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| 00:53.54 | C4away | is there any analog cards that support the "4-wire E&M" analog lines? |
| 00:54.02 | C4away | s/is/are/ |
| 00:54.43 | C4away | of any of the signalling protocols therein? |
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| 00:59.11 | sah-work | so i just installed the lastest dl of * and the gui |
| 00:59.21 | sah-work | i log into the gui |
| 00:59.30 | sah-work | and there is nothing to do. there are no pannels. |
| 00:59.33 | sah-work | what am i msising |
| 00:59.53 | [TK]D-Fender | sah-work: and then realize the GUI has its own support channel as listed in the channel topic |
| 01:00.19 | sah-work | thanks |
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| 01:18.59 | sah-work | yah. no help there |
| 01:22.38 | jaytee | it's usually like a morgue in the gui support channels, for a patient that refuses to die. |
| 01:25.04 | jblack | isn't the point of a gui to make things so good that assistance is unnecessary? So, by definition, anything not understood should be filed as a bug in the gui's bug tracker. |
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| 01:26.43 | C4away | interesting, never noticed that |
| 01:26.51 | C4away | who reads topics anyways? |
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| 01:50.07 | sah-work | with make menuselect, the module embedding is just to compile the module as part of the bin vs loading later correcT? |
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| 02:39.34 | jks | anyone knows where I can download language files for polycom 330 phones? |
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| 02:48.09 | jaytee | jks, did you try Polycom's site? |
| 02:48.52 | jks | jaytee, yes, but it didn't have the right language file for the version I'm running |
| 02:49.08 | jks | perhaps I should just update to 3.1.1b and hope that works as well with asterisk as the older ones |
| 02:49.11 | jaytee | which language and what version? |
| 02:49.30 | jks | I'm running 2.1.1... Danish language |
| 02:55.33 | jaytee | jks, their website appears to be weird. it will let me select Dansk as a language but then switches back to English |
| 02:56.43 | jaytee | you might call your reseller about getting the files you need or if you can find the language format in a newer version then you can upgrade. I haven't heard of anyone having problems with 3.0 or 3.1 on Asterisk |
| 02:57.38 | jks | guess I better call the reseller... weird that they have to keep the files private |
| 02:59.09 | jaytee | jks, they don't keep the english ones private so I don't know why they'd make the dutch ones private. It might be a thing here in the states. |
| 02:59.26 | jks | hehe, I don't want the dutch ones ;-) |
| 03:00.11 | jaytee | I thought you did because you said were running the Danish language |
| 03:01.50 | jaytee | jks, what exactly are you looking for? |
| 03:02.09 | jks | Danish like they speak in Denmark |
| 03:02.15 | jks | not Dutch as they speak in the netherlands |
| 03:03.17 | jaytee | ok, I was ignorant of the fact that Danish and Dutch were two languages. I always thought they were the same. My bad. Y'know how us Americans are :-) |
| 03:04.07 | coppice | A Danish reference means that for the second time today I have pretext to say: |
| 03:04.09 | coppice | jaytee's comments are "a tale told by an idiot, filled with sound and fury, signifying nothing" :-) |
| 03:05.03 | jaytee | bet that makes ya feel all smart, smug and superior don't it? |
| 03:09.45 | coppice | ah, but isn't it interesting that I hadn't used that quote for years, yet within just half an hour it fit nicely in two different contexts. there's some higher power at work here, imposing synchronicity upon us. |
| 03:13.37 | jaytee | damn! synchronicity! all these big words are giving me a headache. Think I'll logoff and lie down for a bit. |
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| 03:25.47 | glaz | I am trying to establing an IAX trunk between two asterisks, I keep getting Registration Refused, can't figure out what I am doing wrong |
| 03:26.05 | glaz | If I paste my two iax.conf file to pastebin, anyone can take a look at them? |
| 03:28.05 | glaz | http://rafb.net/p/0kKHN463.html |
| 03:29.12 | glaz | Seems OK to me :\ maybe a second look could tell me "hey! you did this mistake on line XX!" |
| 03:29.15 | glaz | heh |
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| 03:41.50 | sah-work | anyone else not able to dl the iso for asterisk now |
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| 04:13.50 | JohnnyBeGood | @sah-work try different mirror |
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| 04:37.17 | simprix | Does the cisco 7921 support sip or skinny ? |
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| 04:49.19 | glaz | simprix: both. |
| 04:51.15 | SkramX | yeah, both |
| 04:51.26 | SkramX | been working with skinny on those recently if you have any questions about that |
| 04:56.27 | glaz | SCCP on * ? or with cisco call manager? |
| 04:56.34 | SkramX | Asterisk |
| 04:56.45 | SkramX | Work uses CCM however my project is strictly * |
| 04:57.25 | SkramX | IIRC, I havent gotten XMLDefaults.cnf.xml to work the way I want it yet, so I have to use individual SEPXXXX.cnf.xml files (yuck!) |
| 04:57.43 | SkramX | you? |
| 04:57.51 | glaz | I use them with SIP. |
| 04:57.57 | glaz | SIPDefault.cnf works fine. |
| 04:58.09 | SkramX | fully compatible with services, etc.? |
| 04:58.13 | glaz | yes |
| 04:58.22 | SkramX | im trying to remember why I decided to go with SCCP |
| 04:58.23 | SkramX | heh |
| 04:58.35 | glaz | I'm wondering why you when with SCCP :p |
| 04:58.39 | SkramX | pastie SIPDefault? |
| 04:58.41 | glaz | s/when/went |
| 04:58.46 | glaz | sure, hold on. |
| 04:58.58 | SkramX | well, at first I thought it'd be easier and go w/o re-loading firmware |
| 04:59.05 | SkramX | but I ended up upgrading the f/w anyways |
| 04:59.13 | SkramX | k thanks |
| 04:59.32 | glaz | I dont have the one I use at the office from here, but I have the one I use at home. |
| 04:59.46 | SkramX | i really want to support 7940, 7980 (Cisco IP Communicator), and 7921s all in one config file |
| 04:59.49 | glaz | http://rafb.net/p/6h2NhX93.html |
| 04:59.50 | SkramX | k |
| 05:00.06 | glaz | I have 7940 and 7960 all over the office |
| 05:00.19 | SkramX | all on SIP and all in one config file? |
| 05:00.47 | glaz | well, will you configure each line on each phone? |
| 05:01.01 | glaz | you can do this and only use SIPDefault.cnf |
| 05:01.05 | SkramX | thats what i've had to be doing |
| 05:01.13 | SkramX | but i discovered asterisk config templates |
| 05:01.17 | SkramX | need to play around with those |
| 05:01.17 | glaz | or, you create a SIPMA:CA:DD:RE:SS.cnf for each phone |
| 05:01.28 | SkramX | yeah- that's annoying |
| 05:01.38 | glaz | can be yeah. |
| 05:01.53 | SkramX | right now each p hone has a SEPMA:CA...cnf file and then a couple lines in skinny.conf |
| 05:01.56 | glaz | or you can use Services to login as an agent |
| 05:02.00 | glaz | like I did for a client |
| 05:02.10 | SkramX | wel |
| 05:02.17 | SkramX | my whole project is actually about agents |
| 05:02.30 | SkramX | but i have each phone get a line then the agent logs in on the line |
| 05:02.54 | glaz | ok |
| 05:03.04 | glaz | there are plenty of ways to do it heh |
| 05:03.24 | SkramX | yeah |
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| 06:33.29 | carbocalm | bbryant: thank you, it works this way |
| 06:33.55 | bbryant | welcome |
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| 09:04.22 | mort_gib | Morning |
| 09:06.06 | mort_gib | Any recommendations for a nicely priced POE switch?? |
| 09:22.39 | x86 | mort_gib: Linksys |
| 09:22.58 | mort_gib | Linksys?? -Yeah?? |
| 09:23.07 | x86 | anyone know of a decent web-based CDR reporting tool? |
| 09:23.16 | x86 | preferably with graphs and stuff |
| 09:23.28 | x86 | I can't get Areski's to display graphs it seems |
| 09:25.04 | mort_gib | I think that you might have to look at some of the commercial solutions..... |
| 09:27.02 | SwK | cdrtool |
| 09:27.24 | SwK | http://www.voip-info.org/wiki/view/CDRTool |
| 09:30.43 | x86 | SwK: free? |
| 09:30.55 | x86 | w00t! I got call pickup working with BLF.... |
| 09:31.11 | x86 | works to pick up ringing extensions, wonder about established... |
| 09:33.03 | SwK | yes its free |
| 09:34.36 | x86 | looks like it's for OpenSER |
| 09:34.44 | x86 | CDRTool is an Open Source solution that provides mediation, accounting and tracing for Call Detail Records enerated by OpenSER by using RADIUS protocol and OpenSER siptrace facility. |
| 09:34.55 | SwK | it does asterisk too |
| 09:34.57 | x86 | (from CDRTool's website) |
| 09:35.01 | SwK | keep reading |
| 09:37.01 | x86 | is there a demo of it? |
| 09:37.08 | x86 | from the screenshots I don't see any graphs |
| 09:38.56 | mort_gib | x86: How did you get BLF pickups to work?? What handsets |
| 09:41.00 | x86 | Linksys 962 with 932 sidecar |
| 09:41.09 | x86 | BLF pickups only work with ringing calls it seems |
| 09:41.54 | x86 | I can't get the line keys to call regular extensions it seems |
| 09:42.33 | x86 | they will show presence on regular extensions, but if I try to call them by hitting the line key, the phone says invalid extension, without even attempting to send the call to Asterisk |
| 09:42.57 | mort_gib | Yeah, that's what I get |
| 09:43.12 | mort_gib | Snom 3xx and Asterisk of course |
| 09:43.57 | mort_gib | I can choose to use the buttons as speed dials or as pickup.. |
| 09:44.01 | mort_gib | Not both |
| 09:48.39 | x86 | it works for both for me |
| 09:48.53 | x86 | but only when the line is in ringing state |
| 09:49.11 | x86 | also, when I try to page all phones from any linksys phone, I get this: SIP/2.0 487 Request Terminated |
| 09:49.29 | x86 | but when I page TO the linksys phones, from say X-Lite, it works fine |
| 09:50.44 | x86 | like the Linksys phones are cancelling the inbound call for some reason |
| 09:50.47 | x86 | any ideas why? |
| 09:55.56 | x86 | also, I have a working call parking line key, but I can not transfer a call to it |
| 09:56.11 | x86 | when I hit transfer, it will only let me input an extension it seems, I can't hit a line key? |
| 09:59.25 | mort_gib | Hmm, I use Snoms transfer, not call parking |
| 10:00.04 | x86 | well I can't get the line keys that have local extensions on them to call that extension |
| 10:00.24 | x86 | line keys with local extensions are only working for presence and call pickup (while ringing) |
| 10:00.35 | mort_gib | There ARE things that some of the old PBX's do better... |
| 10:02.58 | x86 | well Cisco Call Manager does a great job of traditional key system emulation |
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| 10:17.23 | mort_gib | Never had a look at Cisco's Call manager... Don't like Cisco much |
| 10:17.48 | mort_gib | Had this discussion with a MS/Cisco administrator about command line |
| 10:18.23 | mort_gib | Lame clown claims that (then) was 2007 and we should not have to use the command line anymore... |
| 10:19.28 | coppice | they are usually the kind who believe the same about brains |
| 10:19.37 | mort_gib | -Saying this while he is using telnet to configure our border gateway! -So what are you doing there?? That looks like you are using a commandline over an unprotected connection! |
| 10:19.37 | Nugget | telnet is eeeeeeevil! |
| 10:19.47 | mort_gib | Yep... |
| 10:23.40 | mort_gib | Anyways, Sunday blues see you all later... |
| 10:31.19 | x86 | coppice: hey, ever use Linksys phones? specifically a 962? |
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| 10:59.30 | respecting | please what' i want an asterisk version for windows ? |
| 11:00.05 | respecting | i have a company equiped with windows Pc's and i want to use asterisk but it works only on linux? is there any version for windows |
| 11:02.01 | orly_owl | http://www.asteriskwin32.com/ |
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| 11:03.03 | respecting | wawwwwwwwwwwwwwwwww |
| 11:03.08 | respecting | that's wonderfull |
| 11:03.15 | respecting | thank you |
| 11:03.15 | respecting | Mr orly_owl |
| 11:03.25 | respecting | you have just saved my ass |
| 11:13.12 | tzafrir_laptop | respecting, hmm... I'm not really sure how well that works |
| 11:14.14 | tzafrir_laptop | IIRC it's a dated version |
| 11:15.58 | C4away | wow, for some odd reason the concept of running asterisk on a windows server NEVER crossed my mind |
| 11:16.50 | C4away | kinda like running a porsche engine in a gremlin |
| 11:17.31 | C4away | technically, with enough work I suppose you could get it functional, but it's just an odd concept |
| 11:22.12 | C4away | in the case of asterisk on windows, you get to pay more for licensed operating sytem to run an app that was not designed to run in that environment, probably get buggy and questionable results ... when there is a free operating system the software was designed to run on |
| 11:27.05 | respecting | wait a second i think that's like asterisk for linux |
| 11:27.11 | respecting | they did the same job |
| 11:27.17 | respecting | i was on their website |
| 11:27.26 | respecting | and they did the same as in linux |
| 11:27.47 | tzafrir_laptop | respecting, give it a shot. Tell us what it was like |
| 11:28.05 | tzafrir_laptop | Though I'm not sure how well supported it actually is |
| 11:28.38 | tzafrir_laptop | Generally I would recommend you to set up a separate Linux system |
| 11:28.50 | respecting | the * for windows works as a gatekeeper(PABX) and it has the ability to work as a gateway |
| 11:29.12 | respecting | between RTC(ISDN in english) and voip network |
| 11:29.45 | tzafrir_laptop | What ISDN adapter do you have? |
| 11:31.00 | respecting | zapatel(i'm gona make my intership about * :the company manager tell me the whole idea that he want to deploy * on windows) |
| 11:31.44 | tzafrir_laptop | I have no idea if they actually ported zaptel to windows. It takes a huge ammount of work. And I have heard of no one who did that |
| 11:32.20 | tzafrir_laptop | Maybe they have drivers for some specific cards |
| 11:32.39 | tzafrir_laptop | Generally test it yourself before even mentoining it to anyone else |
| 11:32.48 | respecting | yeah the problem that suppose a bank that have only windows laptops and pc's how can we make a voip server without a windows version of *? |
| 11:33.39 | tzafrir_laptop | people's laptops don't need to have asterisk servers. Do you want to put osftware voip phones on them? |
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| 11:34.09 | respecting | no here's the problem.in a bank there's only windows |
| 11:34.19 | respecting | how can u make voip server their? |
| 11:34.39 | respecting | you can not say to them please buy a new pc and install linux then install * |
| 11:34.49 | respecting | you must have a windows version of * |
| 11:35.08 | tzafrir_laptop | respecting, the PBX should be on a dedicated server in most setups |
| 11:35.29 | respecting | yes and that server must be linux is that correct? |
| 11:35.44 | tzafrir_laptop | For Asterisk: yes (if you value your time) |
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| 11:56.23 | dfas | What companies provide managed asterisk hosting? |
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| 13:00.08 | *** join/#asterisk GNUtoo (n=GNUtoo@host135-29-dynamic.17-79-r.retail.telecomitalia.it) |
| 13:02.52 | GNUtoo | hello, I had bad luck with siproxd and ekiga...so could asterisk be used as sip proxy? if yes is there any infos on that: http://it.slashdot.org/it/08/12/06/1914242.shtml ? |
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| 13:10.58 | GNUtoo | by the way does I still need dahdi in order to make meetme work? |
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| 13:22.19 | Ko_deZ | Hi. I have a problem with my incoming DID (SIP). The people calling me are not getting a ring tone, just silence. I have been trying to figure this out, and the only thing I find is that I should add a 'r' option to the dial command for the incoming connection, but it does not help me much. |
| 13:22.44 | Ko_deZ | There is still silence at the calling end. |
| 13:22.52 | Ko_deZ | If I pick up, everything is OK though. |
| 13:24.23 | Ko_deZ | my extensions.conf is here: http://pastebin.com/d55ebe9e1 |
| 13:24.27 | tzafrir_laptop | GNUtoo, yes |
| 13:24.45 | GNUtoo | tzafrir_laptop, yes for dahdi? |
| 13:25.01 | tzafrir_laptop | Yes |
| 13:25.08 | tzafrir_laptop | What version of Asterisk do you use? |
| 13:25.25 | GNUtoo | i'm actually compiling the 1.6.0.2 |
| 13:25.31 | GNUtoo | s/i/I/ |
| 13:29.03 | Ko_deZ | haha, cool bot. |
| 13:29.14 | Ko_deZ | s/cool/awsome/ |
| 13:29.26 | Ko_deZ | I like it! |
| 13:34.37 | orly_owl | s/like/don't like/ |
| 13:38.09 | Ko_deZ | fail! |
| 13:38.29 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 13:38.34 | orly_owl | Worth a try. |
| 13:38.37 | Ko_deZ | my favorite: http://failblog.org/2008/10/07/cow-curiosity-fail/ |
| 13:38.43 | Ko_deZ | sure. |
| 13:38.54 | Ko_deZ | I would have done that too =) |
| 13:39.26 | Ko_deZ | Need to get some of those fail/win stamps they are selling. |
| 13:40.49 | orly_owl | I want a kb with no OS specific keys. >_> |
| 13:41.05 | Ko_deZ | I figured out my "silence" problem when people call me btw. I removed the Answer before using Dial to redirect to the internal numbers. |
| 13:41.56 | orly_owl | I have to figure out * first. |
| 13:42.41 | Ko_deZ | orly_owl: http://www.doobybrain.com/2007/10/29/computer-keyboard-calculator/ <-- probably not what you where looking for though... |
| 13:43.27 | orly_owl | No, but that's a really nice looking calculator. |
| 13:44.03 | Ko_deZ | http://www.kingmedia.com.au/product_info.php?manufacturers_id=16&products_id=79&osCsid=ae1227a8b4fff8aa1185dfcd86a7118f <-- even smaller |
| 13:44.13 | orly_owl | Now show me the flipped version for use with the left hand. |
| 13:44.14 | Ko_deZ | it sure is |
| 13:44.23 | Ko_deZ | haha. |
| 13:45.22 | orly_owl | I saw an apple alu kb recently. didnt like the OS keys and some other key labels, but DAMN is it a nice looking keyboard. felt well built too |
| 13:46.14 | orly_owl | a happy hacking kb in that style would be nice |
| 13:46.25 | Ko_deZ | Humm. Can you find a link? |
| 13:47.01 | orly_owl | http://en.wikipedia.org/wiki/Image:Apple_wired_thin_keyboard-2007-08-11.jpg |
| 13:47.13 | orly_owl | dodgy photo though |
| 13:47.28 | orly_owl | http://www.apple.com/keyboard/ |
| 13:48.33 | orly_owl | http://ln-s.net/2Y+x |
| 13:49.18 | Ko_deZ | oh... |
| 13:49.25 | Ko_deZ | that _is_ nice |
| 13:49.50 | orly_owl | =D |
| 13:50.07 | Ko_deZ | how are the buttons? |
| 13:50.13 | orly_owl | felt ok |
| 13:50.29 | orly_owl | i didnt use it though. |
| 13:50.36 | orly_owl | wasnt plugged in |
| 13:50.41 | Ko_deZ | right. |
| 13:50.50 | orly_owl | my current kb http://notzzap.zzzzz.ws/model-a-kb.jpg |
| 13:51.02 | Ko_deZ | keytronic ftw! |
| 13:51.09 | orly_owl | keytronic? |
| 13:51.29 | Ko_deZ | http://www.testipenkki.net/atk/artikkelit/keytr/kuvat/kta2001-022.jpg |
| 13:51.41 | Ko_deZ | best there ever was... |
| 13:51.59 | orly_owl | bleh. OS keys |
| 13:52.16 | Ko_deZ | ah, yes, the newer ones had those. Mine did not. |
| 13:52.22 | Ko_deZ | from the early 90's |
| 13:52.30 | orly_owl | ah ok |
| 13:53.14 | orly_owl | i like the gap betwen alt+ctrl |
| 13:54.33 | orly_owl | why are ip phones so expensive? POTS phones can be had for $10 |
| 13:57.56 | coppice | A simple POTS phone costs about $1 to make. A fancy caller ID phone with a nice LCD display costs about $8. An entry level IP phone with, and simple LCD display, costs in the high 20s. |
| 13:57.59 | Ko_deZ | orly_owl: http://communication.howstuffworks.com/telephone.htm. POTS phones a _very_ basic. |
| 13:58.12 | orly_owl | that's true |
| 13:58.54 | Ko_deZ | I mean, a speaker, a mic, and a hook switch, and you are good to go. Does not get any cheaper than that. |
| 13:59.03 | orly_owl | so if an entry level ip phone costs high 20s, why are they selling for $100 minimum? |
| 13:59.40 | coppice | you can get entry level IP phones for about $40 in reasonable numbers |
| 14:00.11 | orly_owl | ah ok |
| 14:00.23 | orly_owl | reasonable would be 50 at least? |
| 14:00.45 | coppice | yeah. I'm not talking about the 1M price break :-) |
| 14:01.58 | Ko_deZ | orly_owl: you can get a voip adapter for far less than 100$. |
| 14:17.40 | orly_owl | but if a voip adapter is over $40, might as well get an ip phone |
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| 16:34.50 | invalidrecord | hi guys is AMI solid enough to use for billing info or should i do that from a custom application for stability? |
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| 16:45.25 | x86 | good morning |
| 16:46.00 | invalidrecord | morning |
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| 16:52.55 | jaytee | invalidrecord, I'd recommend using CDR data from either mysql or postGRE sql. you can store the data in a local sql database or use the unixODBC connector for writing to a sql database on another system. |
| 16:54.32 | x86 | jaytee: cdr_addon_mysql.so doesn't have to use a local database, wtf are you talking about |
| 16:54.42 | x86 | jaytee: niether postgres |
| 16:55.09 | invalidrecord | ok ill do that may be simpler |
| 16:55.14 | invalidrecord | thanks |
| 16:55.27 | jaytee | x86, so you don't need the unixODBC connector? I store locally so I haven't tried setting an external database. good to know, thanks |
| 16:55.47 | invalidrecord | you still need odbc but db dont hve to be local |
| 16:56.16 | invalidrecord | you guys used realtime? |
| 16:56.22 | jaytee | no |
| 16:56.31 | invalidrecord | damn |
| 17:03.41 | [TK]D-Fender | Why are we comparion a DATABASE with a LIVE PBX MONITORING PROTOCOL? |
| 17:04.06 | jaytee | we weren't or at least I wasn't |
| 17:04.45 | jaytee | and I'm assuming comparion is Quebecian for comparing :-) |
| 17:05.51 | [TK]D-Fender | jaytee: No its "OMG I only got up a few minutes ago, have a headache and am not entirely there, lets just hybrid-conjugate everything!" |
| 17:06.06 | jaytee | lol |
| 17:06.12 | jaytee | go get some coffee |
| 17:07.14 | jaytee | x86, so you're sayin if I want to write out my CDR data to an external mysql database I don't need the unixODBC connector? all of that is built into the cdr_addon_msyql.so? |
| 17:08.03 | x86 | jaytee: right |
| 17:08.32 | x86 | jaytee: hence the "hostname" configuration option in cdr_mysql.conf (for example) |
| 17:09.13 | x86 | the ODBC stuff is an abstraction layer to allow you to connect to any database that your ODBC driver supports |
| 17:09.15 | jaytee | x86, must have missed that in the docs. I have it running local on my * box but eventually I want to move the data to another system because I'll have 2 * boxes, one for primary and one for failover and I'll want the CDR collection to be "seamless" |
| 17:10.14 | x86 | [TK]D-Fender: ODBC is not a "live pbx monitoring protocol", it's Open DataBase Connctivity |
| 17:10.21 | jaytee | x86, aha! so I'd only need the unixODBC connector if I was going to write out the data to another database like MS SQL Server 2005 or something else non-mysql. |
| 17:10.58 | jaytee | x86, I think he knows that. I think he was referring to AMI vs sql storage of CDR data |
| 17:11.22 | x86 | jaytee: Asterisk comes with a driver for pgsql, and the addons package has a driver for mysql... anything else you'll need the ODBC driver for (but ODBC also works with MySQL and Postgres if you'd want that) |
| 17:11.32 | x86 | jaytee: ah |
| 17:11.51 | [TK]D-Fender | x86: I was referring to invalidrecord's mention of AMI <- |
| 17:11.56 | x86 | [TK]D-Fender: gotcha |
| 17:12.13 | [TK]D-Fender | x86: ..... wanna join me for coffee? ;) |
| 17:12.37 | jaytee | x86, I knew that Postgres was included but I used the mysql addon. I just didn't know I could do it direct from the addon without using ODBC. Glad you pointed that out to me, it'll save alot of work. |
| 17:12.38 | x86 | I wish i had some... used the last of it last night (this morning actually, around 2am) |
| 17:12.57 | x86 | told the wife to know her role and go get some, but she snottishly declined :( |
| 17:13.30 | jaytee | pours x86 and [TK]D-Fender each a cup of his freshly roasted Kenya AA Gjanika roasted last night. "Cream and sugar?" |
| 17:14.02 | x86 | jaytee: the only reason mysql's not included in the default distro is something about licensing... i think Digium want's everything distributed in the core of Asterisk to be truely free, while the mysql code is not truely free, or something like that |
| 17:14.28 | x86 | skip the cream and sugar, I use hot cocoa mix instead :) |
| 17:14.36 | x86 | yum yum dim sum |
| 17:15.12 | jaytee | x86, cool! kinda makes it a mocha |
| 17:15.35 | jaytee | I sometimes add a dollop of Hersheys syrup for that |
| 17:15.57 | x86 | ok, is there a way I can use PickupChan with a dynamic channel name? such as SIP/7002-08faa7a0 (where 7002 is my extension)? |
| 17:16.29 | x86 | jaytee: the cocoa makes it chocolatey, the marshmallows make it sweet :) |
| 17:16.32 | x86 | works great |
| 17:18.29 | jaytee | x86, I sometimes mix beans from different crops when I roast but usually I roast and drink just the single varietal bean so as to pick up all the nuances. I'm a coffee snob. Won't drink Starbucks because they overroast all their stuff on the dark side. |
| 17:19.02 | x86 | hmm, perhaps I can write an AGI to do get the dynamic part of the channel nam |
| 17:19.04 | x86 | name* |
| 17:19.31 | invalidrecord | ok i have a pbx which supports multiple companies i am driving extensions from a realtime config each company has one context does this sound right |
| 17:20.01 | [TK]D-Fender | invEach company should probably have multiple contexts each |
| 17:20.38 | [TK]D-Fender | invalidrecord: and that's just for minimal separation of class |
| 17:20.54 | invalidrecord | what dialout long distance dial out local etc etc? |
| 17:21.29 | [TK]D-Fender | invalidrecord: etc * 50 |
| 17:21.47 | [TK]D-Fender | invIVR's, etc |
| 17:21.52 | invalidrecord | i was going to have some global contexts for that which i would include the companies in so dialout-local Xcomp ycomp zcomp |
| 17:22.21 | invalidrecord | no ivr's needed at this time |
| 17:22.25 | [TK]D-Fender | invalidrecord: If you even have to ask about that well... you might want to reconsider your role in the project ;) |
| 17:22.52 | jaytee | and if you want to split out the billing you probably want separate outbound contexts for each company for CDR data to segregate the billing and make it easier to run queries against the DB for a single customer. |
| 17:22.54 | invalidrecord | leartning on my feet im a rails dev |
| 17:23.10 | invalidrecord | only really palyed with avaya b4 |
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| 17:23.39 | jaytee | never had a chance to work on Avaya systems. I'm an old Nortel retread |
| 17:23.59 | invalidrecord | ipoffice was ok but fking licence nightmare |
| 17:24.20 | invalidrecord | splicecom was good |
| 17:24.25 | jaytee | Nortel's licensing scheme is just as bad or probably worse |
| 17:24.46 | invalidrecord | we replaced the avayas with that in the end as bos had no confidence in asterisk |
| 17:25.10 | invalidrecord | he wanted to pay thousands gave him a warm cosey feeling |
| 17:25.42 | jaytee | with Nortel systems you need licenses for the TN's (terminal numbers=physical analog or digital ports) and also licenses for the DN's (directory numbers or "extensions") |
| 17:28.12 | [TK]D-Fender | invalidrecord: Bosses do things like that |
| 17:28.56 | invalidrecord | yeah true he also hadf played with slackware so thought he was a guru, I used to have to rebuild ldap or other systems almost every monday tosser! |
| 17:29.24 | invalidrecord | insisted he had root even when i wouldnt trust him with windows but he signed my cheques what do you do |
| 17:29.52 | jaytee | look for another job |
| 17:30.00 | invalidrecord | heheh exactly what i did |
| 17:30.21 | invalidrecord | doubled my money halfed my stress overnight |
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| 17:30.53 | jaytee | if only the economy wasn't in the shitter I'd be looking more in earnest than I am. |
| 17:32.10 | beek | Hi jaytee -- have a moment for a PRI question? |
| 17:32.10 | invalidrecord | i was so lucky i just switched job and it took me 24hrs and i got a nice slice of equity and a good package, can believe i was that lucky! |
| 17:32.25 | invalidrecord | it amazes me isdn is still about |
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| 17:32.32 | andreaf | hi * |
| 17:33.27 | beek | jaytee -- I took your advice and configured a test loopback for my PRI problem -- diagrammed here: http://www.pastebin.ca/1278904 |
| 17:34.37 | beek | I used port 4 instead of the telco and have the dialplan simply answering, then repetively playing the "congratulations" message. |
| 17:35.27 | beek | I have twenty-three phones dialed in, thus am using all 23 channels on spans 1,2 & 4. It has been happily running since last night at 7:00pm. With the office phones all playing this, you ought to hear the cacophony. |
| 17:35.35 | invalidrecord | is it me or is the orilley asterisk book a lot more superficial than mpst orilley books therte seems verry little in in rerally |
| 17:36.18 | beek | My question is: given that I have had zero errors since this began, can I assume that the problem is definitely the Telco, despite their protestations to the contrary? |
| 17:36.23 | [TK]D-Fender | invalidrecord: I find it a little to "open theory" without enough practical explanation and implementation |
| 17:36.31 | [TK]D-Fender | invalidrecord: Fails to drive home some important points. |
| 17:37.09 | invalidrecord | [TK]D-Fender: where should i be looking this has fallen on me and I have to work it out but this ook dosent have what i need really |
| 17:37.33 | [TK]D-Fender | invaWhat, just for billing? |
| 17:37.33 | jaytee | beek, yes barring the one point that it still could be problems with the CSU and the only way to determine that is to replace it with a known good one. |
| 17:38.13 | beek | jaytee: I used the CSU that is currently hooked up. I just ran a cross-over cable from span4 to the "net" connection on the CSU I was connected to the telco with. |
| 17:38.26 | jaytee | invalidrecord, check some of the items on this page, it may be of help: http://www.voip-info.org/wiki/view/Asterisk+billing |
| 17:38.35 | beek | I configured span 4 to use _net signalling. |
| 17:39.03 | invalidrecord | well i work for a virtual office company and we provide an voip network handeling call answering for multiple companies and the associated billing no outgoing calls except the odd forwarding of an extension to a pre difined number |
| 17:39.58 | beek | jaytee: Thanks for the tip. Now I go back to do battle with the Telco tomorrow. |
| 17:40.04 | jaytee | so you're currently using the same CSU as part of the loopback? that's the part of the diagram that threw me off but now it makes sense. If you're not getting errors now I'd document it thoroughly and then call your telco and somewhere in the conversation mention the word "lawyers". :-) |
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| 17:41.47 | beek | jaytee: That's what I'm going to do. I had a gal from the telco here on Thursday and she showed me on her T-Berd that the errors were coming from "my side." I showed her the errors on the CSU that pointed to the "Net" side. Hooking things up this way at least allows me to test all of the cabling and equipment that's my property. |
| 17:43.39 | beek | Oh, and Allison just counted the number of times she's repeated that prompt on my phone: 1,828. |
| 17:43.42 | jaytee | beek, if she says her T-Berd was showing the errors coming from your side, which point did she say the errors were introduced? if they're sending timing and you're getting frame slips but that goes away when you loopback how could it be your end? especially if you changed nothing in the configuration of the cards. |
| 17:44.41 | beek | jaytee: All she showed me were the error counts on the equipment in their office. The "outgoing" line showed a couple but the "incoming" side showed a metric shitload. |
| 17:45.05 | andreaf | hi, i'm looking for information about to routing all outcoming call to a SIP provider with asterisk. I'm looking for tips and link on google (i'm googling also with not so good success) |
| 17:45.52 | andreaf | googling on voip-info.org |
| 17:46.05 | jaytee | and yet if you eliminate the span to them, it all works perfectly? what does that tell you? |
| 17:47.17 | beek | jaytee: exactly. I wanted to ensure that my test harness was valid, which you confirmed, so now I have the amunition to go back into battle. |
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| 17:48.05 | jaytee | beek, good luck. getting most telcos to cooperate takes patience and a certain amount of luck. They always want to blame the CPE side for faults. |
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| 17:48.40 | beek | I assumed that configuring span4 as a master, using the internal clock, then inserting that into where the telco plugs in should give me a conclusive and convincing test. |
| 17:49.08 | jaytee | beek, it should |
| 17:49.55 | beek | jaytee: well, thanks again for all of your help. I'm a T1/PRI newbie and the telco can easily blow smoke up my ass. It's nice to have a resource such as yourself who has done this before to help debunk what they tell me. |
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| 17:51.44 | jaytee | beek, you can also enable debug info in the logs and turn on pri debug or pri intense debug at the CLI to capture the info to the log files but the log files will get BIG fast so you don't want to leave it like that for too long. |
| 17:52.22 | beek | jaytee: I'll do that. I have a terabyte available for logging and if that's what it takes to capture the problem, I'm willing to wallow through it. |
| 17:52.25 | etfonhomey_ | jaytee, that should be a capital, bolded, underlined ALWAYS. It's SOP for them. |
| 17:52.58 | jaytee | beek, I'm still something of an * newbie myself. I've had over ten years working with T1's but not constant enough to consider myself an "expert" but enough to get by on. |
| 17:53.01 | [TK]D-Fender | andreaf: "routing all outcoming call to a SIP provider with asterisk" ? Where are the calls coming from? And then going to? What do you have currently? |
| 17:53.42 | jaytee | etfonhomey_, yeah, I'll have to remember that :-) |
| 17:53.45 | etfonhomey_ | jaytee, although I actually got some halfway decent service from the telco the other day with a telecommuter's DSL connection |
| 17:54.13 | beek | jaytee: Thanks again and I'll let you know how I make out. |
| 17:54.54 | jaytee | etfonhomey_, to be honest I haven't really run into major headaches with my current provider, Time Warner Telecom. Their people at their NOC have actually been very helpful and didn't automatically assume it was a problem at my end when I had issues. |
| 17:55.01 | etfonhomey_ | jaytee, the connection was: DSLmodem --> ASA --> PC and the user kept losing their Internet connection. The telco guy after coming out and telling us it was my Cisco ASA, left us an extra DSL modem. |
| 17:55.15 | jaytee | beek, your welcome. good luck wrestling with them. |
| 17:55.29 | jaytee | etfonhomey_, lol |
| 17:55.56 | etfonhomey_ | jaytee, I configured the second DSL modem for the default (no bridge mode) so that when the user lost Internet they could undock their computer and move the phone cable to the 2nd DSL modem and plug directly in to test the connection. |
| 17:56.13 | etfonhomey_ | jaytee, Doing so proved it wasn't the ASA. |
| 17:56.15 | andreaf | TKD-Fender are originate inside the asterisk (with the sample.call) and have to go to a "real" number tought a SIP provider |
| 17:56.34 | andreaf | sorry outgoing not outcoming, my fault |
| 17:56.39 | jaytee | etfonhomey_, don't ya love it when they actually hand you the ammo to shoot their legs out? |
| 17:57.17 | etfonhomey_ | jaytee, I hate PPPoE, but for this user's location, DSL is faster than cable Internet. But anyway, back on topic... |
| 17:57.22 | [TK]D-Fender | andreSo you want * ot Originate a call like an automated message system? |
| 17:57.31 | andreaf | tkd-fender: something like a "default route" in ip network. All call generate inside the asterisk are routed to a SIP provider. |
| 17:58.00 | andreaf | yes: i can originate the call (with originate or the sample.call script) but I have to route it to a SIP proxy (external) where I have an accoutn |
| 17:58.21 | [TK]D-Fender | andreaf: What would these calls be doing? |
| 17:59.01 | andreaf | TK-DFender: polls, adv, automatic outbound callcenter something like that. |
| 17:59.39 | andreaf | tk-dfender: but the problem is that i can't configure my asterisk to "peer" to another sip proxy, apparently failed the authentication |
| 18:00.19 | [TK]D-Fender | andreOk, well you should go install *, install a soft-phone and sit down with the BOOK and start learning how the dialplan works. This is the hard part. |
| 18:00.55 | [TK]D-Fender | andreaf: Ok, failing to authenticate is another matter. |
| 18:01.22 | andreaf | yep, i'm full of docs, i can originate the dial (more or less) with "originate" cli command. but apparently the authentication with the SIP proxy fail (and the call is not routed) |
| 18:01.30 | [TK]D-Fender | andreaf: pastebin your SIP.conf masking only passwords and removing all comments, and include the CLI output of your failed attempt at verbose 10, SIP DEBUG enabled. |
| 18:01.32 | [TK]D-Fender | ~pb |
| 18:01.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 18:01.34 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 18:01.56 | andreaf | ok, |
| 18:02.50 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 18:03.50 | andreaf | tk-d fender: it works, sorry I use the "magic" parameter in sip.conf ";auth=mark:topsecret@digium.com" |
| 18:03.55 | andreaf | yeppa! |
| 18:04.41 | andreaf | with my username and password on the sip proxy esternal server. quite a woodo I think, but my mobile handset rings :) |
| 18:05.46 | andreaf | tkd-fender with in the CLI originate SIP/+MYNUMBER@PROVIDER application Dial(SIP/+MYNUMBER) |
| 18:12.05 | *** part/#asterisk Akiyuki (i=TuxGuy@cpe-065-184-194-136.ec.res.rr.com) |
| 18:14.10 | *** join/#asterisk Mark17 (n=mark@vnc.tt.streamservice.nl) |
| 18:16.22 | x86 | I'm trying to do 'core show channels concise' from an AGI, but i'm not having any luck... keeps failing saying "unable to find application 'core'" |
| 18:16.39 | x86 | any ideas what I can do to get the current channel stats? |
| 18:17.10 | Mark17 | hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device |
| 18:17.24 | Mark17 | if needed i am willing to pay for the help with paypal |
| 18:18.34 | [TK]D-Fender | x86: And you're failing to show us exactly HOW you're calling that. |
| 18:19.14 | seanbright | AND cross-posting to -dev |
| 18:19.28 | [TK]D-Fender | to DEV? Good greif |
| 18:19.29 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 18:19.33 | jaytee | tsk tsk |
| 18:19.34 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 18:19.53 | invalidrecord | [TK]D-Fender: can i ask you a q bassed on the orilley book do you have a copy? trying to get my head round contexts |
| 18:20.01 | seanbright | ~ask |
| 18:20.01 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:20.04 | [TK]D-Fender | ^^ |
| 18:21.47 | invalidrecord | ok well on page 133 they have a dialplan with 2 contexts defined the employees one defines 101 and 102 extensions, my question is couldnt the same ability to dial 101 and 102 in the emplyes context be acvhived with just the internal context and dialing 1 & 2 i dont see how they are that diffrent |
| 18:22.34 | invalidrecord | must be missing something |
| 18:23.30 | [TK]D-Fender | invalidrecord: this is showing how to make an IVR. |
| 18:24.04 | [TK]D-Fender | invalidrecord: When you dial a company, do you expect to dial somehting like "1" to reach the sales dept? Or expect to have to puch 10 digits? |
| 18:24.21 | seanbright | x86: get_variable |
| 18:24.27 | x86 | seanbright: ok |
| 18:24.39 | [TK]D-Fender | invalidrecord: Yes they lead to the same person, but they may have a more direct & privately known number so you can JOE who just happens to be in the SALES dept |
| 18:24.57 | x86 | seanbright: [Dec 7 12:24:56] ERROR[27187]: pbx.c:2772 ast_func_read: Function CHANNELS not registered |
| 18:25.14 | seanbright | x86: pastebin the actual script |
| 18:25.15 | invalidrecord | cool ok i was just checking i wasnt midssing some nuance |
| 18:25.18 | [TK]D-Fender | invSo somebody calling your company only has to press "1" to talk to joe in sales and not have to know his longer "direct" like number |
| 18:25.47 | invalidrecord | so they are practicaly aliases which also define routing |
| 18:26.27 | x86 | seanbright: http://pastebin.ca/1278932 |
| 18:26.35 | [TK]D-Fender | invalidrecord: Not really... the don't do the exact same things... just SOME things in common. |
| 18:26.35 | Corydon76-dig | x86: are you using 1.6 or 1.4? |
| 18:26.36 | seanbright | x86: and actually, the error is pretty clear. you don't have the CHANNELS function loaded. |
| 18:27.10 | invalidrecord | but calling 101 would be effectivley the same as just dilaing 1 in this example |
| 18:27.35 | x86 | Corydon76-dig: 1.6.0.1 |
| 18:27.47 | [TK]D-Fender | invalidrecord: No.... 101 dials them INDEFINITELY. the other will only ring for 10 SECONDS. And then even play a MESSAGE. Is this the SAME? I don't think so. |
| 18:28.06 | [TK]D-Fender | x86: Where did you invent that parameter-less function from? |
| 18:28.07 | x86 | seanbright: does it ship with astrisk 1.6.0.1? |
| 18:28.10 | invalidrecord | ok i wasnt specific enough, i have it thznks@! |
| 18:28.20 | x86 | [TK]D-Fender: from Corydon76-dig |
| 18:28.24 | [TK]D-Fender | x86: this is not something that AGI does.. this is for AMI <- |
| 18:28.34 | x86 | [TK]D-Fender: not according to Corydon76-dig |
| 18:28.51 | [TK]D-Fender | x86: Continuing on your "how do I dump my channels |
| 18:29.10 | [TK]D-Fender | x86: AGI is not a means of dumping channels. AGI controls dialplan apps. |
| 18:29.14 | invalidrecord | good news is for now i only need one context per company :-) no ivr or anything clever |
| 18:29.30 | [TK]D-Fender | x86: AGI is a way of ening up in an outside script that you can go make AMI calls through |
| 18:29.36 | x86 | [TK]D-Fender: cory said in 1.6.0.1 I can use GET VARIABLE CHANNELS() to get a list of active channels |
| 18:29.41 | Corydon76-dig | x86: it's only in 1.6.1 |
| 18:29.50 | x86 | oh, weak! |
| 18:30.09 | x86 | 1.6.1 is stable? |
| 18:30.15 | invalidrecord | how far from stable is 1.6 if im developing a new app should i target 1.6? |
| 18:30.15 | seanbright | not released |
| 18:30.16 | Corydon76-dig | Yes |
| 18:30.18 | [TK]D-Fender | invalidrecord: Every device can dial the exact same things? No special inbound processing? |
| 18:30.41 | x86 | Corydon76-dig: no way I can compile just that module to handle CHANNELS() and plug it into 1.6.0.1? |
| 18:30.54 | invalidrecord | no no special processing we have multiple companies that have one main extension that incomming is routed to and the others can only call eachother |
| 18:31.11 | Corydon76-dig | x86: It might or might not work |
| 18:31.18 | Corydon76-dig | I have not tried it. |
| 18:31.20 | x86 | :( |
| 18:31.37 | [TK]D-Fender | x86: Just use AMI and get it DONE already... |
| 18:31.49 | x86 | well I would use AMI, but Asterisk::AMI seems to have trouble getting full responses back from 1.6 |
| 18:32.04 | [TK]D-Fender | x86: You're wasting time on stuff they will get released "eventiually" and stable... who know how much firther past that... |
| 18:32.08 | Corydon76-dig | I don't see anything in the module that depends upon 1.6.1 features |
| 18:32.13 | invalidrecord | no outgoing or anything all we want is to have one incomming number that goes to their first extension and the ability for that to transfer to other extenbsiuons in their context |
| 18:32.26 | invalidrecord | extensions |
| 18:32.36 | [TK]D-Fender | invalidrecord: Guess you could do that cleanly enough in 1 context. |
| 18:32.40 | x86 | Corydon76-dig: not sure, but it doesn't work... |
| 18:32.43 | [TK]D-Fender | invalidrecord: Still kind of crappy |
| 18:32.54 | x86 | Corydon76-dig: Asterisk::AMI is a bit crappy anyway |
| 18:33.04 | invalidrecord | [TK]D-Fender: how would you attack it then |
| 18:33.15 | x86 | Corydon76-dig: also, it's event-based... |
| 18:33.18 | Corydon76-dig | x86: you asked for a way to do it |
| 18:33.22 | x86 | Corydon76-dig: I'd like to use this from an AGI |
| 18:33.40 | x86 | Asterisk::AMI has to run an eventloop, which sucks |
| 18:33.40 | Corydon76-dig | CHANNELS() is the only way from AGI |
| 18:33.45 | x86 | hmm |
| 18:33.58 | x86 | is there a tarball of 1.6.1? |
| 18:34.03 | [TK]D-Fender | invalidrecord: Well if every device is just as functional as the rest, no IVR's, or anything at all more that what you've described then sure. |
| 18:34.11 | Corydon76-dig | Sure, there's a beta tarball |
| 18:34.17 | x86 | w00t |
| 18:34.18 | x86 | url me? |
| 18:34.27 | Corydon76-dig | downloads.digium.com |
| 18:34.29 | [TK]D-Fender | x86: Eventloop? No need to sit for the response. |
| 18:34.30 | x86 | hehe |
| 18:34.31 | invalidrecord | ok cool all we do is provide an answering service for time being |
| 18:34.41 | Corydon76-dig | pub/telephony/asterisk/releases/ |
| 18:34.50 | x86 | [TK]D-Fender: I'm relying on the response, why wouldn't I want to wait for it? |
| 18:35.12 | [TK]D-Fender | x86: not as an event. What are you doing exactly to try this currentky? |
| 18:36.36 | x86 | [TK]D-Fender: Asterisk::Manager's sendcommand() |
| 18:37.24 | [TK]D-Fender | x86: AMI "COMMAND"? |
| 18:37.28 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 18:37.38 | x86 | [TK]D-Fender: looks like, yes |
| 18:37.47 | [TK]D-Fender | x86: Works really straightforward for my use of dumping queues. |
| 18:37.57 | [TK]D-Fender | x86: Don't come back as an "event" |
| 18:38.24 | x86 | [TK]D-Fender: from the source code of the module, looks like I can capture the output to a hash by calling it like: %myhash = $ami->sendcommand('foo') |
| 18:38.42 | x86 | my %myhash only ends up with some status line and "channels will follow" |
| 18:38.58 | x86 | think it's a bug with Asterisk::Manager, seems to be trauncating output |
| 18:39.31 | [TK]D-Fender | x86: Don't forget EOL stuff in there |
| 18:39.42 | [TK]D-Fender | x86: I parse the string out the hard way |
| 18:40.31 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:43.05 | x86 | [TK]D-Fender: opened the output in a text editor, there is nothing extra in the file :( |
| 18:43.20 | x86 | i'll upgrade to 1.6.1-beta3, no biggie |
| 18:44.48 | [TK]D-Fender | x86: Congratulations on jumping from 1.2 right though "bleeding edge" |
| 18:47.20 | x86 | well I've been using 1.4 in an outbound call center for some time |
| 18:47.50 | x86 | ugh, 1.6.1-beta3 doesn't compile |
| 18:48.49 | x86 | http://pastebin.ca/1278949 |
| 18:52.06 | x86 | gah, all this is wasting my time, it's not possible to do what I want anyway |
| 18:52.18 | x86 | I want to pickup a channel that's already answered, using PickupChan |
| 18:52.57 | x86 | I pinned a call up, got the channel name, put an extension in my dialplan to pickup that channel name, and it still fails with "No target channel found for SIP/7001-08f5ff60." |
| 18:53.06 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:54.22 | x86 | I want to barge in on a bridged call, with full two-way audio (which rules out ExtenSpy / ChanSpy) |
| 19:03.50 | [TK]D-Fender | x86: BRIDGE <- |
| 19:07.27 | x86 | [TK]D-Fender: does that require me to know the full channel name? like SIP/7000-abcdef12? or can I use it like SIP/7000? |
| 19:08.40 | [TK]D-Fender | SIP/7000 is not a channel |
| 19:09.33 | *** join/#asterisk j-tek (n=NTech@86.105.21.220) |
| 19:10.22 | x86 | so I run into the same problem that I must somehow extract the channel name |
| 19:12.42 | [TK]D-Fender | x86: Why is it we never see exactly what's going on? |
| 19:14.05 | invalidrecord | how do i specify a prefix for a context so say a global operator can dial 123 exten 1000 or 124 exten 1000 |
| 19:15.21 | [TK]D-Fender | invalidrecord: that makes no sense |
| 19:15.56 | [TK]D-Fender | invalidrecord: I think I see what you're getting at.... |
| 19:16.18 | invalidrecord | im prob using wrong terms :-S |
| 19:16.30 | [TK]D-Fender | invalidrecord: You are using numbered contexts? |
| 19:16.50 | invalidrecord | was thinking it would be best as we have deskphones |
| 19:16.51 | [TK]D-Fender | invalidrecord: like [123] for company "123"? |
| 19:16.56 | invalidrecord | yeah |
| 19:17.18 | invalidrecord | so our main operator can dial each company by prefix then exten number |
| 19:17.48 | [TK]D-Fender | invalidrecord: then something like "exten => _XXXXXX,1,Goto(${EXTEN:0:3},${EXTEN:3},1) |
| 19:18.29 | invalidrecord | ok ill go work out how that matches and implement thanks |
| 19:20.35 | hardwire | pokes Carlos_PHX |
| 19:21.22 | *** part/#asterisk j-tek (n=NTech@86.105.21.220) |
| 19:21.35 | invalidrecord | [TK]D-Fender: is numbered contexts the wasy to go? |
| 19:21.38 | invalidrecord | way |
| 19:21.38 | *** join/#asterisk ShaunWing (n=chatzill@dsl-243-95-10.telkomadsl.co.za) |
| 19:22.31 | *** join/#asterisk andresmujica (n=andresmu@201.244.108.160) |
| 19:22.48 | [TK]D-Fender | invalidrecord: You need to sit down and think out your deployment. You should alrady be loking at the ups & downs of each. Go try stuff. |
| 19:23.16 | farkus | I'm having a problem where my call audio sounds fine, but the monitor recordings have short periods of silence ( afew milliseconds) occasionally that makes the recording sound jumpy |
| 19:23.19 | ShaunWing | say, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas? |
| 19:23.43 | invalidrecord | yeah im just asking as i think it through should prob test more ask less. Thanks man! |
| 19:26.21 | *** part/#asterisk ngeloa (n=ang@81-208-36-86.ip.fastwebnet.it) |
| 19:30.43 | x86 | [TK]D-Fender: because you suck at asking for what you want to see heh |
| 19:32.04 | x86 | I'm having a terrible time with a Linksys SPA962 and a Linksys SPA942 with paging... I can page FROM X-lite to both Linksys phones and it works like a champ, but if I page FROM either Linksys phone, the call is cancelled almost immediately |
| 19:32.36 | x86 | any idea why that might be? |
| 19:33.04 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:33.44 | jaytee | x86, because you suck at configuring Linksys phones, heh |
| 19:33.59 | x86 | jaytee: tis why I'm asking for help? :) |
| 19:34.11 | jaytee | x86, was just kiddin ya man! |
| 19:34.16 | x86 | hehe |
| 19:34.30 | x86 | jaytee: have you ever done paging with linksys phones? |
| 19:34.55 | jaytee | x86, if ya want a real hemorrhoid try the SPA-3102 ATA. Works like a champ when you get it working but configuring it is a pain in the ass |
| 19:35.05 | jaytee | x86, no I use Polycom phones |
| 19:35.44 | x86 | wish I could say the same... boss bought these POS linksys phones without first asking me |
| 19:36.12 | x86 | of course, the salesman always knows more than the technical guy implementing the system... |
| 19:36.15 | [TK]D-Fender | x86: Can't see your code, can't see the resulting variables to see if you extra crap chars. Don't see channel dumps to back up the one's you're trying to grab |
| 19:36.21 | [TK]D-Fender | .x8you suck at showing ANYTHING. |
| 19:36.43 | x86 | sure do, because I have no idea what you need until you ask |
| 19:37.00 | [TK]D-Fender | x86: Your code doesn't work? SHOW US THE CODE |
| 19:37.10 | x86 | here, I'll do a sip debug on the peer (linksys 962) trying to page |
| 19:37.12 | [TK]D-Fender | x86: is this the right channel? DUMP THEM |
| 19:37.19 | jaytee | don't know what the equivalent setting is on Linksys but on a Polycom it's ring-answer. I modified that in the Polycom configs to use a particular ring type and a duration of 1200 ms and then use SipAddHeader to send Alert-Info: Ring Answer to the phone when dialing, then I've got it setup in the dialplan so any extension you dial you just add an 8 in front of it for paging, intercom. |
| 19:38.23 | *** join/#asterisk viliar (n=viliar@viliar.dialup.corbina.ru) |
| 19:40.38 | etfonhomey_ | jaytee, like this: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 19:40.40 | etfonhomey_ | ? |
| 19:40.41 | x86 | [TK]D-Fender: http://pastebin.ca/1278984 |
| 19:41.45 | viliar | hello all |
| 19:41.50 | jaytee | etfonhomey_, yep in fact I used that as a rough guide to doing it |
| 19:42.21 | [TK]D-Fender | x86: This isn't the channel dumping issue... |
| 19:42.24 | jaytee | viliar, hello |
| 19:42.29 | x86 | [TK]D-Fender: 611 is my paging extension, 7000 is my Linksys 962, 7001 is my Linksys 942, 7002 is X-Lite |
| 19:42.38 | viliar | i have a little q. |
| 19:42.45 | x86 | [TK]D-Fender: it's a SIP debug of 7000 calling the paging extension |
| 19:42.46 | [TK]D-Fender | x86: and WAY too much excess debug crap |
| 19:42.50 | viliar | probably someone can point me to right sie |
| 19:42.59 | [TK]D-Fender | x86: broken EOL chars, etc |
| 19:43.10 | x86 | [TK]D-Fender: doesn't look like broken EOL to me |
| 19:43.12 | viliar | quistion is,as i think, is noob. |
| 19:43.22 | x86 | [TK]D-Fender: actually, I think like 11 says the problem |
| 19:43.24 | jaytee | ~ask |
| 19:43.24 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:43.29 | [TK]D-Fender | x86: #334 begs to differ. Normal CLI doesn't do shit like that |
| 19:43.40 | [TK]D-Fender | x86: 1 line = 12 line. no stupid ^M's visdsible |
| 19:43.58 | x86 | [TK]D-Fender: the ^M is the EOL sent from the phone |
| 19:44.06 | viliar | there is some of astersik users. Everyone have sip acc ount and mobile phone |
| 19:44.09 | x86 | [TK]D-Fender: inside the SIP header |
| 19:44.13 | [TK]D-Fender | x86: You saying it looks like that from * CLI? |
| 19:44.22 | x86 | *nod* |
| 19:44.26 | [TK]D-Fender | x86: W-T_F |
| 19:44.37 | viliar | so I want to route call to sip, if it's unreachable - route to mobile |
| 19:44.51 | viliar | here is no problem |
| 19:45.07 | Kobaz | how would i reset the queue information in 'queue show', without having to unload/load the module |
| 19:45.26 | [TK]D-Fender | x86: But yes, a 401 sure looks like something to fix |
| 19:45.48 | x86 | not sure why it'd be 401'ing though... lemme show you my dialplan |
| 19:46.05 | viliar | but scheme is more complicated if I want to move some calls to more than one account. |
| 19:46.21 | [TK]D-Fender | x86: no need |
| 19:46.30 | [TK]D-Fender | x86: 401 has nothing to do with dialplan |
| 19:46.31 | Kobaz | walnut*CLI> queue show |
| 19:46.31 | Kobaz | group3 has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:4, C:5, A:123, SL:4.0% within 60s |
| 19:46.40 | Kobaz | how would i reset those numbers? |
| 19:46.46 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 19:46.52 | viliar | so it's good to be, if was a they to describe abstract user and send call to him, not directly to sip or mobile. |
| 19:47.02 | viliar | *if was a way |
| 19:47.27 | jaytee | viliar, try reading up on local channels in the WIKI or in the book |
| 19:47.33 | [TK]D-Fender | viliar: Makes no sense. "abstract" what? for who? What do you want to have happen exactly? |
| 19:47.57 | viliar | ok |
| 19:47.57 | [TK]D-Fender | viliar: And there is no "direct". Everything goes through the dialplan. It does what you tell it to |
| 19:48.19 | etfonhomey_ | Kobaz, I don't see any way to do that from the CLI. |
| 19:48.20 | x86 | [TK]D-Fender: hmm, ok... what am I missing then? |
| 19:48.27 | [TK]D-Fender | x86: DEVICE AUTH. |
| 19:48.38 | x86 | [TK]D-Fender: 7000 can call any other extension just fine (and subscribe to all hints) |
| 19:48.39 | [TK]D-Fender | x86: 401 = fix your damn sip.conf |
| 19:48.40 | Kobaz | etfonhomey_: me neither... other than 'module unload app_queue' 'module load app_queue' |
| 19:48.51 | Kobaz | etfonhomey_: which seems kinda bad |
| 19:48.55 | x86 | [TK]D-Fender: what's wrong with it if it's only breaking on a single extension? |
| 19:49.07 | [TK]D-Fender | x86: turn down all that extra psycho-shit debug and show 2 SMALL samples for working vs non-working |
| 19:49.13 | etfonhomey_ | Kobax, do a "module reload app_queue" that'll do it quickly. |
| 19:49.20 | x86 | [TK]D-Fender: also, it DOES ring the other devices, but as soon as other devices start ringing, the call is cancelled BY the originating Linksys phone |
| 19:49.24 | Kobaz | etfonhomey_: it doesn;t |
| 19:49.30 | viliar | example. As if was context [user1]. where is 1,1,Dial(SIP/User1). 1,2,Dial(SIP/Out/User1Mobile) |
| 19:49.32 | Kobaz | etfonhomey_: a reload doesn't clear the stats |
| 19:49.32 | [TK]D-Fender | x86: And you'd better stop calling DEVICES as EXTENSIONS... |
| 19:49.43 | viliar | so I can route call to this context |
| 19:49.53 | x86 | [TK]D-Fender: show me example of how I did that? |
| 19:49.57 | etfonhomey_ | Kobaz, hmm, always thought a reload just did an unload/load at the same time. Guess not. |
| 19:49.59 | [TK]D-Fender | viliar: What call? From where? What did they dial? |
| 19:50.11 | x86 | [TK]D-Fender: http://pastebin.ca/1278995 |
| 19:50.33 | Kobaz | etfonhomey_: reload will call the module's internal reload function, which can do whatever it wants... it depends on how the module was written |
| 19:50.38 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 19:51.00 | [TK]D-Fender | x86: <sip:10.42.69.252>\ <-- having a fixed value like this looks bad |
| 19:51.17 | [TK]D-Fender | x86: and go show me the working call, and the failing call. |
| 19:51.19 | jaytee | shudders |
| 19:51.23 | viliar | [TK]D-Fender: call from menu is going to some user. It can be done via context user1. Main target - to move call SIP or Mobile. |
| 19:51.30 | x86 | [TK]D-Fender: the IP will never change... guess I could put it in a global var, but that's just vanity at this point, not the problem |
| 19:51.39 | [TK]D-Fender | viliar: "move call = huh? |
| 19:51.53 | x86 | [TK]D-Fender: ok, you have failing already... let me make a working call... |
| 19:51.58 | viliar | [TK]D-Fender: sorry for a bad english |
| 19:52.04 | [TK]D-Fender | x86: I said a SMALL sample |
| 19:52.16 | [TK]D-Fender | viliar: and this : 1,1,Dial(SIP/User1) will ring forever |
| 19:53.12 | viliar | [TK]D-Fender: no. after timeout it's going to next hop. It's simplified example |
| 19:53.29 | viliar | [TK]D-Fender: next hop - is mobile. |
| 19:53.41 | x86 | [TK]D-Fender: not sure how to get what you want! |
| 19:54.00 | x86 | [TK]D-Fender: sip set debug peer FOO is giving you all that crap, not sure how to make it less chatty... |
| 19:54.08 | x86 | [TK]D-Fender: tell me what you want |
| 19:54.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:54.58 | x86 | here's the big one of the working call |
| 19:54.59 | x86 | http://pastebin.ca/1279001 |
| 19:55.46 | viliar | I try to re-describe. |
| 19:55.59 | x86 | [TK]D-Fender: this is with core set verbose 0 and sip set debug peer 7002 (x-lite phone) |
| 19:56.18 | [TK]D-Fender | Ok, this crap is too much. I'm stepping out for a while. |
| 19:57.05 | viliar | Incoming call - it's going to main menu. From it it going to user1. if it unreachable - call going to user2,user3, |
| 19:57.39 | viliar | user1,user2,user3 can be registered sip account or their mobile phones as next step |
| 19:57.42 | x86 | gah, why does he have to be such an asshole?! |
| 19:57.51 | *** join/#asterisk SiberAIR (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 19:58.14 | x86 | how hard is it to just freaking tell me what you need?! |
| 19:58.20 | viliar | so i just search for apropriate way to create such scheme |
| 20:01.23 | viliar | the more simple way, as i think here is to describe in some way users and all their phones in macros or context, and than point to it from dial plan. |
| 20:01.33 | beek | viliar: would the FollowMe application do it? |
| 20:02.05 | viliar | thx. wait a second, i try to look. |
| 20:05.48 | viliar | <beek>: thx. looks good. do you think it can be used with Dial(user1,user2)? |
| 20:08.01 | beek | viliar: I'm not sure. I've never used it before. It would be easy to try it, though. |
| 20:08.08 | ShaunWing | say, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas? |
| 20:08.52 | viliar | <beek>: thanks again. will thiks around this :-) |
| 20:14.17 | ShaunWing | >say, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas? |
| 20:22.12 | x86 | ShaunWing: looks like you don't know what redirect=yes actually does |
| 20:23.07 | ShaunWing | I understand it tries to send rtp from ednpoint (Quintum) to VSP |
| 20:23.31 | ShaunWing | However c=IN shows me that redirection not taking palce |
| 20:23.34 | ShaunWing | place |
| 20:24.24 | ShaunWing | The Quintum is on piblic ip - nonat. The asterisk 1.4 server is also on public ip with nonat. The vSP is also on public ip. |
| 20:24.44 | ShaunWing | I'm chewing up bandwidth on my hosted Asterisk server....... |
| 20:29.56 | ShaunWing | Any ideas? |
| 20:30.31 | x86 | ok so you still don't understand redirect=yes |
| 20:30.36 | x86 | as it doesn't touch RTP |
| 20:30.59 | x86 | redirect allows two endpoints to re-invite and bypass the proxy |
| 20:31.57 | x86 | you might get further telling us what the end goal is |
| 20:34.12 | etfonhomey_ | ShaunWing, read about canreinvite=yes (or no), maybe that's what you're looking for. |
| 20:35.56 | ShaunWing | Sorry, Its as you say, just expressing it incorrectly as new to Asterisk. What I require is that the Asterisk in teh middle (the proxy) does nto carry the rtp through it |
| 20:37.36 | ShaunWing | The Quintum is on public ip - nonat. The asterisk 1.4 server is also on public ip with nonat. The VSP is also on public ip. I understand that adding in sip.conf reinvite = yes with allow the end points to redirect rtp directly to each other and not need the proxy..... |
| 20:37.46 | ShaunWing | but its not working in my case.... |
| 20:39.27 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 20:40.01 | Mark17 | hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device. if needed i am willing to pay with paypal. i use debian and asterisk (only configuring the part so i can create a new trunk for the bluetooth device and everything else is done is also an |
| 20:40.01 | Mark17 | option) |
| 20:40.13 | x86 | ShaunWing: canreinvite=yes is the option |
| 20:42.36 | ShaunWing | Tx, but its not working. |
| 20:43.35 | etfonhomey_ | ShaunWing, pastbin your sip.conf |
| 20:44.54 | *** join/#asterisk LND (n=LND@89.193.180.58) |
| 20:47.04 | ShaunWing | Tx. http://pastebin.com/m54934d39 |
| 20:52.14 | etfonhomey_ | ShaunWing, I've never seen trun=yes in a sip.conf before. |
| 20:52.21 | etfonhomey_ | trunk=yes* |
| 20:53.16 | ShaunWing | I understand that allows multiple channels |
| 20:53.40 | etfonhomey_ | ShaunWing, you don't need it. |
| 20:53.55 | ShaunWing | ok |
| 20:54.18 | ShaunWing | Will this fox the redirect issue? |
| 20:54.59 | ShaunWing | fix sahll I say |
| 20:55.05 | etfonhomey_ | Is this a production system? |
| 20:55.10 | *** join/#asterisk LND (n=LND@89.193.214.255) |
| 20:55.12 | etfonhomey_ | Or something you're playing around with? |
| 20:55.55 | ShaunWing | Its my server that is live but I can play aorund with it |
| 20:57.38 | etfonhomey_ | ShaunWing, then I would simply your config and add features slowly. I'm assuming RoB5768 is a local extension? |
| 20:58.09 | ShaunWing | no |
| 20:58.22 | ShaunWing | Its the registration for the Quintum |
| 20:58.36 | ShaunWing | that is sending the traffic to the Asterisk |
| 20:59.19 | ShaunWing | the reinveite=yes is supposed to tell it to redirect its rtp to the VSP (voipsw1) |
| 20:59.29 | ShaunWing | ok will make changes slowly |
| 20:59.34 | ShaunWing | tx for the advice |
| 21:00.01 | ShaunWing | but at the moment I have tried all I can think of the get the redirect to work... |
| 21:00.05 | ShaunWing | Any ideas please? |
| 21:00.06 | x86 | reinvite has nothing to do with RTP, again |
| 21:02.37 | etfonhomey_ | ShaunWing, here's my simple SIP line for my ITSP: http://pastebin.com/d310edd33 Also, you should include your [general] section and any register => xxx lines in your pastebin. |
| 21:03.37 | etfonhomey_ | ShaunWing, and don't forget to mask your usernames and passwords. |
| 21:05.27 | ShaunWing | tx. My general section is default. |
| 21:05.43 | ShaunWing | Can you please tell me what insecure=port,invite |
| 21:05.44 | ShaunWing | does? |
| 21:06.32 | etfonhomey_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure |
| 21:09.23 | ShaunWing | tx |
| 21:09.37 | ShaunWing | I see you don't use reinvite=yes |
| 21:12.05 | etfonhomey_ | That's because I want my * box between my SIP peers. |
| 21:12.31 | etfonhomey_ | RTP peers to be more accurate (and it is canreinvite=yes NOT reinvite=yes). |
| 21:13.14 | ShaunWing | ok I understand. Sorry - tx. Can you make any suggestions to get my system working the way I need , please? |
| 21:15.52 | etfonhomey_ | ShaunWing, you need to start from scratch and get a basic call working with canreinvite=no and then change to canreinvite=yes, then debug. |
| 21:16.12 | ShaunWing | Basic calls are working perfectly |
| 21:16.29 | ShaunWing | I just want the reinvite to work |
| 21:16.45 | etfonhomey_ | You have tons of other crap in your sip.conf. |
| 21:17.13 | andrewn | anyone know of a provider that can complete calls to toll free numbers that are restricted to Canada? |
| 21:21.41 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 21:25.31 | rhousand | This is alittle out side of the realm of this channel but I am not sure where to ask. I have set up my managed switch to mirror the traffic on the port where my Voip(MGCP) traffic goes out to a prot where i am running tcpdump to grab and play it with wireshark. If i grab the traffice from my router i can play it but if i grab it from the mirrored port it will not play. any ideas why? |
| 21:26.03 | rhousand | Please feel fre to recomend another channel for me to try. |
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| 21:27.33 | *** join/#asterisk ShaunWing (n=chatzill@dsl-243-95-10.telkomadsl.co.za) |
| 21:27.55 | ShaunWing | ok, happy to remove |
| 21:27.56 | ShaunWing | what should I remove please? |
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| 23:18.55 | C4colo | does any know if it is possible to have more than one audio device available to asterisk, and if so how would you 'address' them? |
| 23:19.34 | C4colo | would they be console/1 console/2 etc? |
| 23:20.28 | C4colo | I have a customer that wants up to 5 line-out channels |
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| 23:36.38 | C4colo | so is there no way to address multiple console audio ports? |
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| 23:51.19 | Mark17 | hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device. if needed i am willing to pay with paypal. i use debian and asterisk (only configuring the part so i can create a new trunk for the bluetooth device and everything else is done is also an |
| 23:51.19 | Mark17 | option) |
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