| 00:02.25 | *** join/#asterisk killfill (~killfill@200.63.96.244) |
| 00:02.52 | killfill | hey guys.. im doing requests to asterisk ajam (8088) how do i arrage thing so i can debug in asterisk whats happening?.. i see nothing in aterisk -vvvvvv... |
| 00:03.03 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.08 | *** join/#asterisk Defraz (~t0tal@corp.fuzecore.com) |
| 00:05.05 | jaytee | ~itsp-uslist |
| 00:05.33 | *** join/#asterisk TJNII (~TJNII@207.189.199.58) |
| 00:05.45 | jaytee | ~itsp |
| 00:05.46 | infobot | [~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 |
| 00:06.01 | jaytee | ~itsplist-us |
| 00:06.02 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 00:07.23 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 00:07.28 | Katty | howdy |
| 00:10.57 | Katty | anyone familiar with port 2000 'phonebook'? |
| 00:11.10 | hardwire | yeh :) |
| 00:11.12 | hardwire | it's old skewl |
| 00:11.50 | Katty | heh. i googled it and found Nugget in an old irc log |
| 00:12.10 | Katty | Nugget: ping |
| 00:13.03 | Katty | looks like some sort of cisco thing |
| 00:13.55 | Katty | Skinny |
| 00:14.04 | *** join/#asterisk bobisa (~boboboboo@modemcable065.109-21-96.mc.videotron.ca) |
| 00:14.13 | Katty | yay for google |
| 00:14.16 | Katty | infobot: google |
| 00:14.17 | infobot | extra, extra, read all about it, google is http://lmgtfy.com/?q=google |
| 00:14.37 | *** part/#asterisk killfill (~killfill@200.63.96.244) |
| 00:14.48 | bobisa | does starfish is a good choice for ipbx ? |
| 00:15.22 | Katty | why are you asking that in an asterisk channel |
| 00:15.26 | Nugget | hi katty |
| 00:15.36 | bobisa | starfish is not based on asterisk ? |
| 00:15.52 | Katty | i've never heard of starfish |
| 00:15.56 | Katty | Nugget: how're you deary |
| 00:15.56 | Nugget | either |
| 00:15.57 | bobisa | k |
| 00:16.01 | bobisa | just asking |
| 00:16.01 | Nugget | life is good |
| 00:16.28 | Nugget | back to the track on saturday and the weather's supposed to be doubleplusawesome |
| 00:16.35 | Katty | really? |
| 00:16.39 | Nugget | yup yup |
| 00:16.40 | Katty | good. i didn't want a rainy easter anyway |
| 00:17.06 | Katty | Nugget: telnet |
| 00:17.10 | Katty | not first :< |
| 00:17.18 | Katty | i will have to try again tomorrow |
| 00:17.21 | Nugget | it's rainy there today, though, so that means it'll be muddy. no spins |
| 00:17.29 | Katty | was nice here |
| 00:17.33 | Katty | just got back from a walk. |
| 00:17.39 | Katty | apparently i average 14min 20 seconds per mile |
| 00:18.17 | Nugget | brisk |
| 00:18.21 | Katty | i guess. |
| 00:18.27 | Katty | dunno. it's the pace i always walk with the pup |
| 00:18.36 | Katty | just used my ipod to record it this time |
| 00:18.50 | Katty | still need to time my run |
| 00:18.59 | Nugget | got an iPhone? RunKeeper is awesome. |
| 00:19.09 | Nugget | but it needs gps I think, so it wouldn't work on a touch |
| 00:19.13 | Katty | nah, it's an ipod nano |
| 00:19.31 | Katty | has a little transmitter you put on the shoe, and a reciever into the phone as an attachment |
| 00:19.41 | TJNII | So, last night the power went out. My * server went down with it, as you might expect. Sometime today (I was at work) power was restored and the server came back on. Asterisk seemed to come up okay, except it didn't put the two peers for my ITSP in the right incoming context and they didn't show up in "sip show peers" when I logged in. I restarted Asterisk and now it is fine. Anyone heard of something like that? |
| 00:20.41 | Katty | hun, i've heard of crazier stuff than that with power outages |
| 00:20.51 | Katty | i've had a power outage and my entire xserver took a crap |
| 00:21.12 | TJNII | I need to stop being cheap and buy batteries for the UPS. |
| 00:21.33 | Katty | well linux sure doesn't like to be shut down abruptly |
| 00:21.41 | Katty | if it will save you some stress and headaches, you should do it |
| 00:21.48 | Katty | if for no other reason than your peace of mind |
| 00:22.17 | Katty | mmmkay, shower time for me. |
| 00:22.34 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 00:22.35 | TJNII | I should, the power isn't as clean out here as it was back in Iowa. |
| 00:23.00 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 00:23.42 | TJNII | Though at least the puzzle of why I kept getting milliwatt whenever I called my home phone is solved... |
| 00:25.26 | Katty | hehe |
| 00:25.34 | Katty | i'm glad the stove didn't turn itself on when you called ;) |
| 00:26.14 | TJNII | No, no. The appliances arn't on the network. |
| 00:26.17 | TJNII | .... yet. |
| 00:26.31 | TJNII | (Firmware is still alpha.) |
| 00:27.16 | TJNII | I'm not joking, either. I want the dryer networked into * so it calls me when it is time to change the load. |
| 01:39.38 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:39.38 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 01:39.59 | p3nguin | MeetMe will either keep or pass DTMF, depending on what options you use. |
| 01:44.47 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 01:44.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:47.02 | p3nguin | I'll sure be glad when my new mattress gets here. |
| 01:47.58 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 01:53.41 | rizwank | thanks for option p. |
| 01:53.48 | *** join/#asterisk randomuser (~pete@97-121-222-209.blng.qwest.net) |
| 02:39.39 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:39.39 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 02:40.20 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 02:44.09 | ChannelZ | splat |
| 02:53.09 | *** join/#asterisk OrNix (~ornix@host89-251-107-3.hnet.ru) |
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| 03:11.28 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 03:13.42 | TJNII | Google is making shit up. The text in the search window says "4.25.6 Selecting the serial flow control scheme". I go the page and there is no section 4.2.6, and google's header frame says "These terms only appear in links pointing to this page: serial flow" |
| 03:14.54 | Micc | wow, this is a new one. I got an extension change to state Ringing when nothing is happening. |
| 03:16.25 | Micc | so I guess I can't make a hint have multiple SIP devices and seperate them with & |
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| 03:18.18 | *** part/#asterisk megalomano (~jun-kame@189.191.55.100) |
| 03:23.18 | *** join/#asterisk coppice (~chatzilla@202.62.81.147) |
| 03:29.03 | p3nguin | micc: While extensions don't have states, channels do... and I thought you could hint two channels at a time, but I could be mistaken on that part. |
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| 03:55.45 | *** join/#asterisk javisj (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 03:56.21 | javisj | I jsut installed asterisk I more o less understand but it is a nightmare to configure from scratch ... |
| 03:57.06 | p3nguin | ~book |
| 03:57.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 03:57.07 | ChannelZ | hmm nightmare? |
| 03:57.36 | javisj | not exactly a nightmare actually i am pretty excited about ti because all the possibilities... |
| 03:57.39 | p3nguin | I don't know so much about nightmare, but you can certainly spend a lot of time on it. |
| 03:57.56 | p3nguin | years |
| 03:58.05 | javisj | but that makes me wonder too much. I need to get to the point. I am looking for someone with capabilities of configuring SIP trunk, Skype Trunk, Fax2email, and 5 users. Kind of a freelancer just for the initial setup |
| 03:58.30 | ChannelZ | Skype ain't free. But the rest is not so bad |
| 03:58.37 | javisj | p3nguin: It is a total new world so is amazing all the new things and how excited I get thinking each thing |
| 03:58.40 | ChannelZ | With a basic understanding of what you're doing it doesn't take that long |
| 03:58.41 | p3nguin | How much does the job pay? |
| 03:58.55 | javisj | ChannelZ: I know it is not free I will pay the trunk and for labor of setting all that up |
| 03:59.16 | javisj | I am looking for someone confident with their skills than can give me an estimate cost of this |
| 03:59.34 | javisj | is for a familyhomebusiness |
| 03:59.35 | ChannelZ | oh you want someone else to do this |
| 04:01.21 | javisj | just for the initial setup |
| 04:01.35 | javisj | later on I will be able to mock around with it |
| 04:01.37 | javisj | and learn |
| 04:01.48 | javisj | but I am overwhelmed with information.... |
| 04:02.46 | p3nguin | Do you need it done right this minute? |
| 04:05.11 | javisj | p3nguin: no rush at all |
| 04:05.43 | javisj | p3nguin: time is not a contstraint I just want to learn slowly and I have plenty to read... |
| 04:05.57 | sawgood | In a nutshell, what is the Asterisk Manager account used for? |
| 04:06.30 | p3nguin | javisj: If you start with The Book, you'll be able to configure most of the stuff yourself. |
| 04:06.57 | javisj | p3nguin: i got it so I am going to dive into it see what I can get |
| 04:07.09 | p3nguin | Unless you don't understand any of it, it's not that bad. |
| 04:07.14 | javisj | I am just having some issues with the default config files that come with so much... shit... |
| 04:07.37 | p3nguin | The default configs are ONLY for example/documentation. They are not to be used. |
| 04:07.37 | javisj | I was thinking to use asterisk-gui but dont know if just going for doing it by hand... |
| 04:07.44 | p3nguin | asterisk-gui is dead. |
| 04:07.51 | javisj | ohh i see |
| 04:07.57 | p3nguin | Hasn't been maintained for some time. |
| 04:08.07 | javisj | no worries will not use then |
| 04:09.07 | javisj | so I will start reading the book then |
| 04:09.07 | javisj | so why there is so many config files? |
| 04:09.07 | p3nguin | Asterisk has lots of capability. |
| 04:09.07 | javisj | absolutely but I learnt to keep things simple |
| 04:09.20 | p3nguin | So you won't need to bother with some of them. |
| 04:09.22 | javisj | even that can do EVERYTHING I just need it to do a few things |
| 04:09.35 | javisj | could I delete or moved the ones I dont need? |
| 04:09.55 | p3nguin | You probably could, but I would just leave them alone. |
| 04:10.26 | javisj | hehe having that many files just makes me wonder around heheh... |
| 04:11.01 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 04:11.06 | p3nguin | Are you going straight VoIP, or are you going to connect to the PSTN too? |
| 04:11.17 | javisj | I have PSTN too |
| 04:11.28 | javisj | that is all the cool stuff |
| 04:11.34 | p3nguin | How will you connect your Asterisk system to it? |
| 04:11.44 | javisj | yes will be connected |
| 04:11.58 | p3nguin | The question was "How?" |
| 04:12.11 | javisj | the users are via analog phones |
| 04:12.19 | javisj | Well I have a card installed |
| 04:12.35 | javisj | at the moment I have asterisk installed drivers in place etc |
| 04:13.22 | p3nguin | Do you have a regular residential phone line hooked to the card? |
| 04:13.24 | ChannelZ | Doin' it fo tha shortiesssss |
| 04:15.06 | p3nguin | Or are you going to use a PRI/T1? |
| 04:15.35 | javisj | yes will be hooked to card it is not hooked as of yet |
| 04:15.40 | javisj | just need to plug the cable |
| 04:15.49 | javisj | and will have hooked 4 phones for users |
| 04:15.50 | p3nguin | Yeah, I understand you're going to hook "it" to the card... |
| 04:15.54 | javisj | and 1 fax for sending fax |
| 04:15.59 | p3nguin | I'm asking WHAT will be hooked to the card? |
| 04:16.23 | javisj | 1 residential line, 4 telephones, 1 fax |
| 04:16.24 | p3nguin | Lots of technology available -- you need to be specific. |
| 04:16.34 | javisj | telehpones are normal analogue telephones |
| 04:16.58 | p3nguin | Do you have FXS cards/ports for the phones, or are you going to use ATAs with them? |
| 04:17.21 | javisj | I have FXS ports for the phones and fax and FXO for the input line |
| 04:17.35 | p3nguin | Okay, sounds good so far. |
| 04:17.42 | javisj | then one SIP line innbound/outbound call |
| 04:17.47 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 04:17.52 | javisj | and the skype |
| 04:18.13 | p3nguin | You're going to buy a DID to get calls over the internet in addition to the PSTN? |
| 04:18.35 | p3nguin | or only SIP URI calling over the internet? |
| 04:18.57 | javisj | my telephone company already has a voip phone setup at home. And I can use it with sipphone on computer so I thought that asterisk will be able to use it as well |
| 04:19.22 | p3nguin | oh it will, you're right about that. |
| 04:20.10 | javisj | so at least i have a slight idea of what asterisk is capable of... |
| 04:20.32 | p3nguin | It sounds like you've at least got part of a plan thought up. |
| 04:20.40 | javisj | yes part of the plan |
| 04:20.47 | javisj | even hardware its set |
| 04:20.53 | javisj | and software installed |
| 04:20.57 | p3nguin | You're on the right road. |
| 04:21.03 | javisj | now it comes part to tackle the 105 .conf's! |
| 04:21.05 | javisj | heheh |
| 04:21.14 | javisj | hopefully i can start doing some rm here and there |
| 04:21.18 | javisj | and reduce that large number |
| 04:21.18 | p3nguin | probably only less than 10, actually. |
| 04:21.44 | javisj | that made me smile! with luck in a few days i have this up and running |
| 04:22.51 | p3nguin | sip.conf, extensions.conf, voicemail.conf, queues.conf, dahdi.conf |
| 04:23.16 | p3nguin | Maybe a few more. |
| 04:24.31 | p3nguin | The book is pretty good, so you'll do fine and learn a lot from doing it yourself. |
| 04:25.40 | javisj | yes definetly i wanted myself but infor was so scattered around was frustrating |
| 04:25.44 | p3nguin | On the other hand, if you're in a spot where you need it done right away and don't have the time, I'm sure we can work out something. |
| 04:25.54 | javisj | this book at least has everything in one source so I am doing it muself |
| 04:25.56 | p3nguin | Hang around here and ask good questions. |
| 04:26.04 | p3nguin | ~answers |
| 04:26.05 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 04:26.29 | javisj | no worries ! I will formulate complete questions with all the info just in case |
| 04:26.38 | javisj | Thanks lot for help p3nguin |
| 04:27.30 | p3nguin | If you stay here long enough, you'll see when the most people are active to get more help. |
| 04:28.03 | javisj | good sometimes lot of plataforms users are not active so its hard to get things done |
| 04:28.08 | p3nguin | Is it almost 6:30 PM where you are? |
| 04:28.11 | javisj | so for faxtoemail I can find free solutions? |
| 04:28.30 | p3nguin | Sure, there are some free options. |
| 04:28.42 | javisj | skype I will need to pay i was considering Skype for SIP (form skype) that has an ongoing cost of $5 month or A trunk from digium that is just $66 forever |
| 04:29.00 | javisj | p3nguin: it is around 3pm here |
| 04:29.10 | p3nguin | Ah, I miscalculated. |
| 04:29.10 | javisj | and around 5am where I am setting up asterisk hehe |
| 04:29.41 | p3nguin | I think daytime in north america is when the most people are active. |
| 04:29.43 | javisj | I live in aussie land but help mum back home in Spain where things are not going great at the moment.. |
| 04:29.54 | javisj | so the son needs to do some work from the other side of world |
| 04:30.18 | javisj | oh well i am going to dive into this will share some concerns in the next few days! and probably years! hehehe |
| 04:30.19 | javisj | cheers! |
| 04:30.39 | p3nguin | Read and ask questions... you'll be done in no time. |
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| 05:41.29 | ChannelZ | The wheels on the bus go round and round! |
| 05:47.25 | ChannelZ | and the wheels have fallen off. |
| 05:49.00 | sawgood | ha! |
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| 05:49.28 | sawgood | If I wanted to hire someone to 'help' with my dialplan (to add an 'application') is this called programming? |
| 05:49.50 | ChannelZ | Yes |
| 05:50.13 | sawgood | So, I would like the Asterisk application called ExtenSpy added to my dialplan |
| 05:50.23 | sawgood | Is this the correct terminlogy? |
| 05:51.02 | ChannelZ | well sort of.. I thought you were asking about having a custom application written for you |
| 05:51.29 | sawgood | I think ExtenSpy is an application for Asterisk (which exists already) but is not part of the standard build |
| 05:51.32 | sawgood | Is that right? |
| 05:51.34 | ChannelZ | But yeah you are 'programming the dialplan' so to speak by adding things to make it do stuff |
| 05:51.53 | ChannelZ | Well it's in 1.6.1 anyways. It might be in a module you just don't have loaded |
| 05:52.06 | sawgood | really? ... its not part of 1.6.0? |
| 05:52.37 | sawgood | I have Asterisk 1.6.0.26 ... and I've checked online for modules, and it is not part of that database |
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| 05:52.49 | ChannelZ | I have no idea |
| 05:52.49 | sawgood | maybe it is in another database, or it has to be loaded manually? |
| 05:53.04 | ChannelZ | I'm looking |
| 05:53.52 | ChannelZ | it should be in app_chanspy |
| 05:54.09 | ChannelZ | module load app_chanspy |
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| 05:55.41 | sawgood | ChannelZ: How come I don't see this module when I ask for updates? |
| 05:55.48 | sawgood | Is it only availabe IF you have 1.6.1.x? |
| 05:55.52 | ChannelZ | ask who for what updates? |
| 05:56.07 | sawgood | right ... I am speaking of checking for modules inside of FreePBX |
| 05:56.14 | sawgood | You probably mean from the CLI, right? |
| 05:56.19 | ChannelZ | yeah |
| 05:56.25 | ChannelZ | hell if I know about freepbx |
| 05:56.33 | sawgood | So, you can load modules from the CLI? |
| 05:56.43 | sawgood | How can you 'scan' to see if there are available modules? |
| 05:56.44 | ChannelZ | the asterisk console yes |
| 05:56.59 | sawgood | Interesting ... |
| 05:57.13 | sawgood | Can I get a 'list' of modlues from the Asterisk console? |
| 05:57.25 | ChannelZ | 'module show' which show every loaded module. Some might not be loaded which live in the asterisk lib directory |
| 05:57.47 | sawgood | wow a morse code module |
| 05:57.55 | ChannelZ | yeah |
| 05:58.08 | sawgood | Are the modules in module show already installed in the Asterisk build I have ? |
| 05:58.30 | ChannelZ | Yeah they are actually loaded |
| 05:58.57 | sawgood | dumb question ... but is is 'possible' a module is loaded in Asterisk that FreePBX knows nothing about? |
| 05:59.01 | ChannelZ | As I said there could be more that are not loaded - you typically don't load up a bunch of crap you're not using |
| 05:59.29 | sawgood | module show = what modules are currently installed on your box ... |
| 05:59.35 | ChannelZ | Yeah it's entirely possible, fpbx confines you to doing whatever it is they let you do. |
| 05:59.37 | sawgood | what command shows available modules from online? |
| 05:59.57 | ChannelZ | no, not just 'installed' but ones that are actually currently loaded and resident in the running asterisk |
| 06:00.18 | ChannelZ | There's no command that shows others.. they (generally) live in /var/lib/asterisk/modules |
| 06:00.40 | sawgood | So, how does one 'know' if there are additonal modules which they might want/need |
| 06:00.49 | sawgood | like with yum list |
| 06:01.07 | ChannelZ | Two different things |
| 06:02.07 | ChannelZ | If you are using packaged asterisk, depending on who packaged it they might have split some modules up into different pacakges.. which I have no idea |
| 06:02.32 | ChannelZ | Generally all modules are built with asterisk though with the exception of a couple as part of asterisk-addons |
| 06:02.42 | sawgood | ChannelZ: are 'modules' installed and removed sort of like RPM packages? |
| 06:03.26 | ChannelZ | "modules" in freepbx are actually plugins to freepbx its self |
| 06:03.35 | sawgood | right ... I got that part ... |
| 06:03.41 | sawgood | from the CLI when I do a module show |
| 06:03.48 | ChannelZ | as I said, it depends on who build the package. Maybe they are, maybe they aren't, I don't know. |
| 06:03.51 | sawgood | these have nothing to do with FreePBX, right? |
| 06:03.55 | ChannelZ | right |
| 06:04.18 | sawgood | excellent ... so a module show from the CLI tells me what modules are 'loaded' on my box currently? |
| 06:04.22 | ChannelZ | asterisk is generally compiled with almost everything as a dynamically loadable module, so you can only run the bits you need |
| 06:04.32 | sawgood | I have 156 loaded modules it sayws |
| 06:04.33 | sawgood | says |
| 06:04.46 | ChannelZ | Yeah that's fairly average |
| 06:05.05 | sawgood | I was wondering if there was a 'place' on the Internet where 100's or 1000's of additional modules are located |
| 06:05.14 | sawgood | Like for example ChanSpy and ExtenSpy |
| 06:05.27 | ChannelZ | Chanspy and extenspy are both standard in asterisk |
| 06:05.46 | sawgood | Ok ... since they are 'standard', how come module show does not list them on my system? |
| 06:05.49 | ChannelZ | There are 3rd party modules but I wouldn't say 100's and certainly not 1000s |
| 06:06.10 | ChannelZ | because you probably just don't have it loaded - as I said earlier, type "module load app_chanspy" |
| 06:06.37 | sawgood | ok ... now we are getting closer to my original question ... |
| 06:06.56 | sawgood | since the modules are not loaded ... how to I see a list of non loaded but available modules? |
| 06:07.05 | sawgood | can one do this from the CLI? |
| 06:07.14 | ChannelZ | and I told you that twice too.. they are in /var/lib/asterisk/modules (generally) |
| 06:07.17 | sawgood | or, do you simply 'look' in a directory? |
| 06:07.24 | sawgood | thank you!!! |
| 06:08.12 | sawgood | I don't have a /var/lib/asterisk/modules directory |
| 06:08.12 | Dovid | morning y'all |
| 06:08.16 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 06:09.03 | Dovid | sawgood: try /usr/lib/asterisk/modules/ |
| 06:09.12 | ChannelZ | *now must toss it elsewher |
| 06:09.25 | sawgood | they they are ... |
| 06:09.56 | sawgood | Is there a way with ls to 'output' the number of files in a directory? |
| 06:10.00 | ChannelZ | oh no that is default, I'm just typing crazy shit |
| 06:10.35 | ChannelZ | ls -l |wc -l |
| 06:11.03 | sawgood | 158 in the directory |
| 06:11.39 | sawgood | I think 'two' fiiles exist in every directory to make up the 'folder' .... |
| 06:11.49 | sawgood | so these must be the 156 which are loaded already |
| 06:12.03 | ChannelZ | no -l doesn't show . and .. |
| 06:12.28 | sawgood | well, I guess there is only 2 modules which are not loaded than |
| 06:12.30 | sawgood | maybe? |
| 06:12.44 | ChannelZ | Probably. It doesn't have to be 1:1 |
| 06:17.10 | sawgood | ChannelZ: how does one get a module 'loaded' in Asterisk? |
| 06:17.27 | ChannelZ | module load xxxx |
| 06:17.46 | sawgood | ok cool ... does that module than 'install' and stay as part of your build? |
| 06:17.50 | ChannelZ | and/or you can add it to modules.conf to load (or not) |
| 06:20.34 | ChannelZ | (and actually from earlier, ls -l outputs a total on the first line so the ls -l | wc -l will show +1 |
| 06:21.41 | sawgood | thank you ... |
| 06:21.46 | sawgood | I found modules.conf ... |
| 06:21.51 | sawgood | this is exciting news to me ... |
| 06:22.30 | javisj | sawgood: seems you are as excited as I am! with all this new info... I have enough to dream for a month! |
| 06:22.37 | sawgood | So, when someone wants the Asterisk dialplan to do 'something' it does not do by default ... you would put in the syntax to the file(s) ... and this would point to the module needed |
| 06:22.38 | ChannelZ | I just jizzed in my pants |
| 06:22.40 | sawgood | does that sound right? |
| 06:22.47 | sawgood | me too .. |
| 06:23.08 | *** join/#asterisk Da-Geek (~Da-Geek@85.64.58.187.dynamic.barak-online.net) |
| 06:23.14 | sawgood | Is this what is called 'AGI programming'? |
| 06:23.14 | ChannelZ | no it's sort of backwards |
| 06:23.19 | ChannelZ | no |
| 06:23.50 | javisj | So the modules loaded in asterisk that in my case are 174, I most likely dont use even half of them... how can I tell asterisk to forget about them? So as well I could take out their conf files and make my asterisk cleaner |
| 06:23.55 | ChannelZ | A module offers 'services' to asterisk.. app_*.so are dialplan applications, func_*.so are dialplan functions, format_*.so are file format handlers, etc. |
| 06:24.19 | sawgood | nice |
| 06:24.24 | ChannelZ | javisj: you can 'module unload' them and/or tell them to 'noload' in modules.conf |
| 06:24.43 | sawgood | So, what type of 'thing' would ChanSpy and ExtenSpy be 'called' |
| 06:24.47 | ChannelZ | with autoload turned on, * will generally load just about everything it finds |
| 06:24.54 | ChannelZ | they are dialplan applications |
| 06:25.04 | javisj | what is recommended to load as few as possible right? |
| 06:25.23 | sawgood | Does an application mean something different than a module? |
| 06:25.26 | ChannelZ | well it's a waste of resources having things loaded you're not using |
| 06:25.26 | javisj | is there anything that is essential to asterislk that I should be not unloading? |
| 06:26.08 | ChannelZ | sawgood: yes, a module is a module.. like a DLL on Windows.. it's a shared library of code that can be loaded dynamically by a program - a plugin |
| 06:26.37 | ChannelZ | javisj: yeah quite a few |
| 06:28.00 | javisj | ok I see I will keep reading then, with time I will start understanding whichones I really need |
| 06:28.12 | ChannelZ | sawgood: a dialplan application is a term specific to asterisk. |
| 06:28.16 | sawgood | So, are applications actual files/directories, etcs? |
| 06:28.33 | sawgood | Or is an application a set of 'code' which pulls its information from modules? |
| 06:28.45 | ChannelZ | more or less |
| 06:29.30 | ChannelZ | Rather than 'ChanSpy' being built into the core of asterisk, it's externalized into a dynamic loadable module (.so stands for 'shared object') so that someone who doesn't need it doesn't have to load it |
| 06:29.49 | sawgood | If someone wanted an application for Asterisk ... which did not exist ... would this be the work of a programmer skilled AGI? |
| 06:30.05 | ChannelZ | After it's loaded though it basically becomes a part of the main application. In fact, you can compile asterisk yourself and 'internalize' any modules you want |
| 06:30.51 | ChannelZ | Yes and no, AGI just lets Asterisk execute an external program.. whether it's written in C, PHP, perl, a shell script, etc. |
| 06:32.07 | sawgood | ChannelZ: are you familar with the Asterisk (Digium) support policy ... for sale to the public |
| 06:32.09 | ChannelZ | And that program can interact to a certain degree with asterisk |
| 06:32.18 | ChannelZ | No |
| 06:32.26 | sawgood | Basically, for $595 it covers one server with 2 support tickets for 12 months |
| 06:33.32 | sawgood | steep price, but it might be what I am looking for to help with ExtenSpy ... |
| 06:33.44 | ChannelZ | uhhhh |
| 06:33.50 | ChannelZ | if you say so. |
| 06:34.05 | ChannelZ | Do you have app_chanspy loaded? |
| 06:34.12 | sawgood | I will look now ... brb |
| 06:34.29 | ChannelZ | module show like chanspy |
| 06:35.20 | sawgood | yes, it is loaded |
| 06:35.40 | sawgood | Do you think ExtenSpy is a application module included with Asterisk? |
| 06:35.51 | ChannelZ | ok now we're just going in circles |
| 06:36.21 | ChannelZ | ExtenSpy() is provided by app_chanspy (as is ChanSpy()) so it should already be useable in your dialplan |
| 06:36.27 | sawgood | sorry!!!!!! |
| 06:36.29 | ChannelZ | core show application extenspy |
| 06:36.55 | sawgood | very nice! |
| 06:36.57 | ChannelZ | Now how you actually hook that up into freepbx I have no idea. |
| 06:37.10 | sawgood | I don't care about hooking it up in FreePBX ... |
| 06:37.21 | sawgood | I am going to study how to write it in my dialplan |
| 06:37.30 | sawgood | you've given me the faith I needed |
| 06:38.14 | ChannelZ | well that's easy |
| 06:39.05 | sawgood | ChannelZ: where are you located at? |
| 06:39.17 | ChannelZ | exten => 555,1,ExtenSpy(222) would try to listen to extension 222 in the same context when you dial 555 |
| 06:39.23 | ChannelZ | CO USA |
| 06:39.43 | sawgood | AMAZING!!!! |
| 06:39.50 | sawgood | Are you a Denver Broncos fan? |
| 06:40.07 | ChannelZ | I don't really care about football |
| 06:40.23 | sawgood | So the syntax you gave me ... its that 'simple' to do? |
| 06:40.45 | sawgood | What does the 1 do in the statement? |
| 06:40.59 | ChannelZ | That's the extension priority |
| 06:41.23 | ChannelZ | Every 'step' has a sequential number |
| 06:41.37 | ChannelZ | ~book |
| 06:41.38 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:41.40 | *** join/#asterisk AeroCloud (~aero@ip72-222-149-220.ph.ph.cox.net) |
| 06:41.41 | sawgood | sometimes I see a 's' in these statements (near the front of the syntax) ... what does that do? |
| 06:41.43 | ChannelZ | You should really read that |
| 06:41.47 | sawgood | ok |
| 06:41.58 | sawgood | I have it actually |
| 06:42.04 | AeroCloud | is there an asterisk command to hangup the callee of a bridged channel? |
| 06:42.11 | ChannelZ | s is a special extension which is the absence of an extension |
| 06:42.18 | ChannelZ | AeroCloud: soft hangup |
| 06:42.19 | *** join/#asterisk diegomad (mad@186.0.4.182) |
| 06:42.20 | AeroCloud | only the callee.. |
| 06:42.49 | ChannelZ | So only give it their channel |
| 06:43.07 | ChannelZ | But I dunno who the other person will be left talking to either way.. |
| 06:43.20 | AeroCloud | example: you call person b.. after talking for a bit.. you want to hangup on b and call c |
| 06:43.26 | AeroCloud | without redialing |
| 06:43.35 | ChannelZ | huh? |
| 06:43.53 | AeroCloud | this is incoming did's |
| 06:43.58 | ChannelZ | Is it supposed to read your mind and dial c by its self? |
| 06:43.58 | AeroCloud | like calling cards |
| 06:44.02 | AeroCloud | no.. |
| 06:44.06 | AeroCloud | it would send back to ivr |
| 06:44.08 | AeroCloud | to dial |
| 06:44.16 | AeroCloud | I just cant get it to only hangup on b |
| 06:44.23 | AeroCloud | it hangs up both sides.. or noone |
| 06:44.39 | AeroCloud | if I blind transfer to a bad channel.. it makes the channel hang |
| 06:44.45 | ChannelZ | you can use the 'h' flag in Dial() |
| 06:45.00 | ChannelZ | and then make it continue in the dialplan, and program accordingly |
| 06:45.04 | ChannelZ | core show application Dial |
| 06:45.09 | AeroCloud | I was unsure of that |
| 06:45.16 | AeroCloud | cause it says * will hangup the caller |
| 06:45.26 | AeroCloud | not the callee |
| 06:45.29 | ChannelZ | (and actually I think you want H not h) |
| 06:45.40 | ChannelZ | read a little |
| 06:45.42 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 06:46.05 | AeroCloud | # H: Allow the caller to hang up by dialing * |
| 06:46.24 | AeroCloud | that does not say, hangup the callee by pressing * |
| 06:46.33 | AeroCloud | I will try it.. thanx for the info |
| 06:48.07 | ChannelZ | see F |
| 06:49.05 | ChannelZ | or you can use 'g' and just depend on the other end to hang up |
| 06:49.38 | ChannelZ | I'm not positive if H/h terminate the channel immediately regardless if you are using F (or g), never tried |
| 06:50.37 | sawgood | exten => s,n,ExtenSpy(${EXTENSION}@from-internal,b) |
| 06:50.45 | sawgood | What does the s do in this statement? |
| 06:50.57 | ChannelZ | s is the extension |
| 06:51.13 | ChannelZ | which means 'no extension'. |
| 06:51.48 | ChannelZ | IE on a POTS line, my incoming calls go into an 'incoming' context and the dialplan starts on the 's' extension when someone calls.. because there is no extension on an incoming call in that case |
| 06:52.32 | sawgood | I see 's' in a lot of dialplan syntax |
| 06:52.40 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 06:52.45 | sawgood | for example: exten => s,1,Playback(vm-extension) |
| 06:53.28 | ChannelZ | yeah.. if you are creating a bunch of IVRs in separate contexts it's common to see the 's' extension used for each context. |
| 06:54.46 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:58.18 | ChannelZ | AeroCloud: I just tried it, using gH I can press * and continue on in the dialplan (which in your case would Goto or something to jump back to your IVR letting people make a call or whatnot) |
| 06:58.24 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:00.32 | AeroCloud | ok cool |
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| 07:04.06 | ChannelZ | goes to poo and shower |
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| 07:15.11 | AeroCloud | well that kinda works the way I wanted it |
| 07:15.55 | AeroCloud | sleeptime |
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| 07:21.51 | ChannelZ | kinda.. |
| 07:26.01 | *** join/#asterisk troubled (~troubled@unaffiliated/troubled) |
| 07:29.53 | AeroCloud | it hung me up too |
| 07:30.01 | AeroCloud | :) |
| 07:30.42 | ChannelZ | hmm does your dialplan actually do something after the Dial? |
| 07:32.04 | AeroCloud | yes |
| 07:32.36 | AeroCloud | this is the line right after dial |
| 07:32.37 | AeroCloud | exten => _X.,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE}) |
| 07:32.48 | AeroCloud | I wanted to see what it would return |
| 07:32.59 | AeroCloud | it did not do the NoOp, just hung me up |
| 07:38.15 | AeroCloud | next test.. was different result |
| 07:38.29 | AeroCloud | first time.. it displayed 0 as hangupcause |
| 07:38.35 | AeroCloud | and continued on |
| 07:38.43 | AeroCloud | then 2nd time.. it hung me up |
| 07:39.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:40.53 | ChannelZ | hmm |
| 07:42.55 | AeroCloud | I waited 3 more seconds the 2nd time |
| 07:43.05 | AeroCloud | first time I did it within 2 seconds of pickup |
| 07:43.34 | AeroCloud | oh well, going to bed now.. will work on this tomorrow... |
| 07:46.40 | ChannelZ | WTF |
| 07:46.56 | ChannelZ | if I cp -R '/home/server/Archives/archive_store/store_20081212/DS MXF' '/home/server/Archives/outbox' |
| 07:47.29 | ChannelZ | it tells me that it cannot stat a file in the DS MXF dir, permission denied. Yet I can manually stat it, look at it, see it, copy it... |
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| 08:38.52 | Polysics | hello |
| 08:38.58 | Polysics | two different tasks at hand |
| 08:39.22 | Polysics | 1 - store SIP users' status in a MySQL table (online/busy/offline) |
| 08:39.43 | Polysics | 2 - users have different "languages", create a queue for each language and stick calls into it |
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| 08:40.07 | Polysics | i have a running EventMachine service to handle FastAGI and AMI |
| 08:40.11 | Polysics | jsut need some pointers |
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| 08:55.33 | Polysics | hello ManxPower |
| 08:55.50 | Polysics | i need a few pointers :-) |
| 08:56.19 | Polysics | first, what is the best way to store users' status in a table? i need online/busy/offline, nothing more |
| 08:56.42 | Polysics | i have an AMI client logging events around |
| 08:56.54 | Polysics | users will be mainly reached through queues |
| 08:57.09 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 08:57.21 | Polysics | i also need to put users in the proper language queue when they log in, but i suppose that's something i can do on login |
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| 09:02.30 | javisj | before I been told that asterisk-gui is old... so it is recommended do the config no-gui? or is there any gui I could install that is simple and just manages asterisk? |
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| 09:25.57 | tuxx- | hey guys, when im in a conference with another person (meetme), is it possible to transfer the other user to some extension? When i try it, the person im trying to redirect the call to doesnt hear anything, and the person in the meetme im trying to transfer is still in the meetme on his own. |
| 09:26.19 | tuxx- | maybe its better to use callparking with out switchboard instead of a meetme |
| 09:26.41 | tuxx- | the only thing were stuck on now is the transfer from a meetme, anyone got a clue? :) |
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| 09:46.34 | tamiel | tuxx-: res_features don't work in app_meetme |
| 09:46.56 | tamiel | tuxx-: res_features only work in bridged situation |
| 09:48.08 | tamiel | (when channel is bridged with another one, not the bridge new api from 1.6.2) |
| 09:56.09 | tuxx- | mkay |
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| 10:17.39 | adnc | the disposition field of cdr records in the database are with numbers like: ANSWERED=8, NO ANSWER=4, BUSY=2, FAILED=1 but i also have some entries with 0 as value. what does 0 refer to? |
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| 12:06.01 | ManxPower-work | ~answers |
| 12:06.02 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:09.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:10.11 | *** join/#asterisk jkroon (~jkroon@dsl-244-28-03.telkomadsl.co.za) |
| 12:10.44 | jkroon | hi guys, having an issue connecting a Digium Quad BRI (all ports in NT mode) to another Quad BRI (all in TE mode). |
| 12:11.46 | jkroon | signalling is set to respecitively bri_net (NT) and bri_cpe (TE), however, NT side (pri show spans) reports "Provisioned, Down, Active" whilst TE side reports "Provisioned, Up, Active" |
| 12:12.01 | jkroon | any ideas on what I can do to figure this out and get it to work? |
| 12:14.03 | russellb | contact http://www.digium.com/en/supportcenter/ |
| 12:14.26 | jkroon | I've been through that before. |
| 12:15.05 | jkroon | on a few occasions. I've yet to actually receive a reply on any of them. |
| 12:17.41 | russellb | wow, seriously? |
| 12:17.49 | russellb | that is certainly not right |
| 12:17.55 | russellb | try calling |
| 12:18.21 | ManxPower-work | russellb, try contacting Digium support sometime via e-mail from an account that they would not recognize. |
| 12:18.23 | russellb | or if you get me your contact info, i'll walk down there myself later today and make sure someone calls you :-) |
| 12:18.35 | ManxPower-work | "secret shopper" |
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| 12:21.35 | jkroon | russellb, pvt? |
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| 12:22.39 | jkroon | i lie. i got a reply to issue #WEI-945710 |
| 12:22.47 | jkroon | but it never got resolved. |
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| 12:36.10 | roe | I am using a snom370, I realize that my issue is probably a phone specific issue and thus this might not be the correct place. On speaker phone the output of the speaker seems to change drastically if the microphone detects noise (the speaker gets lower) |
| 12:36.56 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 12:38.00 | kombi | after a year of faultless operation suddenly I get "Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) What could have happened? |
| 12:43.43 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:45.28 | tzafrir | kombi: missing digits? |
| 12:45.46 | kombi | tzafrir: how would I tell? |
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| 12:46.13 | tzafrir | kombi, is it analog? PRI? |
| 12:46.22 | kombi | yip, B410P |
| 12:46.53 | tzafrir | kombi, one thing to try would be to use 'pri debug span NN' |
| 12:47.20 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 12:47.35 | pentanol | hi every1 |
| 12:47.36 | kombi | Sending Set Asynchronous Balanced Mode Extended |
| 12:47.53 | pentanol | I've got warning on conference server Maximum retries exceeded on transmission szguunqpdlbpltu@laptop for seqno 536 (Critical Response) -- See doc/sip-retransmit.txt |
| 12:48.10 | pentanol | conferencing kind of works, but may I filtering this? |
| 12:48.44 | pentanol | I've readed this document, thefore I decided to ask here.... |
| 12:49.02 | kombi | tzafrir: that gives the same error (and uhm,.. how do I stop the polling?..;) |
| 12:49.18 | tzafrir | kombi, if you only see those, you don't even have layer 2 up |
| 12:49.24 | tzafrir | do incoming calls work? |
| 12:49.33 | kombi | jeez.. no they don't |
| 12:49.37 | tzafrir | Does an outgoing call work immediately after an incoming call? |
| 12:49.49 | kombi | let me try.. |
| 12:50.35 | kombi | nope.. doesn't... |
| 12:52.01 | kombi | tzafrir: can I tell whether layer 2 is up with dahdi show status? |
| 12:52.50 | Faustov | dahdi show channels shows it |
| 12:53.27 | kombi | says "In Service".. weired.. |
| 12:54.20 | tzafrir | kombi, what signalling have you set for it? |
| 12:54.20 | kombi | pri show span 1 gives "Status: Provisioned, Down, Active" (whatever that means) |
| 12:55.03 | kombi | tzafrir: where is it set? |
| 12:55.24 | kombi | never mind.. fxo_ls |
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| 12:56.05 | tzafrir | kombi, that can't be |
| 12:56.19 | tzafrir | it has to be bri_<something> |
| 12:56.50 | kombi | tzafrir: I'll best pastebin my configs... |
| 12:57.32 | *** join/#asterisk sroth (~57eaca08@gateway/web/freenode/x-eurvyhpgezubamyh) |
| 12:58.24 | kombi | http://pastebin.se/200852 |
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| 12:58.45 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:59.30 | tzafrir | kombi, can you include dahdi-channels.conf as well ? |
| 12:59.45 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:59.56 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:00.06 | tzafrir | also: is that connection PtP or PtMP? |
| 13:00.39 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 13:01.21 | kombi | http://pastebin.se/200853 ptp/ptmp where do I tell again? |
| 13:02.10 | kombi | bri_cpe_ptmp it says in dahdi-channels.conf |
| 13:03.39 | *** join/#asterisk ManxPower-work (~manxpower@139.sub-75-234-63.myvzw.com) |
| 13:03.44 | tzafrir | is it a connection you can connect a standard ISDN phone to? |
| 13:04.21 | kombi | must be ptp, but inclusion of dahdi-channels.conf is commented out |
| 13:04.28 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 13:04.53 | tzafrir | If so: it's ptmp (point to multi-point - allow connecting multiple phones on the same line) |
| 13:05.25 | tzafrir | change it back to _ptmp and use: dahdi restart |
| 13:05.33 | tzafrir | in the starisk CLI |
| 13:05.41 | tzafrir | or restart asterisk |
| 13:06.04 | kombi | tzafrir: but it already says ptmp in dahdi-channels.conf, no? |
| 13:06.47 | kombi | sorry, actually it says pri_cpe there.. |
| 13:07.07 | kombi | file unchanged for some nine month.. |
| 13:07.22 | kombi | ok, I'll try that |
| 13:07.39 | tzafrir | kombi, ptp is mostly used in business-grade connections |
| 13:07.55 | tzafrir | (it's simpler and more reliable, but you can't connect ISDN phones to it) |
| 13:08.36 | kombi | that is actually kind of what we are, the B410P is used to connect to PSTN and our line is PTP |
| 13:09.01 | kombi | do you want to see the output of dahdi restart? |
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| 13:10.25 | kombi | http://pastebin.se/200854 |
| 13:12.20 | kombi | tzafrir: I'll do an ultra hard reboot including disconnecting power now... |
| 13:17.44 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 13:18.38 | kombi | tzafrir: when doing dahdi restart, which configs are read exactly? |
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| 13:27.29 | tzafrir | kombi, configs of chan_dahdi.so . That is: basically chan_dahdi.conf (but also users.conf . You don't use it, right?) |
| 13:27.43 | jkroon | tzafrir, if I have two digium BRI cards and I connect an NT port to a TE port - do I just set signalling to bri_net on the NT side and bri_cpe on the TE side? |
| 13:28.35 | tzafrir | jkroon, IIRC, yes |
| 13:29.00 | jkroon | i was hoping you're going to tell me i'm being stupid. |
| 13:29.12 | tzafrir | speaking of BRI cards: anybody has any missing PCI IDs in that driver? |
| 13:29.20 | jkroon | TE side says link is up, NT side says to bugger off. |
| 13:29.43 | jkroon | using a normal straight cable. both sides has a green light up. |
| 13:30.05 | *** join/#asterisk dinesh___ (~dinesh@84-73-120-175.dclient.hispeed.ch) |
| 13:30.32 | dinesh___ | hi all, I have a kind of general question: Are phone number prefix-free? (i.e. there is no such numbers such that one is prefix of another) |
| 13:31.09 | jkroon | dinesh___, you can do whatever you want there. |
| 13:31.34 | dinesh___ | yup, but I mean real phone numbers you can find |
| 13:32.03 | dinesh___ | i've never seen a number prefix of another |
| 13:32.27 | dinesh___ | perhaps there are even limitations regarding this on older hardware |
| 13:32.35 | [TK]D-Fender | dinesh___: What prefix? |
| 13:32.52 | jkroon | he means like my number is 786 and your is 7865, then my number is a prefix of yours. |
| 13:32.52 | dinesh___ | let's say 911 is a valid number, and then another valid number would be 911453210 |
| 13:33.12 | dinesh___ | yes |
| 13:33.14 | jkroon | there is such problems with very, very old analog exchanges. |
| 13:33.21 | jkroon | those that still used linefinders. |
| 13:34.18 | [TK]D-Fender | dinesh___: Generally you won't get multiple length patttern overruns like that |
| 13:35.10 | jkroon | nor would i recommend configuring them, even though you can if you wanted to. |
| 13:36.27 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:36.30 | dinesh___ | ok thanks ;) |
| 13:37.13 | dinesh___ | i'm super happy, since today I have no analog phone line anymore (saves 25 USD/month), and my old number got ported to an internet-based SIP provider, and it's all working ^^ |
| 13:37.51 | dinesh___ | i got still have to optimize on the asterisk side, like dialing "9"[number] would force to dial through SIP provider 2 instead of 1 |
| 13:39.12 | dinesh___ | (but the linksys wireless adapter for their sip ATA was total crap, I had to remove it) |
| 13:39.49 | dinesh___ | for some reason it gets a signal strenght of just 30% when still in the same room than the access point .. |
| 13:40.05 | tuxx- | oi guys, when using the ReceiveFax application, the dialplan doesnt continue, it exitst after receivefax has been executed. Does anyone know how this is possible? |
| 13:40.52 | Chainsaw | tuxx-: You do your cleanup in the h priority. |
| 13:40.54 | jkroon | yes. ReceiveFax will terminate when the channel gets hung up. |
| 13:41.26 | ManxPower-work | tuxx-, correct. When the channel hangs up the dialplan will jump to the "h" extension. If there is no "h" extension, the dialplan terminates. |
| 13:41.41 | tuxx- | thanks! :) |
| 13:41.54 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:43.34 | jkroon | tzafrir, this looks weird: http://pastebin.co.za/97548 - garqi 1-4 and c3po 1-2 is PRI and works, what messes with my head is the BRI links (all the others). garqi 5-8 is connected to 9-12 and 13-16 to c3po 3-6 ... all the TE ports are Up, and their connected NTs is down. |
| 13:43.41 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:45.22 | tzafrir | jkroon, "garqi"? huh? |
| 13:45.40 | jkroon | machine names, look at the pastebin. |
| 13:45.53 | jkroon | it's with those BRI cards we mentioned earlier. |
| 13:46.07 | tzafrir | oh, ok. using a search engine helps |
| 13:47.15 | jkroon | i've tried that. |
| 13:47.28 | jkroon | either i don't know what to feed google or i'm looking at my eyelids. |
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| 13:50.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.16 | *** join/#asterisk jaytee (~465bd509@gateway/web/freenode/x-uxbssulrmlouoaiu) |
| 13:51.22 | sroth | hi, i have a question. During a conference i want to start an agi script when '*' ist pressed. The AGI Script is a DTMF menu und make some mysql selects. I search since 2 days to solve the problem. But i can't find a way to start a agi script by pressing a key. I only find features.conf, but this only work wenn i use DIal to a local Channel before i get in to the conference room. |
| 13:52.22 | jaytee | anyone have a recommendation for a ITSP providing cheap 800 Toll Free inbound that services the state of Indiana? |
| 13:52.51 | [TK]D-Fender | jaytee: I don't recall 800 #'s being state specific. |
| 13:53.15 | [TK]D-Fender | jaytee: Or ITSP's that care where you are |
| 13:53.33 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:53.33 | pentanol | anybody here use opensips\ser kamailio so as to load balance client between asterisk servers? |
| 13:54.31 | tzafrir | jkroon, is it ptp or ptmp? |
| 13:54.36 | jkroon | ptp |
| 13:54.43 | jkroon | bri_net and bri_cpe |
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| 13:55.25 | tzafrir | what do you see in 'pri debug' on one of those spans? |
| 13:55.42 | tzafrir | actually: intense debug (or debug level 2) |
| 13:57.04 | Skeeter- | does anyone recommend sipp to test the capacity of an asterisk system? |
| 13:57.08 | jkroon | would it confuse you to see them on the same machine or would you prefer I use one of the cross-linked spans? |
| 13:59.15 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 13:59.18 | Katty | hi |
| 14:01.45 | Skeeter- | Katty |
| 14:02.43 | Katty | so who wants to hold my hand with a pg_restore this morning |
| 14:02.55 | Katty | i just wanna learn how to do it |
| 14:02.59 | Katty | nothing is broken :P |
| 14:06.24 | jkroon | tzafrir, i have no idea how to interpret this: http://pastebin.co.za/97638 |
| 14:06.54 | jkroon | from my very much non-existent experience it looks like TE receives stuff from NT but not the other way round ? |
| 14:07.11 | tzafrir | HDLC Abort (6) on Primary D-channel of span 16: something wrong in the D-channel data |
| 14:08.00 | Katty | i recently swapped an A101 and an A101D around |
| 14:08.07 | Katty | without reconfiguring anything |
| 14:08.14 | Katty | and spazed about the D channel |
| 14:08.18 | Katty | just throwin that out there |
| 14:08.35 | jkroon | ok ... how do I figure out what? |
| 14:08.50 | tzafrir | A101D is a A101 with an extra D channel? |
| 14:09.04 | Katty | i believe D is some sort of additional echo module |
| 14:09.15 | Katty | it's basically the same card, but with echo cancelation |
| 14:09.18 | Katty | and a bigger price tag |
| 14:09.38 | Katty | the folk at sangoma had me swap them around for testing purposes, said no reconfiguring was required |
| 14:09.51 | jkroon | these don't have echo cans on them afaik but I can quickly rip open the case to check. putting the cards in TE mode gets them to link with my providers though. |
| 14:09.55 | Katty | but i did notice that for some unusually odd and seemingly unrelated reason, the D channel constantly spazed. |
| 14:09.58 | tzafrir | jkroon, don't know. Contact support? :-( |
| 14:10.11 | jkroon | dropped greg an email. |
| 14:10.20 | jkroon | support doesn't respond. |
| 14:10.30 | Katty | probably because they've been helping me ;) |
| 14:10.38 | Katty | let me get you another contact email |
| 14:10.55 | Katty | jkroon: -> |
| 14:11.03 | jkroon | also seeing some strange congestion (34) issues when trying to route with pri. |
| 14:11.55 | Katty | i've also been having congestion issues |
| 14:12.02 | Katty | but, that's obviously related to snickerdoodled D channel |
| 14:12.42 | jkroon | well, I am basically plugging a PRI on a NT configured channel straight back into a TE and looping my calls 30 times through that. |
| 14:12.48 | jkroon | still, not expecting it to go belly up. |
| 14:13.05 | Katty | i don't suppose you have another quad bri card |
| 14:13.08 | Katty | to swap out and test |
| 14:13.19 | jkroon | i've got 4. two of them in NT mode, two in TE mode. |
| 14:13.25 | jkroon | both do exactly the same thing. |
| 14:13.28 | Katty | nods |
| 14:13.32 | jkroon | so I'm quite sure it's software. |
| 14:13.38 | Katty | certain sounds like it |
| 14:13.54 | jkroon | probably some config issue... |
| 14:14.09 | Katty | anyone from i9 technologies lurking this morning |
| 14:14.19 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 14:14.28 | Katty | dont' make me call you! |
| 14:14.28 | tzafrir | jkroon, what do you see with pri intense debug (on both sides)? |
| 14:14.48 | jkroon | tzafrir, see the pastebin @ http://pastebin.co.za/97638 |
| 14:14.58 | jkroon | oh wait, for the PRI with 34 ... ? |
| 14:15.54 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:16.05 | tzafrir | c3po sends a SABME. It gets no SABME from the other side |
| 14:16.14 | Katty | how do i see the timestamp on a particular file |
| 14:16.29 | tzafrir | garqi gets a SABME and properly responds with a UA |
| 14:16.40 | tzafrir | That UA likewise does not get to the other side |
| 14:17.07 | tzafrir | So c3po can send to garqi, but not the other way around |
| 14:17.20 | *** join/#asterisk giesen (giesen@dirtypackets.net) |
| 14:17.23 | jkroon | ok, both sides link with no alarms (both says OK). |
| 14:17.33 | jkroon | protocol error or config? how do I check the config? |
| 14:17.35 | tzafrir | I vaguely recall some bug in that driver that had that symptom |
| 14:17.53 | jkroon | dahdi 2.2.0.2 ?? |
| 14:17.53 | tzafrir | But then again, I figure the support people would be more familiar with it |
| 14:18.00 | jkroon | let's hope. |
| 14:18.03 | tzafrir | trying a later version might help |
| 14:18.10 | Katty | Qwell: ping |
| 14:18.26 | Katty | file: YOU |
| 14:21.56 | Skeeter- | i need something to test the capacity of simlutanous call of my PBX, how can i check it out? |
| 14:22.42 | Naikrovek | Skeeter-: make a call, put it on hold. make another call, put it on hold. make another call, put it on hold |
| 14:22.49 | Naikrovek | Skeeter-: what phone do you have |
| 14:23.02 | Katty | hi Naikrovek |
| 14:23.08 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:23.08 | Naikrovek | hi katty |
| 14:23.33 | Skeeter- | Naikrovek, polycoms |
| 14:23.37 | *** join/#asterisk shader (~user@nom26990d.nomadic.ncsu.edu) |
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| 14:23.42 | *** part/#asterisk Malky (~oliv@lon92-1-82-67-212-172.fbx.proxad.net) |
| 14:23.58 | Naikrovek | Skeeter-: if you have recent models you can get up to 4 calls going per line key |
| 14:24.16 | Naikrovek | make a call, put it on hold. repeat |
| 14:24.34 | Katty | call a meetme, put it on hold. repeat |
| 14:24.54 | Katty | most fun i've had all week, annohying the crap out of a whole room of people with my on hold music |
| 14:24.56 | Skeeter- | that sounds not professional |
| 14:25.01 | Naikrovek | Skeeter-: or you can use SIPp to do that kind of thing i think |
| 14:25.13 | Naikrovek | Katty: lol yeah |
| 14:25.20 | Skeeter- | SIPp is weird |
| 14:25.41 | Katty | i need to learn how to do a pg_restore |
| 14:25.41 | Naikrovek | Skeeter-: if you want to test how many hundreds of calls you can have on your box, that i don't know how to do. i'm sure there's a way though |
| 14:25.46 | tzafrir | Skeeter-, use another Asterisk, on a stronger system :-) |
| 14:26.00 | Katty | maybe Kobaz will help me later. |
| 14:26.08 | tzafrir | sipp is also known to help |
| 14:26.08 | Skeeter- | i just wanna test the capacity |
| 14:26.21 | Skeeter- | its running on a dual CPU xeon quad with 8gb of ram |
| 14:26.25 | tzafrir | Skeeter-, are you familiar with 'originate' ? |
| 14:26.42 | tzafrir | MEmory is probably not the bottleneck. CPU power may be |
| 14:27.35 | Skeeter- | tzafrir, no |
| 14:28.12 | Skeeter- | im getting 100% cpu with 12 calls |
| 14:28.15 | Skeeter- | impossible |
| 14:28.31 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:28.58 | Naikrovek | Skeeter-: what!? something's up |
| 14:29.09 | Naikrovek | i know of no codec that would do taht |
| 14:29.21 | Katty | wav |
| 14:29.34 | Katty | i have no idea |
| 14:30.35 | *** join/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 14:32.34 | Katty | anyone ever heard about a network card turning itself off after a period of time? i have to ifdown and ifup againf or it to work |
| 14:32.47 | Katty | happens with the onboard, so i added another nic. it does the same thing |
| 14:33.06 | devoid | Katty: Power management? |
| 14:33.33 | beek | Hi Katty. I was going to suggest a BIOS 'sleep' setting, but with the inserted NIC? Nevermind... |
| 14:33.49 | Skeeter- | Katty, which distro, do you have a GUI? |
| 14:34.09 | Naikrovek | Katty: i also suspect power management. |
| 14:34.09 | Katty | i just xset -dpms |
| 14:34.12 | Katty | we'll see what happens |
| 14:34.19 | Katty | i'll check bios too |
| 14:34.27 | Katty | Skeeter-: it's debian, and yes it is running xserver |
| 14:34.32 | Skeeter- | voila |
| 14:34.35 | Skeeter- | network manager |
| 14:34.38 | Katty | oh? |
| 14:34.39 | Skeeter- | the new one does that |
| 14:34.43 | Katty | bummer |
| 14:34.46 | Skeeter- | i got it on my laptop |
| 14:34.50 | tzafrir | dpms is for the screen only, right? |
| 14:34.51 | Skeeter- | deibna squeeze |
| 14:34.52 | Katty | how do i uhh |
| 14:35.08 | Katty | politely tell it to stop screwing up my connecton |
| 14:35.13 | Skeeter- | i modprobethe b43 driver and bring it back |
| 14:35.19 | Skeeter- | seems the only work around |
| 14:35.22 | Katty | good lord |
| 14:35.34 | Katty | is it croned? |
| 14:35.44 | Skeeter- | croned? |
| 14:35.46 | Katty | cron job |
| 14:35.46 | Skeeter- | ?croned |
| 14:35.50 | Skeeter- | no |
| 14:35.51 | Katty | crontab |
| 14:35.57 | Skeeter- | but |
| 14:36.23 | Skeeter- | u could make a pidof that check every minute if the netowrk is up , if not issue ifdown and up |
| 14:36.39 | Katty | yeah but the phones will still go down |
| 14:36.42 | Katty | right? |
| 14:36.49 | Katty | at least at night they would |
| 14:36.57 | Katty | what if i just kill xserver |
| 14:37.17 | Katty | tho i'm not quite sure how to do that |
| 14:37.38 | Katty | maybe a croned ping |
| 14:37.54 | Skeeter- | ur asterisk has a desktop/GUI? |
| 14:38.24 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:38.35 | Katty | Skeeter-: it's there in case someone needs to reboot the system |
| 14:38.42 | Katty | Skeeter-: i don't want them tinkering about the terminal |
| 14:38.46 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 14:38.50 | Katty | Skeeter-: but i can deal with it |
| 14:39.33 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 14:39.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.35 | Skeeter- | Katty, i dont recommend you GUI on an asterisk server for CPU peaks and less process uses too, but that just me |
| 14:39.38 | Skeeter- | reboot system? |
| 14:40.19 | Katty | looks like i'm running at rc2 |
| 14:40.29 | Katty | runlevel2, i mean |
| 14:41.05 | Skeeter- | i dont know what that means |
| 14:41.22 | Katty | one of the run levels executes the gui |
| 14:41.24 | p3nguin | katty: Seriously, you're giving someone far too much control by giving them console access. Use ssh and only give them access to the reboot or shutdown stuff. |
| 14:41.40 | Katty | p3nguin: i'm not giving them any console acess, that's the point. |
| 14:41.40 | [TK]D-Fender | Skeeter-: Hire a system admin. One who will also make sure you're not running FreePBX like yuo do |
| 14:41.48 | p3nguin | katty: You just said you were. |
| 14:41.56 | Katty | p3nguin: the gui is there simply for them to click Actions and restart |
| 14:42.10 | p3nguin | katty: The physical box is the console. You're giving them console access. |
| 14:43.04 | p3nguin | And runlevel 2 in Debian has squat to do with Xorg running. |
| 14:43.22 | p3nguin | 2-5 should all be exactly the same. |
| 14:43.56 | p3nguin | Xorg runs because of gdm being told to start as a service. |
| 14:44.10 | *** join/#asterisk dajhorn (~dajhorn@206.16.96.160) |
| 14:44.32 | *** join/#asterisk patrb (~asdf@64-150-178-3.kansascity.abac.net) |
| 14:44.36 | patrb | 'ello 'ello |
| 14:44.38 | Katty | p3nguin: how would you recommend keeping gdm from running then? |
| 14:44.56 | devoid | p3nguin: delete /etc/init.d/rc2.d/Sxxgdm (or something like that) |
| 14:45.13 | Katty | k |
| 14:45.31 | p3nguin | I would use whatever tool debian wants me to use, such as the update-rc.d thing or dkpg-reconfigure. |
| 14:46.02 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:46.05 | wcselby | o/ |
| 14:46.25 | Katty | hi wcselby |
| 14:46.31 | p3nguin | I also wouldn't have a monitor or a keyboard on the server, since that only facilitates console logins (which should never be allowed). |
| 14:47.06 | p3nguin | As a matter of fact, gdm and xorg shouldn't even be installed at all. |
| 14:47.39 | wcselby | console logons are sometimes the only that can save you if you have issues with network connectivity |
| 14:47.56 | wcselby | but hey, I jumped into the middle of a conversation here |
| 14:47.57 | p3nguin | And that's when you grab a monitor and keyboard and go to the SECURED server room. |
| 14:47.59 | wcselby | so ignore me :) |
| 14:48.09 | Katty | k, i found the debian services manager, and unchecked gdm (= |
| 14:48.10 | wcselby | i was about to say, I'd have them all in a secure server room |
| 14:48.27 | wcselby | that only authorized people had access to |
| 14:48.32 | Skeeter- | [TK]D-Fender, whats your point? |
| 14:48.56 | Katty | Skeeter-: i'll see what happens. |
| 14:49.02 | p3nguin | skeeter-: It seemed more like a suggested command rather than a point. |
| 14:49.09 | Katty | Skeeter-: hopefully with gdm not running, that silly network manager thing will not be an issue |
| 14:49.28 | Skeeter- | p3nguin, then im sorry |
| 14:49.33 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 14:49.34 | Skeeter- | i didnt mean it |
| 14:49.39 | p3nguin | Network Mangler is one of the first thing that normally gets removed. |
| 14:50.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:51.07 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:51.31 | [TK]D-Fender | Skeeter-: Point is its loaded up with AGI's and other monitoring processes that bog down your capacity. Then again I don't recall you stating any goals you are looking to acheive. |
| 14:52.25 | Skeeter- | [TK]D-Fender, are you referin to Katty's problem if not what are you refering to? |
| 14:53.42 | [TK]D-Fender | Skeeter-: You want to test capacity. I'm listing things working against yours |
| 14:54.00 | Katty | katty has no problems |
| 14:54.02 | Skeeter- | [TK]D-Fender, oh that, i gave up, thanks for you time tho |
| 14:54.21 | Skeeter- | Katty, Issue*? |
| 14:54.23 | Katty | i'm very healthy! |
| 14:54.33 | Katty | Skeeter-: that's just an inconvience |
| 14:54.53 | *** join/#asterisk moy (~chatzilla@74.12.121.207) |
| 14:55.02 | Katty | Skeeter-: a problem would be like...me breaking a leg |
| 14:55.17 | Katty | Skeeter-: gotta take it all in stride |
| 14:55.41 | *** join/#asterisk McBoingbo (~Galabaga@mail.hrsg.ca) |
| 14:56.14 | McBoingbo | Guys, how can I add (if I can) multiple extensions to this line "exten => 1,1,Goto(hrsg-extensions,228,1)"? |
| 14:56.42 | McBoingbo | exten => 1,1,Goto(hrsg-extensions,SIP/262&SIP/238,1)? |
| 14:57.40 | [TK]D-Fender | McBoingbo: You can't Goto a DEVICE |
| 14:57.48 | [TK]D-Fender | mcgGoto = jump to EXTENSION |
| 14:57.52 | p3nguin | He thinks phones are extensions. |
| 14:58.01 | McBoingbo | you know what I mean |
| 14:58.05 | p3nguin | Do we? |
| 14:58.13 | Kobaz | you know what he means |
| 14:58.21 | p3nguin | We really only know what you show us or tell us. |
| 14:58.26 | [TK]D-Fender | McBoingbo: Goto = jump to line in dialplan. Not "make these devices ring. That would be a DIAL |
| 14:58.35 | Kobaz | just explain that extensions are bits of dialplan, and devices are phones... and quit pretending :P |
| 15:00.39 | Katty | what's the name of the voicemail module |
| 15:00.51 | Katty | voicemailsomethingsomething.so |
| 15:00.57 | [TK]D-Fender | Katty: app_voicemail.so |
| 15:01.05 | Katty | ohah |
| 15:01.07 | McBoingbo | so with GOTO, you can not send to multiple extensions then? |
| 15:01.32 | p3nguin | I usually just run "module show like voice" to find it. |
| 15:01.33 | [TK]D-Fender | McBoingbo: You jump to AN extension. SIP/262 is NOT an extension. |
| 15:01.35 | Kobaz | McBoingbo: sending a call to multiple devices requires Dial() |
| 15:01.53 | p3nguin | mcboingbo: With Goto, you can only Go To another extension. |
| 15:02.06 | p3nguin | mcboingbo: Phones are NOT extensions. |
| 15:02.19 | p3nguin | mcboingbo: You cannot Goto() phone devices. |
| 15:02.24 | McBoingbo | so unless I made an extension that dialed several DEVICES :P, I need to use dial |
| 15:02.29 | Kobaz | [TK]D-Fender: in the regular world, 262 on your desk would be an extension... in the asterisk world... an extension is a bit of code in extensions.conf telling the system how to handle when someone dials a number |
| 15:02.33 | p3nguin | mcboingbo: exactly |
| 15:03.03 | [TK]D-Fender | [11:02]<McBoingbo>so unless I made an extension that dialed several DEVICES :P, I need to use dial <- hmmm... |
| 15:03.07 | Kobaz | dial(SIP/123&SIP/124&SIP/125) |
| 15:03.09 | [TK]D-Fender | [10:58]<[TK]D-Fender>McBoingbo: Goto = jump to line in dialplan. Not "make these devices ring. That would be a DIAL |
| 15:03.11 | Kobaz | will dial three phones |
| 15:03.16 | McBoingbo | Can you mix parameters like GOTO on priority 1 and Dial on priority 2? |
| 15:03.21 | p3nguin | kobaz: My phone can be reached by pressing in extension 262... that does not make my phone extension 262. |
| 15:03.27 | Kobaz | p3nguin: sure it does |
| 15:03.33 | p3nguin | no, it doesn't. |
| 15:03.42 | Kobaz | p3nguin: to anyone who isn't a phone guru... that makes perfect sense |
| 15:03.47 | [TK]D-Fender | McBoingbo: Priority 2 won't get executed... you are GOTO-ing AWAY from it |
| 15:03.58 | p3nguin | kobaz: That's just how people rationalize it. |
| 15:04.05 | p3nguin | kobaz: They see it logically rather than literally. |
| 15:04.25 | McBoingbo | is that the same case for Dial for the first line then a goto next? |
| 15:04.27 | Kobaz | p3nguin: sure in an avaya system i can set up exten 2000 to ring 20 phones |
| 15:04.28 | Skeeter- | [TK]D-Fender, do you give any asterisk training? |
| 15:04.33 | Kobaz | but the rest of the world doesn't think like that |
| 15:04.34 | p3nguin | kobaz: They think, "Oh, I call 262 to ring that phone, so that phone must be extension 262," but they are wrong. |
| 15:04.39 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:05.06 | [TK]D-Fender | Skeeter-: I don't have a formal class, but have consulted for specific training. |
| 15:05.16 | Kobaz | people see 262 on their desk... they are sitting at extension 262 |
| 15:05.20 | Kobaz | no ifs ands or butts |
| 15:05.27 | Kobaz | (to the lay person) |
| 15:05.30 | Skeeter- | [TK]D-Fender, ok, contact infos? |
| 15:05.35 | jkroon | ok ... this is a weird one, setting up span 4 of my quad pri in net mode and looping it back to one of the cpe ports I can't make any calls over that. |
| 15:05.38 | p3nguin | kobaz: You're trying to rationalize it just like other people do. |
| 15:06.08 | jkroon | however, just moving the exact same config to a different port (and obviously adjusting the numbers accordingly) it works sweet. patlooptest isn't picking up any problems either. |
| 15:06.11 | McBoingbo | so when setting up a menu system for your frontend (reception, etc) GOTO is ideal because it GOTO's-away, but if I used DIAL for the menu choices, does that create an issue? |
| 15:06.30 | Kobaz | p3nguin: i'm giving you the perfectly good reason why you should accept that people think of extensions the way they do... and it'll make it less frusterating for people asking questions |
| 15:06.54 | p3nguin | kobaz: If my phone is found internally by SIP/0011-fe32-6ab1, that's the phones ID... not any of the other ways the phone can be rang. |
| 15:07.12 | Kobaz | i'm not arguing over how it works |
| 15:07.19 | McBoingbo | ladies, ladies |
| 15:07.23 | McBoingbo | Im over it |
| 15:07.40 | Kobaz | i'm saying if you guys quit saying "huh? extension?".... and instead just explain what the difference is |
| 15:07.47 | p3nguin | If 262 causes it to ring, that doesn't make the phone extension 262 -- it means there is an exten 262 that rings my device. |
| 15:07.51 | Kobaz | it will be easier on everyone, and it will make us look less arrogant |
| 15:08.00 | McBoingbo | lol |
| 15:08.12 | p3nguin | From the IVR, 2 also rings my device. That doesn't make my phone into extension 2. |
| 15:08.16 | Kobaz | p3nguin: I KNOW THIS |
| 15:08.22 | p3nguin | okay then |
| 15:08.29 | Kobaz | p3nguin: i'm saying that people who aren't phone gurus... do not know this |
| 15:08.41 | p3nguin | Stop trying to justify the words of those who don't know and should be learning. |
| 15:08.51 | p3nguin | If they need to learn it, teach them. |
| 15:08.53 | Kobaz | i'm not justifying anything, you're the one who said that |
| 15:08.58 | Kobaz | anyways |
| 15:09.00 | Kobaz | my only point was |
| 15:09.00 | McBoingbo | oh my god, I will never call 238 an extension EVER again |
| 15:09.05 | Kobaz | be easy on the new people :P |
| 15:09.31 | McBoingbo | I was about to hang myself with my shoelaces p3nguin, and thats on your sarcastic ass |
| 15:09.41 | Kobaz | heh |
| 15:09.51 | Naikrovek | McBoingbo: you're not the only person that feels that way. many have /ignore'd him |
| 15:09.56 | p3nguin | hahahaha |
| 15:10.11 | p3nguin | My feelings are DEEPLY hurt. |
| 15:10.19 | Naikrovek | suuure they are |
| 15:10.22 | Kobaz | hehe |
| 15:10.26 | Naikrovek | you relish it |
| 15:10.27 | wcselby | lol |
| 15:10.32 | Kobaz | it's like... you guys get offended when someone calls sip/123 an extension |
| 15:10.35 | Katty | yes he does. |
| 15:10.37 | Kobaz | who just started using asterisk a week ago |
| 15:10.38 | wcselby | McBoingbo - that's not really how IVRs work |
| 15:10.44 | McBoingbo | hey p3nguin, go fly...errr ohhh sorry p3nguins cant fly.... /BOOT |
| 15:11.00 | Katty | but i think anyone around here is looking for any reason to go off on someone |
| 15:11.05 | Katty | something in the air i say |
| 15:11.06 | Kobaz | Katty: hehe |
| 15:11.16 | wcselby | you setup a menu that accepts DTMF, then create new extensions in your extensions.conf in the same context that have actions based on the number entered |
| 15:11.17 | p3nguin | I'd rather let them know that phones are not extensions. |
| 15:11.21 | Kobaz | i vote for a happier atmosphere |
| 15:11.27 | Katty | i second the motion |
| 15:11.31 | Kobaz | p3nguin: yeah, exactly... |
| 15:11.48 | McBoingbo | ok so back to my original question, if I change GOTO to dial for both my menu choice priorities 1 and 2 will it get borked? |
| 15:11.49 | bmoraca_work | oh for the love of god, not this bullshit again |
| 15:12.01 | Katty | bmoraca_work: welcome to #asterisk! |
| 15:12.03 | Katty | bmoraca_work: ;P |
| 15:12.13 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:12.20 | Kobaz | McBoingbo: menu option 1... Dial(foo) menu option 2... dial(Bar) |
| 15:12.20 | bmoraca_work | there's a place for pedants. IT is not one of them. |
| 15:12.40 | Katty | 3 System(eject!) |
| 15:12.56 | McBoingbo | Kobaz: thanks, I was concerned that GOTO properly moves away from the context and DIAL didnt, might not make much sense, but want to be safe |
| 15:12.59 | Kobaz | McBoingbo: keep in mind... asterisk can only process one line of dialplan at a time... so if you're dial()ing... it's not doing *anything* else, until it's finished dial()ing |
| 15:13.00 | Katty | Skeeter-: so far so good |
| 15:13.22 | Skeeter- | Katty, nice |
| 15:13.24 | Kobaz | McBoingbo: well... one like of dialplan at a time... per call |
| 15:13.33 | Kobaz | McBoingbo: you can be running multiple calls at the same time etc |
| 15:13.40 | Kobaz | s/like/line |
| 15:13.40 | Katty | Skeeter-: this calls for tea |
| 15:13.57 | p3nguin | You can certainly Dial() more than one device at a time, though. |
| 15:13.58 | Kobaz | yeap |
| 15:14.04 | wcselby | McBoingbo - i'll try to show you a snippet of an IVR / menu / whatever you want to call it |
| 15:14.19 | p3nguin | You just can't Goto() more than one extension at a time. |
| 15:14.19 | McBoingbo | I just thought menu choices are typically better to use GOTO, if it is then I will create an extension that points to 2 devices, then GOTO it |
| 15:14.32 | Kobaz | McBoingbo: sooo. in conclusion... if on line 1, you goto... and then on line 2 you dial... well.. that dial wont get hit, since your doing your goto first |
| 15:14.41 | Kobaz | sure |
| 15:14.55 | Kobaz | make extension 2000, or whatever... and then it dials sip/123 and sip/124 |
| 15:14.57 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 15:14.59 | McBoingbo | its a menu system, I dont want anymore choices, once a choice is made, go away |
| 15:15.17 | McBoingbo | yeah prolly the best way to go, keeps it generic too |
| 15:15.19 | p3nguin | 1,1,Goto(sales,steve,1) |
| 15:15.48 | p3nguin | Then [sales] has exten => steve,1,Dial(SIP/steve,30) |
| 15:16.14 | p3nguin | Dialplan logic isn't too difficult. |
| 15:16.31 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 15:17.22 | guax | Hi folks. Got a simple scenario but have a problem with assisted xfers. My notebook as server, asterisk 1.4.29. Three users, 2 ip phones 1 sofphone. In any association when A calls B and B xfer to C. C answers the call the call ends and B goes back to A, clean log, no hangup. Asterisk just sends a BYE. Its a local network completely isolated. |
| 15:17.40 | guax | first i tought a nat related but disabled nat for everyone and nothing |
| 15:18.00 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 15:19.49 | wcselby | guax - please provide us with console output and a sip debug (assuming these are sip phones) |
| 15:20.23 | guax | ok |
| 15:22.22 | guax | wcselby, http://pastebin.com/BqegrV6m |
| 15:22.31 | l2trace99 | is there away to see the pri signalling w/ dahdi ? |
| 15:22.49 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:23.24 | l2trace99 | or at least to see the channels in use ? |
| 15:23.46 | leifmadsen | guax: try 1.4.30 as a bunch of transfer issues have been resolved recently |
| 15:24.57 | guax | leifmadsen, ok. |
| 15:27.08 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:31.12 | guax | downloading on a dead slow conection =/ |
| 15:32.38 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 15:35.18 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:37.39 | tzafrir | l2trace99, what do you mean? see where? |
| 15:38.25 | l2trace99 | in the cli |
| 15:39.01 | l2trace99 | or even in dahdi_tools |
| 15:39.27 | *** join/#asterisk farkus (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 15:40.24 | jkroon | tzafrir, dahdi 2.2.1 - same problem. |
| 15:41.17 | tzafrir | l2trace99, you can check the signalling of spans |
| 15:41.42 | tzafrir | the signalling of channels would show "PRI" or something similar, IIRC |
| 15:41.54 | tzafrir | (in 'dahdi show channel NN') |
| 15:42.01 | tzafrir | what do you need it for? |
| 15:43.52 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 15:44.26 | l2trace99 | dahdi show channel [channel #] doesn't show any thing different from a channel inuse and a channel not in use |
| 15:44.44 | l2trace99 | unless I am missing something |
| 15:45.37 | guax | building |
| 15:48.15 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 15:49.25 | michael-i | Hi everyone. I'm having trouble getting transfers working on my analog phone. When I use threewaycalling=yes and transfer=yes in chan_dahdi.conf, pressing the flash key initiates a three way call. I just want transfers to work so I removed threewaycalling and now it no longer reacts to the flash key. |
| 15:50.08 | michael-i | I hear a tone (not dtmf) but nothing happens. Nothing is logged by asterisk either. (using 1.6.1.18) |
| 15:52.02 | guax | leifmadsen, its solved, thank you |
| 15:52.10 | leifmadsen | thanks for testing :) |
| 15:52.43 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 15:53.27 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:54.35 | guax | leifmadsen, it solved my other problem with xfers as well. im officially happy now. |
| 15:54.41 | leifmadsen | w00t :) |
| 15:55.35 | guax | that make me a CDR away of perfection, aeuhuhe, will keep developing my awesome AMI listener of doom |
| 15:57.26 | kombi | has 1.6.2 changed in terms of variables in the dialplan? I use _XXX to tell internal calls from outside calls but that doesn't seem to work in my newly installed 1.6.2 |
| 15:58.39 | guax | kombi, be more specific, you mean _XXX are not matching with your internals? |
| 15:59.00 | bmoraca_work | is t38 support markedly superior in 1.6.2.6 vs. 1.6.2.0? |
| 16:01.17 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:01.22 | kombi | quax: our internals are one or two digits, so everything longer than that goes out. Now, when one dials outside the first digit is used to call an inside phone instead. |
| 16:02.06 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 16:02.47 | guax | kombi, dialplan on pastebin please |
| 16:03.08 | kombi | quax: coming up quax;) |
| 16:03.43 | *** join/#asterisk Z_God (~julius@wlan231217.mobiel.utwente.nl) |
| 16:04.59 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:05.12 | kombi | quax: http://pastebin.se/200857 |
| 16:05.29 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 16:08.08 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 16:08.22 | guax | kombi, first of all, your code is a mess =P |
| 16:08.30 | kombi | quax: thanks..;) |
| 16:09.02 | kombi | outgoing calls are dealt with in lines 30 and 31... |
| 16:09.29 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 16:09.50 | guax | ok, dialplan reload and asterisk log please |
| 16:09.57 | BCS-Satori | I just wanted to check if "limitonpeers" still a valid command in sip.conf for Asterisk 1.6.2.6? |
| 16:10.02 | E-bola | they arent if what ur saying is true |
| 16:12.05 | AeroCloud | From asterisk Dialplan & bridged call, is it possible to execute a command while on the call to disconnect ONLY the callee, not the original channel? |
| 16:12.25 | leifmadsen | BCS-Satori: look at the sip.conf.sample for 1.6.2 |
| 16:13.37 | Slugs_ | <PROTECTED> |
| 16:14.07 | BCS-Satori | leifmadsen: I searched the file didnt see any record of that command |
| 16:14.08 | ManxPower-work | ~answers |
| 16:14.09 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 16:14.12 | kombi | quax: hmm, I run tail -f on /var/log/asterisk/messages but absolutely nothing appears when I try to dial out |
| 16:14.18 | ManxPower-work | BCS-Satori, no, it is not. read the UPGRADE*.txt files. |
| 16:14.28 | leifmadsen | BCS-Satori: then it has likely been removed -- check UPGRADE.txt and CHANGES |
| 16:15.05 | kombi | quax: as does core set verbose 1000 show anything siginificant.. |
| 16:15.12 | guax | is guax |
| 16:15.17 | AeroCloud | Slugs_ I dont know about oh323... but you can get the original DID by using ${EXTEN}, you might want to set a variable |
| 16:15.19 | guax | quax will not highlight me |
| 16:15.23 | kombi | sorry... |
| 16:16.02 | guax | leave the rasterisk open, run the test, and paste the result |
| 16:16.02 | kombi | guax: I can't get any good output, neither from messages nor from cli... |
| 16:16.15 | guax | and do a dialplan show |
| 16:16.21 | guax | :| |
| 16:16.41 | guax | a dialplan show will show who the priorities where interpreted by asterisk |
| 16:16.53 | guax | how the pr...* |
| 16:17.03 | Slugs_ | AeroCloud, DID is what the outside caller dialed? |
| 16:17.10 | AeroCloud | yes |
| 16:17.14 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 16:17.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.36 | AeroCloud | for example you send them to [incoming-did] context.. first line might be.. |
| 16:17.54 | AeroCloud | Set(__DID=${EXTEN}) |
| 16:18.08 | AeroCloud | then later you can grab it in other contexts |
| 16:18.36 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 16:19.09 | AeroCloud | the __ before the variable name, makes it global on that channel, and is passed to any future channels opened by that channel |
| 16:19.26 | kombi | guax: http://pastebin.se/200858 |
| 16:19.42 | Slugs_ | AeroCloud, awesome ill try that |
| 16:19.43 | Slugs_ | ty |
| 16:19.50 | AeroCloud | np, goodluck :) |
| 16:20.02 | BCS-Satori | ManxPower-work: Thanks; it has been renamed to "counteronpeer" |
| 16:20.16 | guax | kombi, can you make a test call and paste the cli log? |
| 16:20.23 | guax | a core set verbose 3 should do it |
| 16:21.06 | Slugs_ | AeroCloud, whats DNIDDigits? |
| 16:21.06 | kombi | guax: == Using SIP RTP CoS mark 5 <- is all I get in CLI... |
| 16:22.52 | AeroCloud | I have no idea |
| 16:23.11 | AeroCloud | but I get the incoming number like you need in my dialplan |
| 16:23.14 | guax | are you with core set verbose 3 |
| 16:23.15 | guax | ?? |
| 16:23.26 | guax | are this phone really using that asterisk? =P |
| 16:23.57 | AeroCloud | I store it in a mysql database, so I know where a person called and if a specific DID is the issue |
| 16:24.03 | kombi | guax: but the dialplan show thing was good, must I maybe now simply put the _XXX. before everything else in priority? I would wonder how to do that though, dialplan does not seem to be parsed in a procedural way.. | yes, core set verbose 3 (was 10000). Phones absolutely use that very *, just had a few incoming calls that display nicely.. |
| 16:24.29 | Slugs_ | ok ty |
| 16:25.11 | Naikrovek | oh yeah |
| 16:25.27 | Naikrovek | AeroCloud: you're the one with the ginormous asterisk system |
| 16:25.30 | guax | kombi, incoming and outgoing calls are the same thing for asterisk if you got no log on the call something is wrong, not on dialplan, its a sip phone? show its sip configuration? |
| 16:25.44 | AeroCloud | yeah tons of servers load balanced |
| 16:27.08 | AeroCloud | I need to be able to disconnect the called party and remain on the line and continue dialplan.. anyone know a way the H inside Dial() disconnects both parties |
| 16:27.25 | kombi | guax: http://pastebin.se/200859 -- bear in mind that all this worked fine in 1.6.1 only hours ago... (weired, isn't it..) |
| 16:27.58 | AeroCloud | <- using 1.6.2 |
| 16:28.07 | Naikrovek | nice |
| 16:29.16 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 16:30.06 | leifmadsen | AeroCloud: Bridge() ? |
| 16:30.19 | AeroCloud | bridge it to a bad channel? |
| 16:31.23 | leifmadsen | I think you bridge to an application |
| 16:31.28 | leifmadsen | or a location in the dialplan |
| 16:31.32 | leifmadsen | could be Hangup() I guess |
| 16:31.41 | AeroCloud | 1 minute, got a phone call |
| 16:31.48 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 16:32.13 | leifmadsen | I'm just throwing out a random idea. I haven't spent a lot of time thinking about the problem since I'm working on other things |
| 16:33.04 | ManxPower-work | AeroCloud, Sounds to me like you don't have a priority after the dial, or your screwed up your features.conf |
| 16:34.26 | guax | kombi, then i dunno |
| 16:34.50 | Naikrovek | he wants to disconnect a call, keeping the originating caller on the line, disconnecting the called party |
| 16:35.26 | kombi | guax: maybe you can show me how you tell outgoing from internal calls? |
| 16:35.37 | leifmadsen | AeroCloud: with 'H' you can't use 'g' because the calling party was hung up, which is where the dialplan execution is happening |
| 16:36.04 | guax | kombi, i have a special agi of awesomeness that makes the request process and all the management of incoming and outgoins via routing table |
| 16:36.05 | leifmadsen | AeroCloud: so once the calling party is hung up, there is nothing to continue on in the dialplan. The far end wasn't executing anything in the dialplan. |
| 16:36.16 | guax | kombi, sourceforge.net/project/snep |
| 16:36.30 | kombi | guax: awesomeness.. i like that..,) |
| 16:36.30 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 16:36.32 | guax | its not internacionalized |
| 16:36.36 | guax | so you will suffer |
| 16:36.36 | *** join/#asterisk p4nther (~63e04505@gateway/web/freenode/x-ipeaenpqkrewtcus) |
| 16:36.44 | guax | its an on going project |
| 16:36.48 | kombi | guax: as i always do...;) |
| 16:37.01 | guax | =D |
| 16:37.12 | guax | kombi, but anyways it was made for 1. |
| 16:37.14 | guax | 1.4 |
| 16:37.31 | kombi | guax: quite liked my approach with the pattern though, nice and simple.. |
| 16:37.35 | p4nther | sorry if this is an obvious question but I've looked and haven't found anything out there ... |
| 16:37.45 | leifmadsen | AeroCloud: it really seems like you need to use 'g' with Dial() along with a blind transfer type of feature that transfers the called party to Hangup() |
| 16:37.48 | guax | p4nther, the truth is out there |
| 16:37.50 | *** join/#asterisk megalomano (~klonstein@38.124.169.126) |
| 16:37.56 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:38.02 | p4nther | is rmirror.digium.com no longer valid ? |
| 16:38.06 | kombi | guax: I'll deve into voip-info for an hour then, thanks guax! |
| 16:38.13 | kombi | *delve |
| 16:38.18 | guax | then you have time |
| 16:38.20 | guax | =P |
| 16:38.32 | ariel_ | hello folks |
| 16:39.10 | p4nther | @guax ... :-) |
| 16:39.14 | megalomano | hi people , one question , how to assign a number for mi asterisk box ( PSTN ) |
| 16:39.36 | devoid | megalomano: Get a landline number? |
| 16:40.17 | ariel_ | I have an issue with call-limit=1 in asterisk 1.6.0.21, for sip. I need to limit the amount of calls to a sip phone is there any other way to do this or a simple dial plan for counting calls up on the sip phone? |
| 16:40.23 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 16:40.29 | guax | p4nther, SETI's been looking for years for something out there and nothing. You cant blame yourself. |
| 16:40.43 | Mhaddog | hello... I need some help... |
| 16:40.50 | p4nther | @guax ... lol |
| 16:40.53 | leifmadsen | ariel_: GROUP() and GROUP_COUNT() ? |
| 16:41.00 | megalomano | <devoid>: i.e ... 5514540099 is the number to my asterisk box |
| 16:41.31 | p4nther | conary has been timing out since FEB ... not sure if it's on my end or not .... asteriskNOW 1.4.18 ... |
| 16:41.31 | megalomano | <devoid>:local dialing |
| 16:41.56 | Mhaddog | My server had a bad shutdown and now asterisk does not recognizes the sangoma cards... at startup I got a loading ec image oct6116-64s.ima... and centos gets stuck in there... |
| 16:42.14 | Mhaddog | I have reinstalled dahdi and wanpipe (sangoma drivers)... any ideas? |
| 16:45.20 | *** join/#asterisk voipmonk (~shido6@dsl-69-172-110-65.acanac.net) |
| 16:46.36 | ariel_ | leifmadsen: any sample on how to use it? |
| 16:46.50 | leifmadsen | check google, but it should just be: |
| 16:47.01 | leifmadsen | Set(GROUP()=my_sip_peer) |
| 16:47.21 | leifmadsen | Set(RESULT=${GROUP_COUNT(my_sip_peer)}) |
| 16:48.19 | leifmadsen | you'll have to enable it on both terminating and originating sides of the dialplan |
| 16:48.29 | leifmadsen | i.e. when you call the peer, and when the peer places a call |
| 16:49.28 | ariel_ | ok let me see how to get that working on a dial plan. Just seems a really nutty way to only allow one call at a time to a phone. |
| 16:50.22 | hardwire | http://www.voip-info.org/wiki/view/Zycoo+ZP302 |
| 16:50.24 | hardwire | uhm..... |
| 16:50.30 | hardwire | that probably needs fixed |
| 16:51.04 | Slugs_ | morning hardwire |
| 16:51.14 | hardwire | ariel_: I have that argument as well, however it's better to handle things like that from the dialplan than it is from the channel driver |
| 16:51.18 | hardwire | modularity and all! |
| 16:51.24 | hardwire | god bless local channels |
| 16:51.31 | hardwire | mornin |
| 16:52.22 | *** join/#asterisk VEc (~Vector@84.12.253.146) |
| 16:52.29 | AeroCloud | leifmadsen: so your saying blind transfer to say extension 1, and its Hangup() |
| 16:52.50 | leifmadsen | AeroCloud: possibly something like that could work -- not sure if you can automate that blind transfer with a single key or not |
| 16:52.54 | AeroCloud | I will give that a try, but I think that hung me up before |
| 16:53.13 | leifmadsen | AeroCloud: ya... you could potentially do a Bridge() perhaps on the OTHER channel, and bridge it to the Hangup() application |
| 16:53.26 | VEc | Whats everyone favourite T1/E1 -> SIP GWs ? I am looking for something with greater than 4 E1s per GW ? |
| 16:53.42 | AeroCloud | this is my current blind transfer |
| 16:53.43 | AeroCloud | ;Transfer Extensions |
| 16:53.44 | AeroCloud | exten => 1,1,Bridge("HANGUP_NOW!!${RAND(1,999999)}") |
| 16:53.44 | AeroCloud | exten => 1,n,Wait(4) |
| 16:53.44 | AeroCloud | exten => 1,n,Goto(${DIALEXT},1) |
| 16:53.49 | AeroCloud | but it does not hangup the callee |
| 16:54.15 | AeroCloud | it does send the main caller back with the goto() |
| 16:54.26 | AeroCloud | but leaves the callee idle there |
| 16:54.30 | leifmadsen | Bridge() doesn't let you specify which channel to run it on? Thought it did. |
| 16:54.40 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:54.46 | AeroCloud | bridge does.. |
| 16:54.52 | AeroCloud | I send them to a bad channel |
| 16:54.58 | AeroCloud | so asterisk should disconnect them |
| 16:55.09 | AeroCloud | but it doesnt |
| 16:55.14 | leifmadsen | not sure... I'd have to play around with it on a test system to know any more than what I'm saying now :) |
| 16:55.22 | leifmadsen | never done that before |
| 16:55.26 | AeroCloud | I have been messin with this for awhile |
| 16:55.36 | AeroCloud | the H inside dial works if done within 2 second of answer |
| 16:55.43 | AeroCloud | once after 2 seconds.. it hangs up both |
| 16:56.02 | leifmadsen | probably because the calls have been bridged in the core |
| 16:56.08 | AeroCloud | yeah |
| 16:56.26 | AeroCloud | I have thought about storing the callee sipcallid |
| 16:56.27 | Katty | hmm. network card is still shutting it self off :< |
| 16:56.27 | *** join/#asterisk pmhaddad (~pmhaddad@71-13-218-72.dhcp.mrqt.mi.charter.com) |
| 16:56.45 | AeroCloud | and then issuing a hangup on that channel via manager |
| 16:56.52 | AeroCloud | but doesnt work right |
| 16:57.44 | AeroCloud | it sounds like such a simple thing to do, there should be a cmd to do that |
| 16:59.02 | Katty | has anyone ever heard of a situation where after a system goes idle, the network card turns itself off? I thought perhaps at Skeeter's suggestion it had something to do with gdm, so i've disabled gdm. Unfortunately it's still happening, regardless of whether it's the integrated card or the pci card i stuck in there. |
| 16:59.41 | AeroCloud | Katty, sorry never heard of that issue before |
| 16:59.45 | Katty | bummer. |
| 16:59.50 | AeroCloud | maybe there is a motherboard issue with power |
| 17:00.09 | p4nther | don't mean to interrupt but does anyone else still run the older AsteriskNOW and get the 'rmirror.digium.com' timeout ? |
| 17:01.26 | Qwell | p4nther: nobody sane |
| 17:01.38 | Qwell | You should very seriously consider upgrading |
| 17:01.46 | spenguin[work] | Katty: turns itself off .. does the interface show up with ifconfig? |
| 17:02.12 | Katty | spenguin[work]: yes, ifconfig shows it. but i can't ping it or ping from it. |
| 17:02.18 | Katty | spenguin[work]: an ifup then ifdown sorts it out |
| 17:02.21 | [TK]D-Fender | p4nther: The old one = dead |
| 17:02.25 | p4nther | @Qwell, true ... I was letting the conary thing do the updating until it stopped |
| 17:02.29 | Katty | spenguin[work]: sadly, i can't really do that in the middle of a production server all the time :/ |
| 17:02.44 | spenguin[work] | Katty: what network card is it |
| 17:03.01 | spenguin[work] | lspci | grep -i ethernet |
| 17:04.10 | Katty | waits |
| 17:04.11 | p4nther | Thanks guys ... just the answer I suspected ... I was presuming it was on my end cuz there was nothing mentioned in the forums anywhere ... |
| 17:04.49 | Katty | spenguin[work]: should the output really take this long? |
| 17:04.53 | spenguin[work] | nope |
| 17:06.42 | p4nther | TK-D-Fender & Qwell ... thanks again ... would you suggest switchvox or stay with AsteriskNOW ? |
| 17:07.15 | Qwell | p4nther: If you're willing to pay for switchvox, definitely go with that. I've never used the free version, so can't comment. |
| 17:07.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:08.00 | spenguin[work] | Katty: or lspci and look in there for ethernet |
| 17:08.02 | p4nther | It's no prob's building from scatch, 15+ years doing Unix Sys Admin (Solaris speciality...) but am quite lazy !!! |
| 17:08.08 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 17:08.30 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 17:09.09 | p4nther | Qwell --- ooops, perhaps I should read things first !! didn't realize it was a pay thing !!! Free = me !!!! |
| 17:09.32 | Qwell | p4nther: well, like I said - there is a free edition, but I've never used it |
| 17:10.50 | Katty | spenguin[work]: i added a ping -i 300 myrouter command to rc.local |
| 17:11.10 | Katty | spenguin[work]: if it's really shut it down because of being idle, that should at least prove or disprove it |
| 17:11.32 | spenguin[work] | hrm, yeah but never heard of a ethernet card going off to sleep before |
| 17:11.44 | Katty | me either |
| 17:13.06 | p3nguin | Does ethtool say anything useful about it? |
| 17:15.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:16.13 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:17.56 | Katty | nothing useful in dmesg |
| 17:17.59 | Katty | p3nguin: i'll get it |
| 17:18.02 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 17:18.31 | Katty | hrmm |
| 17:18.38 | Katty | supports wake on: g, wake on: d |
| 17:18.43 | Katty | not sure what that really means. |
| 17:18.56 | Katty | but i just noticed my resolf.conf was overwritten by network manager |
| 17:19.01 | p3nguin | g is magic packet, d is disabling it |
| 17:19.40 | p3nguin | Should have removed network mangler already. |
| 17:20.02 | Katty | well it's certainly gone now |
| 17:20.08 | Katty | i don't appreciate Managers overwriting my conf files |
| 17:20.31 | p3nguin | It's really bad about doing things people don't want it to do. |
| 17:20.47 | p3nguin | Even more so when using wifi stuff, it seems. |
| 17:20.49 | Katty | we'll see if that has any affect on it. |
| 17:21.16 | p3nguin | Network Manager does like to suspend network interfaces. |
| 17:21.25 | *** part/#asterisk l2trace99 (~jr@74.118.40.1) |
| 17:21.37 | Katty | i'll suspend IT in a minute |
| 17:21.41 | Katty | well, you know |
| 17:21.45 | p3nguin | :P |
| 17:21.56 | p3nguin | hammer and chisel? |
| 17:22.37 | *** part/#asterisk McBoingbo (~Galabaga@mail.hrsg.ca) |
| 17:24.10 | p4nther | once again, thanks guys ... and I'm off ... I'll check back once I've got the lastest AsteriskNOW installed ... I like it cuz it's simple ... it uses the 'rpath' appliance method of Linux underneath lessening the admin and maintenance time |
| 17:25.39 | Qwell | p4nther: New versions of AsteriskNOW no longer use rPath as a base. |
| 17:25.49 | Qwell | They use CentOS, which should provide much more timely (and easier..) updates. |
| 17:25.55 | p3nguin | Oh no, now he's going to hate on it. |
| 17:26.18 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 17:26.19 | p4nther | not at all, CentOS makes it even easier !!! |
| 17:27.29 | Katty | dear polycom, please boot faster. |
| 17:28.59 | spenguin[work] | 8:37 Katty: supports wake on: g, wake on: d |
| 17:29.06 | spenguin[work] | thats for wake on lan |
| 17:29.38 | spenguin[work] | Id suspect the card being faulty for acting up |
| 17:29.40 | elred_ | Katty : i used to talk INTO a phone, not TO a phone |
| 17:29.42 | p3nguin | With Cisco phones, they are constantly checking the tftpd for the files. If the files aren't there, it takes a LOT longer for the phones to boot; when the files are present, phones boot quickly (usually under 1 minute). Could similar things be going on with your Polycoms? |
| 17:30.54 | Katty | spenguin[work]: i would too, except it happens with both the integrated and the other pci network card i put in there |
| 17:31.01 | p3nguin | If the NIC goes wonky again, I would check to see if ethtool shows any errors or other helpful info while the NIC is in the messed up state. |
| 17:31.13 | Katty | p3nguin: i'll check that for sure |
| 17:31.21 | *** join/#asterisk FabiOne (~fabi@151.13.190.14) |
| 17:31.34 | p3nguin | Meanwhile, it's lunch time! |
| 17:31.36 | FabiOne | hi all |
| 17:31.39 | Katty | p3nguin: i'm just impatient |
| 17:31.58 | FabiOne | i've a little problem on my dialplan |
| 17:32.33 | AeroCloud | leifmadsen: the blind transfer to Hangup worked |
| 17:32.54 | AeroCloud | thank you |
| 17:33.23 | FabiOne | how to drop a unanswered call? |
| 17:33.46 | hardwire | is autofallthrough=no in your dialplan? |
| 17:34.16 | FabiOne | uhmm |
| 17:34.24 | FabiOne | i use freepbx fronten |
| 17:34.27 | FabiOne | *d |
| 17:34.36 | AeroCloud | wrong channel here then |
| 17:34.38 | hardwire | yeh |
| 17:34.43 | hardwire | before you get flamed out of the channel |
| 17:34.45 | FabiOne | olo |
| 17:34.47 | FabiOne | lol |
| 17:34.47 | hardwire | you should check with them first |
| 17:34.52 | FabiOne | ok, i'm sorry |
| 17:35.21 | AeroCloud | when you do a transfer, does a variable get sent or saved.. like transferstatus ? |
| 17:36.03 | Katty | alright. time to let this box go idle |
| 17:40.31 | [TK]D-Fender | AeroCloud: Only on blind transfers, not attended |
| 17:43.37 | guax | wait |
| 17:45.55 | guax | AeroCloud, http://pastebin.com/9vbstXVJ |
| 17:46.05 | guax | this is worth a 2l coke |
| 17:46.29 | AeroCloud | TK, I got it, thanx |
| 17:46.52 | guax | when you need this kind of information do a ChanDump() |
| 17:47.09 | guax | it will tell you every variable available at that point on dialplan execution |
| 17:47.16 | AeroCloud | ok |
| 17:47.35 | AeroCloud | just hate to do that on production servers lol |
| 17:48.33 | *** join/#asterisk GreyFoxx (greg@out.of.phaze.org) |
| 17:48.37 | guax | production servers are the best testing servers. aways have real case scenarios. aeuheauhe |
| 17:48.42 | AeroCloud | thanx for the info guax |
| 17:49.12 | AeroCloud | now I just gotta figure out how not to execute the h, on the blind transfered channel |
| 17:49.27 | AeroCloud | since its within the same context.. |
| 17:49.52 | c0rnoTa | goodbye all |
| 17:49.58 | GreyFoxx | Anyone here a "heavy" user of the Asterisk Manager interface ? I've implemented several things in our network to use the manager to gather data but one of the guys in charge of the box fears that too many calls to the manager could make asterisk unstable |
| 17:50.20 | GreyFoxx | And I'm looking to find out if people are using some sort of proxies to minimize connections or if his fears Are just unrealistic |
| 17:50.22 | *** join/#asterisk jtodd (vhfgxemwjr@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:50.22 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:50.26 | AeroCloud | I do 100's of calls to the manager every few minutes |
| 17:50.32 | AeroCloud | servers are stable for over a month now |
| 17:50.34 | GreyFoxx | directly to the manager? |
| 17:50.37 | GreyFoxx | ok |
| 17:50.49 | ariel_ | hardwire: sorry for a late reply. But I got called out to a customer. I have a macro that I made for call counts already. Just was trying not to use a macro for this gateway we are putting up. |
| 17:51.03 | guax | i usually do hundreds of requests per second |
| 17:51.09 | GreyFoxx | The majority of our network is running 1.4.x but we have one box running 1.2.x and maybe he had a bad experience with it |
| 17:51.19 | AeroCloud | I'm using 1.6 |
| 17:51.26 | AeroCloud | so not sure about 1.2 or 1.4 |
| 17:51.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:51.50 | GreyFoxx | guax: Directly to the manager? What version of asterisk ? |
| 17:52.33 | guax | 1.4 |
| 17:52.43 | AeroCloud | guax: you know how hard ChanDump() would be to read with 200 concurrent calls on production server lol |
| 17:53.00 | AeroCloud | hard enough finding what I'm doing sometimes |
| 17:53.07 | guax | AeroCloud, thats why we have full log with grep |
| 17:53.14 | AeroCloud | good idea |
| 17:53.16 | GreyFoxx | heh |
| 17:53.22 | ManxPower-work | 1.2 had such bad problems with manager connections that someone actually wrote a proxy |
| 17:53.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:54.18 | GreyFoxx | Manx: Yeah. And I think that is why he is worried. He found even the copy of astmanproxy he used was crashy :) |
| 17:56.00 | guax | i saw some lock problems with manager once, but never tryed again neither remember what do to test it =P |
| 17:56.21 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:56.23 | Micc | The current manager stuff has to be good. Digium sells a lot of features that use it in their asterisk appliances. |
| 17:57.04 | Micc | Not necessary sound logic, but it may have some merrit. |
| 17:57.30 | hardwire | ariel_: gotcha |
| 17:58.18 | guax | Micc, my switchvox test shows some cdr inconsistencies. Not sure that means something =P |
| 17:58.31 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 18:01.48 | *** join/#asterisk batphone (~will@rrcs-24-153-211-180.sw.biz.rr.com) |
| 18:01.58 | batphone | i have a customer asking me questions about sip privacy headers |
| 18:02.24 | batphone | im running a wholesale carrier switch and they register phones to some other box that connects to another switch, which then connects to me |
| 18:02.42 | batphone | they are saying that MY switch needs to be recognizing the *67 dial code and procesing it |
| 18:02.47 | batphone | changing up the caller id, etc |
| 18:02.51 | batphone | rather than their asterisk box |
| 18:02.54 | batphone | is this true? |
| 18:03.52 | Naikrovek | depends |
| 18:04.26 | *** join/#asterisk V4mpire (~Gary@82.118.111.252) |
| 18:04.30 | batphone | we reject calls that dont begin with a 1 anyway |
| 18:04.31 | Naikrovek | if the downstream provider doesn't allow people to block outgoing callerid then maybe it's up to you |
| 18:04.45 | V4mpire | Hi does anyone know anywhere for free UK geographical DID's to forward to a sip account ? |
| 18:04.54 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 18:05.26 | batphone | being a class 4 switching provider i dont think its my responsibility to employ feature sets on my gear |
| 18:05.49 | batphone | so if upstream's upstream class 5 isnt doing it, i dont see how i can help |
| 18:07.48 | Naikrovek | we should be asking you this question, not vice versa |
| 18:08.15 | guax | 8100388 cdr entries leave selects quite slow =P |
| 18:08.40 | florz | ... then you are doing something wrong ... |
| 18:08.56 | dinesh___ | so far i didn't find any free sip number provider to work properly (sipcall.ch or 12voip.com), numbers are not reliable, sometimes cannot be called, etc |
| 18:10.13 | dinesh___ | well sometimes, actually it's more most of the time |
| 18:13.43 | dinesh___ | hmm i think i'll try to add skype as an inbound number to my asterisk server |
| 18:13.55 | dinesh___ | i heard that there was a module to handle skype's protocol |
| 18:14.58 | dinesh___ | erf it has to be bought :( |
| 18:16.27 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 18:17.34 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.01 | highvoltz | so I have a remote phone over vpn thats going to voice mail when you call it. The phone can call out no problem and it shows registered at the phone, but its not showing a subscription on the server. what might be the problem? |
| 18:18.31 | highvoltz | DND is off |
| 18:18.36 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 18:18.44 | V4mpire | dinesh___ is this any use to u http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 18:19.07 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 18:19.57 | sbrath | so is it possible that chan_sip can deadlock, and not prevent calls in general from comeing in, just prevent SIP endpoints from being able to register? |
| 18:20.09 | *** join/#asterisk newsmafia (~newsmafia@207-114-163-134.static.twtelecom.net) |
| 18:20.49 | batphone | Naikrovek: class 4 and class 5 services are completely different. they can be looked at almost like the OSI in networking. class 4 services are plain, wholesale traffic swtiching which might correspond to layer 2 or 3 in the networking world. |
| 18:21.19 | sbrath | I guess I need to brush up on dumping core when it happens next, as currently when this occurs their is nothing in the logs to signal the problem. sip show peers looks ok... It's wierd. |
| 18:21.20 | batphone | Naikrovek: class 5 on the other hand is things like voicemail, call forwarding, find me/follow me, call queus, etc. and would be more analogous to the application layer of networking. |
| 18:21.45 | batphone | Naikrovek: in telecom there is a distinct difference between the two and a clear line dividing the contracts held between parties along these numbers. |
| 18:22.24 | batphone | Naikrovek: i was just wondering if anyone in here really expected someone like the local telco or SIP provider to pass your *67 code to the remote CO for processing. |
| 18:23.26 | batphone | i am just not aware of any inherent functionality in the SIP protocol itself to cause a class 4 switch to modify the caller ID based on the dialed number without some outside agreement stating that this functionality will be worked into the product. |
| 18:27.24 | *** join/#asterisk knarfly (~vlad@98.242.237.166) |
| 18:28.06 | knarfly | it's been a while since I configured a dial plan, what's the latest incarnation for |
| 18:28.06 | knarfly | exten => s,1,Wait,20 ; Wait 20 seconds |
| 18:28.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:28.35 | slacker775 | anyone try sip trunks from bandwidth.com? we need sip outbound termination that just works and other providers we have tried so far like to go to crap when we need to make important calls |
| 18:28.44 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 18:29.01 | Katty | well i just tried calling the new asterisk box |
| 18:29.03 | Katty | after letting it idle |
| 18:29.05 | Katty | it worked. |
| 18:29.08 | knarfly | slacker775: I used them before and it worked well, they ar kind of expensive |
| 18:29.22 | slacker775 | yeah, the $$ is my biggest issue w/ them really |
| 18:29.33 | Micc | batphone, your provider should have a way to handle *67 with a sip header, but it doesn't get passed on to the CO. |
| 18:29.53 | Katty | it's been sitting idle for about 40 minutes |
| 18:29.58 | knarfly | still $30/month is not that bad, but you can get less expensive service |
| 18:30.03 | *** join/#asterisk flapjacks (~flapjacks@wsip-70-166-201-90.ph.ph.cox.net) |
| 18:31.07 | Micc | batphone, you might have to process the *67, strip it out and add the sip header before dialing out your provider. |
| 18:31.32 | Micc | batphone, some might do that for you. |
| 18:31.41 | slacker775 | yeah, i'd like to pay less, especialyl since our usage fluctuates a lot, # active lines and minutes/mo |
| 18:32.12 | knarfly | where is the latest TFOT book |
| 18:32.20 | Micc | slacker775, where are you located? |
| 18:32.30 | slacker775 | we've been using vitelity & voicepulse to-date and they can work great at times, but then sometimes echo, noise, or all circuits busy |
| 18:32.32 | knarfly | \book |
| 18:32.40 | dinesh___ | oh localphone looks great, anyone tried it already ? |
| 18:32.43 | slacker775 | we're in tampa here but call out all over the US & canada |
| 18:33.04 | dinesh___ | their DIDs are cheap, 0.75 euros/month and are geographical ones |
| 18:33.23 | Micc | slacker775, we are in Seattle. We use vitelity for long distance, I know what you mean. its not always 100%. |
| 18:33.44 | slacker775 | yeah, and it CAN work great for periods of time and it's costing us almost nothing |
| 18:33.50 | Micc | slacker775, we have a local provider for all local calls. And that is 100% |
| 18:33.56 | batphone | Micc: being an intermediary i would think this sort of functionality would be up to the endpoints |
| 18:34.10 | slacker775 | unfortunately, 90% of our calls are LD |
| 18:34.13 | batphone | Micc: you wouldnt want your ISPs core routers doing deep packet inspection on your Youtube videos |
| 18:34.21 | batphone | Micc: inserting ads, etc |
| 18:34.46 | batphone | Micc: you want your ISPs core routers to find the best path to for your traffic to route. |
| 18:35.31 | Micc | batphone, your right, but some ITSPs will do features like that for you so you don't have to impliment them yourself. |
| 18:35.43 | Micc | batphone, but for the most part, your right, its up to the end points. |
| 18:35.52 | batphone | Micc: in which case you would be purchasing class 5 services |
| 18:36.47 | Micc | slacker775, we are looking for a better long distance provider too. A local one if possible. I don't think its vitelity's fault. It seems to be problems on the public internet most of the time. |
| 18:37.40 | Micc | slacker775, if you find a good cheap provider, let me know. The only thing I can find thats 100% would cost about 5 times as much. |
| 18:38.36 | slacker775 | voicepulse seemed pretty good when i first set them up a month or so ago, but lately i get nothign but all circuits busy.... yes we've paid the bill! lol |
| 18:39.32 | Micc | batphone, in your example, that would be correct. But I don't think most ITSPs classify themselves that way. |
| 18:40.05 | Micc | slacker775, I haven't tried voicepulse, but I have heard stories, not good oens. |
| 18:40.08 | newsmafia | i have the same problem with Voicepluse...lots of intecept. I've used teliax. they are pretty solid. |
| 18:40.37 | Micc | slacker775, vitelity is pretty good, we rarely have a problem with them. |
| 18:41.03 | Micc | slacker775, its usually a problem with the internet. Although some of their defaults might limit your calling internationally. |
| 18:41.32 | slacker775 | yeah, we can usually deal w/ canada but canadians can't call our vitelity 800 #'s and such... |
| 18:41.34 | Micc | slacker775, they have you turn it on first, then they have a rate limit that it will reject if your call is going to be more than 20 cents a minute. |
| 18:42.09 | Micc | slacker775, thats also an option I think. It should work from canda, but it'll cost more and vitelity has to enable that for you manually I think. |
| 18:42.11 | slacker775 | oh yeah? didn't know that.. |
| 18:42.29 | slacker775 | yeah, i did enable at least basic intl calling... |
| 18:42.54 | slacker775 | i wouldn't mind falling back to vitelity or whatnot if other 'premium' outbound lines are used up... aint workign w/ trixbox so far it seems tho |
| 18:43.02 | slacker775 | cfg issues on my end i'm sure |
| 18:43.16 | Micc | slacker775, vitelity is pretty good about responding to open trouble tickets. I've opened probably 50 or so and they all get taken care of pretty quickly. Usually they make an adjustment to my account and fix the problem. |
| 18:44.13 | Micc | slacker775, yeah, we fail over to vitelity for all calls. |
| 18:44.18 | slacker775 | i probably need to bug them more often when i'm havign issues... |
| 18:44.29 | slacker775 | any particular trick for failing over to a 2nd/3rd provider? |
| 18:44.38 | Micc | even local calls, sometimes theres a local routing problem. |
| 18:45.06 | Micc | slacker775, just add more Dial's in the dial plan. |
| 18:45.31 | slacker775 | jeez (on trixbox 2.6) just noticed the trunk sequence part on outbound routes |
| 18:45.41 | Micc | slacker775, I check the dialstatus, but you can just dial them without checking it too. |
| 18:45.47 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:46.42 | Micc | slacker775, good dialplans take constant grooming, kind of like good hair. |
| 18:47.00 | Micc | at least thats been my experience. |
| 18:47.01 | slacker775 | yeah, no kidding... that's why i keep my hair short! |
| 18:47.28 | Micc | I am always looking for ways to improve my dialplan to make it more fail safe and user friendly. |
| 18:48.30 | Micc | I watch logs and sometimes the console and when I see stuff happen that I don't like, I get to work on fixing it. Weird stuff happens that you don't think about the first time through. |
| 18:48.52 | Micc | trixbox probably has a good starting dialplan though. |
| 18:48.55 | slacker775 | especially if you have diff providers for different intl calls and all of that fun |
| 18:49.09 | slacker775 | i'm on a pretty std, basic dialplan, especially since i only have the one out really |
| 18:49.32 | ManxPower-work | trixbox talk? ta ta |
| 18:49.34 | *** part/#asterisk ManxPower-work (~manxpower@139.sub-75-234-63.myvzw.com) |
| 18:50.30 | Micc | opinions and veiwpoints on trixbox are not necessary the view of this channel or the asterisk community. |
| 18:51.14 | Micc | I don't use trixbox, so I don't know much about it. |
| 18:51.47 | Katty | everyone sure has been cranky lately |
| 18:52.22 | Micc | Well, I have to get going. I was suppose to leave a while ago. I've got customers to go see and tweak some adapter settings. |
| 18:52.29 | Katty | byebye |
| 18:52.43 | spenguin[work] | im going home |
| 18:52.46 | spenguin[work] | goodnight |
| 18:53.01 | Katty | ninite penguin |
| 18:53.15 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 18:53.24 | spenguin[work] | waves |
| 18:54.21 | highvoltz | so I have a remote phone over vpn thats going to voice mail when you call it. The phone can call out no problem and it shows registered at the phone, but its not showing a subscription on the server. what might be the problem? DND is off |
| 18:54.26 | *** join/#asterisk bmoraca (bmoraca@66.242.162.254) |
| 18:55.14 | bmoraca | so I've got an E&M wink trunk here that's immediately hanging up on some (not all) calls. that's likely a timers issue between me and the telco, right? |
| 18:55.24 | Katty | hi bmoraca |
| 18:55.26 | bmoraca | wishes this customer really had a PRI like they said they did |
| 18:55.32 | bmoraca | hey |
| 18:56.19 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 18:59.21 | bmoraca | i don't need to worry about E&M types with dahdi, do I? i don't recall finding anywhere to configure types 1-4 |
| 19:01.50 | knarfly | like I said, it's been a while, the caller called my providers number, the call came through to my SIP phone inside my LAN. I could hear the caller but they could not hear me. |
| 19:02.28 | beek | hugs Katty |
| 19:02.35 | Katty | hi beekers |
| 19:02.37 | Katty | hugs beek |
| 19:02.47 | beek | How are you today Katty? Is the crittercam fired up? |
| 19:03.00 | Katty | no crittercam is off. it's all raining and gross outside today |
| 19:03.26 | beek | Damn... we have the same. I was hoping for a nice sunshine and wildlife video feed. |
| 19:04.06 | Katty | yeah, sorry :< |
| 19:04.49 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 19:05.13 | knarfly | wow, I can't recall what the fix for this was...I can here the callers but they can't hear me! |
| 19:05.57 | Katty | that usually have to do with natting and rtp ports |
| 19:06.03 | Katty | infobot: nat? |
| 19:06.04 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:06.07 | Katty | hrm no |
| 19:06.10 | Katty | infobot: sip nat? |
| 19:06.14 | Katty | infobot: nat sip? |
| 19:06.18 | Katty | infobot: natsip? |
| 19:06.26 | p3nguin | ~sipnat |
| 19:06.26 | infobot | [~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 |
| 19:06.30 | Katty | there we go |
| 19:07.31 | Katty | frowns |
| 19:07.39 | Katty | my workstation is Ticking |
| 19:07.43 | Katty | not like hard drive ticking |
| 19:07.47 | p3nguin | IT'S A BOMB! |
| 19:07.48 | Katty | it's like...clock ticking |
| 19:07.50 | Katty | IT"S A BOMB |
| 19:08.14 | bmoraca | p3nguin's been watching too much Live Free Die Hard |
| 19:08.17 | Katty | probably because i'm installing this piece of software |
| 19:08.31 | p3nguin | Bomb software? |
| 19:08.36 | Katty | writeTICKwriteTICKwriteTICK |
| 19:08.56 | p3nguin | How To Turn Your Workstation Into A Bomb v3.6 |
| 19:09.28 | Katty | requires 2 cellphones, or a garage door opener and reciever |
| 19:09.59 | Katty | i'm actually installing quickbooks |
| 19:11.02 | chuckf | that's just the timer counting down to when the license on quickbooks will expire |
| 19:11.09 | p3nguin | haha |
| 19:11.26 | chuckf | it'll quiet down soon |
| 19:11.32 | Katty | i doubt that |
| 19:11.34 | Katty | mister smart tail |
| 19:11.43 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 19:11.48 | Katty | hi fifer |
| 19:11.55 | fifer | Afternoon |
| 19:13.40 | fifer | When I last worked extensively with * a 2.4ghz pentium or even celeron would easily handel 1-2 PRI's and 60 phones, even quite a bit more. I'm setting up a new system with * 1.6.0 and the indicated load but I'm having issues with the machine I now have it in. |
| 19:14.03 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 19:14.34 | fifer | It is a Pentium 4 2.8ghz but I have another dell box that has either a P42.4ghz or a celeron (bios says P4, dell config info (from service tag) says Celeron) |
| 19:14.48 | fifer | Should this box still very easily handle this load? |
| 19:15.19 | fifer | It might actually be a single PRI and 10-20 chanells of sip trunking |
| 19:15.40 | Naikrovek | fifer: yeah |
| 19:15.44 | Naikrovek | should be more than fine |
| 19:15.48 | fifer | In reality the load would not likely rize above using halfe that even once a month |
| 19:16.02 | Naikrovek | but P4s *suck* in comparison to a modern intel proc |
| 19:16.03 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 19:16.06 | fifer | That is what I thought, just wanted to pose the question |
| 19:16.18 | *** join/#asterisk maximCH (~maximCH@adsl-ecom-4-c15-p038.vtx.ch) |
| 19:16.25 | p3nguin | What if the moderm processor is a P4? |
| 19:16.38 | fifer | I'm well aware! We are actually only starting out with 4 pstn lines, by the time we have everyone on board we can likely buy a new machine to replace this one |
| 19:16.41 | fifer | just can't now |
| 19:16.43 | Naikrovek | well the p4 arch is quite old now |
| 19:16.45 | maximCH | anyone here use SfA? ever since I upgraded to 1.6.1.18 I have trouble |
| 19:17.01 | Katty | come on quickbooks! come on! you can DO EET |
| 19:17.08 | Katty | stares at installation progress bar |
| 19:17.10 | Katty | sighs |
| 19:17.14 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.14 | p3nguin | When did they stop making the P4? I know it started a LONG time ago. |
| 19:17.25 | fifer | I'm having motherboard chipset issues with my a1200p and the existing machine |
| 19:17.53 | Qwell | fifer: Call your manufacturer. |
| 19:17.58 | fifer | I even have a new machine I could use if I had to but it will not accept a full length card |
| 19:18.35 | Katty | call Qwell |
| 19:18.39 | Katty | i would. |
| 19:18.45 | Katty | rings Qwell |
| 19:18.52 | fifer | @Qwell: You have said that multiple times, and I'm sure your point is that you know I CAN't. I would love to have bought a digium card but it was not an option. |
| 19:19.03 | Qwell | fifer: Why can't you? |
| 19:19.17 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 19:19.24 | fifer | The ONLY way I was going to get them on * was to keep the cost bellow a certain amount. I simply can not afford even an 8 port DIgium card right now. |
| 19:19.26 | fifer | Not my money |
| 19:20.02 | Katty | money's tight everywhere, all over the country, right now |
| 19:20.12 | p3nguin | Why does that prevent you from calling the manufacturer of that card? |
| 19:20.15 | fifer | I won't go into details, just not an option, and I know most are not running into this issue, I just have the bad luck to have a motehrboard that conflicts |
| 19:20.20 | Qwell | You've probably spent more money trying to fix the problem than buying non-crap hardware. |
| 19:20.27 | *** part/#asterisk newsmafia (~newsmafia@207-114-163-134.static.twtelecom.net) |
| 19:20.29 | Katty | perhaps he got it off craigslist :P |
| 19:20.46 | p3nguin | The mfgr should still support it. |
| 19:20.55 | Katty | watches the polycom bootrom update breat the quickbooks install |
| 19:21.03 | Katty | breat? |
| 19:21.09 | Naikrovek | BREAT |
| 19:21.10 | Katty | infobot: breat? |
| 19:21.15 | p3nguin | breast? |
| 19:21.28 | Katty | while both are verbs... |
| 19:21.32 | Katty | i don't really think this applies |
| 19:21.54 | Katty | FORMATTING FILE SYSTEM PLEASE WAIT |
| 19:21.59 | p3nguin | lol |
| 19:22.14 | p3nguin | Hope it's the phone and not your hard drive. |
| 19:22.18 | fifer | I'm dealing with Openvox via the net, but there in Beijing so I have not had the ability to call them yet due to the time diference. I plan on trying tonight |
| 19:22.25 | Katty | p3nguin: nothing's wrong with my phone |
| 19:22.30 | Katty | p3nguin: just updated the bootrom |
| 19:22.45 | Katty | p3nguin: now that the dtmf is definately not phone related |
| 19:23.32 | Qwell | fifer: there's nothing anybody in here can do to help you with a hardware problem... |
| 19:23.47 | Katty | fifer: i can offer a hug. |
| 19:23.48 | *** join/#asterisk wvds-nl (~wvds-nl@245-138-045-062.dynamic.caiway.nl) |
| 19:24.16 | *** part/#asterisk wvds-nl (~wvds-nl@245-138-045-062.dynamic.caiway.nl) |
| 19:24.41 | fifer | @Qwell: Sure there is, I have helped myself. Often it is not at all for sure that it is a hardware problem. Informed/expert troubleshooting can often provide a solution without changing hardware. I'm just running out of things to try |
| 19:25.15 | *** join/#asterisk Elad (~dalemccra@174-23-31-194.slkc.qwest.net) |
| 19:25.18 | fifer | I have already reduced the issue by a factor, but it is still there ( audible clicking once every 30-120 sec) |
| 19:25.53 | fifer | I'm just trying to do the best I can with the situation and I do apreciate the help I have already recieved here during the past week. |
| 19:26.22 | *** join/#asterisk ManxPower-work (~manxpower@139.sub-75-234-63.myvzw.com) |
| 19:26.34 | Katty | i sure these polycoms had a fun lil icon for when their httpd is up |
| 19:26.44 | Naikrovek | audible click once every 30-120s? I get 120 clicks per second. |
| 19:26.46 | fifer | Katty: Thanks! :-) |
| 19:27.04 | Naikrovek | Katty: it comes up when the main phone ui comes up for me |
| 19:27.28 | Katty | Naikrovek: takes mine a couple minutes |
| 19:27.38 | Naikrovek | diff't model than me |
| 19:27.46 | Katty | probably. these are older |
| 19:27.59 | Naikrovek | i'm 100% 3[32][01] |
| 19:28.11 | Naikrovek | 320, 321, 330, 331 |
| 19:28.18 | Katty | this particular one sitting on my desk is a 501 i believe |
| 19:28.21 | Katty | but it could be a 500 |
| 19:36.14 | *** join/#asterisk nightrid3r (~kvirc@adsl196-63-161-217-196.adsl196-14.iam.net.ma) |
| 19:36.21 | Elad | I am looking at setting up an asterisk server and was hoping to have some questions answered. Are these systems cost efficient for an office of 10 or less people? Seems like you need to get a T1 to provide the ability for multiple calls to take place at once. Or am I misinformed? |
| 19:36.35 | AeroCloud | misinformed |
| 19:36.50 | seanbright | question |
| 19:37.08 | seanbright | when you have multiple people join a MeetMe |
| 19:37.12 | AeroCloud | each call uses a certain amount of bandwidth based on the codecs you are using |
| 19:37.23 | AeroCloud | you just need to have internet to support that amount of bandwidth |
| 19:37.27 | seanbright | actually, nevermind. |
| 19:37.31 | AeroCloud | you can use SIP termination |
| 19:37.58 | [TK]D-Fender | Elad: Cost effective compared to what? * can use the same kinds of connectivity as any other system. * doe not make such connectivity CHEAPER. * just makes the PBX core & control YOURS |
| 19:38.35 | AeroCloud | seanbright: you answered yourself? |
| 19:38.45 | [TK]D-Fender | Elad: Want to use a T1? Sure. What to use an ITSP? Sure. Want to use analog POTS lines? Sure. |
| 19:38.53 | seanbright | AeroCloud: yes |
| 19:38.57 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 19:38.59 | Elad | [TK]D-Fender, we currently use Packet8, and we are looking at adding on 8 new users, so I am trying to figure out which method will be cheaper for phones. Pay $30/per month/per user or switch to something like Asterisk |
| 19:39.11 | AeroCloud | we used a Business Cable connection at one of my past work places |
| 19:39.16 | [TK]D-Fender | Elad: Wire-line connection require an appropriate piece of interface hardware of course |
| 19:39.18 | AeroCloud | cheaper than T1 for sure |
| 19:40.08 | [TK]D-Fender | Elad: What does that $30/mo actually give you? See they probably bundle support for a certain number of phoens & lines , plus LD, etc al together making it harder to evaluate the value they offer you |
| 19:42.56 | Kobaz | Elad: how many users do you have already? 8 users at 30 bucks a month is $240, which is about the price of a t1 |
| 19:43.04 | Kobaz | Elad: and you'll get much better call quality |
| 19:43.10 | Naikrovek | Elad: i have a T1 carrying voice and data, and about 80 phones. I pay for the data T1, I paid $1500 for the server, and $100/phone. I have 8 channels (think of them as simultaneous calls in to or out of the building) and I pay $44 each, monthly, with free calls to US, Canada, and parts of Europe |
| 19:43.40 | AeroCloud | Dont forget the cost of hosting/managing the server etc |
| 19:44.12 | Naikrovek | well i host mine in the office and that but yes, he'll need to consider that |
| 19:44.30 | AeroCloud | if you arent a strong server administrator, know * |
| 19:44.42 | Naikrovek | just had to show his accountant how to use quickbooks. |
| 19:44.43 | AeroCloud | you might just be better off paying someone else, unless your willing to spend hours learning it |
| 19:44.44 | [TK]D-Fender | People please be SPECIFIC baout what your T1 carries. Is it DATA, or VOICE. You can get 8 channels over DSL easily enough |
| 19:44.58 | [TK]D-Fender | T1 Voice requires an interface card. Data does not. |
| 19:45.14 | AeroCloud | if you use SIP phones, you can do DSL or multiple DSL's |
| 19:45.21 | AeroCloud | lower your cost |
| 19:45.25 | Kobaz | [TK]D-Fender: data t1 still requires hardware to interface... via either card or gateway |
| 19:45.52 | bmoraca | data needs a DSU, voice needs a CSU |
| 19:46.04 | bmoraca | fractional data also needs a CSU |
| 19:46.05 | [TK]D-Fender | Kobaz: Most ISP's will provide you a router, etc... I'm talking about not needed a T1 Voice card for your * server <- |
| 19:46.22 | Elad | We already have a spare server that I was going to put everything on (in the office), we have a bunch of Packet8 phones (that I am researching to see if they are compatible with Asterisk), and I was looking at different connection methods to determine which card I need to buy. Then I was going to look at overall cost for the different options we are considering |
| 19:46.42 | Kobaz | packet8 is just sip |
| 19:46.48 | [TK]D-Fender | Correct |
| 19:46.56 | AeroCloud | asterisk supports SIP, and SIP is cheapest |
| 19:47.30 | *** join/#asterisk jtodd (mq2ii8rnmh@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 19:47.30 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:47.35 | AeroCloud | you will need to find SIP termination and buy enough channels to support your maximum concurrent users |
| 19:49.31 | Elad | sorry, I am not finding a lot of information on google, is SIP Termination a service I pay for that will allow some company to route phone numbers to my IP where the Asterisk machine would catch them, and then distribute to the phones in my office? |
| 19:50.24 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:50.46 | Naikrovek | Elad: yes. |
| 19:50.54 | Naikrovek | Elad: companies that do that are called ITSPs |
| 19:50.55 | wcselby | Elad - look for ITSP's or VoIP providers. A lot of them use the term "Sip Trunk", even though that's really a misnomer |
| 19:50.56 | Naikrovek | I think.. |
| 19:51.07 | wcselby | Naikrovek - yeah it's an ITSP |
| 19:51.14 | wcselby | internet telephony service provider |
| 19:51.16 | wcselby | or something like that |
| 19:51.45 | wcselby | anyone in here use vitelity.net? |
| 19:51.48 | AeroCloud | each channel you purchase allows that many concurrent calls |
| 19:52.17 | AeroCloud | some providers sell flat unlimited rates for us/canada, others cheap monthly and per minute rates |
| 19:52.40 | Naikrovek | and you can have as many internal calls as your server can handle, it's the outgoing and incoming calls that count against that channel number. |
| 19:52.42 | AeroCloud | you will have to figure out what works best for your company |
| 19:53.16 | AeroCloud | Naikrovek: thanx for clarifying what could be confusion |
| 19:53.49 | AeroCloud | if you have 16 phones, doesnt mean you need 16 channels |
| 19:53.57 | AeroCloud | you might be good with 4 or 5 |
| 19:54.05 | Naikrovek | i remember one dude coming in here getting confused by it then swearing off of voip forever |
| 19:54.21 | Naikrovek | i don't think he'll stick to that but i felt it needed clarification |
| 19:54.30 | AeroCloud | thanx |
| 19:54.31 | Elad | AeroCloud, so you are saying if the most people that will be on the phone at once is 5 out of the 10, then I only need 5 channels? |
| 19:54.36 | AeroCloud | yes |
| 19:54.40 | Naikrovek | Elad: yes |
| 19:54.48 | Naikrovek | Elad: i have 80 phones, but only 8 channels |
| 19:54.53 | Elad | I should clarify, 5 outbound/inbound calls at once |
| 19:55.05 | AeroCloud | outbound ^ |
| 19:55.07 | Naikrovek | every phone in the place could call another phone in the office and not use a single channel |
| 19:55.14 | Elad | got it |
| 19:55.15 | AeroCloud | inbound is limited by your DID providers channels |
| 19:55.20 | Naikrovek | but only 8 people can call home at once |
| 19:55.29 | wcselby | Naikrovek - you ever setup international digitmaps for polycoms? |
| 19:55.38 | Naikrovek | wcselby: yeah |
| 19:55.46 | Elad | I appreciate you guys answering my questions, I was kind of confused by everything and wasn't sure where to start |
| 19:55.55 | Naikrovek | wcselby: but i shortcut them, i don't account for every possible digit length |
| 19:56.01 | AeroCloud | * can be cheaper, but it takes more management time |
| 19:56.02 | wcselby | that's what I need |
| 19:56.07 | wcselby | could you share a snippet? |
| 19:56.11 | AeroCloud | once you get it setup its easy :) |
| 19:56.16 | Naikrovek | wcselby: stand by |
| 19:56.30 | Elad | I am sure we will just use a "vanilla" install for a while |
| 19:56.40 | AeroCloud | it doesnt quite work that way |
| 19:56.55 | AeroCloud | unless your using a GUI, then we cant support that in this channel |
| 19:57.04 | Katty | Skeeter-: so far the network card hasn't idled out |
| 19:57.12 | Naikrovek | wcselby: lol what i have won't help you. hehe i'm so stupid. 011xxx.T |
| 19:57.13 | Naikrovek | lol |
| 19:57.13 | Katty | Skeeter-: i ended up completely apt-get removing the networkmanager thing |
| 19:57.18 | Skeeter- | Katty, good to hear |
| 19:57.18 | AeroCloud | you will have to learn about dialplans and build a dialplan that works for your company |
| 19:57.20 | Katty | Skeeter-: just disabling gdm didn't do much |
| 19:57.23 | Naikrovek | wcselby: i guessed i shortened that up a while ago |
| 19:57.38 | Katty | Skeeter-: we're about to slide the server in tonight at 5PM...about 2 hours |
| 19:57.39 | p3nguin | elad: Termination is for calling out of your system to the PSTN; origination is when calls from from the PSTN through your ITSP to you. |
| 19:57.40 | wcselby | Naikrovek - yeah that's what we have, but my user isn't able to enter anything other than 011+ plus the country code before the dialplan tries to dial it |
| 19:57.46 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 19:57.49 | Katty | Skeeter-: the real test will be tonight (= |
| 19:57.59 | p3nguin | elad: And you don't have to use SIP, you can easily use IAX2. |
| 19:58.05 | Skeeter- | Katty, keep me up to date |
| 19:58.08 | Skeeter- | tomorrow.. |
| 19:58.10 | Skeeter- | :) |
| 19:58.22 | Naikrovek | wcselby: really. the . should allow you eo enter any number of digits as long as you don't wait T seconds between digits |
| 19:58.23 | cusco | what test Katty ? |
| 19:58.39 | cusco | any number > 1 |
| 19:58.41 | Katty | cusco: just updating our production server |
| 19:58.52 | wcselby | Naikrovek - where is T defined, that's probably my error |
| 19:59.11 | cusco | Katty: from what to what? |
| 19:59.13 | Naikrovek | wcselby: T defaults to 3 but you can override it |
| 19:59.13 | cusco | svn? |
| 19:59.13 | cusco | :p |
| 19:59.15 | AeroCloud | _X. matches anything |
| 19:59.35 | Katty | cusco: 1.4.somethingsomething to 1.6.2 |
| 19:59.44 | cusco | hmmm... |
| 19:59.48 | *** join/#asterisk jmacz (~jmacz@190.25.40.70) |
| 19:59.55 | cusco | manny cli commands have changed meanwhle |
| 19:59.59 | cusco | and functions |
| 20:00.01 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 20:00.02 | cusco | slight changes... |
| 20:00.11 | Katty | not many |
| 20:00.14 | cusco | like CALLERID(number) is now CALLERID(num) |
| 20:00.19 | rocksfrow | anybody have customer service headset suggestions? (good noise cancel, good voice quality, comfortable) |
| 20:00.19 | Katty | and the ones that have changed are plastered all over 1.4 cli |
| 20:00.24 | Katty | WARNING CHANGING TO BLAHBALH |
| 20:00.29 | *** join/#asterisk KingDavidNYC (~Chris1232@rrcs-69-193-218-18.nyc.biz.rr.com) |
| 20:00.30 | cusco | ok |
| 20:00.42 | KingDavidNYC | Hello |
| 20:00.42 | Katty | just little stuff like stop now is now core stop now |
| 20:00.50 | AeroCloud | cucso didnt CALLERID(num) work in 1.4 also |
| 20:01.14 | cusco | AeroCloud: maybe, I don't know. All I know is that it is deprecated in 1.6.2 and no longer works |
| 20:01.24 | Katty | cusco: my backup script has changed slightly |
| 20:01.30 | Katty | cusco: due to directory names |
| 20:01.53 | cusco | Katty: I guess you will have to know about changes you did not predict once you move on to 1.6 :P |
| 20:02.02 | cusco | maybe you got them all! |
| 20:02.10 | p3nguin | rocksfrow: monaural or binaural? |
| 20:02.16 | rocksfrow | bi |
| 20:02.22 | cusco | just yesterday I found out soft hangup no longer works |
| 20:02.32 | rocksfrow | well, i was thinking binaural..what do you prefer? |
| 20:02.34 | Katty | yeah it's another command |
| 20:02.36 | Naikrovek | rocksfrow: i use my xbox360 headset from time to time, but not noise cancelling |
| 20:02.39 | rocksfrow | i'm thinking binaural will have better noise cancel |
| 20:02.39 | Naikrovek | :D |
| 20:02.40 | cusco | channel request hangup |
| 20:02.41 | Katty | yeah |
| 20:03.06 | p3nguin | rocksfrow: Plantronics H251N for mono, H261N for bi. |
| 20:03.20 | rocksfrow | yeah, the h261N is what i was looking at actually |
| 20:04.08 | cusco | and out last update was more than a week ago |
| 20:04.11 | p3nguin | rocksfrow: Good choice. I personally use an H251 (mono, voice tube (not noise cancelling)), and I've been wanting to upgrade to the H261N. |
| 20:04.18 | cusco | so I might find ou that there other stuff not working :p |
| 20:04.47 | *** join/#asterisk norrec (~norrec@76-201-85-28.lightspeed.frokca.sbcglobal.net) |
| 20:05.07 | rocksfrow | p3nguin, what do you think of the polycom 550's ? |
| 20:05.31 | p3nguin | rocksfrow: My only reserve is spending the $65 or $70 on a headset when I already have a working one. I don't use Polycoms. |
| 20:05.50 | rocksfrow | p3nguin, right..makes sense..what phones do you use? |
| 20:05.54 | [TK]D-Fender | IP 550 = waste |
| 20:06.03 | p3nguin | rocksfrow: I use Cisco 7900 series. |
| 20:06.11 | rocksfrow | oh realllly |
| 20:06.13 | [TK]D-Fender | A bastart inbetween product that doesn't fit in the lineup |
| 20:06.23 | rocksfrow | hrm...im in a debate between cisco and polycom |
| 20:06.32 | rocksfrow | just can't wait to get off these damned grandstreams! |
| 20:06.39 | [TK]D-Fender | IP 335/450 sit in the mid-rage, IP 650 on the high. No place for the 550 |
| 20:06.43 | rocksfrow | p3nguin, could you pursuade me one way or the other? |
| 20:06.48 | sbrath | rocksfrow: I used http://www.smithcoronaheadsets.com/ good quality, they sent me headsets to try free of charge, and 30 days to evaluate. |
| 20:06.52 | Qwell | ~cisco-licensing |
| 20:06.54 | p3nguin | rocksfrow: Depending on your ties with Cisco, you might prefer Polycom. |
| 20:06.55 | socain | Any idea why I can't arrow through on-hold calls with a new Polycom IP650 (SIP 3.2.2)? |
| 20:07.01 | Qwell | ~cisco licensing |
| 20:07.02 | norrec | rocksfrow, ugh, i'm trying to get off grandstreams myself |
| 20:07.04 | sbrath | and nothing over 50$ most were like 39$ |
| 20:07.06 | Qwell | stupid bot |
| 20:07.06 | KingDavidNYC | can anyone please help me with a question I have about queues? |
| 20:07.16 | cusco | ~stupid bot |
| 20:07.17 | infobot | Stupid human. |
| 20:07.17 | rocksfrow | norrec, i feel for you.. |
| 20:07.20 | cusco | :p |
| 20:07.29 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:07.32 | rocksfrow | sbrath, thanks i'll check em out |
| 20:07.49 | *** join/#asterisk Alagar (~Administr@122.164.42.111) |
| 20:07.54 | sbrath | rocksfornow: And if your doing grandstream, make sure you DON't use the rj11 jack, get the 2.5mm plugs. |
| 20:08.01 | norrec | rocksfrow, u should take a look at snom, the inital deployment is kind of a bitch but it seems like a good compromise between polycoms and grandstreams |
| 20:08.38 | rocksfrow | sbrath, no, ordering new phones at the same time of the headsets |
| 20:08.46 | rocksfrow | really leaning towards the polycom phones, and plantronics headsets |
| 20:09.07 | p3nguin | Headsets I definitely say Plantronics, but the phone choice is going to be up to you. |
| 20:09.11 | rocksfrow | i have 20-30 grandstreams i want to go office space on :-p |
| 20:09.43 | sbrath | rocksfornow: which model grandstreams? |
| 20:11.02 | rocksfrow | nice voiplink has bulk discounts1 |
| 20:11.44 | sbrath | rocksfornow: also check telephonydepot.com for phones. |
| 20:12.02 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 20:12.29 | rocksfrow | sbrath, ~10 2000's, and a bunch of the budgetone 200s |
| 20:12.52 | sbrath | ya, I'm using 2020's and 2010's with good luck. For now.. |
| 20:13.04 | rocksfrow | sbrath, no freezing? |
| 20:13.10 | sbrath | what are the 2000's doing for you? |
| 20:13.12 | rocksfrow | you're using PoE? |
| 20:13.19 | rocksfrow | sbrath, freezing.. |
| 20:13.20 | sbrath | I'm not using PoE. |
| 20:13.26 | rocksfrow | sbrath, ah.. |
| 20:13.34 | rocksfrow | well, the 200's aren't on PoE, and still randomly freeze |
| 20:13.35 | sbrath | How do they freeze? How can I tell |
| 20:13.39 | rocksfrow | unplug/plug back in and they come back up fine |
| 20:13.49 | rocksfrow | sbrath, the phones will go inactive in the FOP panel |
| 20:14.23 | rocksfrow | other than that I haven't had major major problems.. |
| 20:14.41 | sbrath | I've had a few occurances of asterisk not being able to deliver a SIP call, and every phone say's it's trying to register, but I'm thinking that's a asterisk issue since if I restart asterisk the phones come back on-line. |
| 20:14.52 | rocksfrow | fortunately most people in the office don't use their phones, i'm replacing all phones that are used a lot |
| 20:14.54 | rocksfrow | (customer service) |
| 20:15.14 | sbrath | FOP, so you're using FreePBX or a distro? |
| 20:15.21 | rocksfrow | sbrath, yeah..the issue i have is total freezing..you can see the clock on the LED screen frozen |
| 20:15.36 | sbrath | what firmware are you running? |
| 20:15.41 | rocksfrow | sbrath, asterisknow/freepbx |
| 20:15.59 | rocksfrow | sbrath, i haven't flashed them |
| 20:16.03 | rocksfrow | (i know) |
| 20:16.07 | rocksfrow | ~gs |
| 20:16.08 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:16.13 | rocksfrow | (lol) |
| 20:16.17 | rocksfrow | i just love that |
| 20:16.55 | sbrath | I wish I had the budget for Polycoms, but my Merlin converts like the "Keysystem" like phones with all the lights... |
| 20:17.02 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.10 | rocksfrow | sbrath, i'm also attracted to the ftp provisioning for the polycom phones |
| 20:17.15 | rocksfrow | would be quite convienient |
| 20:17.29 | nightrid3r | i'm looking for sip/iax phones that don't need to get firmware from a server, |
| 20:17.40 | sbrath | For the grandstreams I'm using gsutil, which is a perl script that will find all the phones and update their configs ;) |
| 20:17.50 | rocksfrow | nightrid3r, i dont think my grandstreams do |
| 20:17.59 | rocksfrow | sbrath, oh? interesting.. |
| 20:18.19 | rocksfrow | [TK]D-Fender, which polycoms do you suggest then? |
| 20:18.19 | sbrath | almost all phones now have the firmware local, except Cisco which are a PITA to setup :) |
| 20:18.30 | *** part/#asterisk GreyFoxx (greg@out.of.phaze.org) |
| 20:18.32 | p3nguin | Cisco phones are simple to set up. |
| 20:18.52 | sbrath | ok, if you have a contract with cisco and the software, then yes .. |
| 20:18.59 | socain | Any Polycom users know how to configure the phone where you press *X, then press speed dial contact, and it sennds the *X and the speed dial number to the PBX? Right now it just wipes out the previous input and dials the extension... |
| 20:19.01 | p3nguin | If you think they are difficult, I believe that you have never done it. |
| 20:19.25 | [TK]D-Fender | rocksfrow: 321/331/335/450/650 depending |
| 20:19.49 | Naikrovek | rocksfrow: i use polycom and i love them. i have 320s, 321s, and 330s. they're inexpensive, excellent, and my users seem to like them, except one old guy who thinks the bell style handset can't be beat |
| 20:19.49 | [TK]D-Fender | rocksfrow: Actually... I probably wouldn't suggest 331... the 335 might be a better option in that bracket |
| 20:19.53 | rocksfrow | [TK]D-Fender, so you like a 3-line phone vs a 4-line? |
| 20:20.02 | Naikrovek | yeah 335 is good |
| 20:20.05 | [TK]D-Fender | rocksfrow: # of lines almost never matters |
| 20:20.14 | Naikrovek | 335s can handle up to 8 calls i think |
| 20:20.14 | sbrath | I've setup a Cisco 7960.... |
| 20:20.19 | [TK]D-Fender | rocksfrow: Every polycom can handle at least 4 calls anyway |
| 20:20.23 | rocksfrow | [TK]D-Fender, yeah..i was going to say i only use one line on my 4-line grandstreams |
| 20:20.28 | rocksfrow | but who knows what i'd want to do |
| 20:20.31 | rocksfrow | i plan on using separate lines |
| 20:20.31 | p3nguin | sbrath: Then you know it's a breeze. |
| 20:20.32 | sbrath | And I use it at home now, but I had to setup a tftp server, configure a XML file, ... |
| 20:20.36 | rocksfrow | for separate calling queues, perhaps |
| 20:20.40 | *** join/#asterisk knarfly (~vlad@c-98-242-237-166.hsd1.fl.comcast.net) |
| 20:20.54 | p3nguin | I think it's a lot less bother than Polycoms. |
| 20:21.16 | [TK]D-Fender | socain: Polycom speed-dials will not do in-call DTMF. its always new call/target for Polycom feature |
| 20:21.28 | knarfly | I'm running asterisk-1.6.0.21 on FreeBSD...I used to run ztdummy in order to use the meetme functions but now it is dahdi that is required??? |
| 20:21.32 | sbrath | I agree the polycoms are a headache in setting up all the XML files for dialplans and such for the phone, but once it's configured they are nice. |
| 20:21.41 | [TK]D-Fender | rocksfrow: I manipulate CID to indicate queue calls |
| 20:21.59 | rocksfrow | [TK]D-Fender, right. |
| 20:21.59 | Naikrovek | sbrath: i have a set of scripts that i'll share with anyone - makes it easy |
| 20:22.01 | knarfly | sbrath: stay the heck away from Grandstream too...I got burned big time on these |
| 20:22.07 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:22.10 | rocksfrow | knarfly, hehe..can you elaborate? |
| 20:22.50 | Naikrovek | i had grandstream. they would echo and not hang up when you put the headset down, one of them never rang. |
| 20:23.03 | Naikrovek | they are now in a landfill |
| 20:23.03 | knarfly | rocksfrow: I bought three phones from GS...a BudgetTone 200 and two GXP2000 |
| 20:23.18 | knarfly | the BT200 is the only one out of the three that works now. |
| 20:23.26 | rocksfrow | knarfly, funny.. |
| 20:23.29 | rocksfrow | i have a bunch of 200's too |
| 20:23.32 | rocksfrow | and some 2000's |
| 20:23.36 | [TK]D-Fender | cehckout time, later all |
| 20:23.37 | sbrath | rocksfornow: I also manipulate CIDNAME to indicate call queues, since the phones can handle 30 characters for CIDNAME, and the phone company only really provides about 15 characers.. |
| 20:23.39 | rocksfrow | the 200's randomly freeze up on me |
| 20:24.08 | knarfly | I used the GXP2000 for a short time, then apparently they did a firmware update and both of them went south within hours of each other...and the vendor said I was SOL |
| 20:24.17 | Naikrovek | nice |
| 20:24.28 | rocksfrow | heh, nice |
| 20:24.53 | knarfly | yes, nice to throw $100 into the garbage cans |
| 20:25.01 | knarfly | make that $180 |
| 20:25.02 | sbrath | put them on ebay :) |
| 20:25.08 | p3nguin | sbrath: My problem with that is that I could manipulate the CID/name immediately before sticking the call into a queue, but then while the caller is waiting (s)he can dial an extension to reach someone directly. It would be confusing. |
| 20:25.15 | knarfly | no, I wouldn't do anything like that |
| 20:26.03 | knarfly | so does anyone know how to run meetme without a timing device? |
| 20:26.12 | *** join/#asterisk cesar_CR (~cesar@190.10.115.176) |
| 20:26.28 | *** join/#asterisk timeshell_atwork (~timeshell@gateway.airnet.ca) |
| 20:26.31 | p3nguin | I suppose I could throw in another context where the CID change would get undone when the caller pops out of the queue. |
| 20:26.41 | sbrath | p3nguin: you mean that others would be confused as to why the callerid(name) had "CS" or something prepended ? |
| 20:27.04 | sbrath | ok, what you just said, that's what I do :) |
| 20:27.08 | p3nguin | sbrath: That was my thought, but I think I could work around it with another context to strop those back off. |
| 20:27.18 | p3nguin | strip, rather |
| 20:27.27 | sbrath | strop is better :) |
| 20:27.38 | p3nguin | Strop tease! |
| 20:28.07 | seanbright | anyone have any first hand experience with this beast: |
| 20:28.07 | seanbright | http://www.howlertech.com/screamer-card/ |
| 20:28.08 | sbrath | I think it's nice to see that the call I'm getting was actually originally desin for CustomerService, I know how to answer then. |
| 20:28.10 | p3nguin | Now I have dialplan changes to design. |
| 20:28.48 | p3nguin | sbrath: Absolutely. |
| 20:29.25 | hardwire | priority jumping (large offsets) is getting me into a lot of trouble lately |
| 20:29.36 | wcselby | anyone have any contacts with AT&T sip sales? |
| 20:29.49 | hardwire | wcselby: I do.. shoot me your email |
| 20:30.32 | p3nguin | sbrath: I hate simply saying, "This is Rob. May I help you?" It would be much better to be able to say, "Support, this is Rob. May I help you?" if I knew they were looking for support rather than having a problem with a bill. |
| 20:31.30 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:31.31 | p3nguin | Of course if there were no problems with billing ever, that could also be useful. :/ |
| 20:32.04 | sbrath | I have my caller-id macro do a mysql dip to lookup the CID and find the customer# and customer type, and prepend it to the callerid-name, I was also messing with the callerid-number but that pissed of the Transfer stuff, and nobody could transfer those calls :) |
| 20:32.57 | sbrath | I also have that macro check calls to direct lines, and if a known person is calling, it steals their call and sends it to a IVR, even if they dialed me directly because they did a Call-Back :) |
| 20:33.57 | *** part/#asterisk simcop2387 (~simcop238@p3m/member/simcop2387) |
| 20:34.15 | rocksfrow | does anybody else think the 335's are ugly? lol |
| 20:35.23 | wcselby | i do |
| 20:35.33 | wcselby | i haven't liked the look of any of the sub 500 polycom phones |
| 20:35.39 | wcselby | but I'm biased I guess :) |
| 20:35.53 | p3nguin | Great. A power outage. |
| 20:35.56 | Naikrovek | i like the 335s |
| 20:36.09 | Naikrovek | but yes the higher end ones look better |
| 20:36.41 | hardwire | it's nice being able to have exten,5000(name),dostuff included into the current context |
| 20:36.48 | hardwire | but there's a lot of room for overlap |
| 20:37.57 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 20:40.57 | rocksfrow | i dono these 550s i like better |
| 20:40.58 | rocksfrow | lol |
| 20:41.30 | rocksfrow | all of these are 2 calls per line |
| 20:41.36 | rocksfrow | do you guys know of any phones that can do more? |
| 20:41.47 | sbrath | rocksfrow: grandstream :) |
| 20:41.49 | sbrath | He He |
| 20:41.55 | rocksfrow | sbrath, i know!! lol |
| 20:41.58 | ariel_ | 2 calls per line? |
| 20:42.01 | sbrath | Why would you need more than 2 calls anyway? |
| 20:42.47 | rocksfrow | sbrath, ..i guess i won't/wouldn't |
| 20:42.50 | sbrath | I guess 2 inbound calls, one on hold, and then doing an attended transfer. |
| 20:42.51 | ariel_ | with asterisk, there is always conference |
| 20:43.08 | ariel_ | parking |
| 20:43.10 | rocksfrow | right..yeah |
| 20:43.21 | rocksfrow | im leaning towards these 550s |
| 20:43.24 | rocksfrow | even though i know its a waste |
| 20:43.30 | rocksfrow | but they're so much nicer looking, heh |
| 20:44.26 | Skeeter- | POLYCOM FTW |
| 20:44.34 | Skeeter- | sry for da caps |
| 20:44.37 | rocksfrow | ftw? |
| 20:44.50 | Qwell | rocksfrow: Free The Whales |
| 20:45.01 | Skeeter- | flick the world or for the win |
| 20:45.05 | Naikrovek | rocksfrow: my Polycom 321 can do 4 calls per line key |
| 20:45.13 | Naikrovek | for a total of 8. the phone cost $80 |
| 20:45.31 | Skeeter- | Naikrovek, refurb/stolen? |
| 20:45.35 | rocksfrow | Naikrovek, • Up to 2 lines with up to 2 calls per line |
| 20:45.36 | Naikrovek | no |
| 20:45.42 | Naikrovek | not stolen or refurb |
| 20:46.01 | Skeeter- | rocksfrow, play with the .cfg files |
| 20:46.05 | Naikrovek | rocksfrow: come over here and watch as this phone does 8 calls simultaneously |
| 20:46.11 | Skeeter- | my 550 takes 6 lines per key |
| 20:46.14 | Naikrovek | Skeeter-: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-321 |
| 20:46.51 | Skeeter- | Naikrovek, Nice. |
| 20:47.00 | Naikrovek | i buy all my phones from these guys |
| 20:47.02 | Naikrovek | they're good |
| 20:47.30 | Naikrovek | and the phones are very capable indeed |
| 20:47.35 | wcselby | Naikrovek - are those backlit? |
| 20:47.37 | Naikrovek | not had a single problem with them that i didn't cause |
| 20:47.41 | Naikrovek | wcselby: no, but the 335s are |
| 20:47.47 | wcselby | yeah |
| 20:47.53 | wcselby | i'd like a backlit phone for my office at home |
| 20:47.58 | Naikrovek | same |
| 20:47.59 | wcselby | i've got a 7941 |
| 20:48.02 | wcselby | at home |
| 20:48.07 | wcselby | it doesn't have the backlit |
| 20:48.14 | wcselby | have to go to 7945 I think for that |
| 20:48.16 | Naikrovek | http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-335 |
| 20:48.19 | wcselby | makes the price go ^^^^ |
| 20:48.32 | Naikrovek | yes |
| 20:48.46 | wcselby | just as you like the polycoms, I like the ciscos |
| 20:48.47 | wcselby | lol |
| 20:48.52 | Naikrovek | picture on that site doesn't show it but the 335 supports G722, Siren7 and Siren14 |
| 20:48.59 | wcselby | although, qwell yells at me about the cisco licensing |
| 20:49.14 | Naikrovek | yeah |
| 20:49.24 | Naikrovek | supposed to have a call manager license even if you don't use call manager |
| 20:49.31 | Qwell | wcselby: Because I respect copyright. |
| 20:49.40 | Naikrovek | which i think is BS but if that's their license, then that's their license |
| 20:50.14 | Naikrovek | gotta follow it or you're vulnerable legally |
| 20:51.13 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:51.15 | sbrath | is anyone using 1.6.2.6 yet? I'm on 1.6.2.0 and thinking to upgrade. |
| 20:51.33 | Naikrovek | sbrath: is there something on 1.6.2.6 that you need |
| 20:51.51 | ariel_ | wow, I am still on 1.6.0.25 |
| 20:52.15 | sbrath | is faxdetect=yes also on 1.6.2.0 ? |
| 20:52.22 | hardwire | yes |
| 20:52.34 | hardwire | I'm using 1.6.2.6 |
| 20:52.41 | hardwire | and faxdetect should be in 1.6.2.0 |
| 20:52.58 | wcselby | i think i'm on 1.6.2.2 still |
| 20:53.41 | ariel_ | hardwire: in 1.6 for single calls to sip device call-limit works but only if you setup the sip as peer. |
| 20:53.43 | sbrath | I've been running 1.6.2.0 since it came out with no issues, then just starting 2 days ago, it started locking all the sip endpoints out. I do sip show peers and they are all there, but no phones can register, I restart asterisk and all the phones are back online. |
| 20:54.26 | sbrath | seems to happen about once a day, usually at 8am or 6pm, so just as I'm waking up, or walking out ... :( |
| 20:54.54 | ariel_ | cron job to restart just prior |
| 20:54.57 | sbrath | what's more exciting, is that there are no indications in the logs of an error. |
| 20:55.17 | sbrath | ariel_: I'd considered the microsoft approach, but it makes me feel dirty... |
| 20:55.31 | ariel_ | I was just kidding |
| 20:55.40 | sbrath | I'd have to slower longer in the morning to make me feel right.. |
| 20:56.08 | sbrath | s/slower/shower/ |
| 20:56.16 | sbrath | neat :) |
| 20:56.22 | *** join/#asterisk timeshell_atwork (~timeshell@gw.lusi.on.ca) |
| 20:56.40 | ariel_ | I just finished 4 days of testing with people from France (Alcatel) trying to show off there system compared to my Asterisk based Gateways, so I am a bit abrasive today |
| 20:56.40 | sbrath | infobot is smart ;) |
| 20:57.17 | sbrath | are they hard to deal with, and was asterisk better? |
| 20:57.23 | hardwire | ariel_: rough |
| 20:57.38 | sbrath | Alcatel == Lucent right ? |
| 20:57.48 | hardwire | they are lovers |
| 20:57.51 | ariel_ | yes but this is an actual Alcatel PBX |
| 20:58.14 | hardwire | ariel_: did they compete well? |
| 20:58.14 | ariel_ | they think there smarter then us America's in there words. and that there system can do no wrong... |
| 20:58.46 | ariel_ | oh, they left blown away with all we can do via your simple setup in there view. |
| 21:00.10 | hardwire | ariel_: yeh.. I like that some PBXs offer absolutely no choice as far as hardware interop |
| 21:00.13 | hardwire | their pbx.. their phones |
| 21:00.17 | ariel_ | we can trunk easier, we can ring devices on other servers like they are connected to same system. We have a better access to caller ID info and much more |
| 21:00.20 | hardwire | you'd think they would have the most amazing stuff in the world |
| 21:00.34 | hardwire | but I like that asterisk allows for interop with anything that likes specs :) |
| 21:01.21 | ariel_ | we ended up having to do the pin colleciton on our gateway, and even do the LCR due to there trunks only like sending called number and not name. |
| 21:01.35 | ariel_ | there attitude was why we need E1 or trunk to send more. |
| 21:01.49 | jksM | is just distracted by there there |
| 21:02.08 | hardwire | ariel_: ew |
| 21:02.20 | hardwire | yuo guys using a custom rate engine? |
| 21:02.33 | p3nguin | If the stupid power doesn't come back on soon, my UPS batteries are going to be exhausted. |
| 21:02.43 | sbrath | any sugestions on a Fax T38 provider that will take a ported-in number in the US? |
| 21:02.58 | ariel_ | but the good part is that we proved to them we can plug into there PBX and not cause any issues to them. So we got a nice Certification from them |
| 21:03.28 | hardwire | p3nguin: this is why I have a farm of hamsters and a large array of hamster wheels |
| 21:03.28 | ariel_ | hardwire: radius engine not really custom but works |
| 21:03.40 | hardwire | ariel_: how does it handle route selection? |
| 21:03.48 | p3nguin | The hamsters got flooded out. |
| 21:03.50 | hardwire | we use a2billing at the moment.. I'm creating my own system. |
| 21:03.52 | ariel_ | by account codes |
| 21:03.57 | p3nguin | I figure that's also why the power is off. |
| 21:04.08 | hardwire | ariel_: that wouldn't be LCR then right? |
| 21:04.22 | ariel_ | the way we use it yes |
| 21:04.28 | hardwire | interesting |
| 21:04.31 | p3nguin | I'm down to about 11 minutes left on battery, then the gateway and modem will be offline. :( |
| 21:04.41 | hardwire | p3nguin: I'll pray for you |
| 21:04.52 | sbrath | when you disconnect we know the battery is gone... |
| 21:05.01 | AeroCloud | backup generator :) |
| 21:05.05 | ariel_ | we do mysql look up for the contry code then city then get the account code then route that via our dialing rules |
| 21:05.06 | p3nguin | I don't even know the runtime on the battery for the Asterisk box, but once the gateway is offline, it won't matter anyway. |
| 21:05.37 | ariel_ | our mysql has a store policy that updates info via the Radius |
| 21:05.42 | sbrath | we just had that same thing happen, so I went out and bought some mondo batteries, I can go for 3-4 hours on battery now, and if I shutdown all the dev servers probably more like 6.... |
| 21:06.09 | rocksfrow | does anybody know any telephonydepot coupon codes? :-p |
| 21:06.10 | *** join/#asterisk Madoc (~Madoc@bas3-ottawa23-1177800811.dsl.bell.ca) |
| 21:06.18 | seanbright | a2 infotech : problem solved |
| 21:06.24 | seanbright | is that irony? |
| 21:06.26 | seanbright | :-) |
| 21:06.43 | Madoc | Is there a an easy way to alter the makefile or something so I can link libsox to my custom app? |
| 21:07.10 | ariel_ | if there is a rate change it just gives us the new account code to use and we route on that. Then post our cdr's to the mysql db in i_calls then the store procedure updates the Radius server. |
| 21:07.17 | p3nguin | I don't know about irony, but it's certainly annoying. |
| 21:07.27 | AeroCloud | I want a new feature: HangupAndGoto(), this will be able to be run via application map, and disconnect the current callee, then goto the context, extension, priority specified. |
| 21:07.32 | p3nguin | It'll throw the phone number into the failover server which is voicemail only. |
| 21:09.10 | wcselby | any way to quickly skip to voicemail number 71? |
| 21:09.38 | nightrid3r | p3nguin: go to the parking lot and grab all the car battry's you can find :) |
| 21:09.49 | p3nguin | haha, good idea! |
| 21:10.31 | p3nguin | I guess if the power doesn't come back on soon, I'll go find something else to do, possibly in the rain. |
| 21:11.03 | sbrath | Do I have access to look up the Email address of a SIP endpoint to use it to send them their fax? |
| 21:11.19 | nightrid3r | p3nguin: like connect a long power cable to a broom stick and stand on the roof ? |
| 21:11.21 | sbrath | or do I have to configure that lookup outside the voicemail stuff. |
| 21:11.45 | p3nguin | nightrid3r: Unfortunately there is no lightning. :/ |
| 21:11.50 | *** part/#asterisk Madoc (~Madoc@bas3-ottawa23-1177800811.dsl.bell.ca) |
| 21:13.46 | Gugge | is it possible to have Dial play music, and a dialtone at the same time, or do i just have to make my musicfiles sound like they have a dialtone? |
| 21:14.16 | ariel_ | wow dial tone and music together, why? |
| 21:14.46 | knarfly | is there anyway to keep Asterisk from reporting that it's mapped to my ITSP every three minutes and clogging the console with unwanted messages about it? |
| 21:14.47 | bmoraca_work | is there any way to have Asterisk monitor packet loss and jitter? |
| 21:15.17 | Gugge | ariel_: because music sounds strange, but low music with a dialtone sounds fine :) |
| 21:15.38 | Gugge | knarfly: what messages? |
| 21:16.14 | knarfly | Gugge -- ast_get_srv: SRV lookup for '_sip._UDP.sip.callwithus.com' mapped to host sip.callwithus.com, port 5060 |
| 21:16.44 | knarfly | this keeps repeating over and over again...! |
| 21:17.33 | Gugge | enable dnsmgr |
| 21:18.08 | knarfly | Gugge: can you explain a little more in detail on how one does this? |
| 21:18.18 | Gugge | look in dnsmgr.conf |
| 21:19.10 | knarfly | Gugge: ok thanks got it....now to figure out how to get meetme working? |
| 21:19.47 | Gugge | enable a timer :) |
| 21:20.28 | knarfly | I installed dahdi with this FreeBSD server...I don't know how to do that with zaptel and zaptel does not work with *-1.6.x |
| 21:20.33 | knarfly | AFAIK |
| 21:20.47 | sbrath | zaptel == dahdi |
| 21:20.51 | Gugge | dahdi is the new zaptel |
| 21:20.56 | Gugge | just use dahdidummy |
| 21:21.12 | knarfly | ah,,, you mean dahdi has a dummy? |
| 21:21.24 | Gugge | dahdi is "just" a renamed zaptel |
| 21:21.46 | Gugge | so yes, theres a dahdidummy like there was a ztdummy |
| 21:22.12 | knarfly | then I guess I'm the dummy for now figuring that one out myself 8-) |
| 21:23.23 | knarfly | still getting that annoying dns message even though I've restarted * |
| 21:24.10 | knarfly | Gugge: didn't find the dahdidummy.ko file,,,how so I kldload it |
| 21:24.19 | Katty | Skeeter-: still no luck |
| 21:24.25 | Katty | Skeeter-: however the results of nmap are very interesting |
| 21:24.51 | Katty | Skeeter-: i ran nmap locally on itself, 0.12, and it reports about 6 things running. when i nmap from the other server, going TO 0.12, it says only 80 and 5060 are open |
| 21:25.30 | knarfly | Gugge: can dahdi be loaded after Asterisk is complete or must it go in ahead of asterisk like zaptel required? |
| 21:25.41 | Katty | Skeeter-: the mac address also doesn't match |
| 21:26.01 | Gugge | knarfly: you should only get that dns msg when asterisk makes a dns lookup |
| 21:26.09 | Gugge | and it should cache that for a longer time with dnsmgr |
| 21:26.24 | Gugge | i think dahdi needs to be there before you compile asterisk, but i dont know |
| 21:26.40 | Gugge | i dont use it :) |
| 21:26.51 | knarfly | Gugge: yes, but I guess it's doing it every 300 seconds, so I'll have to reset it....looks like I will have to reinstall * as well... |
| 21:27.22 | knarfly | I can't recall if I set dahdi to install, I thought I did but I don't find / -name "dahdi*" -print comes up empty |
| 21:28.40 | Katty | Skeeter-: if i ifup and down eth0, and run nmap again, all the ports are back ;) i'm guessing one of our toshiba test ip phones is sitting on 0.12 playing with my head all day ;) |
| 21:29.02 | Katty | giggles at the entire situation |
| 21:29.04 | knarfly | oops, I just found dahdidummy.ko |
| 21:29.38 | knarfly | actually it's dahdi_dummy.ko |
| 21:30.47 | knarfly | Gugge: yippee! Meetme works |
| 21:30.59 | knarfly | that was painless |
| 21:31.20 | knarfly | unlike the pinched nerve in my shoulder from typing all day! |
| 21:31.55 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 21:32.46 | knarfly | Gugge: how do I start dahdi_dummy.ko at boot time? |
| 21:33.04 | Gugge | dont know |
| 21:33.08 | Gugge | i use freebsd :) |
| 21:34.01 | knarfly | yes, I speak FreeBSD here too....I will dig into this...I used to have enable_zaptel in /etc/rc.conf and ztdummy loaded from the file in /etc/rc.d |
| 21:35.25 | Gugge | you could just make a script in /usr/local/etc/rc.d that loads it |
| 21:36.09 | knarfly | Gugge: the script is already there,,,I will edit it to include dahdi_dummy.ko |
| 21:36.26 | Gugge | ahh yes, the dummy module isnt added as default |
| 21:39.33 | knarfly | actually it just needs the tweak to /etc/rc.conf dahdi_enable="YES" and dahdi_modules="dahdi_dummy.ko" |
| 21:39.49 | knarfly | now to reboot and test it out |
| 21:44.40 | knarfly | Gugge: works great |
| 21:47.07 | p3nguin | sighs |
| 21:47.11 | Katty | hi p3nguin |
| 21:47.15 | Katty | do you need a hugeroonie |
| 21:47.19 | p3nguin | Ameren has a huge outage area. |
| 21:47.26 | Katty | major bummer. |
| 21:47.30 | Katty | i hope my house is still okay |
| 21:47.33 | p3nguin | Quite. |
| 21:47.36 | Katty | is it just the STL area? |
| 21:47.47 | Katty | not that you would probably know |
| 21:47.51 | Katty | considering you have no power ;) |
| 21:48.02 | p3nguin | STL area plus a few extenuating areas |
| 21:48.27 | Katty | one way to find out |
| 21:48.29 | Katty | calls her house |
| 21:48.31 | p3nguin | Well, I only called the Ameren IP number. |
| 21:48.42 | Katty | oh, auto attendant. hmm, guess we have power ;) |
| 21:48.53 | p3nguin | I think the other side of the river is on Ameren UE. |
| 21:48.59 | Katty | possibly |
| 21:49.04 | knarfly | do polycom phones suck as bad as Grandstream phones? |
| 21:49.09 | bmoraca_work | no |
| 21:49.11 | Katty | knarfly: not at all |
| 21:49.14 | p3nguin | There's also Ameren CIPS, which I have no clue where they are. |
| 21:49.15 | Katty | knarfly: polycoms ar eone of the bestest |
| 21:49.43 | knarfly | I need a new 4 line phone to replace the POS phones that GS sold me |
| 21:50.37 | *** part/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 21:50.54 | Katty | most phones these days don't really work with 'lines' |
| 21:51.08 | Katty | you usually have the phone registered as 1 extension, and then the system just gives it the next open line |
| 21:51.45 | p3nguin | When I called the number to report/inquire the outage, they have a recording listing all the cities without power rather than having a person actually answering the phone, so I'm expecting several hours without power. |
| 21:51.47 | Katty | i'm using a polycom 500 or a 501, don't recall, and it has enough for 3 different registerations |
| 21:52.04 | Katty | i guess i could setup my dialplan to make each one be a different 'line' |
| 21:52.06 | p3nguin | And phones cannot "register as extensions" |
| 21:52.07 | Katty | but that seems pretty silly |
| 21:52.31 | p3nguin | Phones can certainly register, but that doesn't magically make it an extension. |
| 21:52.38 | Katty | knarfly: your best bet is to call telephonydepot, and tell them you need a phone that will support x concurrent calls at a time |
| 21:53.02 | Katty | knarfly: they have a very wide variety of phones...i tend to prefer polycoms.. |
| 21:53.13 | Katty | knarfly: they will no doubt share their best sellers with you |
| 21:53.16 | Slugs_ | im not grapsing 'context' fully. Is it true that the 'context' only has to show up in the configuration of the channel, ex. sip.conf and the dialplan. So if i had a context called 'asdf' it would have to appear in sip.conf as context=asdf and extension.conf as [asdf] and underneath that have exten => 1000,1,Dial(SIP/{$EXTEN}) ? |
| 21:53.44 | Katty | Slugs_: first you tell your phone what context it's supposed to be in, in sip.conf |
| 21:53.50 | Katty | Slugs_: then you pickup the phone and it goes OKAY I GO HERE |
| 21:53.51 | p3nguin | slugs_: The dialplan context starts with the square-bracketed letters/numbers. |
| 21:53.57 | Katty | Slugs_: then it starts at [OKAY I GO HERE] |
| 21:54.03 | Katty | Slugs_: and whatever you punch it, it matches |
| 21:54.39 | p3nguin | The peer/device/phone context simply says where to send the call. |
| 21:54.41 | p3nguin | nothing more. |
| 21:54.50 | Katty | Slugs_: so if your phone is set to be [from-internal] in your sip.conf, and you pick up your phone and dial 911... it will go directly to [from-internal] in extensions.conf and look for an entry (or a matching pattern) for 911 |
| 21:55.08 | Katty | Slugs_: well more like context=from-internal in sip.conf, but you get the idea |
| 21:55.24 | p3nguin | On an incoming call from the ITSP, the context for the ITSP's peer is where the call is sent (as long as the call is correctly matching that peer definition). |
| 21:55.43 | Katty | Slugs_: p3nguin's right, same thing happens in zaptel.conf or chan_dahdi.conf |
| 21:55.49 | Slugs_ | yes definitily ty both so much |
| 21:55.54 | Katty | Slugs_: you assign your channels, or lines, or whatever to a group, and you give the group a context |
| 21:56.10 | wcselby | time to head out i think |
| 21:56.11 | Katty | Slugs_: so when you get an incoming call, it goes directly to [from-pstn] or whatever in extensions.conf and looks for something to do |
| 21:56.19 | p3nguin | If there is no match in the context, it will then check in any included contexts. |
| 21:56.23 | p3nguin | in order. |
| 21:56.41 | Slugs_ | got ya |
| 21:56.47 | Slugs_ | ty ty ty |
| 21:57.38 | p3nguin | You can then use lots of other applications to jump around to different contexts based on the extension that matches. |
| 21:57.46 | Katty | 3 minutes till i swap the servers around |
| 21:57.47 | knarfly | OK I found a po;ycom phone that looks right. It says it's PoE so will that mean my standard Ethernet cables and connections will work with it to draw power? |
| 21:57.51 | p3nguin | GoSub, Goto, GotoIf, etc. |
| 21:57.56 | Slugs_ | right |
| 21:58.07 | Katty | knarfly: if it's what i'm thinking, probably not |
| 21:58.31 | Katty | knarfly: the 500 and 501 that we have here, have an ac power adaptor, and from that power adaptor is a little thing to run power through the cat5 on, but one end is notched |
| 21:58.33 | Slugs_ | p3nguin, i really need to make sure i understand this b4 anything ;/ |
| 21:58.37 | Slugs_ | ty |
| 21:58.44 | knarfly | http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-331 |
| 21:58.46 | Katty | knarfly: the notched in goes directly into the phone, so you don't accidnetally put power into wall jack |
| 21:58.47 | p3nguin | The book explains most of it pretty good. |
| 21:59.06 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:59.24 | Katty | knarfly: call them and make sure |
| 21:59.44 | p3nguin | Friggin' power is spazzing out here, now. I'm going to be non-existent before long. |
| 22:00.12 | Katty | well have a nice evening |
| 22:00.15 | p3nguin | heh |
| 22:00.16 | Katty | time for me to swap servers myself |
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| 22:00.48 | *** join/#asterisk TimeRider (steve@5ac3181a.bb.sky.com) |
| 22:00.51 | p3nguin | I guess I can mess around with some hardware as well, since there's nothing else to do. |
| 22:01.46 | knarfly | Katty: yep, you need a PoE switch to run this otherwise I have to purchase a power supply for it |
| 22:01.58 | knarfly | my switch is standard, not PoE |
| 22:02.41 | p3nguin | Oh, wait, I think the power must be back on over there. |
| 22:02.56 | p3nguin | That's probably why it's spazzing out here. |
| 22:06.28 | knarfly | Gugge: this dns message is killing me...it pops up even when I'm trying to ee a file...I know it won't be included but it's making it difficult to read what I'm editing...is this thing neccessary? |
| 22:06.39 | *** join/#asterisk geneticx_wrk (~geneticx_@c-65-34-240-65.hsd1.fl.comcast.net) |
| 22:07.32 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 22:08.10 | freezey | trying to setup dundi between two PBX systems... i can do dundi show peers and the peer comes up but when i try to call it gives me all circuits are busy now |
| 22:08.59 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:09.49 | russellb | test with *CLI> dundi query |
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| 22:13.34 | knarfly | rolls up another one and blazes into the night 8-) |
| 22:19.00 | freezey | russellb: the query works and retruns the info needed |
| 22:19.18 | freezey | when attempting to call still get all circuits are busy |
| 22:21.34 | freezey | pulls all the proper info as well when dialing just gets all circuits are busy now |
| 22:21.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:24.50 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:25.55 | jdoe | can anyone here help me with a quick polycom question? built-in (non asterisk) transfer is immediately trying to transfer after 3 digits of the extension are entered, I don't see anything relevant in the config. What am I missing? |
| 22:27.00 | cusco | could use features.conf *8 |
| 22:27.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:28.24 | jdoe | cusco: I could, but most people are going to press the button instead of remember the feature code, and the phones are old enough that I can't use the speeddial hack (because there's no efk) |
| 22:29.59 | freezey | ok so i can query via cli and that works... when i type the extension on the phone it pulls the correct name information for the user i am trying to call.. the only problem is now it still says all circuits are busy |
| 22:38.17 | p3nguin | What's the fix for Dial(SIP/username@172.16.255.21/3149691077) saying "no such host 172.16.255.21/3149691077"? I tried changing the IP address to a peer name rather than the IP address, but it didn't help any. |
| 22:38.48 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 22:42.31 | dinesh___ | hm that's a weird syntax, usually I do SIP/username@sip_provider where sip_provider is registered in sip.conf |
| 22:44.37 | p3nguin | Sure. But how do you send a phone number? |
| 22:44.51 | [TK]D-Fender | p3nguin: Dial is broken for that... |
| 22:44.55 | [TK]D-Fender | p3nguin: make a peer |
| 22:45.03 | p3nguin | I did make a peer. |
| 22:45.43 | [TK]D-Fender | p3nguin: well you're dialing by IP. |
| 22:45.47 | p3nguin | Let me see what it does without the username in the Dial. |
| 22:45.55 | [TK]D-Fender | p3nguin: you aren't using the peer in there |
| 22:47.04 | *** join/#asterisk smooth_penguin (~smoove@59.95.0.113) |
| 22:47.42 | p3nguin | Dial(username@newpeer/3149691077) fails the exact same way... but if I take out the username and make sure the username is in the sip entry, then I do not get the error. I guess that fixes it, huh? |
| 22:48.21 | p3nguin | e.g., Dial(SIP/newpeer/3149691077) |
| 22:48.23 | freezey | [TK]D-Fender: one who knows everything want to try and assist with my small issue? |
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| 22:50.32 | *** join/#asterisk nix8n82 (~bob@63.162.27.14) |
| 22:50.36 | [TK]D-Fender | p3nguin: You're doing it wrong.. the peer should have the user.. you should not be specifying it in the dial |
| 22:50.51 | [TK]D-Fender | freezey: Ask and we'll see |
| 22:50.55 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:51.11 | freezey | its a dundi issue... so i can query via cli and that works... when i type the extension on the phone it pulls the correct name information for the user i am trying to call.. the only problem is now it still says all circuits are busy |
| 22:51.12 | p3nguin | Yeah, I changed that and the no such host error disappeared. |
| 22:52.15 | p3nguin | But now the device must not know what to do with the phone number because the channel gives a circuit-busy response. |
| 22:52.22 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 22:52.31 | p3nguin | This is still progress. |
| 22:54.09 | *** join/#asterisk undertuga (~undertuga@99.48.103.87.rev.vodafone.pt) |
| 22:54.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:54.56 | freezey | [TK]D-Fender: what i do notice is the dial plan on _53XX(extensions for other office) it looks like this _53XX,1,Macro(outisbusy,) i tried adding lookupdundi which i specificed in extensions.conf but it still bombs... when i added the lookupdundi it gave me a different error message... which leads me to believe it has something to do with this option |
| 22:57.42 | freezey | [TK]D-Fender: nm i got it |
| 22:57.44 | nix8n82 | Some one asked me a question if it was possible to send text data to a ip phone and have a key or two make a choice from that phone and have asterisk call a number or send to a meetme, or just log information into a database. If so what is it called so I may google it to get more info? |
| 22:57.55 | freezey | was that |
| 22:58.14 | freezey | sweet |
| 23:00.44 | [TK]D-Fender | nix8n82: No. there is no generic "sent text toa phone and get some input back". |
| 23:01.29 | [TK]D-Fender | nix8n82: Several IP phones have browsers of some kind on them that you can push information /pages to by other means |
| 23:03.57 | nix8n82 | Cool xml/xhtml? |
| 23:04.35 | nix8n82 | Do you know of an example I could look at? |
| 23:05.32 | nix8n82 | for like polycoms and grandstreams? |
| 23:05.41 | [TK]D-Fender | nix8n82: Depends on the phone. There is no standard |
| 23:10.05 | nix8n82 | [TK]D-Fender thanks for your response and input, I appreciate it. |
| 23:11.00 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:14.24 | nix8n82 | Have you heard of pushing an xml stream to a phone and changing the meaning of possible programable buttons on the fly? |
| 23:16.53 | nix8n82 | probably with polycom or grandstream, the ones with out the xhtml browsers? |
| 23:17.30 | hardwire | on the fly eh? |
| 23:17.51 | hardwire | has a hard time understanding which part of a pbx handles the "on the fly" part. |
| 23:17.55 | *** join/#asterisk bmoraca (bmoraca@66.242.162.254) |
| 23:18.19 | *** join/#asterisk TJNII (~TJNII@207.189.199.58) |
| 23:18.42 | nix8n82 | with out having to re provision from the boot server |
| 23:18.54 | hardwire | nix8n82: what initiates that? |
| 23:19.47 | nix8n82 | a button being pressed on the phone that isn't part of the standard keypad on a phone |
| 23:19.54 | hardwire | this may be entirely possible.. I'm just not sure how to approach it as a standard based on general user requirements. |
| 23:20.09 | hardwire | nix8n82: so bam.. you're a different extension? |
| 23:20.11 | hardwire | etc.. |
| 23:20.25 | nix8n82 | no |
| 23:20.33 | hardwire | or "press this button to remap all your screen buttons to the weather" |
| 23:23.08 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:23.44 | nix8n82 | press this button to punch a time clock, or a call is here, send it to place a or b, or add to blacklist while in call before hanging up, your boss is a tool? button 1 = yes button 2 = no, button 3 plead the 5th |
| 23:24.11 | hardwire | plead the 5th should be the 5th button |
| 23:24.21 | hardwire | 4th should be a golf joke |
| 23:24.41 | nix8n82 | good call |
| 23:25.04 | nix8n82 | 4=duck |
| 23:25.07 | nix8n82 | ? |
| 23:26.14 | hardwire | heh |
| 23:26.24 | hardwire | nix8n82: I'm not familiar with a method to remap anything on the fly |
| 23:26.34 | nix8n82 | yeah neither am I |
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| 23:30.39 | *** join/#asterisk norrec (~norrec@76-201-85-28.lightspeed.frokca.sbcglobal.net) |
| 23:31.35 | hardwire | nix8n82: I had a hunch |
| 23:32.18 | upb | googles hardwire |
| 23:32.22 | norrec | i'm running asterisk 1.6.0.24 and have SpanDSP installed but i'm going to swtich to fax for asterisk and i assume i need to removed SpanDSP but i'm not really sure how... do i just do make menuselect and unselect the fax or what... |
| 23:32.29 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 23:32.31 | hardwire | upb: I wouldn't do that at work |
| 23:32.36 | upb | hahaha |
| 23:32.39 | upb | its 1 am here |
| 23:33.32 | p3nguin | lol |
| 23:33.57 | hardwire | 15:33 CTCP TIME reply from upb: Fri Mar 26 01:33:14 2010 |
| 23:34.03 | hardwire | it appears you are mostly accurate |
| 23:34.08 | ChannelZ | norrec: probably just need to not load the driver/module/whatever it is |
| 23:34.15 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:35.21 | manxpower | ~answers |
| 23:35.21 | infobot | i guess answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 23:35.33 | norrec | ChannelZ: where do i set what loads at startup? |
| 23:35.38 | thehar | manxpower: see. |
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| 23:36.03 | ChannelZ | norrec: well I'm assuming there's a module you load into asterisk, in which case /etc/asterisk/modules.conf |
| 23:36.27 | ChannelZ | norrec: I don't really know anything about SpanDSP... |
| 23:37.07 | ChannelZ | norrec: if it's just a library of support code and an asterisk module, then you can just 'noload' the module to make sure |
| 23:38.13 | norrec | ChannelZ: alright, thanks |
| 23:45.31 | bmoraca | damn telco |
| 23:45.37 | bmoraca | supposed to be here an hour ago |
| 23:47.02 | bmoraca | norrec, i'm pretty sure that fax for asterisk uses spandsp. that said, having it installed won't hurt anything. |
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| 23:53.31 | p3nguin | I don't think I have spandsp support and I do use FFA. |
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