:14.00 | jeffspeff | hi, i just setup an asterisk now system at home... just playing around. i'm using voip, and was wanting to know of a recommended windows client. |
| 03:22.34 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 03:22.57 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:23.08 | jeffspeff | does anybody have a recommended soft-phone? |
| 03:26.15 | jameswf-home | Wierd I can get sip to work with ipkall but not iax... seems bass ackwards |
| 03:28.02 | jameswf-home | I like moxiax |
| 03:34.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 03:37.39 | gitguy | how come that SIP and even IAX2 has problems with NAT on Amazon EC2? |
| 03:37.53 | gitguy | when I try to connect with my softphone I don't even see activity in the CLI |
| 03:43.46 | gitguy | I told my boss to switch from server provider but he wont listen. |
| 03:43.55 | gitguy | what a ****er |
| 03:50.03 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.229) |
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| 04:35.32 | *** join/#asterisk shital (n=shital@122.167.93.2) |
| 04:35.41 | shital | Hello All |
| 04:36.11 | killmel8tr | hi |
| 04:36.42 | shital | i have TDM11B card with my Intel p3system |
| 04:37.03 | shital | how can check its correct configuration |
| 04:37.34 | shital | is it necessary to plug in 12 V power supply to card? |
| 04:38.13 | JT | what modules are installed? |
| 04:39.38 | shital | both FXO and FXS |
| 04:39.52 | shital | one each |
| 04:39.58 | killmel8tr | I'm sure you have to plug in the power regardless |
| 04:40.04 | JT | killmel8tr: no |
| 04:40.10 | JT | only if you have FXS modules |
| 04:40.20 | killmel8tr | oh, interesting |
| 04:40.22 | JT | then the molex connector must be plugged in |
| 04:40.45 | killmel8tr | ahhh.. ya i guess that makes sense |
| 04:40.52 | *** join/#asterisk talntid (n=t@71-213-240-195.spkn.qwest.net) |
| 04:40.58 | killmel8tr | since you are generating the power out in that case |
| 04:41.17 | JT | right, and it probably can't pull enough from the pci bus |
| 04:54.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:02.56 | shital | i wanted just to check whether the card is configured correctly, even then also it needs power supply? |
| 05:03.23 | shital | without any phone connections |
| 05:03.34 | JT | why don't you just plug it in? |
| 05:04.19 | shital | bcz the problem is in my system that extra connector is not there, so only |
| 05:06.23 | shital | now the output of ztcfg command is ZT_CHANCCONFIG failed on channel 1: Invali argument (22) can u tell me what exactly this is? |
| 05:07.30 | shital | JT: r u there? |
| 05:09.18 | JT | it means there is an error in the configuration file |
| 05:10.43 | shital | ok thank you |
| 05:14.55 | lmadsen | howdy's the room |
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| 05:25.39 | x86 | heya lmadsen |
| 05:28.15 | *** part/#asterisk GoldenFish (n=is@AC9E4665.ipt.aol.com) |
| 05:54.34 | x86 | wtf cant say hi back nub? |
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| 06:15.58 | jeffspeff | hi, is anybody familiar with asterisknow ? |
| 06:15.58 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 06:17.15 | jeffspeff | *does anybody have experience with asterisk now? |
| 06:19.33 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:27.03 | drmessano | jeffspeff: wrong channel |
| 06:28.02 | jeffspeff | what channel is that in? asterisknow? |
| 06:28.14 | drmessano | yes |
| 06:28.15 | jeffspeff | ahh, it is. :) |
| 06:28.18 | jeffspeff | thanks |
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| 06:35.50 | jeffspeff | drmessano: maybe you can answer part of my question |
| 06:35.57 | jeffspeff | drmessano: i'm wanting to do it all through voip. Do I have a VoIP service provider just to test the features inside my network, like from extension to extension? and also, at the moment i'm just testing using x-lite, but when i get the IP phones, do i have to have any of the special cards? thanks. |
| 06:36.39 | jeffspeff | *Do I have to have a VoIP service.... (sorry for the typo). |
| 06:37.35 | drmessano | Depends, depends, depends |
| 06:37.50 | drmessano | No, you do not need a VoIP provider to call extension to extension |
| 06:38.05 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:38.11 | drmessano | You only need a provider if you are wanting to connect to the PSTN via IP |
| 06:38.27 | drmessano | If you want to connect via a copper line or T1, you need a card |
| 06:38.54 | drmessano | If you want to connect analog phones to your system you may need a card as well |
| 06:39.34 | jeffspeff | but using cat5 cables, i won't need i card, i can just plug it into the network, correct? |
| 06:39.43 | drmessano | Or Analog Telephone Adapters, which are boxes that take a network connection and connection from a standard phone, and make it an IP connection |
| 06:39.48 | drmessano | yes |
| 06:39.59 | drmessano | Vo *IP* |
| 06:40.36 | jeffspeff | I'm new to the telephone stuff. just making sure i cover my bases and trying not to assume anything. |
| 06:40.46 | drmessano | ok, np |
| 06:41.44 | jeffspeff | i'm not getting any repsonses in the asterisknow channel, would you mind helping me with some other questions? |
| 06:42.18 | jeffspeff | i understand if you can't due to asterisknow being different in some parts or what not. |
| 06:42.27 | drmessano | that's correct |
| 06:44.00 | jeffspeff | when adding a user, there's a field that says "Insecure", and the tool tip description of that is "Insecure: Matching of IP for a peer without matching port, do not require authentication of invites.". Is that wanting to know the IP of the device the corresponding user will use to connect? |
| 06:45.10 | drmessano | jeffspeff: This is not the place for Asterisknow questions |
| 06:45.53 | jeffspeff | ok, thought i'd see if you might know anything at all about it. |
| 06:46.01 | *** join/#asterisk DaveCanoe (n=Dave@63.250.111.37) |
| 06:46.39 | drmessano | That's irrelevant |
| 06:46.44 | drmessano | This is the WRONG CHANNEL |
| 06:47.39 | jeffspeff | ok, sorry |
| 06:48.20 | jameswf-home | ~asterisknow |
| 06:48.21 | jbot | asterisknow is, like, based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 06:49.55 | jeffspeff | oh, ok... i thought they were developed closer than that. |
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| 07:38.08 | eddy122 | need help installing dundi, anyone alive here please? |
| 07:39.43 | eddy122 | need help installing dundi, anyone alive here please? |
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| 08:31.31 | eddy122 | need help installing dundi, anyone alive here please? |
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| 08:38.15 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 08:38.43 | EmleyMoor | Are there any tools available for analysing a dialplan so that I can spot errors, redundancy etc. before I make it "live"? |
| 08:44.28 | eddy122 | EmleyMoor all people dead here |
| 08:51.43 | EmleyMoor | has just rewritten his dialplan and it's a bit complicated |
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| 09:10.11 | eddy122 | how to install dundi? in which module it is available |
| 09:11.47 | EmleyMoor | I don't have any dealings with dundi |
| 09:12.58 | unpaidbill | dundi rocks the crocodiles |
| 09:17.25 | servettas | <PROTECTED> |
| 09:17.29 | servettas | can anyone help me |
| 09:17.32 | servettas | ? |
| 09:17.39 | eddy122 | i need to link 4 servers together and i dont know how to guys |
| 09:19.26 | servettas | ihave a sound problem and looking frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end error msg can anyone help me pls thanks |
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| 09:32.33 | servettas | ihave a sound problem and looking frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end error msg can anyone help me pls thanks |
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| 09:50.54 | Qapf | anyone else here using VP? i think their service just died |
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| 11:00.50 | EmleyMoor | Are there any tools for analysing a dialplan - to look for errors, redundancy and also to check what it would actually do? (assuming it's on a late 1.2) |
| 11:01.45 | ac1djazz | lol yea thats funny |
| 11:01.53 | ac1djazz | asterisk only debugs at a line by line basis |
| 11:02.31 | EmleyMoor | ac1djazz: I have done a major rewrite of three sections of my dialplan and want a way to "dry run" it |
| 11:02.40 | ac1djazz | yea |
| 11:02.45 | ac1djazz | i was thinking about this earlier |
| 11:02.54 | ac1djazz | none of the dialplan is realistically runtime |
| 11:05.26 | EmleyMoor | Some calls I am unlikely to make, I have nevertheless tried to allow for |
| 11:09.02 | EmleyMoor | I am preparing for a switch of provider priority in the near future |
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| 13:54.09 | Marquel | morning |
| 13:54.21 | gr0mit | afternoon |
| 13:54.42 | Marquel | whatever |
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| 13:55.16 | EmleyMoor | Next, a policeman will come in and say "Evenin' all" |
| 13:55.41 | gr0mit | ello ello, ello, wot's goin on 'ere? |
| 13:55.52 | Marquel | EmleyMoor: that's the problem w/ IRC - you can _never_ have all the cool movie scenes reproduced ;) |
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| 14:09.00 | *** join/#asterisk rerzerty (n=chatzill@dyn-91-165-213-44.ppp.tiscali.fr) |
| 14:09.24 | rerzerty | hi |
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| 14:10.27 | rerzerty | anyone use "dsl router + ata" in one device ? |
| 14:11.19 | killmel8tr | we used to sell the linksys ones |
| 14:11.35 | killmel8tr | I think vonage used them (maybe still do) for a while as well |
| 14:13.07 | killmel8tr | I guess a benefit would be easy QoS setup |
| 14:15.12 | rerzerty | my question is : is it exist one device which is capable to do "dsl + ata" in one device ? |
| 14:15.29 | rerzerty | if yes what is model ? |
| 14:16.19 | killmel8tr | you want the modem itself in there as well or just a router? |
| 14:17.23 | Marquel | rerzerty: f.ex. Linksys SPA2102 or Allnet ALL7902 - both are ATA with integrated dsl routers. |
| 14:18.20 | rerzerty | thx marque1 |
| 14:18.55 | killmel8tr | just go to voipsupply.com |
| 14:19.45 | killmel8tr | of course you could just get comcast digital voip.... errr voice |
| 14:19.52 | killmel8tr | :) |
| 14:21.06 | rerzerty | ok thx killmel8tr |
| 14:22.15 | rerzerty | i find this one http://www.voipsupply.com/images/gs486topo.gif |
| 14:23.34 | rerzerty | is it capable to do dsl router + ata ??? |
| 14:23.40 | killmel8tr | i had some bad experiences with grandstream... (guessing thats what the link is to by the name)... I met some Jerry guy (think that was his name) at VON one year, made a huge deal and he reneged. Anyhow, they have probably improved, but they are the yugos of the voip world... or at least they were. |
| 14:24.02 | killmel8tr | if I remember right that adapter supports a pots line which is actually pretty cool |
| 14:24.27 | killmel8tr | they sent me one to play with but i never opened it up. |
| 14:25.47 | rerzerty | ok |
| 14:26.03 | rerzerty | is it capable to do dsl router + ata ??? |
| 14:26.06 | rerzerty | or not ? |
| 14:26.15 | killmel8tr | ya |
| 14:26.39 | killmel8tr | not very robust though and you will need a switch if your hooking up more than one device behind it |
| 14:27.31 | killmel8tr | your better off getting a good router like a linksys wrt54gx that you can flash with dd-wrt or something and then getting you ip voip device to put behind it |
| 14:27.36 | killmel8tr | the spa2102 is good. |
| 14:27.48 | rerzerty | yeah |
| 14:27.53 | rerzerty | very expensive |
| 14:27.59 | killmel8tr | even though it can be a router, I would still use the linksys (just router) for wireless and stuff. |
| 14:28.02 | rerzerty | than grandstream |
| 14:28.17 | killmel8tr | where you at? (US?) |
| 14:28.25 | rerzerty | in europe |
| 14:28.55 | rerzerty | france excatly |
| 14:29.24 | killmel8tr | I have like 2000 spa2000's with euro power adapters so I was gonna just send you one, they will be trashed eventually |
| 14:29.25 | killmel8tr | err 200 |
| 14:29.28 | killmel8tr | but i am in the us |
| 14:30.14 | killmel8tr | you could have the gs486 too, I dont want it.... how much does it cost to mail a little box to france? |
| 14:30.17 | rerzerty | i got spa3000 is it possible to use dsl router +ata ? |
| 14:30.39 | killmel8tr | yes |
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| 14:30.48 | rerzerty | thx a lot |
| 14:30.56 | rerzerty | i just buy it for fun |
| 14:31.02 | rerzerty | very happy now |
| 14:31.06 | rerzerty | thx a lot |
| 14:31.50 | killmel8tr | cool, i like that it has the pots interface |
| 14:31.56 | killmel8tr | talk to ya l8tr |
| 14:32.36 | rerzerty | if i understand correctly the spa3000 is a router with ata adaptor am i right ? |
| 14:36.35 | rerzerty | ok |
| 14:37.28 | rerzerty | i just connect dsl line to my sipura 3000 |
| 14:37.47 | rerzerty | and i also connect my analogue phone |
| 14:37.55 | rerzerty | and rj45 to my pc |
| 14:38.08 | rerzerty | internet not working is it normal ? |
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| 14:38.37 | rerzerty | just beginner |
| 14:38.41 | rerzerty | plz |
| 14:41.08 | rerzerty | killme r u there N |
| 14:42.15 | rerzerty | helhello |
| 14:42.57 | *** join/#asterisk harryv (n=fufufu@3e6b0130.rev.stofanet.dk) |
| 14:46.52 | Marquel | i have a little technical question - what may be wrong if i need to add my local dial prefix for asterisk to work where i wouldn't need it normally? |
| 14:47.40 | harryv | hey. when i try to call '36946811' from the outside the call gets through, but is rejected and the sound file is not played, this is my conf and sip debug output: http://pastie.org/187226 |
| 14:50.07 | rerzerty | killme r u there ? |
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| 14:58.53 | Corydon76-dig | Marquel: what kind of trunk? |
| 14:59.46 | Marquel | Corydon76-dig: ZAP (ISDN, EDSS1, works with another provider without adding the local prefix) |
| 15:00.19 | Corydon76-dig | You probably have it set up for national number plan |
| 15:00.40 | Corydon76-dig | You might try setting the pridialplan to unknown |
| 15:00.45 | Corydon76-dig | or dynamic |
| 15:01.41 | Corydon76-dig | unknown is usually better, though. dynamic assumes NANPA conventions |
| 15:01.59 | Marquel | Corydon76-dig: pridialplan is unset (and it's a bri-card anyway) |
| 15:02.10 | Corydon76-dig | Oh, okay |
| 15:02.49 | Corydon76-dig | Well, it's a question you'll need to ask your provider |
| 15:04.29 | Marquel | i tried - either they're ignorant or didn't listen (i'll retry monday anyway). seems they understood "i have to dial local prefix on the phone" instead of "the box has to dial the local prefix on the outside, b/c not doing so doesn't work"... :( |
| 15:05.44 | Marquel | but i'll try setting pri*dialplan. that's anyway better than this ugly workaround in extensions.conf. |
| 15:09.43 | Marquel | little other question: my phones record a missed call if the call is answered by another phone - is there something i can do about it or do i need to learn to live with it? |
| 15:10.33 | *** join/#asterisk ManxPower (n=manxpowe@6.sub-70-223-36.myvzw.com) |
| 15:11.05 | Corydon76-dig | Learn to live with it |
| 15:11.29 | Marquel | humpf. |
| 15:11.57 | Corydon76-dig | You can generally turn off the missed calls, but that's probably not what you want |
| 15:12.04 | Corydon76-dig | It's an all-or-nothing deal |
| 15:12.29 | Corydon76-dig | The phone cannot tell the difference between missing a call and some other phone picking up the call |
| 15:12.29 | Marquel | since isdn normally is able to notice all ringing phones not to record a missed call if the call is answered. i thought there would be something similar i could send to a sip-phone.... |
| 15:12.42 | ManxPower | You can turn off missed calls on a per-line-appearance basis with the Polycoms |
| 15:13.10 | Corydon76-dig | Marquel: from the phone's perspective, it looks like the same thing |
| 15:13.22 | ManxPower | Marquel: That sort of thing is still pretty new in the VoIP world. |
| 15:13.50 | ManxPower | Corydon76-dig: Polycoms support server based missed call lists, but I'm sure it's not a standard defined protocol |
| 15:14.32 | Marquel | ManxPower: i guess my siemens gagaset phones then will be far from supporting that - the dect base is not even able to handle more than two calls over IP - where cheaper phones have no problems managing four concurrent calls.... |
| 15:14.38 | ManxPower | Marquel: I'm sure that in 3 years it will be a standard feature. |
| 15:16.52 | Marquel | very frustrating. |
| 15:17.50 | ManxPower | Marquel: Regardless of what the "experts" say, VoIP is still an immature technology and does NOT have all the features of many systems that have been around for 20 years. |
| 15:18.21 | ManxPower | Fortunately, VoIP (mainly SIP) is mature and useful enough for large numbers of people. |
| 15:18.52 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.76) |
| 15:19.54 | Marquel | ManxPower: at least there's one major advantage of voip: you don't need to redo all the cabling previously badly put together by cheap idiots... ;) |
| 15:20.35 | ManxPower | Marquel: Yes, but people also want phones where there was never a computer or network jack. |
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| 15:21.25 | Marquel | ManxPower: that's right, and i won't trade some of the advantages of a pots- or isdn-connection for voip. |
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| 15:33.11 | NeonLevel | Good morning, I have a sip provider in mexico if i try to register asterisk with this provider i get all the time Conflict, but if I use a sip client in my linux like ekiga the client will register without a glitch, can anyone give me an advise... |
| 15:35.40 | jameswf-home | sip debug,,, turn up verbosity.... pastebin error message |
| 15:39.50 | NeonLevel | http://pastebin.com/m54eb98f5 |
| 15:40.11 | NeonLevel | this is what i get http://pastebin.com/m54eb98f5 |
| 15:40.14 | NeonLevel | sorry |
| 15:40.23 | NeonLevel | this is what i get -- Got SIP response 409 "Conflict" back from 200.76.111.57 |
| 15:42.59 | harryv | hey. when i try to call '36946811' from the outside the call gets through, but is rejected and the sound file is not played, this is my conf and sip debug output: http://pastie.org/187226 |
| 15:44.13 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 15:45.33 | harryv | just updated the sip debug stuff |
| 15:45.35 | harryv | the other one was messy |
| 15:46.39 | NeonLevel | there was no 1234 exten in incoming context, could that be? |
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| 15:47.08 | harryv | .X_ should catch that one |
| 15:47.35 | harryv | eh, forgot to put that in the paste.. but it's there |
| 15:47.55 | harryv | _X. even |
| 15:52.59 | harryv | and when i call internally it's fine |
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| 15:58.35 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:59.22 | acxty | hi guys, I got a ip telephone line. The ISP gave me theh server, user and password for a proxy server. Where can I find information on it |
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| 16:01.10 | lmadsen | acxty: huh? |
| 16:01.14 | lmadsen | they gave you everythign you need... |
| 16:01.29 | lmadsen | now you have to read some documentation |
| 16:01.32 | lmadsen | ~thebook |
| 16:01.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:02.20 | Ron56 | hie, there is my extension.conf here: http://87.98.151.62/asterisk/extensions.conf.txt and i dont understand why when i call 100/101/102 numbers it doesn't work :s |
| 16:05.21 | acxty | lmadsen, thanks that is what I was looking for |
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| 16:16.14 | NeonLevel | Good morning, I have a sip provider in mexico if i try to register asterisk with this provider i get all the time Conflict, but if I use a sip client in my linux like ekiga the client will register without a glitch, can anyone give me an advise... http://pastebin.com/m54eb98f5 |
| 16:22.13 | lmadsen | NeonLevel: maybe their proxy is looking for signs that it is an asterisk box registering, and rejecting the registration |
| 16:22.22 | lmadsen | if you changed the user-agent to something else, it might work |
| 16:22.28 | lmadsen | throws out a random guess |
| 16:23.06 | NeonLevel | thank you for your response lmadsen, i tryed changing useragent = Firefly in my general inside sip.conf |
| 16:23.11 | NeonLevel | and didnt work |
| 16:23.22 | NeonLevel | is driving me crazy.... |
| 16:23.55 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 16:24.28 | *** join/#asterisk blq (n=Bl@dslb-088-064-155-025.pools.arcor-ip.net) |
| 16:25.00 | lmadsen | tzanger: you might be interested in this: http://www.opengpstracker.org/ |
| 16:25.23 | davidcsi | hello all, I'm testing chan_h323 on 1.4, is there no way to use it BEHIND NAT? it doesn't seem to be doing it rith, it ends with "cause EndedByTransportFail" |
| 16:26.58 | *** join/#asterisk frieze (n=frieze@cpe-69-203-15-230.nyc.res.rr.com) |
| 16:27.00 | lmadsen | davidcsi: you're the first person I've even heard using it :) |
| 16:27.15 | lmadsen | like... who came in here and didn't ask how to get it working, but actually has something operational |
| 16:27.24 | frieze | Is there any brand or model of sip phone that's easier to live with using asterisk? |
| 16:27.28 | lmadsen | so you might actually be the most qualified person to answer your own question :) |
| 16:27.33 | frieze | sip wireless phone I meant to say |
| 16:27.52 | lmadsen | frieze: the sip client in my Nokia E61i is really good -- best wifi sip phone I've ever used |
| 16:27.55 | lmadsen | the rest seem to suck |
| 16:29.41 | plik | lmadsen: do you know if the E61i is unique, or similarly good accross other nokias? |
| 16:29.50 | lmadsen | I own one, and love it |
| 16:30.01 | lmadsen | no idea about the sip client vs. other nokia phones |
| 16:30.47 | plik | ok, just wondering if the sip client on E65 or 6110i might be as good... may get to find out one day |
| 16:31.08 | *** join/#asterisk solar_ant (n=John@122.164.243.97) |
| 16:31.09 | andrew` | hi, trying to use AEL but it seems to be quite different than the old style..or maybe I'm just confused...why does pressing 8 during the Background sounds in s not work? http://www.pastebin.ca/998509 |
| 16:31.34 | harryv | so -- if you don't aim for a smartphone (i always end up throwing my phones away or damaging them or something, so i stick with cheap $50-ones) are there any somewhat-cheap wifi sip's ? going to equip the house with themm |
| 16:32.08 | davidcsi | lmadsed, i've had it working for some time now, it works perfectly.... i'm having the problem now because it is behind nat... :S |
| 16:32.49 | plik | harryv: prolly better off with a DECT sip phone like the Siemens gigaset *IP range |
| 16:34.03 | harryv | plik: you're probally right. better batterytime etc.. |
| 16:34.06 | harryv | and cheaper. |
| 16:34.22 | *** join/#asterisk JamminJud (n=JamminJu@208.84.220.251) |
| 16:34.59 | plik | if you want properly cheap and durable, stick a regualr corless on an ATA like a linksys PAP2 |
| 16:35.57 | JamminJud | calling from a zap1 interface to iax trunk... where do I set the callerid so that it shows up properly at called party? |
| 16:36.12 | NeonLevel | I have two sip registrar accounts with the same provider in mexico, i can register the asterisk with one of this accounts and works ok, but i cannot register the other one i've done some sip debug and the only difference i've seen is in the response they told me that the sip proxy ip is 200.76.111.57 and here is the response with the one that doesnt work From: <sip:523338391548@200.52.137.115> as you can see i get a different IP , couldnt that be a pr |
| 16:36.12 | NeonLevel | oblem? |
| 16:36.17 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 16:37.13 | JamminJud | NeonLevel: if you're going to the same ip, are you using different ports? |
| 16:37.28 | NeonLevel | no no different ports |
| 16:37.48 | NeonLevel | that response is being generated by my sip provider |
| 16:38.27 | JamminJud | I would think you would need a different port for the second connection |
| 16:38.36 | JamminJud | use 5061 instead of 5060 |
| 16:38.41 | NeonLevel | how? |
| 16:38.51 | JamminJud | hmmm, that's a good question |
| 16:39.00 | NeonLevel | :-) |
| 16:39.06 | outtolunc | port=xxxx |
| 16:39.08 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 16:39.14 | JamminJud | ahh, there you have it |
| 16:39.21 | MrTelephone | how many people have MWI working with realtime and openser? |
| 16:39.23 | NeonLevel | trying that |
| 16:39.53 | MrTelephone | sip show users doesn't show any users even though they are in the mysql database |
| 16:39.59 | MrTelephone | this is version 1.2.9.1 |
| 16:40.10 | MrTelephone | does 1.4 work better? |
| 16:41.52 | JamminJud | have 1.4 but not using mysql |
| 16:42.35 | andrew` | or is AEL just crap and nobody really uses it? :) |
| 16:43.00 | MrTelephone | what is AEL? |
| 16:43.04 | MrTelephone | jamminjud are you using obdc? |
| 16:43.05 | andrew` | lol |
| 16:43.49 | lmadsen | AEL is an alternate method of writing dialplan logic |
| 16:43.58 | MrTelephone | oh |
| 16:44.05 | lmadsen | AEL is actually parsed and converted to Asterisk internally |
| 16:44.24 | andrew` | it just doesn't seem to follow the documentation |
| 16:44.40 | JamminJud | I'm using plain text |
| 16:44.44 | andrew` | I was trying to be 'modern' but I guess I'll revert to the old style |
| 16:45.18 | JamminJud | no database.... mine is just for home |
| 16:45.20 | MrTelephone | i wouldn't use it until it's a year old |
| 16:45.49 | MrTelephone | its someones dumb idea to improve things when the old methods worked fine |
| 16:46.56 | JamminJud | calling from a zap1 interface to iax trunk... where do I set the callerid so that it shows up properly at called party? |
| 16:47.34 | MrTelephone | does it show up at all? |
| 16:48.13 | JamminJud | yes, shows up with providers default trunk. provider is teliax |
| 16:48.43 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:48.55 | MrTelephone | look for some kind of option to sendrpid |
| 16:49.02 | MrTelephone | that may be only for sip though |
| 16:50.38 | MrTelephone | do you have callerid set in zapata.conf? |
| 16:50.41 | JamminJud | I'm thinking it's a service provider problem |
| 16:51.32 | JamminJud | they have an option on their website to set it.... but it's not working |
| 16:52.06 | ManxPower | JamminJud: Put a Noop(CALLERID(all) is ${CALLERID(all)}) somewhere in your dialplan so you can see what Asterisk's idea of callerid is that at point in the dialplan |
| 16:52.33 | MrTelephone | its spanxpower1 |
| 16:52.34 | JamminJud | k |
| 16:52.37 | MrTelephone | ! |
| 16:52.41 | ManxPower | Also read "core show application dial" Pay special attention to the callerid related options (I think option "o") |
| 16:53.33 | ManxPower | JamminJud: Is this "zap trunk" an FXO, FXS, PRI, E&M/Wink? |
| 16:53.46 | MrTelephone | winks at manxpower |
| 16:53.52 | JamminJud | fxs... pots line |
| 16:54.05 | ManxPower | JamminJud: It can't be both FXS and a POTs line. |
| 16:54.15 | JamminJud | provides dialtone... I forget whether its fxs or fxo... must be fxo then |
| 16:54.17 | MrTelephone | he means pots style lines |
| 16:54.18 | ManxPower | ~fxofxs |
| 16:54.19 | jbot | fxofxs is probably An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 16:54.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:54.44 | ManxPower | JamminJud: if it is an FXS port then you want the line callerid=Robert Dobbs <5045551212> |
| 16:54.59 | ManxPower | Notice no quotes and no special chars like - and no 1 in the phone number |
| 16:55.12 | ManxPower | that would set the callerid for calls made from that port to that callerid |
| 16:55.15 | JamminJud | that's in the zapata.conf file? |
| 16:55.27 | ManxPower | yes, right before the channel => line, just like every other option |
| 16:57.13 | JamminJud | awesome, that's what I was looking for |
| 16:57.30 | ManxPower | zapata.conf.sample should have had examples of that |
| 16:58.25 | *** join/#asterisk vlsoft (n=vlsoft@ai4.inf.elte.hu) |
| 16:58.32 | vlsoft | Hi! |
| 16:58.50 | MrTelephone | ok |
| 16:59.37 | MrTelephone | 2006-12-01 12:40pm <MrTelephone> do you have callerid set in zapata.conf? |
| 16:59.57 | JamminJud | lol, 2006? |
| 17:00.00 | MrTelephone | haha |
| 17:00.04 | MrTelephone | just kiddin around |
| 17:00.09 | MrTelephone | ;P |
| 17:00.48 | JamminJud | I appreciate the help on that one |
| 17:01.18 | *** join/#asterisk UQlev (n=kvirc@ykulyev.logos.cy.net) |
| 17:01.23 | JamminJud | zapata.conf format is a bit different than sip.conf |
| 17:01.33 | JamminJud | but ManxPower is right about the examples |
| 17:02.01 | MrTelephone | might still not work, thats the sad part |
| 17:02.11 | vlsoft | Is some of the devs of iaxclient present here? I would like to ask for some help compiling iaxclient (and utlimately iaxcomm) under a cygwin environment (I have already compiled wxwidgets, but having some problems running ./configure on the iaxclient trunk) |
| 17:02.45 | JamminJud | MrTelephone: it worked |
| 17:03.56 | MrTelephone | does your iax provider go out to pstn? |
| 17:04.00 | JamminJud | now I just need to reroute my incoming calls to my Zap1 line and all will be good |
| 17:04.04 | JamminJud | yes |
| 17:04.31 | andrew` | MrTelephone, it goes back like 3 years... |
| 17:04.35 | MrTelephone | what is your iax peer context? |
| 17:06.15 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:06.33 | JamminJud | I've but one context and everything is in it for simplicity sake |
| 17:07.20 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 17:08.51 | *** join/#asterisk rerzerty (n=chatzill@dyn-91-165-213-44.ppp.tiscali.fr) |
| 17:09.22 | MrTelephone | what are all these new errors in asterisk 1.4? undefined symbol: option_priority_jumping? |
| 17:09.25 | rerzerty | hi |
| 17:09.33 | rerzerty | i got sipura 3000 |
| 17:09.35 | jameswf-home | heh i have one context just for my ex wife :) |
| 17:09.47 | MrTelephone | JamminJud. add a line in there such as exten => s,1,Dial(ZAP/1) |
| 17:10.09 | MrTelephone | i got a context for your xwife too |
| 17:10.16 | MrTelephone | [golddigger] |
| 17:10.17 | MrTelephone | j/k |
| 17:10.21 | rerzerty | my isp router not working is it possible to replace my isp router with my sipura 3000 to access internet ? |
| 17:10.25 | jameswf-home | sadly not farr off |
| 17:10.52 | jameswf-home | I couldnt name it what I wanted to incase the logs had to go to court |
| 17:11.02 | vlsoft | Okay, from the lack of interest, I assume noone ever compiled iaxclient here under a cygwin environment; so I have to solve this problem myself (but not much luck so far... :( ). |
| 17:11.14 | outtolunc | use [1hotmamma] that should get you some |
| 17:11.34 | jameswf-home | my current wife would not approve |
| 17:11.38 | rerzerty | hello ? |
| 17:11.43 | outtolunc | hgaha |
| 17:12.03 | jameswf-home | has not done xyz under cygein as he dont do windoze |
| 17:12.42 | rerzerty | sipura 3000 is it act as a router ? plz |
| 17:13.11 | rerzerty | can be replace with an isp router ? |
| 17:13.33 | jameswf-home | My 3 desktops + 1 server at home all linux my laptop and work desktop linux as well..... who needs windoze... |
| 17:13.46 | MrTelephone | option_priority_jumping ?? |
| 17:14.24 | MrTelephone | is that changed from piriorityjumping |
| 17:14.43 | rerzerty | hello |
| 17:14.47 | rerzerty | cooperate psls |
| 17:15.40 | rerzerty | killme r u there ? |
| 17:16.06 | vlsoft | jameswf-home: well, I'm a Linux system administrator, so I won't need it; but a friend of mine want to use my asterisk server with an iax softphone, and he's got a few "needs", as he's blind. iaxcomm works well with his talking software, but I need to put a few things into it - the first step for me is to create a build env. where the original can be compiled. Am I right? |
| 17:17.26 | vlsoft | jameswf-home: oh and yeah: he uses windows, so I have to compile iaxcomm for windows... |
| 17:18.08 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 17:18.57 | outtolunc | hmm, i thought diax had some features like that |
| 17:19.08 | outtolunc | search 'dante diax' |
| 17:19.19 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 17:19.21 | jameswf-home | moziax is cross platform |
| 17:19.42 | SteveTotaro | James, have you tested moziax? |
| 17:20.04 | SteveTotaro | it looks very promising but the lack of chatter has me worried that it may not be very good |
| 17:20.32 | SteveTotaro | it would be awesome on thin clients |
| 17:20.53 | jameswf-home | i use moxiax on my laptop I like it no fluff does what its suppose to |
| 17:21.06 | SteveTotaro | any negatives? |
| 17:21.13 | rerzerty | eyuilm |
| 17:21.22 | vlsoft | Well, my friend had diax, but he doesn't liked it (who knows why; but i'm not blind, i can't tell what was his problem with it). |
| 17:21.27 | SteveTotaro | besides being firefox dependant |
| 17:21.29 | jameswf-home | It's ugly |
| 17:21.34 | outtolunc | negatives would be the lack of module control, enable/disable, and access control |
| 17:21.35 | rerzerty | adiga oya chidi bin eich bin |
| 17:21.59 | SteveTotaro | it seems to load every time you open a new instance of firefox too |
| 17:22.21 | SteveTotaro | but i have not done any real testing yet |
| 17:23.07 | SteveTotaro | outtolunc: what do you mean by access control? |
| 17:23.33 | SteveTotaro | you could always place it in a context with authenticate if that is what you mean |
| 17:23.52 | outtolunc | access control, meaning password protect the settings |
| 17:24.01 | vlsoft | For blind people; the first thing is: what the program does when you can't use the mouse; does the GUI recognizable with "Jaws" (a popular talking program here for blinds, almost all blind people i know use that) |
| 17:24.32 | vlsoft | What does it do when you press tab, does it cycle through buttons as it should, or not. |
| 17:24.39 | SteveTotaro | vlsoft: go on, this is great info |
| 17:25.24 | outtolunc | needs to pay attention as my eyes are getting worse everyday |
| 17:25.31 | outtolunc | <- old fart |
| 17:25.31 | SteveTotaro | is there a site that details how to program apps for the blind? |
| 17:25.42 | vlsoft | I don't know moziax, so i'll try it out. |
| 17:26.38 | SteveTotaro | like i said, i installed it and used it once, it looked and worked OK, that is why i would expect more community chatter |
| 17:27.07 | vlsoft | SteveTotaro: they don't have "extra" needs, if the program uses the standard windows GUI components, then Jaws will talk to them, it says the button currently selected (like "1" "3"), but if the buttons have only icons on them, well, that's a catastrophe. |
| 17:28.39 | vlsoft | well, for diax, pressing tab does not do anything (and pressing space or enter, it's not working - so that's why my friend couldn't use it) |
| 17:28.46 | SteveTotaro | http://www.nanopac.com/JAWS.htm |
| 17:28.49 | outtolunc | maybe you can get access tothe ribbit api |
| 17:29.13 | outtolunc | checking |
| 17:29.14 | SteveTotaro | i see a few companies in google when searching for "jaws blind" |
| 17:29.30 | SteveTotaro | is the above link the correct Jaws? |
| 17:30.03 | jameswf-home | thats nasty miller had a commercial with this spanish girl trying to act all sexy and she rides up her skirt and WOAH she needs to shave her legs |
| 17:30.14 | jameswf-home | who are they marketing to |
| 17:30.28 | SteveTotaro | many parts of the world |
| 17:30.32 | SteveTotaro | french maybe |
| 17:30.33 | vlsoft | http://www.freedomscientific.com/ <- this is the jaws page my friend showed me once |
| 17:30.35 | outtolunc | 1hotmamma |
| 17:30.39 | outtolunc | haha |
| 17:31.44 | SteveTotaro | thanks for the tip vlsoft, i also need to look into TTY for the deaf |
| 17:32.02 | SteveTotaro | ADA and all |
| 17:32.33 | SteveTotaro | not sue what can be done for the deaf and blind.... |
| 17:32.34 | vlsoft | (And yes, Jaws is definitely NOT cheap... Ehhh... Actually it's too much money from blind people, but they use it.) |
| 17:32.39 | jameswf-home | A sip tty... seems redundant if you have the interweb and all but that actualy sounds kinda cool |
| 17:32.53 | jameswf-home | oh wait thats TDD |
| 17:33.04 | SteveTotaro | TDD? |
| 17:33.28 | jameswf-home | Telephond device for the deaf |
| 17:33.48 | SteveTotaro | so it is voice to text? |
| 17:34.08 | vlsoft | Linux/Gnome has some support for blind people, KDE does not (but that should change with QT4, QT3 haven't got the required API stuff for screen readers). |
| 17:34.44 | jameswf-home | SteveTotaro: http://en.wikipedia.org/wiki/Telecommunications_devices_for_the_deaf |
| 17:34.49 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:35.05 | SteveTotaro | you would think there would be free or at least very inexpensive programs for the blind. |
| 17:35.23 | vlsoft | You might be surprised, how blind people use Windows; almost like You or me... But without mouse. They use IE/Firefox, and the Office stuff (like Word, Excel) |
| 17:35.49 | SteveTotaro | out of need, kindness, government subsidy |
| 17:36.08 | ManxPower | SteveTotaro: there are |
| 17:36.16 | vlsoft | And Jaws can recognize bitmaps too, so it's not impossible to "learn" the buttons on diax, but it's a pain in the ass - so to say. |
| 17:36.20 | SteveTotaro | i would rather see tax dollars to help the blind use a computer rather than pay for illegals to get healthcare |
| 17:36.42 | ManxPower | At least in the USA, in fact I believe that every phone bill has a small fee on it for the fund that helps with TTY/TTD needs. |
| 17:36.44 | jameswf-home | One of our customers is a blind asterisk admin... does pretty well Bling folks excell pretty well in this type of deal as it is 99% text based |
| 17:37.03 | jameswf-home | s/bling/blind/ |
| 17:37.21 | vlsoft | Well, the downloadable iaxcomm client works pretty good, it's Jaws-able... But I have to put a few features into it that my friend asked. |
| 17:37.49 | SteveTotaro | iaxcom is a rename of an older client? |
| 17:38.10 | SteveTotaro | or is it a browser plugin too? |
| 17:38.30 | vlsoft | (Like different ringtone for different callers - and iaxcomm has some ring problem when two iaxcomm call each other; if i call an iaxcomm from my hardphone, it works okay, but not between two iaxcomms...) |
| 17:38.31 | SteveTotaro | i really like the browser plugin idea for thin clients |
| 17:39.09 | vlsoft | iaxcomm is a "sample client" for the iaxclient library : iaxclient.sourceforge.net |
| 17:40.06 | vlsoft | http://iaxclient.sourceforge.net/iaxcomm/ <- the binary here works almost perfect for blind people (as stated above, with minor problems...) |
| 17:40.27 | SteveTotaro | too bad i didn't get the code (or authorization) to release an updated JIAX client |
| 17:41.26 | SteveTotaro | i had it at one point but had to adhere to my contract |
| 17:43.24 | outtolunc | i was wondering whatever happened to jiax, i liked it |
| 17:43.41 | ManxPower | hugs his Polycom |
| 17:44.06 | outtolunc | closes the curtain so manx can be alone with his phone |
| 17:44.15 | vlsoft | :) |
| 17:45.56 | vlsoft | This is a cygwin problem I think, but maybe someone here encountered similar; the first error is: |
| 17:46.16 | vlsoft | ./configure: line 18793: PKG_PROG_PKG_CONFIG: command not found (and yes, pgk_config is installed) |
| 17:46.34 | vlsoft | sed s/gk/kg/ |
| 17:46.58 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-86-17.vif.net) |
| 17:47.34 | JamminJud | is happy that everything is working now |
| 17:48.48 | vlsoft | the remaining 2 errors are: |
| 17:49.03 | vlsoft | ./configure: line 19003: syntax error near unexpected token `PORTAUDIO,' |
| 17:49.06 | vlsoft | and: |
| 17:49.15 | vlsoft | ./configure: line 19003: `PKG_CHECK_MODULES(PORTAUDIO, portaudio-2.0 >= 19,,{ { echo "$as_me:$LINENO: error:' |
| 17:49.47 | vlsoft | so this is where i am at the moment, and i don't know why can't i even ./configure this iaxclient library... |
| 17:50.18 | vlsoft | this of course would work flawlessly under any Linux environment - grrrr.... |
| 17:52.15 | jameswf-home | heh token |
| 17:53.02 | *** join/#asterisk codefreeze-lap (n=murf@ip68-109-175-69.ph.ph.cox.net) |
| 17:57.31 | vlsoft | Oh and I forgot to mention what an iax client program should NOT do EVER when a blind uses that computer: mess with the volume controls!!! (So they won't hear ever :D as a nasty plus over their blindness - aww) |
| 17:58.39 | Marquel | possibly a stupid question and not strictly asterisk-related, but is it possible to have two sip-phones sharing the same account at the same time? |
| 17:58.48 | vlsoft | If the volume gets too low; then they can't really find the mixer; if it gets too high, then they get a nasty surprise. |
| 17:58.53 | SteveTotaro | i hate mingw and cygwin |
| 17:59.37 | vlsoft | Well, I start to hate it now too... |
| 17:59.44 | *** join/#asterisk decaf (n=mehmet@85.108.247.15) |
| 18:00.03 | SteveTotaro | i developed the hate while working on the JIAX stuff |
| 18:00.26 | outtolunc | i was watching a show on hulu.com a week ago, and damn near blew my speakers when an advert came on |
| 18:00.39 | SteveTotaro | tv does the same thing |
| 18:00.47 | outtolunc | this was twice what tv does |
| 18:00.51 | vlsoft | Now multiply it by 2, put on a headphone, and ouch... |
| 18:00.56 | SteveTotaro | the show is a good volume and then the commercials blow your speakers |
| 18:01.01 | vlsoft | This is what a blind gets... |
| 18:01.19 | SteveTotaro | wake you from a good sleep if you are like me and have to have noise to sleep |
| 18:01.33 | SteveTotaro | i cannot sleep in dead silence |
| 18:02.03 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:02.12 | SteveTotaro | not only blind but going deaf from the headphone blast |
| 18:02.20 | outtolunc | places an ipod in one of those rocking water displays, minus the water, the ipod slides back and forth playing <G> |
| 18:02.23 | vlsoft | Okay, i figured out how to proceed with that ./configure problem, it was an environment problem: export PKG_CONFIG=/usr/bin/pkg-config |
| 18:02.36 | vlsoft | cygwin forgots to set up environment variables??? |
| 18:02.49 | SteveTotaro | yes cygwin and mingw are terrible in that regard |
| 18:02.56 | SteveTotaro | you need to pass all kinds of flags |
| 18:03.05 | vlsoft | argh... nasty stuff... |
| 18:03.31 | SteveTotaro | but the joy of success is that much more satisfying! |
| 18:04.02 | vlsoft | configure: error: [new line] portaudio is required to build this package! |
| 18:04.04 | vlsoft | :D |
| 18:04.10 | SteveTotaro | the first time i got jiax to compile the jars, i was jumping around and yelling (all alone of course) |
| 18:04.26 | SteveTotaro | just another flag you need to pass |
| 18:04.31 | vlsoft | well, i thought that was included with iaxclient... have to run a few rounds again :D |
| 18:04.57 | *** join/#asterisk steveaj (n=steve@82-71-61-44.dsl.in-addr.zen.co.uk) |
| 18:05.11 | vlsoft | but yes, success (although a small one) |
| 18:05.25 | RobH | i forget, someone tell me how to turn on color in the CLI? |
| 18:05.28 | RobH | please =] |
| 18:05.51 | SteveTotaro | it takes many small success to make a big one |
| 18:09.12 | vlsoft | Well folks, thanks for the support; it seems i'm on the track again... I have to go now, but it was a pleasure to talk here with You. |
| 18:14.17 | jameswf-home | many small farts lead to a big shart and maybe a poo :) |
| 18:16.15 | ruied | I normally use the "make samples" to have a working asterisk" than I change sip.conf and extensions.conf... now I don't want to make the files from the samples, I've made my extensions.conf and sip.conf. but it simes that I'm missing some files since * cli doesn't have the 'sip' command... |
| 18:17.07 | ruied | I only have the asterisc.conf, sip.conf and extensions.conf files in /etc/asterisk ... |
| 18:18.26 | *** join/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 18:18.51 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:19.40 | MrTelephone | where do you include ODBC_STORAGE Cflag in the asterisk 1.4? |
| 18:19.48 | MrTelephone | is that set in the configure line? |
| 18:25.49 | *** join/#asterisk chendy (n=chatzill@58.61.8.58) |
| 18:28.20 | jameswf-home | Maybe its cause I have done it 1000 times but I cant remember ever having an issue compiling... |
| 18:29.41 | Qwell | eXistenZ: Private personalized help is $250/hour. Ask here instead. |
| 18:29.41 | eXistenZ | ok |
| 18:29.41 | Qwell | (and don't message an op for help) |
| 18:29.41 | eXistenZ | I have a problem here, I have some stupid anonymous caller (on pots line), I cannot change my number because it is already known to all people. I am trying to find a solution without changing the number to block anonymous callers, or just redirect them somewhere else. |
| 18:30.49 | SteveTotaro | the telco here will do this for you |
| 18:30.57 | eXistenZ | I am in Israel |
| 18:31.21 | eXistenZ | There is no such service |
| 18:31.29 | jameswf-home | exten => s,n,GotoIf($["${CALLERID(num)}" = " "]?telemarket-torture,begin,1) |
| 18:31.29 | plik | eXistenZ: don't they Anonymous Call Rejection over there? |
| 18:31.37 | eXistenZ | plik, only for mobiles |
| 18:31.43 | eXistenZ | I've already done it for mobiles |
| 18:31.51 | SteveTotaro | then dump any call without callerid to Hangup() |
| 18:32.09 | eXistenZ | that's not the problem |
| 18:32.12 | plik | eXistenZ: bah, over here (UK) we can get it on landlines but not mobiles, despite it being a legal requirement |
| 18:32.19 | eXistenZ | the problem is that they always call |
| 18:32.23 | eXistenZ | the ring-ring :/ |
| 18:32.32 | SteveTotaro | just dump them before they ring |
| 18:32.52 | eXistenZ | I don't want a sidejob |
| 18:32.55 | plik | eXistenZ: asterisk can answer all calls with no callerid, before you hear it ring.... |
| 18:33.02 | plik | and do whatever you want with it |
| 18:33.06 | jameswf-home | eXistenZ: exten => s,1,GotoIf($["AA${CALLERID(num)}" = "AA "]?telemarket-torture,begin,1) << this should work change dest to whatever |
| 18:33.28 | SteveTotaro | just send them to Hangup() |
| 18:33.37 | SteveTotaro | don't tie up POTS lines |
| 18:33.47 | jameswf-home | I use telemarketer torture as hangup is too nice |
| 18:33.52 | eXistenZ | plik, connecting asterisk to POTS lines needs a serious budget :) |
| 18:33.53 | plik | sound advice there |
| 18:34.16 | plik | not really - linksys spa3102 is a good low budget option |
| 18:34.36 | eXistenZ | plik, for the line and for the phones? |
| 18:34.36 | SteveTotaro | Even openvox if Martin will ship to you |
| 18:35.26 | SteveTotaro | well, an Asterisk PBX with phones isn't that expensive compared to proprietary |
| 18:35.28 | jameswf-home | actualy create a context that plays sit and ss-noservice would be better then an outright hangup() |
| 18:35.28 | SteveTotaro | how many phones? |
| 18:35.42 | eXistenZ | SteveTotaro, 5 |
| 18:35.46 | MrTelephone | odbc is fuckin shit to setup |
| 18:35.57 | MrTelephone | man that pisses me off .. brb |
| 18:35.57 | SteveTotaro | odbc is better than the mysql addon |
| 18:36.09 | *** join/#asterisk gitguy (n=diego@adsl-152-204.click.com.py) |
| 18:36.10 | MrTelephone | there is barely any verbose when it can't connect |
| 18:36.10 | eXistenZ | spa3102 is enough for the external line and 1 phone line? |
| 18:36.11 | gitguy | hi |
| 18:36.21 | gitguy | i have a asterisk server on amazon ec2, i tried SIP and IAX2 |
| 18:36.23 | MrTelephone | I had it working 2 days ago and now i can't connect with isql |
| 18:36.27 | plik | eXistenZ: yes... |
| 18:36.31 | gitguy | but i don't see any activity going on when i try to register a phone |
| 18:36.34 | gitguy | to the server |
| 18:36.37 | SteveTotaro | it can only be so many things |
| 18:36.41 | gitguy | with both protocols |
| 18:36.51 | gitguy | why could that be? |
| 18:36.51 | SteveTotaro | firewall, creds... |
| 18:37.00 | plik | you can add 2 X PAP2 for the additional 4 FXS (analog phones) |
| 18:37.07 | SteveTotaro | sip debug, iax debug |
| 18:37.13 | gitguy | i have that on |
| 18:37.17 | gitguy | i don't see activity |
| 18:37.19 | gitguy | when i try to connect |
| 18:37.38 | gitguy | iptables -t nat -L -v ; iptables -t filter -L -v also don't show nothing |
| 18:37.51 | SteveTotaro | stop iptables for a bit and try |
| 18:38.10 | gitguy | iptables don't have any rules set |
| 18:38.10 | plik | erk, time to go... laters |
| 18:38.14 | plik | & |
| 18:38.31 | eXistenZ | plik, p3 computer + spa3102? |
| 18:38.44 | SteveTotaro | that would work e |
| 18:38.47 | gitguy | SteveTotaro: the only thing I know is that amazon has some weird network |
| 18:39.00 | SteveTotaro | amazon? |
| 18:39.05 | gitguy | SteveTotaro: and my server is probably behind their firewalls/nats |
| 18:39.06 | gitguy | yes |
| 18:39.07 | gitguy | ec2 |
| 18:39.23 | SteveTotaro | ah, territory i would be weary of |
| 18:39.27 | gitguy | i told my boss to use a better server solution but he doesn't listen to me |
| 18:39.37 | SteveTotaro | know maybe he will |
| 18:40.13 | SteveTotaro | if you can get a tech from amazon to confirm it wont work, then you get "i told you so points" |
| 18:41.10 | SteveTotaro | those points can be dangerous depending on your boss though ;) |
| 18:42.18 | gitguy | i already mentioned him that it doesn't work and he said "let me do research, i remember there is some tutorials" |
| 18:42.21 | gitguy | bah |
| 18:42.38 | gitguy | too idiotic |
| 18:44.31 | gitguy | where can i find a list of SIP/IAX2 ports? |
| 18:44.55 | gitguy | these protocols works over udp right? |
| 18:45.56 | *** join/#asterisk paulc (n=paulc@S01060013102c9156.vc.shawcable.net) |
| 18:48.10 | jameswf-home | wtf is the command to kill the peer cache |
| 18:49.02 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:52.52 | ManxPower | gitguy: IAX2 is UDP port 4569, SIP is UDP port 5030, RTP (audio) is dynamically determined at call setup time. You can control Asterisk's side of the RTP ports in /etc/asterisk/rtp.conf |
| 18:53.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:54.09 | Strom_C | ManxPower: SIP is UDP 5060 |
| 18:54.32 | ManxPower | Strom_C: I sit corrected. |
| 18:54.54 | gitguy | ManxPower: ok |
| 18:55.03 | gitguy | ManxPower: i think that's the problem, I had to open port 80 for apache and all tha ttoo |
| 18:55.06 | gitguy | that too* |
| 18:55.19 | ManxPower | gitguy: you're going to have to open much more than that to run an Asterisk server |
| 18:55.23 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 18:55.44 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:56.05 | gitguy | ManxPower: yes, I opened UDP/4569 for IAX2 and i see activity on iax2 set debug |
| 18:56.06 | Marquel | retrying from earlier: possibly a stupid question and not strictly asterisk-related, but is it possible to have two sip-phones sharing the same account at the same time? |
| 18:56.09 | gitguy | ManxPower: when I try to register |
| 18:56.17 | Strom_C | Marquel: no |
| 18:56.24 | ManxPower | Marquel: not with Asterisk, yes with some other systems. |
| 18:56.29 | Strom_C | not if you want them to receive calls |
| 18:56.48 | Strom_C | for placing outbound calls only, you can have them use the same credentials if you do it right |
| 18:57.10 | Strom_C | but, really, why bother |
| 18:57.11 | Marquel | Strom_C, ManxPower: then i'll give them different accounts and make asterisk fake they're both the same. thx. |
| 18:57.22 | Strom_C | what are you trying to do, exactly? |
| 18:57.26 | Strom_C | shared line appearances? |
| 18:57.33 | ManxPower | Marquel: stop setting the SIP username to be the same as the extension -- doing that leads to problems like you are seeing |
| 18:59.03 | Marquel | Strom_C: have a mounted and a dect-phone sharing the same extension. in inbound-calls as well as outbound. but if it's not possible to have that done by sharing sip-accounts, i'll program asterisk to act as if it were so, using different accounts. |
| 18:59.30 | eXistenZ | I have some old P1 computer with 64MB, would it be enough for asterisk? |
| 18:59.43 | Strom_C | Marquel: they don't need to be the same extension to both ring for the same call |
| 18:59.50 | Marquel | ManxPower: i won't regard that as a problem, it was just a question to stop me from doing something in completely wrong direction... |
| 18:59.54 | Strom_C | you /can/ ring multiple extensions simultaneously... |
| 19:00.19 | Marquel | Strom_C: that is known, but it also was about them sharing the same CID on outbound calls. ;) |
| 19:00.22 | ManxPower | devices, not extensions |
| 19:00.32 | Strom_C | Marquel: that can be done too |
| 19:00.50 | MrTelephone | ok i setup voicemail with odbc and now asterisk 1.4.19 segfaults.. whats the deal I wonder |
| 19:01.25 | Marquel | Strom_C: i guess by setting the cid through asterisk prior to starting the dial(). that's also known. i was just wondering if i need to undergo that procedure or if accountsharing would spare me that ;) |
| 19:01.57 | ManxPower | Marquel: no, set it in the sip.conf account |
| 19:02.39 | Marquel | ManxPower: ah - _that_ was _not_ known to me. that makes my life even simpler then :) |
| 19:03.01 | ManxPower | Marquel: start reading the stuff in /path/to/src/asterisk/configs and /path/to/src/asterisk/doc |
| 19:07.34 | eXistenZ | does SPA8000 include an FXO port as well |
| 19:11.19 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 19:17.36 | *** join/#asterisk dkwiebe (n=Darren@139.142.18.18) |
| 19:17.52 | eXistenZ | is it possible to use a hardphone over the same line, on which a connection to a router is established? |
| 19:23.54 | EmleyMoor | eXistenZ: What do you mean by the same line? |
| 19:24.09 | eXistenZ | I have a cat5 line from my room to my router |
| 19:24.26 | eXistenZ | It is inside the wall, I don't want to pass another cabel |
| 19:24.28 | eXistenZ | *cable |
| 19:24.44 | Strom_C | eXistenZ: of course you can do that |
| 19:24.50 | Strom_C | that's the joy of packet data |
| 19:24.56 | EmleyMoor | eXistenZ: Yes - if the phone has a built in hub, no problem. If not, add a hub or switch |
| 19:24.59 | *** join/#asterisk SuperGeek (n=SuperGee@unaffiliated/supergeek) |
| 19:25.03 | SuperGeek | Hello |
| 19:25.25 | SuperGeek | I'm rather interested in Asterisk as a whole for use in my home...could someone answer a few of my questions? |
| 19:25.53 | EmleyMoor | SuperGeek: Just try us |
| 19:25.56 | SuperGeek | Alright |
| 19:26.13 | Strom_C | uses asterisk in his home |
| 19:26.21 | EmleyMoor | uses it too |
| 19:26.23 | SuperGeek | Well, you know how in call centers and such they can recieve multiple calls on a single line? |
| 19:26.30 | SuperGeek | How would I go about doing that with Asterisk? |
| 19:26.35 | Strom_C | SuperGeek: it's not a single line |
| 19:26.39 | EmleyMoor | SuperGeek: By "line" do you mean "number"? |
| 19:26.42 | SuperGeek | Yes. |
| 19:26.50 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:26.51 | Strom_C | number != line |
| 19:26.53 | SuperGeek | (I'm new to telephony as a whole, so bear with me) |
| 19:26.58 | SuperGeek | Ok, ok |
| 19:27.01 | EmleyMoor | Many VoIP providers will allow you to receive more than one call at a time |
| 19:27.09 | EmleyMoor | (on the same number) |
| 19:27.12 | Strom_C | SuperGeek: you may want to read "telephony 101" which is a good crash-course in telephony |
| 19:27.14 | Strom_C | ~101 |
| 19:27.14 | jbot | hmm... 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 19:27.19 | SuperGeek | ok. |
| 19:27.32 | SuperGeek | EmleyMoor: But with a landline provider such as Verizon, is that possible |
| 19:28.04 | Strom_C | SuperGeek: it's possible if you order, say, ISDN PRI from them |
| 19:28.15 | SuperGeek | Oh.. |
| 19:28.18 | Strom_C | but for a traditional single-pair analog circuit, the answer is no |
| 19:28.22 | SuperGeek | How much would that run? |
| 19:28.35 | Strom_C | $300-$1000 per month? |
| 19:28.38 | EmleyMoor | I don't know much about ISDN so I can't answer any more on that |
| 19:28.42 | SuperGeek | gah |
| 19:29.01 | Strom_C | SuperGeek: but ISDN PRI is probably overkill for what you need |
| 19:29.01 | SuperGeek | What I'm worried about is that I'll loose voicemail |
| 19:29.16 | Strom_C | ...so you want the voicemail to be tight instead? |
| 19:29.21 | SuperGeek | See, with my current plan when I'm on the line with someone else, any other callers get redirected to voicemail |
| 19:29.29 | EmleyMoor | SuperGeek: If you went VoIP you could implement your own |
| 19:29.41 | SuperGeek | ah |
| 19:29.50 | EmleyMoor | Asterisk has a rather good voicemail system |
| 19:29.52 | SuperGeek | I think I'm beginning to see what you're talking about |
| 19:29.56 | Strom_C | SuperGeek: so get a VoIP account that allows multiple calls, or get a second analog circuit and set up a rotating hunt group |
| 19:30.12 | SuperGeek | Strom_C: A rotating what? |
| 19:30.26 | Strom_C | rotating hunt group |
| 19:30.32 | paulc | two phone lines, 1 number - calls hunt (or rotate) across the free/available phone lines |
| 19:30.44 | SuperGeek | oh |
| 19:30.50 | paulc | actually, "two" could be any number |
| 19:31.07 | SuperGeek | i see |
| 19:31.12 | SuperGeek | I already have something like that |
| 19:31.19 | paulc | then you can decide how you want to hunt.. sequential.. cyclic.. or in north america there's a third variant "most idle" (which I think's weird, but that's cos I'm European) |
| 19:31.37 | SuperGeek | Do you mean two phones hooked up in my house, and when someone calls my number both ring? |
| 19:31.40 | Strom_C | no |
| 19:31.42 | SuperGeek | >_< |
| 19:31.46 | ManxPower | Actually each analog line has it's own number, even if you don't tell people about it. It will show up on the callerid of outgoing calls |
| 19:31.55 | Strom_C | SuperGeek: go read that document i linked you to |
| 19:32.02 | SuperGeek | Strom_C: Alright |
| 19:32.07 | EmleyMoor | SuperGeek: That sounds like one line with two phones on it |
| 19:32.14 | SuperGeek | EmleyMoor: YEah. |
| 19:32.22 | ManxPower | paulc: longest idle was VERY popular in the days of the modem |
| 19:32.38 | SuperGeek | Well, ok then |
| 19:32.50 | SuperGeek | For the sake of simplicity, I'll probably go VoIP |
| 19:32.53 | paulc | multi line = imagine you have 3 phones on the desk. Someone calls your number. The first phone rings. If a second person calls your number, and the first phone is busy, the second phone rings. Ditto for a third caller/third phone. If 3 people are on 3 calls and a 4th person phones you, they get busy tone |
| 19:32.54 | ManxPower | In the event of a problem with a modem, the next call would not hit that modem |
| 19:32.57 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 19:33.13 | SuperGeek | I'm sure you guys all get this question, but could you recommend a particularly good VoIP provide? |
| 19:33.15 | SuperGeek | provider* |
| 19:33.16 | paulc | ManxPower: But why? Why not just use cyclic hunting? s'what we did back in the UK - the "most idle" thing just seemed like extra overhead in the switch to me |
| 19:33.24 | Strom_C | SuperGeek: i like teliax |
| 19:33.31 | EmleyMoor | SuperGeek: It all depends what you want... |
| 19:33.46 | paulc | SuperGeek: I like www.link2voip.com but there's a whole bunch of 'em out there |
| 19:33.51 | EmleyMoor | I use voiptalk but that's mainly because they are British and cheapish |
| 19:33.51 | SuperGeek | ok |
| 19:34.00 | paulc | Where are you, SuperGeek? |
| 19:34.11 | SuperGeek | I live in the US, I'd like something under $20/mo, and I need to recieve multiple calls on a single number |
| 19:34.20 | Strom_C | SuperGeek: teliax |
| 19:34.23 | SuperGeek | Alright |
| 19:34.26 | Strom_C | pay-as-you-go plan |
| 19:34.28 | paulc | what sort of call volume? calls/minutes a month? |
| 19:34.31 | ManxPower | SuperGeek: that's not going to happen with any standard telephone line |
| 19:34.43 | SuperGeek | ManxPower: That's why I'm going VoIP |
| 19:34.52 | ManxPower | SuperGeek: A friend pays about $5/month |
| 19:34.52 | SuperGeek | Well, call volume |
| 19:35.04 | ManxPower | 1 DID number, some not large number of mins |
| 19:35.04 | SuperGeek | If possible, unlimited calling in the US |
| 19:35.10 | Strom_C | SuperGeek: and, seriously, go read that document now. |
| 19:35.10 | SuperGeek | ManxPower: DID = ? |
| 19:35.15 | SuperGeek | Strom_C: I am, haha |
| 19:35.15 | ManxPower | SuperGeek: phone number |
| 19:35.43 | SuperGeek | Well, alright then |
| 19:35.48 | Strom_C | SuperGeek: no...you're chatting on IRC and glancing at the document :P |
| 19:35.49 | SuperGeek | I'm going to go read up on this stuff |
| 19:35.59 | SuperGeek | Strom_C: Now I'm not, heh |
| 19:36.06 | SuperGeek | idles. |
| 19:36.55 | EmleyMoor | I got a block of 10 geographic numbers and have used eight of them so far, with one of the uses trivial enough to relinquish |
| 19:38.48 | EmleyMoor | ... leaving me three available for faxing, if a way ever comes |
| 19:39.05 | paulc | it's funny how fax is still so prevalent |
| 19:39.11 | paulc | but mostly for junk faxes/spam/etc |
| 19:39.21 | EmleyMoor | paulc: My partner depends on it |
| 19:39.38 | ManxPower | Maybe in your world. My users depends on fax every day |
| 19:39.49 | ManxPower | (mostly contract and contract revisions) |
| 19:40.01 | SuperGeek | EmleyMoor: What provider do you use, and how much do you pay? |
| 19:40.15 | jameswf-home | if someone could call 2222@sip.jameswf.info so I can see if it is me or the provider? THX |
| 19:40.38 | EmleyMoor | voiptalk. GBP 17.63 a month for the numbers, and top up my account 10 or 20 at a time when needed |
| 19:40.47 | SuperGeek | hm. |
| 19:40.48 | SuperGeek | Thanks |
| 19:40.57 | paulc | I've got a DID with voiptalk.. they're pretty good, I've been happy with them.. |
| 19:42.24 | EmleyMoor | Not sure if, across my 10, I am limited to only 2 calls at a time or something, but there's only 2 of us here so that is just about enough |
| 19:42.56 | *** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
| 19:43.00 | ManxPower | Most per min plans have unlimited number of calls, most per month plans only allow 1 or 2 calls at a time. |
| 19:43.05 | *** part/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
| 19:44.45 | paulc | I'm still paying Telus $50/month for my fully bundled home phone line.. I should SO port to VoIP and pay $2.50/month + per minute, for the amount I use it |
| 19:44.53 | paulc | internet down? use the cell phone |
| 19:45.06 | paulc | $500/year savings = plenty of beer :) |
| 19:45.24 | EmleyMoor | I still use BT for calls on which they are cheaper |
| 19:45.32 | ManxPower | Emercency services? 911, 113 |
| 19:46.11 | EmleyMoor | Weekend/evening 0[123],, most 0[89]., 0087 (if ever!)... |
| 19:46.23 | EmleyMoor | 999/112 goes out over them too |
| 19:47.23 | paulc | Yeah.. emergency services is the only thing I'd worry about.. |
| 19:47.39 | EmleyMoor | As it happens, I have only accounted for 03 numbers as of today! |
| 19:47.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:47.54 | paulc | the other option is make my smart ring number that I never use my prime line, strip all features, but it's still $25 to 30 a month.. for something I'd barely use |
| 19:48.00 | jameswf-home | who needs 911, |
| 19:48.25 | ManxPower | jameswf-home: only people that will soon be dead anyway, so why worry about it? |
| 19:48.35 | jameswf-home | exactly |
| 19:48.47 | paulc | LOL it's all about risk and probability |
| 19:49.25 | EmleyMoor | There's a strong rumour of emergency calls being allowed over VoIP here soon |
| 19:50.19 | jameswf-home | http://www.voip-info.org/wiki/view/VOIP+911+Service+Providers |
| 19:54.24 | *** join/#asterisk nanex (n=mariano@189.140.132.85) |
| 19:54.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:56.38 | nanex | Evening guys! I'm just about ready to purchase some hardware for a PBX, and I'm between a TDM410 with 4 FXO, or a Grandstream GXW4104 and was wondering if someone here could point me in the right direction. Both seem to do the same, but the TDM is 600+ dollars, and the GXw is about 200. |
| 19:56.57 | ManxPower | ~gs |
| 19:56.57 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:57.03 | ManxPower | ~phones |
| 19:57.04 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 19:58.26 | nanex | Wow, didn't know it was that bad... I'll go with the TDM then, thanks :) |
| 19:59.17 | ManxPower | nanex: there may be similar non-grandstream products out there -- keep searchng |
| 19:59.27 | EmleyMoor | I certainly haven't had a problem with my TDM card apart from the echo, and I gather that was part of the rationale behind the TDM410P |
| 19:59.41 | EmleyMoor | If I could get a bare 410 cheap I'd try it |
| 20:00.00 | nanex | yup, will do, thing is I live in Mexico, and there's not much supply here... need to find a Digium reseller |
| 20:00.20 | EmleyMoor | The supplier I bought mine from seems to have vanished too |
| 20:10.05 | SuperGeek | rofl |
| 20:10.11 | SuperGeek | What's wrong with Grandstream |
| 20:10.20 | paulc | uh.. everything? ;-) |
| 20:10.22 | paulc | nah, I kid.. |
| 20:10.28 | paulc | they're "alright" but they're not great.. |
| 20:10.31 | SuperGeek | I see |
| 20:10.34 | SuperGeek | ~gs |
| 20:10.35 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:10.36 | paulc | I like Sipura ATAs |
| 20:10.39 | EmleyMoor | They have a reputation for poor build quality |
| 20:10.40 | SuperGeek | ^^ That says differently |
| 20:10.41 | SuperGeek | heh |
| 20:10.42 | SuperGeek | Ok |
| 20:10.46 | paulc | jbot speaketh the truth |
| 20:10.55 | ManxPower | The GS HARDWARE is not that horrid, but they could not write stable firmware if their lives depended on it. |
| 20:11.05 | paulc | when I worked for a supplier of VoIP gear, we definitely had more GS returns than any other brand |
| 20:11.22 | ManxPower | With GS products you just keep trying beta firmware versions until you get one that works for YOUR usage patterns |
| 20:12.26 | EmleyMoor | I still like my ancient phone |
| 20:12.36 | *** join/#asterisk dkwiebe (n=Darren@139.142.18.18) |
| 20:13.56 | dkwiebe | greetings everyone. I have a new building of asterisk 1.2.26.1. It seems to work fine except that there is no audio from the sound files. I can talk between phones without a problem. I'm using ulaw on the phones and the sound files are in .ulaw |
| 20:14.06 | EmleyMoor | I am looking, once I've moved perhaps, to get some IP phones |
| 20:14.28 | ManxPower | dkwiebe: you also upgraded zaptel, didn't you? |
| 20:15.24 | dkwiebe | ahhh, that might be my problem. I'm on zaptel 1.2.25 |
| 20:16.02 | dkwiebe | no, 1.2.25 is the most recent version of zaptel in the 1.2 series |
| 20:16.34 | dkwiebe | I'm using Sangoma hardware patched into the zaptel drivers |
| 20:18.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:18.52 | MrTelephone | im not too impressed with any of that realtime stuff |
| 20:19.01 | MrTelephone | how come sip show users doesn't show users that are in the database? |
| 20:20.49 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:20.49 | ManxPower | dkwiebe: rmmod ztdummy, then start asterisk, if that fixes it, then it's a common problem. |
| 20:21.47 | ManxPower | I don't remember the fix, as I don't use ztdummy, but a search of the mailing list should help. |
| 20:21.49 | ManxPower | ~mailinglist. |
| 20:21.49 | jbot | rumour has it, mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 20:21.52 | *** join/#asterisk ruied (n=ruied@89.181.126.230) |
| 20:21.52 | ManxPower | ~mailinglist |
| 20:21.53 | jbot | rumour has it, mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 20:22.09 | ManxPower | Cooolll, jbot ignored the stray . |
| 20:22.12 | dkwiebe | ERROR: Module ztdummy does not exist in /proc/modules |
| 20:22.21 | dkwiebe | K, I'll check the mailling archives. |
| 20:22.23 | ManxPower | dkwiebe: lsmod does not show it either |
| 20:22.28 | dkwiebe | no it doesn't |
| 20:22.35 | ManxPower | then that is not your problem |
| 20:22.43 | dkwiebe | k |
| 20:22.49 | ManxPower | The problem ONLY happens when ztdummy is loaded. |
| 20:23.03 | dkwiebe | k, then I won't try rebuilding it. :-) |
| 20:23.25 | *** join/#asterisk s0lid (n=s0lid@210.213.198.112) |
| 20:27.16 | *** join/#asterisk uqlev (n=uqlev@ykulyev.logos.cy.net) |
| 20:27.36 | *** join/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 20:29.10 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
| 20:40.02 | MrTelephone | does anyone here know how openser handles rpid? |
| 20:40.13 | MrTelephone | because asterisk handles it properly |
| 20:40.30 | MrTelephone | when you set a privacy tag it will mangle the from: header with unkown uknown@ip |
| 20:40.43 | dkwiebe | Interesting, it's definitely something with zap. I set the zaptel module to not load and it works perfectly. |
| 20:40.58 | MrTelephone | dkwiebe what are you trying to do? |
| 20:41.36 | dkwiebe | MrTelephone: This relates to the problem I mentioned earlier. I can talk between phones but the sound files don't give audio |
| 20:42.00 | MrTelephone | try only using ulaw |
| 20:42.30 | MrTelephone | are you reinviting? |
| 20:43.09 | MrTelephone | disallow=all allow=ulaw |
| 20:43.48 | MrTelephone | i spent all morning working on some realtime solutions and they all suckd |
| 20:44.00 | MrTelephone | you might as well just use the conf files |
| 20:45.13 | dkwiebe | MrTelephone: reinvites are turned off and I just tried using only ulaw. |
| 20:45.31 | dkwiebe | MrTelephone: I'll try a different version of the sangoma stuff too. |
| 20:45.51 | MrTelephone | i use sangoma as well |
| 20:46.01 | MrTelephone | what about your phones? |
| 20:46.16 | MrTelephone | are they setup to allow ulaw? |
| 20:47.01 | dkwiebe | yes, wip330 and aastra 480i. They're talking ulaw. In case it was a sip thing I tried zoiper in iax2 mode but it didn't change it. |
| 20:49.17 | MrTelephone | try eyebeam x-lite and see if you get audio |
| 20:49.47 | MrTelephone | did you debug rtp to see if asterisk was sending rtp? |
| 20:50.05 | paulc | a guy I was talking to the other day had some weird problem with audio.. he'd go to voicemail and it would think he wasn't saying anything and timeout after x secs of silence |
| 20:50.13 | paulc | hint was something about a video frame, in the console |
| 20:50.28 | paulc | ended up being some crazy fucked up thing he was doing between 1.4 and 1.2, looping calls round and back again |
| 20:50.47 | paulc | unrelated? probably.. you're doing direct device <--> single server right? |
| 20:51.11 | MrTelephone | does googletalk work or is that another shit option? |
| 20:51.15 | file | waves to paulc |
| 20:51.30 | paulc | chucks a DVD at file |
| 20:51.35 | file | dodges |
| 20:51.41 | gr0mit | MrTelephone, can you explain your prb |
| 20:51.42 | dkwiebe | <PROTECTED> |
| 20:52.10 | gr0mit | or was it dkwiebe ? |
| 20:53.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:53.24 | gr0mit | goedenavond lesouvage |
| 20:53.53 | MrTelephone | no i was just bing an asshole |
| 20:54.05 | MrTelephone | is mad at realtime and MWI |
| 20:54.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:54.15 | gr0mit | hehe! |
| 20:54.20 | MrTelephone | the voip-info wiki about that should be removed |
| 20:54.23 | gr0mit | gave up with realtime peers |
| 20:54.25 | MrTelephone | its the most bogus shit i've seen |
| 20:54.38 | MrTelephone | just make a peer for every mailbox |
| 20:54.50 | lesouvage | Is it possible to prioritise an outbound number in a way that even if all channels are busy, channels are freed and the outbound call can be set up. |
| 20:54.51 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:55.04 | lesouvage | gr0mit: hoi |
| 20:55.30 | gr0mit | i used to live in assen - saw your hostname ;-) |
| 20:56.06 | lesouvage | gr0mit: you see my hostname when I enter the channel? |
| 20:56.10 | gr0mit | yup |
| 20:56.28 | gr0mit | a.assen1.dr.home.nl |
| 20:56.47 | lesouvage | gr0mit: And do miss the anual TT races. |
| 20:56.51 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:57.00 | gr0mit | yup - i remember the TT-weekend |
| 20:57.08 | paulc | I thought that was an Isle of Man thing? |
| 20:57.16 | gr0mit | and Assen |
| 20:57.24 | paulc | who knew? |
| 20:57.28 | paulc | goes off to google Assen |
| 20:57.33 | gr0mit | took my first driving lesson round the assen TT circuit ;-) |
| 20:57.59 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 20:59.28 | lesouvage | gr0mit: I've had a party at the VIP place a couple of month ago. It is really a great place. And the party is real nice. (lots of beer, bands and excitment till dawn) |
| 20:59.45 | lesouvage | VIP place of the TT circuit |
| 20:59.48 | gr0mit | well i lived in Assen in about 1982 |
| 20:59.57 | gr0mit | for 18 months or so |
| 21:00.50 | gr0mit | guess the place has changed beyond recognition |
| 21:01.35 | gr0mit | are you in the centre of assen, lesouvage ? |
| 21:01.43 | lesouvage | paulc: Assen is the only city in the world with a circuit specially build for moter races. This weekend we had Superbikes race with 60.000 visitors. |
| 21:02.06 | lesouvage | gr0mit: yes |
| 21:02.20 | gr0mit | used to live off Beilerstraat |
| 21:02.37 | gr0mit | opposite het Asserbos |
| 21:02.44 | x86 | http://www.boners.com/content/804697.1.jpg <-- my new computer |
| 21:02.57 | lesouvage | gr0mit: just a couple of 100 meters away |
| 21:02.58 | paulc | unrelated, but can anyone recommend a good brand/model of FRS radio? |
| 21:03.24 | lesouvage | paulc: we seem to be in off topic mode |
| 21:03.27 | gr0mit | Taxusplantsoen 6 |
| 21:03.55 | lesouvage | gr0mit:? |
| 21:04.07 | gr0mit | my old address |
| 21:04.26 | gr0mit | porblabyl long since demolishe now |
| 21:04.40 | gr0mit | was an old prefab house with a flat roof! |
| 21:05.20 | lesouvage | gr0mit: they have demolish a lot for the sake of progress |
| 21:05.40 | gr0mit | aah yes, 'progress' |
| 21:12.07 | *** join/#asterisk frenzy (i=user@unaffiliated/frenzy) |
| 21:15.33 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:20.09 | lirakis | x86: "sweet" |
| 21:21.44 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-91.dslextreme.com) |
| 21:22.40 | RobH | Bleh, I do not recall how to do this, how to I turn on colors in the cli? |
| 21:23.16 | RobH | nocolor=no in asterisk.conf |
| 21:23.20 | RobH | but I still have no color |
| 21:32.46 | paulc | I'm laughing at "nocolor=no" - the double negative is silly.. it's like a girl I work with who always codes her features with a disable flag so the default to on.. what's wrong with turning features ON with a flag? |
| 21:35.26 | Guggemand | default should be what you expect is gonna be most used |
| 21:38.33 | paulc | hmm.. depends.. we launch a new feature on our IVR product, we like to configure it off by default so it's out there, ready to be turned on, once it's been deployed.. then the brand team can turn it on when they're ready |
| 21:52.33 | *** join/#asterisk phalcos (i=phalcos@unaffiliated/phalcos) |
| 21:53.08 | lesouvage | What is the english term when key2 on phone 1 has a blinking light when phone 2 is busy (so I can google for more info about this feature) |
| 21:54.38 | _ShrikE | lesouvage: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 21:55.00 | lesouvage | ShrikE: thanks |
| 21:58.01 | lesouvage | ShrikE: what does the abbriviation BLF means? |
| 21:58.06 | paulc | also known as BLF or busy lamp field |
| 21:58.11 | paulc | oh - there you go - good timing :) |
| 21:59.02 | lesouvage | paulc: I don't think ever answered a question that fast. |
| 21:59.19 | paulc | luck of the window flipping :) |
| 22:03.22 | MrTelephone | busy jerking off field |
| 22:09.29 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 22:11.59 | paulc | busy jerking off all over the walls, as my friend Josh once said.. |
| 22:16.33 | MrTelephone | http://blog.tmcnet.com/blog/tom-keating/skype/sip-to-skype-gateway-breaks-skypes-great-wall-of-voip.asp |
| 22:16.36 | MrTelephone | there is one to do it on |
| 22:17.36 | paulc | http://blog.tmcnet.com/blog/tom-keating/skype/sip-to-skype-gateway-breaks-skypes-great-wall-of-voip.asp |
| 22:17.48 | paulc | damn putty and it's right clickedness |
| 22:17.49 | *** join/#asterisk mandd (n=dache@dsl-134-210.aei.ca) |
| 22:17.50 | paulc | sorry :-$ |
| 22:18.36 | mandd | i keed getting http://pastebin.com/m4ca5016e, when trying to install asterisk on freebsd 7 |
| 22:18.55 | mandd | tried everything, updating ports, different versions of openh323 |
| 22:19.02 | mandd | still can't get it to work |
| 22:19.05 | mandd | any ideas? |
| 22:19.30 | RobH | interesting, anytime I get more than a few ougoing calls to teliax, it becomes unreachable... |
| 22:23.42 | MrTelephone | very interesting |
| 22:23.50 | MrTelephone | what is your upstream? |
| 22:32.46 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 22:32.46 | RobH | got it working |
| 22:32.52 | RobH | it seems that i had qualify set |
| 22:33.06 | RobH | and as more calls occured, the latency to the voip provider got higher and higher |
| 22:33.17 | RobH | until * did what qualify tells it to do, and called it unreachable. |
| 22:34.13 | JayTee52 | where do you set qualify? |
| 22:35.39 | paulc | sip.conf |
| 22:35.53 | JayTee52 | thanks |
| 22:38.46 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 22:39.17 | dssman | Anyone know what would cause a: WARNING[2812]: chan_sip.c:12724 handle_response: Remote host can't match request NOTIFY to call '503a10d40b71493844fb1c7c6e9b8123@10.0.164.16'. Giving up. |
| 22:40.49 | RobH | damn cli isnt showing color still, how annoying |
| 22:42.04 | paulc | do you get colours when you do a directory listing? |
| 22:42.44 | RobH | yes |
| 22:48.55 | lesouvage | dssman: cheque your password and username info. I think that causes your problem. |
| 22:50.12 | lesouvage | Is there a way to have a hotdesk logged of automaticaly at a given time |
| 22:52.03 | lesouvage | so when somebody doesn't log off from his flex desk phone callswill not be routed to the phone on the desk the next working day |
| 22:57.11 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 23:12.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:14.16 | paulc | lesouvage: I haven't played with queues in ages but I thought you could auto log agents out if they don't answer calls in X seconds? |
| 23:28.52 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@20150228166.user.veloxzone.com.br) |
| 23:29.16 | [TK]D-Fender | lesouvage, depends how you implemented "hot-desking" |
| 23:35.43 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 23:35.47 | lesouvage | [TK]D-Fender: do you have a link or some proper words to google? |
| 23:36.43 | [TK]D-Fender | lesouvage, can you just clarify what you call "hotdesking" and how it is that you implemented it? Perhaps I can suggest something following that. |
| 23:38.03 | *** join/#asterisk vrtk (n=bruno@201009201011.user.veloxzone.com.br) |
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| 23:39.15 | lesouvage | [TK]D-Fender: It is actualy a demand in a tender paper so I haven't implement de hotdesking yet. What I call hotdesking is when you login on the phone system on any desk and after login the phone works as your phone with all the features of your profile. |
| 23:41.00 | lesouvage | [TK]D-Fender: I have done the logging part before but no one aseked for an automatic logoff. Maybe it doable with a system call using the at command? |
| 23:41.15 | lesouvage | logging=login |
| 23:42.44 | [TK]D-Fender | lesouvage, I'd probably add an entry in a "kick-off" file or DB of some sort with the time, and run a cron process to do this. |
| 23:42.57 | [TK]D-Fender | lesouvage, a lot of relatively easy ways to do that. |
| 23:44.08 | lesouvage | [TK]D-Fender: I got the point, its not that hard as I thought it would be. at is a kind of one time cron job so our ideas fit more or less. Thanks! |
| 23:46.14 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:47.39 | [TK]D-Fender | lesouvage, When would you schedule an auto-kick-out for? |
| 23:49.35 | lesouvage | [TK]D-Fender: it is for use in a flex office. If somebody leaves without lgging out the phone will ring the next day while he/she is on the beach spending aday off. They want the phone to logg off so calls can't be routed on a not available base. |
| 23:50.19 | lesouvage | can't=can |
| 23:50.27 | [TK]D-Fender | lesouvage, So basically just a fixed-hour end of day cleanup? |
| 23:50.36 | lesouvage | yes |
| 23:50.48 | [TK]D-Fender | lesouvage, Oh, thats so much easier.... |
| 23:51.22 | [TK]D-Fender | lesouvage, that tracking in a "relative to login" per-ext basis |
