| 00:12.00 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 00:19.05 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 00:22.55 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 00:23.19 | *** join/#asterisk Gadu (~Gadu@c-71-231-60-73.hsd1.wa.comcast.net) |
| 00:23.54 | dijib | p3nguin, are you here? |
| 00:25.04 | p3nguin | dijib: Yes. |
| 00:25.22 | Gadu | Is asterisk my best bet if I want incoming calls to be forwarded to specific cell phone numbers based on the callers input? |
| 00:25.26 | dijib | ok i have an issue with 2 out of 3 calls with no audio |
| 00:25.35 | Gadu | via a menu or by typing an extension, doesn't matter |
| 00:25.42 | dijib | i believe after that fax script was put in |
| 00:25.44 | p3nguin | gadu: I don't know about "best," but it certainly does that very well. |
| 00:25.53 | Gadu | excellent |
| 00:25.59 | Gadu | now I just gotta go learn how to set that up XD |
| 00:26.23 | Gadu | any recommended links or should I continue to google about? |
| 00:26.51 | p3nguin | gadu: http://pastebin.com/Piqv4Egj Refer to line 59-89. |
| 00:28.40 | Gadu | I have plenty to learn about Asterisk it seems XD |
| 00:30.32 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 00:33.52 | p3nguin | gadu: How are you getting calls and making calls? ITSP, PRI, etc? |
| 00:35.50 | Gadu | Asterisk won't need to make any calls, just receive them and forward them to the selected cell phone. I don't know what method I'll use for the calls to reach Asterisk yet though |
| 00:36.13 | Gadu | I assume I just need a voip account of some kind with a phone number. yes? |
| 00:36.33 | p3nguin | What do you think forwarding a call to a phone is if it isn't making a call to the phone? |
| 00:36.56 | Gadu | oh, you mean that sorry |
| 00:37.27 | Gadu | what voip program do you recommend I use with Asterisk? |
| 00:38.06 | p3nguin | I guess that depends on your definition of voip program. |
| 00:39.13 | Gadu | something like voipbuster but for linux? =P |
| 00:39.21 | p3nguin | Soft phone? |
| 00:39.25 | Gadu | yes |
| 00:39.47 | p3nguin | I like twinkle, but some people complain that it has a Qt3 dependency. |
| 00:40.10 | Gadu | can it be used without a gui? |
| 00:40.24 | p3nguin | I don't think so. |
| 00:40.28 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 00:40.41 | Gadu | hmm, trying to set this up on an ubuntu server that has no gui or even a monitor for that matter |
| 00:40.42 | p3nguin | I can't think of the names of CLI-based SIP soft phones. |
| 00:40.56 | p3nguin | You wouldn't be installing a soft phone on that server anyway. |
| 00:41.08 | Gadu | I wouldn't? |
| 00:41.08 | p3nguin | Asterisk yes. |
| 00:41.26 | p3nguin | It wouldn't make much sense to do so. |
| 00:41.42 | Gadu | I'm afraid I don't understand why that is |
| 00:41.58 | p3nguin | Put asterisk on that server and put the soft phone on your workstation computer. |
| 00:42.36 | Gadu | wouldn't that mean the workstation computer would need to be permanently on? |
| 00:42.55 | p3nguin | Only if you want to be able to receive calls to that phone all the time. |
| 00:43.14 | p3nguin | You don't even have to have a soft phone at all if you don't ever need to make or receive calls. |
| 00:44.14 | Gadu | I only want the soft phone to receive calls for asterisk and for asterisk to be on and able to receive and forward calls at all times |
| 00:44.27 | p3nguin | That doesn't even make sense. |
| 00:44.44 | p3nguin | Asterisk accepts calls all by itself. |
| 00:45.03 | p3nguin | And it'll do with those calls whatever you tell it to do. |
| 00:45.03 | Gadu | so it doesn't need a soft phone? |
| 00:45.24 | p3nguin | A phone is only an interface for you to make and receive calls. If you don't need to do that, you don't need a phone. |
| 00:48.51 | Gadu | does it have a method to obtain a phone number that will reach it or do I use a service like ipkall.com? |
| 00:50.36 | p3nguin | You'll need an ITSP or telco. |
| 00:51.03 | p3nguin | IPkall will give you a free DID, but the number is in Washington state. |
| 00:51.12 | p3nguin | If that's okay with you, that'll work. |
| 00:52.18 | Gadu | works for me. so I won't have to pay for an ITSP? |
| 00:53.37 | p3nguin | IPkall will give you the phone number for free, but you won't have any way to send the call to another phone on the PSTN. |
| 00:54.12 | p3nguin | If you have SIP clients on those mobile phones, then you could do it without PSTN access. |
| 00:55.43 | Gadu | so that free phone number going to a soft phone couldn't be used with Asterisk to forward calls to the cell phones? |
| 00:56.21 | p3nguin | Only if you have a SIP client on the cell phone or if you have PSTN access... but if you have that, there is no use to have the softphone at all. |
| 00:57.13 | p3nguin | A soft phone is just an IP phone, not magic. |
| 00:59.21 | Gadu | for some reason I imagined Asterisk would answer a call that goes to the IP phone on the same machine, then forward that call to a cell phone number |
| 00:59.32 | p3nguin | That's ridiculous. |
| 01:00.01 | p3nguin | Asterisk will answer the call. Or the call will go to the phone, and you can answer it. |
| 01:00.27 | p3nguin | Both conditions together would be unnecessary. |
| 01:01.08 | Gadu | in this case, I need someone to be able to call 3 or more cell phones by calling a single number |
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| 01:01.30 | p3nguin | Do all three cell phones need to ring at the same time? |
| 01:01.35 | Gadu | no |
| 01:01.43 | p3nguin | One, then a second, then a third? |
| 01:01.53 | Korolev | or pick which to ring? |
| 01:01.56 | Gadu | only the cell phone selected by the caller is to ring |
| 01:02.00 | p3nguin | If number one answers, do the others need to ring? |
| 01:02.15 | p3nguin | I see. |
| 01:02.20 | p3nguin | That's easy enough. |
| 01:02.29 | Gadu | sweet |
| 01:02.30 | p3nguin | Asterisk does that quite well. |
| 01:02.33 | Korolev | it is, still looks like he needs a lot of reading |
| 01:03.04 | Gadu | reading I can do, as long as asterisk can do what I want, I will learn ^_^ |
| 01:03.17 | p3nguin | I'd use BackGround() to play a message that says something to the effect of, "For John, press 1. For Alex, press 2." And if you press 1, it dials John's cell number. |
| 01:03.22 | Korolev | yeah, what you need is really simple |
| 01:03.41 | p3nguin | It's but a few lines of dial plan. |
| 01:03.49 | Gadu | I'm incredibly excited now |
| 01:03.55 | Korolev | again, you need some sort of service that will complete your calls into the pstn |
| 01:04.04 | p3nguin | If you're interested in knowing everything there is to know, read The Book. |
| 01:04.06 | p3nguin | ~book |
| 01:04.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 01:04.07 | Korolev | or hardware to do so yourself |
| 01:04.43 | p3nguin | ITSPs are pretty cheap, so I'd probably start with that. |
| 01:05.14 | Gadu | how cheap? |
| 01:05.33 | Korolev | probably about 1c/min for USA |
| 01:05.37 | p3nguin | You can even use your free phone number from IPkall for incoming, and pay only for minutes to the cell phones. |
| 01:05.44 | Korolev | and Canada |
| 01:05.59 | p3nguin | I pay 1.05 cents per minute for outgoing calls to US numbers. Less to Canada numbers. |
| 01:06.09 | Korolev | yeah, canada is far cheaper |
| 01:07.15 | Gadu | so basically, I'd be paying about 1c/min for the calls that asterisk would be forwarding/making? |
| 01:07.26 | p3nguin | yes |
| 01:07.39 | p3nguin | If you use your free number for the calls to asterisk, that is. |
| 01:07.44 | Gadu | yeah |
| 01:07.54 | Korolev | Gadu, thats pretty much what you pay right now |
| 01:08.07 | Korolev | for your cellphone plans anyways |
| 01:08.22 | p3nguin | Get your free number from IPkall, and then refer to lines 67-79 here: http://pastebin.com/tER2jGnY |
| 01:08.29 | Korolev | unlimited usually means about 5000 minutes a month, x 0.01, thats about $50 a month |
| 01:09.31 | p3nguin | I think IPkall will limit you to only two concurrent calls... so if that's a problem, you may want to pay for incoming calls to get more call capacity. |
| 01:09.48 | Gadu | only 1 person will be calling asterisk |
| 01:09.58 | Gadu | in general =P |
| 01:09.59 | p3nguin | Then IPkall should be fine. |
| 01:10.05 | Gadu | cool |
| 01:10.11 | p3nguin | I've used IPkall for several years for free phone numbers. |
| 01:10.50 | Gadu | this is all being done to save money for calls made to a group of us by someone in jail lol |
| 01:11.36 | Korolev | Gadu, dude, then whatever I can do to help, just ask away |
| 01:11.46 | Korolev | that is the best use of asterisk i ever heard off |
| 01:11.52 | Gadu | XD |
| 01:11.53 | p3nguin | If the person can call out to your phone number for cheap, this'll work. |
| 01:12.36 | Korolev | you cant dial a cellphone from jail, mostly, there is credit that you can purchase |
| 01:12.42 | Korolev | its pretty complicated |
| 01:13.03 | p3nguin | If you can only call collect, that's even worse. |
| 01:13.05 | Korolev | and I love speeding, so i know how bad it is to not be able to call anyone |
| 01:13.14 | Korolev | from jail |
| 01:13.50 | Gadu | the person being called has to prepay for the calls and you are charged $6.25 to add money to this prepaid "account" |
| 01:14.24 | Gadu | so to add money to 3-7 phones would end up paying a lot in fees... |
| 01:14.30 | Korolev | yeah |
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| 01:14.42 | p3nguin | It's per number? |
| 01:14.46 | Gadu | yeah |
| 01:14.49 | p3nguin | yuck |
| 01:14.52 | Gadu | a service called PCS |
| 01:15.06 | Korolev | they are making a buck |
| 01:15.15 | Gadu | this will allow us to jointly load a single number |
| 01:15.46 | Gadu | and still allow him to call any of us, and we can add additional numbers at his request whenever |
| 01:15.51 | Gadu | without having to load another phone |
| 01:16.10 | Korolev | actually, well done, you dont even need to add numbers for him |
| 01:16.14 | p3nguin | From what you've said, it's a very simple setup. Just sign up for your free number with IPkall, and register at VoIP.ms and put $25 on your account for making outgoing calls. |
| 01:16.14 | Korolev | he could do it himself |
| 01:16.18 | Gadu | orly? |
| 01:16.43 | Korolev | you need a database for that, but yeah |
| 01:17.07 | p3nguin | I think I'd just hard-code the cell phone numbers. No sense in adding extra steps for every call. |
| 01:17.09 | Gadu | have an sql database if that'll do |
| 01:17.14 | p3nguin | You don't need that. |
| 01:17.16 | Korolev | any database will do |
| 01:17.20 | p3nguin | Asterisk has a built-in database. |
| 01:17.24 | Gadu | ah ok |
| 01:17.25 | p3nguin | But you don't need that either. |
| 01:18.07 | Korolev | or even |
| 01:18.12 | Gadu | if he could punch in a cell phone number for asterisk to call instead of just selecting one from a premade menu, that would give him more flexibility if possible |
| 01:18.16 | Korolev | why would you need to store any numbers? |
| 01:18.16 | Gadu | otherwise the menu style is fine |
| 01:18.22 | p3nguin | It's as easy as "press 1 for John," and exten => 1,1,Dial(SIP/voipms/3145551212); where that is John's cell number. |
| 01:18.29 | Korolev | just give him a prompt: enter th number you wish to call |
| 01:18.40 | Korolev | asterisk even comes with that voice already |
| 01:18.46 | p3nguin | You can give him a password and use DISA. |
| 01:18.47 | Gadu | omg that's amazing |
| 01:18.51 | Korolev | and then read(somevar) |
| 01:18.54 | p3nguin | No |
| 01:19.11 | p3nguin | Just use DISA(). |
| 01:19.27 | p3nguin | After you enter the pass code, it'll give you a dial tone. |
| 01:19.38 | p3nguin | And then you call whoever you want. |
| 01:19.42 | Gadu | O_O |
| 01:19.49 | Gadu | I seriously love you guys right niow |
| 01:19.51 | Gadu | now* |
| 01:22.30 | p3nguin | exten => 2065551212,1,Authenticate(34324); this is the password |
| 01:23.07 | p3nguin | exten => 2065551212,n,DISA(no-password,outgoing-calls); this is where calls go out |
| 01:23.38 | p3nguin | That's an extremely basic implementation, but it'll do what you want. |
| 01:25.46 | p3nguin | Once you have your number and an account with voip.ms, I'll help you set up the rest if you want. |
| 01:26.53 | Gadu | that would be awesome ^_^ |
| 01:29.02 | p3nguin | If you know the phone number(s) he'll call from, you can even tighten it down a bit so that only he can access the authentication prompt. |
| 01:32.10 | Gadu | I do know the number he'll call from in fact |
| 01:37.24 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:38.19 | hypknight | Anone have any insight into skills based routing on Asterisk? I'm looking for insight into a package that's done it, or some idea as to how to successfully build it myself... |
| 01:43.05 | Korolev | I dont know of any packages, actually I had to google skills based routing |
| 01:43.09 | Korolev | because I had no idea |
| 01:43.33 | Korolev | and I tend to reinvent everything myself, but it shouldnt be hard to build |
| 01:44.19 | Korolev | sounds like a couple of tables to me. agents, agents_skillset, agent_cdr |
| 01:44.31 | Korolev | and agent_availability I guess |
| 01:45.08 | Korolev | and for every incoming call, you select against agents available with x skills sorted by last received call, ascending |
| 01:50.18 | hypknight | yea, so I'd have to SQL query in the dialplan or something... |
| 01:50.42 | Korolev | you can use an agi to do that |
| 01:51.38 | hypknight | yea... |
| 01:52.06 | hypknight | gonna have a non-asterisk-guru administering the system... looks like it's going to be a big build |
| 01:52.42 | Korolev | agis are really not a big deal, pretty much any programming language will do |
| 01:53.02 | Korolev | you can have one to query a db in php in no time |
| 01:53.23 | hypknight | oh yea |
| 01:53.30 | hypknight | I mean for the agent skill administration, etc |
| 01:53.55 | *** join/#asterisk i3inary (~i3inary@adsl-71-136-231-186.dsl.sndg02.pacbell.net) |
| 01:54.07 | Korolev | oh |
| 01:55.07 | Korolev | well yeah, if you are going to make it look polished, then you probably need an admin interface, probably an agent interface too so they can update call status |
| 01:55.34 | Korolev | so yeah, you are right, you are probably better off finding a package that already does it :D |
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| 02:07.01 | beta2k | Hello all |
| 02:07.08 | beta2k | Anyone around familliar with setting up a boot server for polycom phones? |
| 02:07.15 | beta2k | There seem to be large holes in their docs :) |
| 02:07.39 | beta2k | Eg, the FW package available on their website doesn't have the files their docs refer to, even the readme in the package! :) |
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| 03:17.26 | dijib | p3nguin, why would my c alls fail o raw |
| 03:17.37 | dijib | svn trunk issues? |
| 03:18.11 | p3nguin | Why would the calls fail... what? |
| 03:18.20 | dijib | did you see my msg earlier? |
| 03:18.34 | p3nguin | Maybe. Regarding what? |
| 03:18.39 | dijib | 2 out of 3 outgoing calls have zero sound. |
| 03:18.49 | p3nguin | I saw that, but I wouldn't have any idea why. |
| 03:18.52 | dijib | what do, do you think? |
| 03:18.58 | p3nguin | Are you still trying to do fax detection? |
| 03:19.17 | dijib | yes sir... but this is outgoing calls only.. all incomming work 100% |
| 03:20.01 | p3nguin | Hmm |
| 03:20.17 | dijib | would you once over my dialplan? |
| 03:20.24 | p3nguin | The last time we spoke, incoming calls had no audio until a phone picked up. Did that ever get solved? |
| 03:21.17 | dijib | yeh i just added the m option to the incomming call so its getting the MOH on line |
| 03:21.21 | dijib | durring that time |
| 03:21.36 | dijib | this issu has still been an issue at that time |
| 03:21.44 | p3nguin | That's not a fix, that's a workaround. |
| 03:21.58 | dijib | ive since got dial by directoryu working. and drank 12 beers tonight |
| 03:22.15 | dijib | ok then. workaround is moh and is good enough |
| 03:23.36 | p3nguin | What happens if you pick up your phone and dial my number? Does it ring while it is calling me? When I answer, can I hear you if you make sounds? Can you hear me if I make sounds? |
| 03:24.09 | dijib | 1 out of 2 times it rings and you can hear me. but 2 out of 3, i dont hear ringing answer or anything |
| 03:24.29 | p3nguin | No audio whatsoever, huh? |
| 03:24.35 | p3nguin | But only sometimes. |
| 03:24.38 | p3nguin | That's very weird. |
| 03:24.45 | p3nguin | Is that box behind NAT? |
| 03:25.00 | dijib | yes surrr |
| 03:25.28 | p3nguin | You've configured all the settings related to proper NAT configuration? |
| 03:25.35 | dijib | and off topic for that transmission are you sure a 4l60e is universal to the gear ratio |
| 03:25.35 | dijib | ? |
| 03:25.39 | dijib | yes sur |
| 03:25.55 | dijib | ill give you /etc/firewall if you want |
| 03:26.01 | dijib | pastebin |
| 03:26.17 | dijib | and i need more beer. and more joints. and fuck you all reading this on the web. |
| 03:27.18 | dijib | is suicide my only option |
| 03:29.15 | p3nguin | To my knowledge, every factory 4L60 has the same gear ratios. |
| 03:29.37 | p3nguin | But even if it doesn't, so what? |
| 03:30.21 | dijib | word up |
| 03:30.47 | dijib | hey penguin can i hear you voice and talk shop since this is an asterisk channel? |
| 03:32.23 | p3nguin | 4L60E gear ratios - 1st 3.059:1, 2nd 1.625:1, 3rd 1.000:1, 4th 0.0696:1, Reverse 2.294:1. |
| 03:32.23 | dijib | ok well hope so. call me on extension 500. just dial as soon as you get menu. |
| 03:32.32 | p3nguin | Not right now. |
| 03:32.35 | dijib | cock |
| 03:32.47 | dijib | then im going to smoke a j and grab another beer |
| 03:33.07 | dijib | oh not ext500, join the confrence im in at ext 8888 |
| 03:33.22 | dijib | be a neighbor |
| 03:33.27 | dijib | back in ten |
| 03:33.39 | dijib | wtf is up with this silence |
| 03:34.53 | p3nguin | Hmm, I see an error in that list of gear ratios. |
| 03:35.14 | p3nguin | 4th is not 0.07, but 0.70 |
| 03:35.32 | p3nguin | 4L60E gear ratios - 1st 3.059:1, 2nd 1.625:1, 3rd 1.000:1, 4th 0.696:1, Reverse 2.294:1. |
| 03:42.25 | p3nguin | And of course there is no contact information on that site and the feedback form is jacked. |
| 03:42.40 | p3nguin | Yay for misinformation on the internet. |
| 03:56.25 | dijib | this is not the 4l60e irc channel sir. |
| 03:56.43 | dijib | just as an fyi, but score on the unlimited 4l60e pawahhhh |
| 03:57.19 | dijib | so if i playtunes a 2600mhz tone sound file, what happends |
| 03:57.39 | dijib | holy shit i think im drinking rat poison |
| 04:02.53 | p3nguin | I guess if you can play a 2600 MHz tone, it would melt your speaker coil. |
| 04:07.24 | dijib | melt my speaker coil ...... i doubt that, im thinking tandul and a possible disconnect? i dont know. im fucking wasted and im going to bed. gnite all. see you another day |
| 04:07.37 | p3nguin | :) |
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| 04:20.11 | Defraz | has anyone seen this error before "tcptls.c: Unable to launch helper thread: Cannot allocate memory" |
| 04:21.11 | Defraz | asterisk just crashes |
| 04:21.16 | Defraz | and then I can't start it |
| 04:21.20 | Defraz | until I kill apache |
| 04:21.21 | dijib | your system full of mem? |
| 04:21.23 | Defraz | on the same server |
| 04:23.01 | Defraz | I wonder if it has something to do with it being on a vm |
| 04:23.16 | p3nguin | Most likely, yes. |
| 04:23.17 | Defraz | but my other vm server hasn't crashed and it is a copy of the same vm |
| 04:24.01 | Defraz | wierd how it is tied into apache |
| 04:24.11 | Defraz | I have apache patched to the latest version |
| 04:29.11 | dijib | how is it tied into apache? |
| 04:29.16 | dijib | i see no mention of this |
| 04:31.00 | Defraz | Well, I try to restart asterisk and it fails until I do a service apache restart |
| 04:31.06 | Defraz | then it will start right up again |
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| 07:26.16 | leftist | morning. does anyone know if there is a vicidial channel anywhere? |
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| 11:39.05 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 11:39.16 | devil_evoxxx | hi all guys |
| 11:39.37 | devil_evoxxx | something use a PRI CARD ( digium ) on a rack server with esx / esxi ? |
| 11:41.32 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
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| 12:56.25 | devil_evoxxx | ehm, something have an idea why with asterisk 1.4 and quescom q401 gateway |
| 12:56.43 | devil_evoxxx | i can call, and upgrading asterisk to 1.8, quescom still saying "Error 503 service unavailable"? |
| 12:56.47 | devil_evoxxx | any idea, suggestion' |
| 12:56.48 | devil_evoxxx | ? |
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| 14:32.11 | *** join/#asterisk punxos (~punxos@149.pool85-60-77.dynamic.orange.es) |
| 14:32.14 | punxos | Hi |
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| 15:01.43 | nutnuts | hi there! has anyone a clue why lcr in the MT_SETUP INDICATION dont get any called_pn? i dont find anything at google or in any config... |
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| 15:13.24 | *** join/#asterisk imox (~imox@p4FC5C560.dip0.t-ipconnect.de) |
| 15:13.57 | imox | hello I need a calling card add-on for asterisk. what is god? |
| 15:14.16 | imox | I want only a add-on not a full software |
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| 15:26.55 | *** join/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld) |
| 15:26.58 | DelphiWorld | hello all |
| 15:27.11 | DelphiWorld | can someone tel me where can i found a firmware for my ST2030? |
| 15:27.13 | DelphiWorld | MGCP no Sip |
| 15:30.26 | ChannelZ | OT: Anyone know if Home Depot sells blue lightbulbs? |
| 15:30.46 | rotten777 | ChannelZ: yessir. LED or CFL? |
| 15:31.09 | ChannelZ | I don't care, it's just for one night |
| 15:31.20 | rotten777 | I believe so |
| 15:31.22 | rotten777 | mine does |
| 15:31.28 | rotten777 | and I live in BFE |
| 15:31.41 | ChannelZ | Cool. I will go look. Figured I'd ask before wasting a trip :) |
| 15:50.43 | imox | someone use a calling card for asterisk? |
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| 16:18.49 | punxos | I have a shevaplug, I can do calls to ext using SPA and GoIP but I can't call other internal extension ??? any idea ??? |
| 16:19.14 | punxos | I installed plugpbx |
| 16:19.16 | p3nguin | Fix the extensions. |
| 16:20.06 | punxos | p3nguin: could are you more accurate ? |
| 16:20.48 | punxos | I am seeing the conf files |
| 16:21.25 | p3nguin | Extensions are configured in extensions.conf. Start by putting the contents of it in pastebin.com so I can see it. |
| 16:21.36 | punxos | ok thanks |
| 16:21.52 | punxos | but maybe I have any wrong in default prefix ... |
| 16:22.07 | punxos | because I can do external calls |
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| 16:22.24 | p3nguin | Let me see your extensions.conf and I'll try to determine why. |
| 16:22.30 | punxos | okis |
| 16:24.17 | irroot | greets folks |
| 16:24.49 | punxos | p3nguin: http://pastebin.com/QtgA8eJm |
| 16:25.13 | p3nguin | Don't ever cat into grep again. |
| 16:26.12 | punxos | I just removed ";" start lines, do you want without grep ? |
| 16:26.33 | p3nguin | No, I just don't want you to ever cat into grep again. |
| 16:27.10 | p3nguin | grep works just fine by itself. |
| 16:27.28 | punxos | grep -v ";" extensions.conf xD |
| 16:27.30 | punxos | maybe ? |
| 16:27.35 | p3nguin | exactly |
| 16:27.37 | punxos | xD |
| 16:27.44 | punxos | yes you are rigth |
| 16:28.13 | punxos | past bad habits |
| 16:28.15 | p3nguin | What is the context assigned to your phone which can make some calls? |
| 16:28.44 | punxos | I don't understand your question |
| 16:28.50 | punxos | I not asterisk expert |
| 16:29.07 | p3nguin | When you configure a phone, which is done in sip.conf, you have to give it a context. What context is given to the phone? |
| 16:32.33 | punxos | p3nguin: http://pastebin.com/V5Gp9pUk |
| 16:32.53 | punxos | from-interal |
| 16:32.59 | punxos | I think |
| 16:33.15 | p3nguin | SIP/505 is the phone that you were having problems with? |
| 16:33.24 | punxos | I have proble with all |
| 16:33.42 | p3nguin | Do they ALL have context=from-internal? |
| 16:33.48 | punxos | Any phone can call to a external number (using a TRUNK) |
| 16:34.01 | p3nguin | You mean using an ITSP. |
| 16:34.10 | punxos | yes all have from-internal |
| 16:34.20 | p3nguin | Okay, let me look at your dial plan now. |
| 16:34.27 | p3nguin | one moment. |
| 16:34.52 | punxos | I'm using spa3102 gateway and any phone can use it perfectly |
| 16:35.16 | punxos | ok |
| 16:35.30 | Korolev | punxos, where is from-internal-custom? |
| 16:35.48 | p3nguin | Give me one example number that you can call which is successful, and one which fails. |
| 16:37.50 | punxos | 91XXXXXXX is ok , 600 is not ok |
| 16:38.00 | punxos | Korolev: mmm |
| 16:38.35 | punxos | Korolev: in extension.conf is defined I think http://pastebin.com/QtgA8eJm |
| 16:40.45 | p3nguin | It's not in that paste. |
| 16:40.53 | punxos | mm |
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| 16:41.21 | punxos | is commented with ";" |
| 16:41.27 | punxos | rigth |
| 16:42.15 | Korolev | punxos, madrid? |
| 16:42.17 | punxos | I have a "from-internal-custom.sample" I should to add ? |
| 16:42.21 | punxos | yes |
| 16:42.27 | Korolev | yeah, add it |
| 16:42.31 | p3nguin | Another way we could go about this is with the dialplan show command on the CLI. Do you want to check things that way? |
| 16:42.44 | p3nguin | You shouldn't use sample files. |
| 16:43.27 | Korolev | put from-internal-custom.sample in a pastebin |
| 16:43.28 | p3nguin | dialplan show 600@from-internal |
| 16:43.34 | Korolev | so we can look at it, too |
| 16:43.43 | punxos | ok |
| 16:44.04 | punxos | http://pastebin.com/Bp5cKXzc |
| 16:45.08 | p3nguin | How about from-internal-additional? |
| 16:45.16 | Korolev | that one too, yeah |
| 16:45.22 | p3nguin | from-internal-custom.sample is incomplete and broken. |
| 16:45.45 | punxos | I'm confused right now |
| 16:45.46 | p3nguin | dialplan show 600@from-internal <---- also this |
| 16:45.56 | Korolev | it will only match 1234 though |
| 16:46.13 | Korolev | you should be, that extensions.conf is confusing :D |
| 16:46.23 | p3nguin | This is why we don't use sample files. |
| 16:46.59 | punxos | I don't used any sample file. I just tell you what files maybe could help me |
| 16:47.21 | p3nguin | If you didn't use a sample file, where did all that macro bullshit come from? |
| 16:47.32 | p3nguin | You certainly did not write it yourself. |
| 16:47.51 | punxos | I used freepbx |
| 16:48.05 | p3nguin | But then you stopped using it? |
| 16:49.04 | p3nguin | dialplan show 600@from-internal <--- still waiting to see this. |
| 16:49.18 | punxos | ok i'm comming |
| 16:52.58 | punxos | http://pastebin.com/N2YigERg |
| 16:53.21 | p3nguin | dialplan show 600@from-internal <--- this. |
| 16:53.27 | punxos | aa |
| 16:53.28 | punxos | sorry |
| 16:54.48 | punxos | http://pastebin.com/VqngUUTd |
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| 16:56.02 | punxos | "I'm sorry that number is not valid" |
| 16:56.14 | punxos | 600 to 505 or 505 to 600 |
| 16:56.28 | ChannelZ | If you feel you have reached this recording in error, please check the number and try again. |
| 16:57.37 | p3nguin | I can't follow that dial plan. |
| 16:57.40 | punxos | I get this message in log http://pastebin.com/wHXxRFnL |
| 16:57.53 | p3nguin | It takes me through at least three macros, and I still didn't find where it dials the device. |
| 16:57.54 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:59.57 | p3nguin | Maybe it happens in macro-dial, but I can't see what it's doing to find the huntmember, huntgroup, and huntloop. |
| 17:00.15 | Korolev | Im still trying to figure out if RT = ringtimer or empty |
| 17:00.30 | p3nguin | There's a dial to the huntmember, but since I don't know what the huntmember is, I can't know if it's the phone or not. |
| 17:00.33 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-34-245.w86-196.abo.wanadoo.fr) |
| 17:00.49 | p3nguin | This is a big part of the reason we don't support FreePBX stuff here. |
| 17:04.02 | irroot | ~freepb |
| 17:04.03 | irroot | ~freepbx |
| 17:04.03 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:04.11 | Korolev | also, we probably need dialparties.agi too |
| 17:04.53 | p3nguin | Or... |
| 17:05.03 | p3nguin | Write your own dialplan and then I'll tell you why it doesn't work. |
| 17:07.00 | punxos | http://pastebin.com/YhyNLAP6 |
| 17:08.25 | Korolev | punxos, did you comment out whatever was inside all those empty ifs? |
| 17:08.55 | punxos | mm I think not |
| 17:09.37 | punxos | not sure |
| 17:09.39 | Korolev | all of a sudden im so glad i never thought of using freepbx :) |
| 17:09.49 | p3nguin | Amen, brother. |
| 17:10.08 | punxos | jeje |
| 17:14.40 | Korolev | seriously, punxos |
| 17:14.51 | Korolev | scratch all that and do as p3nguin said |
| 17:15.16 | Korolev | write your own dialplan, im sure it wont be so hard, given what you need to do |
| 17:15.19 | p3nguin | It doesn't seem like you need a lot of special extensions anyway, so it would take only a short time to write a dial plan. |
| 17:16.01 | punxos | okis thanks you |
| 17:16.07 | Korolev | de nada tio |
| 17:16.20 | punxos | are you spanish ? |
| 17:16.26 | Korolev | no, cuban |
| 17:16.31 | punxos | okis |
| 17:16.33 | punxos | :) |
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| 18:17.10 | ChannelZ | rotten777: Blue lightbulbs procured |
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| 18:32.34 | Lag2 | hello everyone - I am wondering if directmedia is possible with a pap2t behind a dd-wrt router? thank you. |
| 18:33.09 | p3nguin | If you're trying to go through the NAT, don't count on directmedia working. |
| 18:35.07 | Lag2 | ok that is what I thought thank you |
| 18:36.23 | p3nguin | If your call is staying behind the NAT, you can use directmedia. |
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| 19:26.38 | saxa | hi , a question about where to start figureing out, why my phone doesn't hang up the call ? I have a grandstream gxp285 phone which i use with asterisk. When I connect my laptop in the socket of the phone, to use also this possibility of the phone, when I receive a call and I put the handle down, it rings gain advising that the call has not been hanged up, I need to push the EndCall function button on the phone to hang it up. |
| 19:27.01 | saxa | now when the laptop is not connected the phone works ok. |
| 19:27.18 | saxa | any ideas where to start looking ? |
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| 20:47.32 | vezult | good $TIMEOFDAY all. |
| 20:48.16 | vezult | I'm attempting to figure out asterisk by reading through the "getting started" guide for asterisk 1.8. |
| 20:48.30 | p3nguin | ~book |
| 20:48.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 20:48.53 | vezult | unfortunately, I can't seem to register my phone. |
| 20:49.14 | p3nguin | What kind of phone, and what have you done so far to make it register? |
| 20:49.37 | vezult | sorry...I'm working on that :) |
| 20:49.43 | vezult | my sip.conf is here http://dpaste.org/2hOiO/ |
| 20:49.55 | vezult | I've tried using ekiga, and sofsip |
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| 20:50.21 | vezult | ekiga results in an error that I'll post in a sec.. |
| 20:50.36 | vezult | sofsip tells me that the domain is non-local |
| 20:50.42 | vezult | I'll post that as well |
| 20:51.35 | p3nguin | Are you on the LAN with Asterisk? |
| 20:53.30 | vezult | The errors I get are here: http://dpaste.org/7ckAv/ |
| 20:53.36 | p3nguin | I see that your phone's entry has an ACL to only allow from 192.168.1.0/24. |
| 20:53.38 | vezult | p3nguin yes, I'm on the same network |
| 20:53.52 | vezult | the registration attempt is from 192.168.1.119 |
| 20:54.50 | vezult | I'm sure I'm just doing something silly...but I can't seem to spot it yet. |
| 20:54.53 | p3nguin | Just remove the domain line in sip.conf. comment it out or delete it. Then sip reload and try again. |
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| 20:56.37 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:56.42 | vezult | ok, without the domain= line, I get the same error for ekiga as I did for sofsip |
| 20:58.15 | vezult | well, not quite. sofsip says: from is <sip:192.168.1.119:52295>, while for ekiga the error says <sip:test1@192.168.1.1> |
| 20:58.33 | p3nguin | What is asterisk's IP address? |
| 20:58.38 | vezult | 192.168.1.1 |
| 20:59.01 | vezult | both registration attempts are being made from the a host with the IP 192.168.1.119 |
| 20:59.09 | p3nguin | And you have both ekiga and sofsip on 192.168.1.119? |
| 20:59.20 | vezult | p3nguin: yes |
| 20:59.33 | p3nguin | I expect you aren't trying to run them both at the same time. |
| 20:59.57 | p3nguin | I know ekiga works with Asterisk because I've used it, but I am not familiar with sofsip. |
| 21:01.14 | vezult | p3nguin: yeah, I'm not very familiar with sofsip either...I just found that to use as an alternative to ekiga |
| 21:01.18 | vezult | since that wasn't working |
| 21:02.10 | vezult | p3nguin: I am running them both, so sofsip is attempting to register from a different local port |
| 21:02.16 | p3nguin | You removed the domain line in sip.conf, saved the file, and then ran sip reload? |
| 21:02.29 | vezult | if I shut one down and use the other, the behavior doesn't change |
| 21:02.32 | p3nguin | Turn off sofsip. Ekiga will be fine. |
| 21:03.01 | vezult | p3nguin: yes, I removed that line, restarted asterisk |
| 21:03.10 | p3nguin | That's a bit much, but okay. |
| 21:03.55 | p3nguin | And the new error is the same as the old error? |
| 21:04.27 | vezult | I'm pretty sure, but let me try again to be certain. |
| 21:07.37 | p3nguin | In the account settings of ekiga, make sure you filled in the fields correctly. Name is any arbitrary display name. Registrar is the host name or probably the IP address of asterisk. User and Authentication User are both going to be test1 in this case. You know what password is... and timeout of 3600 should be sufficient. Checkmark Enable Account, and press OK. |
| 21:08.08 | vezult | the sip from is different, as I mentioned earlier: sofsip <sip:192.168.1.119:52295>, ekiga <sip:test1@192.168.1.1> |
| 21:08.17 | vezult | otherwise the error is the same |
| 21:09.06 | vezult | I have user/auth user = test1, the registrar is 192.168.1.1, timeout 3600, and password 1234 |
| 21:09.56 | p3nguin | What does %LIMITED mean in that error? |
| 21:10.18 | p3nguin | I'm not familiar with it. |
| 21:10.54 | vezult | p3nguin: I have no idea. I can't find anywhere where that string exists either in the asterisk config files... |
| 21:11.11 | p3nguin | Did you say what asterisk version you're using? |
| 21:11.18 | vezult | and google didn't bring up any hits on asterisk or ekiga source code, that I saw |
| 21:11.25 | vezult | p3nguin: 1.8 |
| 21:11.34 | p3nguin | That's a branch, not a version. |
| 21:11.56 | vezult | however, I don't get that %LIMITED bit now that I removed the domain= line, as you suggested |
| 21:11.58 | p3nguin | core show version |
| 21:12.23 | vezult | 1.8.4.4 |
| 21:12.50 | p3nguin | So what is the new error? Be precise. |
| 21:13.27 | vezult | the new error is quite like the sofsip error except for the difference I described: [Jan 2 18:10:42] NOTICE[2607] chan_sip.c: Registration from '<sip:test1@192.168.1.1>' failed for '192.168.1.119:5060' - Not a local domain |
| 21:14.58 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-34-245.w86-196.abo.wanadoo.fr) |
| 21:15.35 | p3nguin | Change autodomain to no. save, sip reload. |
| 21:16.40 | p3nguin | After you change that, you may end up putting the domain line back in. :( |
| 21:17.45 | p3nguin | I'd first test it without domain and with autodomain set to no. |
| 21:18.25 | vezult | ok, well I got that %LIMITED thing again: http://dpaste.org/K32HZ/ |
| 21:18.28 | vezult | with autodomain off |
| 21:18.47 | vezult | I'll try with that off, and domain=192.168.1.1 again |
| 21:19.51 | vezult | same :( |
| 21:20.10 | *** join/#asterisk Fritz09 (~Fritz09@pop1-469.catv.wtnet.de) |
| 21:20.31 | vezult | I suppose autodomain=yes should just add the local IP to the domains list, so they should be essentially the same thing, right? |
| 21:20.44 | p3nguin | I really don't know what else could cause it. I only manage one 1.8 box, and it is using 1.8.2.2 without autodomain or domain configured in sip.conf. |
| 21:21.09 | p3nguin | I'm a big fan of the 1.4 branch. |
| 21:21.37 | p3nguin | I would think that's what it would do, yes. |
| 21:21.45 | vezult | p3nguin: ok. well, thanks for looking it over for me. |
| 21:21.58 | p3nguin | Stick around a while and maybe more people will be active. |
| 21:22.08 | vezult | will do. thanks! |
| 21:23.56 | p3nguin | I built a 1.8.6.0 package for my own system, which I may test out later today to see if I'm ready to put it into production. |
| 21:24.21 | p3nguin | If I do, I'll look into the autodomain and domain combinations. |
| 21:25.42 | p3nguin | I will say that in my 1.4 systems, I do not configure a domain nor do I set autodomain. I don't know if either has a default value if not configured. |
| 21:27.05 | p3nguin | sip show domains says: SIP Domain support not enabled. |
| 21:27.44 | p3nguin | (on my 1.4.42 system and on the 1.8.2.2 system) |
| 21:35.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:58.00 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 22:08.36 | rotten777 | ChannelZ: feeling blue? |
| 22:23.12 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 22:23.41 | vezult | p3nguin: ok, so registration works now, without specifying a domain or anything. |
| 22:24.24 | vezult | p3nguin: I had a sip proxy set up in ekiga when I was using it with sipxecs |
| 22:24.27 | p3nguin | What else did you have to change to make that work? |
| 22:24.53 | vezult | so I had changed it to point to the asterisk box, then forgot about it. |
| 22:25.05 | vezult | when I removed the sip proxy, registration worked |
| 22:25.15 | p3nguin | I see. |
| 22:27.12 | vezult | other than that, I just got rid of any of the domain, autodomain, etc that I had set in sip.conf |
| 22:27.37 | p3nguin | You can probably go back and add those settings again and it would still work. |
| 22:28.06 | p3nguin | Not that they are really necessary, since I don't use them and everything works fine. |
| 22:29.00 | vezult | Anyway, thanks again for the help. I thought I would just clear up the mystery ;) |
| 22:29.13 | p3nguin | I appreciate the update. |
| 22:44.38 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:52.42 | p3nguin | I just found a typo in asterisk-1.8.6.0/contrib/init.d/rc.archlinux.asterisk |
| 22:52.52 | p3nguin | "Stoping Asterisk..." |
| 22:53.08 | p3nguin | I've never seen something get stoped before. |
| 22:58.24 | *** join/#asterisk lyroy (4a3b7917@gateway/web/freenode/ip.74.59.121.23) |
| 23:00.58 | lyroy | I have an asterisk server and I would like to use it with my family members that already have ata. In my dialout context of my extension.conf i will like to the dial my local users before getting out to pstn. (,1,Dial(SIP/users@myservice.com/${EXTEN})) But the missing part is in my sip.conf . How to be able to create a sip context (myservice.com) where all of my family members will login? |
| 23:01.28 | p3nguin | Don't. |
| 23:01.42 | p3nguin | Create a peer entry in sip.conf for every user who has his own ATA. |
| 23:01.55 | p3nguin | Then Dial(SIP/some-user) |
| 23:03.35 | lyroy | well i would like to be more generic in my dialout context. How a service provider will do it? A peer for every ata and then how do they adress local network dialing? |
| 23:04.07 | p3nguin | If they have ATAs and you have Asterisk, why do you need a service provider? |
| 23:05.44 | *** join/#asterisk salz212 (~chatzilla@182.178.249.81) |
| 23:06.35 | lyroy | I would like to be the service provider of those ATA. They will be registered with my asterisk server, that way local calls will stay in my asterisk network and other calls will be routed to pstn via my pstn lines |
| 23:07.13 | p3nguin | Repeat: Create a peer entry in sip.conf for every user who has his own ATA. |
| 23:09.39 | p3nguin | YAY! I'm running chan_sccp-b and asterisk 1.8.6.0 together! |
| 23:10.09 | Maliuta | p3nguin: why? |
| 23:10.45 | Maliuta | p3nguin: there are better things than sccp :) |
| 23:11.30 | p3nguin | Why? Because I use SCCP phones. I think that's a pretty good reason to use chan_sccp. |
| 23:11.40 | lyroy | and how do you handle local network in dial plan? I know SIP/123456789 will work but is there something more global. Let say I have 2 sip peer (SIP/1112223333 and SIP/1112224444) in my dial out context before trying to get out via my PSTN how can I add only one line to dial those 2 peers? Is there a way to register them in the same context so I will use DIAL(SIP/users@mydomain.com) |
| 23:12.03 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 23:12.06 | p3nguin | You won't use DIAL(SIP/users@mydomain.com) |
| 23:12.08 | p3nguin | at all. |
| 23:12.33 | p3nguin | You'll use Dial(SIP/person1-ata) or Dial(SIP/person2-ata) |
| 23:13.11 | p3nguin | Create an extension for each person you wish to call. |
| 23:13.28 | p3nguin | Put them in a context called "internal" for example. |
| 23:13.46 | p3nguin | Then you will include internal in your phones' context. |
| 23:16.37 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 23:16.42 | scatterp | can any one help me to solve one way calling problems with sipgate? |
| 23:16.54 | lyroy | well i know I can do it that way but i thought I was able to use something like that 1NXXNXXXXXX,1,Dial(SIP/users@myservice.com/${EXTEN}))... Maybe I was draming when I thought it was possible thaat way. Thank you for your time and patience ;) |
| 23:17.20 | p3nguin | No, that's not how you call devices registered to your asterisk. |
| 23:18.10 | p3nguin | That Dial() command dials EXTEN on a host by the name of myservice.com using username users. Completely unnecessary for you case. |
| 23:18.37 | lyroy | lets say you have 100 registered users to your asterisk, in your dialout plan you will dial each of them? |
| 23:19.04 | p3nguin | I'd use an extension pattern. |
| 23:20.03 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 23:22.02 | *** join/#asterisk adeeln (~adeel@184.175.36.92) |
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| 23:44.01 | salz212 | need to know a little bit about .. AMI.. can I use it without telnet. I mean by just connecting a socket to the host..having 5038? |
| 23:44.01 | Nugget | telnet is eeeeeeevil! |
| 23:45.35 | salz212 | ? |