| 00:00.10 | Freman | http://www.yawarra.com.au/product.php?productCode=HW-AX1-M |
| 00:01.00 | *** join/#asterisk pcrane (n=pcrane@120.89.80.110) |
| 00:02.19 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:04.58 | *** join/#asterisk torrikft (n=afj@m85-94-187-141.andorpac.ad) |
| 00:05.10 | Freman | actually, just thought of a DAMN good reason not to use one of them boards - no power for the TDM card :) |
| 00:07.52 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 00:09.25 | coppice | The TDM card only needs external 12V, which is also what the alix card needs |
| 00:10.32 | coppice | they are rather expensive, though, and I wonder how hot they get without a fan. Most ITX boards are completely unrealistic about being fanless. |
| 00:11.29 | tzafrir | I got to look at a sample Atom-based board from Intel. |
| 00:11.52 | tzafrir | That board has no fan on the CPU. THough it has one on the GPU |
| 00:12.10 | coppice | stupid board - lovely cool processor, and stinking hot north bridge :-) |
| 00:12.42 | coppice | the south bridge chip roasts, too, but they don't put any cooling on it |
| 00:12.59 | Freman | The cpu on the alix board is warm to touch |
| 00:14.15 | coppice | Freman: even when its working? The celeron CPU on the Intel 201 ITX board is very cool, until I get some DSP cranking away. they it really needs its little fan |
| 00:14.17 | Freman | at idle |
| 00:14.40 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 00:14.48 | Freman | If anything, I might scavange the cooler from an old dead laptop I have |
| 00:15.06 | torrikft | if u buy cheap u can expect cheap |
| 00:15.16 | Freman | ie: a copper/aluminum lump on top of the cpu to join it to the case |
| 00:15.22 | Freman | meh, it's not so cheap |
| 00:15.28 | Freman | via epia's are cheaper |
| 00:15.35 | torrikft | suer |
| 00:15.59 | coppice | but the Intel 201 board has only passive cooling for its north bridge, and it goes over 100C, unless you take some action |
| 00:16.14 | torrikft | thats a design problem |
| 00:16.25 | *** join/#asterisk Maan (n=Maan@c-76-19-20-210.hsd1.ma.comcast.net) |
| 00:16.38 | coppice | yep. most ITX boards have thermal (lack of) design problems |
| 00:16.43 | torrikft | needs cooling and they should have see that coming |
| 00:16.56 | coppice | and price has little to do with that |
| 00:17.01 | torrikft | well |
| 00:17.30 | torrikft | some board builders pay more attention to cooling than others |
| 00:17.31 | coppice | they aren't being cheap. just really sloppy |
| 00:18.03 | torrikft | if u buy asus or abit u can expect the heatsink glue to come off after say 12 months of use |
| 00:18.05 | coppice | yeah, a cowboy outfit like Intel obviously shouldn't be trusted with thermal management :-) |
| 00:18.32 | Freman | if you buy abit, you can expect the fan to stop operating after 12 hours of use |
| 00:18.40 | torrikft | well the atom processor is not their flagship product |
| 00:19.31 | coppice | and if you buy from intel you can also expect endless trouble. there are no reliable sources |
| 00:19.44 | torrikft | and i can say im using a lifebook p8010 atm with centrino 2 that has thermal issues as well and it wasnt cheap |
| 00:20.38 | torrikft | so true |
| 00:20.46 | torrikft | we buy pure crap |
| 00:20.50 | torrikft | and when it breaks down |
| 00:20.55 | torrikft | we buy more |
| 00:21.22 | torrikft | and they keep renaming it so that we need to upgrade |
| 00:21.29 | Maan | hi all...anyone know if linksys routers (WRT54GS) do any SIP rewriting (messing with the ports)? |
| 00:21.32 | torrikft | only to find the same problems again |
| 00:21.48 | Freman | yeh the alix can provide the 12v nesicary to run the tdm... |
| 00:21.53 | Freman | I'm just not sure it's worth it :) |
| 00:22.11 | Freman | it's not like I'm paying the power bill |
| 00:22.33 | torrikft | Maan whats the problem? |
| 00:22.41 | torrikft | blaming the router are we? |
| 00:23.03 | torrikft | i never liked them linksys routers |
| 00:23.11 | Freman | eigh? I aint blaming nothing, just trying to work out wtf I can be bothered building for $0 profit |
| 00:23.19 | Freman | oh maan not man :) |
| 00:23.20 | Maan | torrifkt: i'm manually sending SIP packets to an outside host, and sure enough the message is different when it reaches the destination |
| 00:23.59 | Maan | torrifkt: so i'm trying to figure out who/what is changing the packet. wondering if comcast could be the culprit... |
| 00:24.08 | torrikft | whats running on the destination? |
| 00:24.11 | torrikft | router firewall? |
| 00:24.12 | nvez | Maan: knowing its a linksys, i think you can install linux on most of them.. |
| 00:24.13 | torrikft | maybe a pix? |
| 00:24.48 | Maan | i have myhost---linksys router---comcast---server |
| 00:24.53 | torrikft | comcast love their packet inspection |
| 00:24.54 | Freman | What sort of 2 port FXO external gateway would ya'll reccommend (or 2 port FXO + 2 port FXS) |
| 00:25.05 | torrikft | Maan pm? |
| 00:25.53 | x86 | Freman: get (1) 2FXO sipura ATA, and (1) 2FXS sipura ATA |
| 00:26.09 | x86 | Freman:or an Astribank ;) |
| 00:26.50 | Freman | hmmm can't find any 4 port astribanks :) |
| 00:28.04 | Freman | How about AUDIOCODES MP114 2FXS 2FXO? |
| 00:28.33 | Freman | or should I stick to a TDM card and be done with it |
| 00:35.07 | coppice | whichever product you say, you'll get some reports of people loving them and some of them hating them |
| 00:36.06 | Freman | I know, but for those saying they hate them I'd hope for a reason why :) |
| 00:38.46 | *** join/#asterisk Yourname`` (i=chatzill@unaffiliated/yourname/x-837320) |
| 00:38.53 | coppice | they usually have what *appears* to be a valid reason. e.g. pick any product and someone will tell you it causes choppy audio. |
| 00:39.26 | Yourname`` | Hi, quick uuestion. Is there a way for an agent to log in and accept calls from multiple queues without the need of them being static? |
| 00:40.28 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 00:40.28 | coppice | "that causes choppy audio" is becoming the new "that's been photoshopped" :-) |
| 00:40.50 | Katty | i'll photoshop you in a minute. |
| 00:40.58 | Katty | <PROTECTED> |
| 00:42.07 | Katty | we had some choppy audio problems. the telco wanted to blame the sangoma card. |
| 00:42.23 | Katty | turned out to be QOS issues. |
| 00:43.00 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167043089.pppoe-dynamic.nb.aliant.net) |
| 00:43.19 | file | snakes in an IRC channel! runnnnnnnnnnn! |
| 00:43.33 | Katty | see this is why i love file. |
| 00:43.33 | tzanger | haha |
| 00:43.52 | file | it was alexhopper's idea |
| 00:43.56 | alexhopper | Calm it down denzel... |
| 00:44.23 | alexhopper | err |
| 00:44.25 | file | hugs on Katty |
| 00:44.25 | alexhopper | samuel |
| 00:45.00 | Katty | pamples file |
| 00:45.09 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 00:45.57 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-67b8588aad00db2d) |
| 00:48.02 | *** join/#asterisk seanmh (n=johndoe@c-68-84-145-133.hsd1.nm.comcast.net) |
| 00:49.16 | Qwell | alexhopper: I hear that file guy is a nub |
| 00:50.20 | alexhopper | You have NO idea! |
| 00:50.22 | alexhopper | :p |
| 00:55.08 | torrikft | anybody have experience with a provider of SIP/IAX termination based in europe? |
| 00:57.57 | angryuser | torrikft dopends on country |
| 00:58.16 | Freman | hehe, a nice embedded system around an alix board will cost me about $500 and limit me severely - a dual core machine with raid will cost me 494 and not limit me... |
| 00:58.46 | torrikft | angryuser spain, france, uk, holand, finland, denmark, germany |
| 00:58.57 | torrikft | all of those i can reach with less that 40ms |
| 00:59.46 | angryuser | torrikft try keyyo or laligne it's in france, and again if the suit you |
| 00:59.59 | angryuser | they* |
| 01:01.30 | torrikft | will have a look at their rates |
| 01:05.59 | nvez | what type of service should I be looking for if I want to run an asterisk box and am going to use a SPA2002 with it for inbound/outbound calls |
| 01:06.30 | angryuser | nvez voip provider |
| 01:07.05 | angryuser | nvez and you dont need asterisk if you got only one spa |
| 01:07.38 | Katty | mmm, spa |
| 01:07.46 | Katty | book me an apointment ;) |
| 01:08.15 | nvez | angler: im going to need other stuff because i need two lines (fax/normal phone) |
| 01:08.15 | angryuser | different rates apply |
| 01:08.57 | nvez | ofcourse. so what type of service (name) am I looking for? |
| 01:09.42 | Katty | nvez: service called mudbath. |
| 01:09.46 | Katty | nvez: or chocolate bath |
| 01:09.50 | Katty | nvez: book me an hour ;P |
| 01:10.33 | *** join/#asterisk exothermic (n=miles@74.85.89.236) |
| 01:11.08 | angryuser | nvez uhh read what i wrote |
| 01:11.15 | exothermic | Is there a good solution out there for agents that need to both be able to be in an inbound queue and make outbound calls? |
| 01:12.13 | angryuser | exothermic use hints on that extensions , and normally they are reported in use when calling out |
| 01:13.18 | angryuser | exothermic maybe i am wrong ;) |
| 01:13.56 | *** join/#asterisk keith4_ (n=kbe2@207-172-236-173.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 01:14.17 | exothermic | ya looking for a solution where the queue system knows the device is on an active call and doesn't try to send another call down to it. |
| 01:14.40 | keith4_ | does DND on linksys SPAs affect all extensions? |
| 01:15.21 | angryuser | try what i suggested, just declare hints, and dont forget to add call-limit=n in sip.conf to make them work |
| 01:15.55 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 01:16.30 | exothermic | but that is on a device basis? |
| 01:16.41 | *** join/#asterisk pcrane (n=pcrane@120.89.80.110) |
| 01:16.47 | Katty | anyone besides me use, or have heard of, isymphony? |
| 01:17.14 | angryuser | exothermic yes |
| 01:17.15 | seanmh | I have |
| 01:17.29 | Katty | seanmh: you like it better than HUD? |
| 01:17.32 | seanmh | but then again I work for the company that makes it ;) |
| 01:17.40 | Katty | oh. |
| 01:17.47 | Katty | you must be the sean that keeps fixing my license problem ;) |
| 01:17.53 | exothermic | exothermic: Ya I'm already using something like that (group count vars etc) works somewhat but still shows agents rejecting calls because the queue system tries to deliver the call. |
| 01:17.55 | seanmh | Can I assist you in some way with it? |
| 01:18.19 | Katty | seanmh: not unless you know how to make my nic stop getting a new mac address after reboot ;) |
| 01:18.31 | seanmh | hrmm.. I don't.. this is a VM? |
| 01:18.38 | Katty | no. |
| 01:19.03 | angryuser | exothermic group counts has nothing to to woth hints |
| 01:19.10 | Katty | it's ok. |
| 01:19.19 | LemensTS | your nic gets a new mac? thats strange |
| 01:19.21 | seanmh | really? that's odd.. if you can call in tomorrow and ask for Mike he should be able to help you in someway 505-246-4220 |
| 01:19.21 | Katty | seanmh: i figured out how to fix it once...i'll find my note somewhere |
| 01:19.39 | seanmh | yeah, that'll definitly cause problems with your iSymphony license |
| 01:20.02 | exothermic | angryuser: Ya but the end result is the same, it is information that is outside of the queue manager, so the queue manager doesn't know the phone is on the call, yes the phone won't actually ring, but queue manager will try. |
| 01:20.08 | *** join/#asterisk UD (n=steve@unaffiliated/underdawg) |
| 01:20.12 | UD | wow this is cool |
| 01:20.17 | Katty | seanmh: it's all good. i called mike on friday |
| 01:20.23 | UD | hi |
| 01:20.31 | jaytee | hi |
| 01:20.32 | Katty | seanmh: and thursday |
| 01:20.45 | angryuser | exothermic are you satisfyed with your end result ? |
| 01:20.47 | seanmh | Katty: Hrmm.. that's a strange problem |
| 01:21.02 | jaytee | yeah, a MAC address should never change on it's own. |
| 01:21.03 | UD | im not even sure where to start with this program |
| 01:21.08 | Katty | indeed. |
| 01:21.16 | jaytee | UD, how about here? |
| 01:21.18 | jaytee | ~book |
| 01:21.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:21.20 | Katty | but everytime i reboot, or network restart, i get a new eth number. |
| 01:21.25 | Katty | we're up to... hrmm, eth11 i think |
| 01:21.34 | jaytee | a new IP address or a MAC address |
| 01:21.42 | UD | i have the paperback beside me |
| 01:21.52 | Katty | well the machine is set staticly |
| 01:21.57 | Katty | but it doesn't think it's the same nic |
| 01:21.58 | UD | oreilly rambles too much and the book's obsolete |
| 01:22.04 | Katty | so it picks a dynamic one from the firewall |
| 01:22.26 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 01:23.34 | angryuser | exothermic if not write some hints for that SIP/ZAP users you have and see if it works better |
| 01:24.04 | UD | is there a good example somewhere of configuring a basic conference room? |
| 01:24.24 | Katty | UD: my blog. |
| 01:24.29 | exothermic | exothermic: Ya not satisfied with the end result, working up the hints now, but 99% sure that isn't going to improve my solution. |
| 01:24.53 | Katty | UD: http://angela.sleekgeek.org/category/geekery/linux/asterisk/ |
| 01:24.56 | exothermic | angryuser: Although it looks like pausing an agent might work |
| 01:25.03 | Katty | UD: there's a lot of... noob posts there, i guess you could say. |
| 01:25.22 | angryuser | exothermic there is a chance that state of devices are reported to queue agent |
| 01:25.46 | exothermic | angryuser: You have some examples of use of the hints? |
| 01:25.48 | angryuser | exothermic another solution is to manyally change state of device before call |
| 01:25.50 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 01:28.12 | angryuser | exothermic example i have ext 30,1,Dial(Sip/30) exten => 100,hint,SIP/peername |
| 01:28.48 | angryuser | exten => 30,hint,SIP/30 |
| 01:30.29 | angryuser | exothermic http://www.voip-info.org/wiki/view/Asterisk+presence |
| 01:30.38 | UD | so the book explains it eh? |
| 01:32.08 | UD | i've found a lot of oreilly books kinda really describe how to use redhat 7.1 more than they do on the subject |
| 01:32.38 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 01:33.20 | UD | like kinda blah blah noshit gettothepoint <old script that doesnt work> blah blah <20 pages that are in the readme anyway and are outdated> blah etc |
| 01:34.23 | angryuser | UD this one will describe you asterisk |
| 01:34.25 | LemensTS | just got the openSER oreilly book friday |
| 01:34.39 | exothermic | LemensTS: It any good? |
| 01:35.27 | LemensTS | ive not had much time to read it this weekend, but from the few pages ive read it really goes into detail on the sip protocol. It would be good for just an asterisk user to read. |
| 01:35.52 | LemensTS | then they would know what invite, ack, etc means |
| 01:36.16 | angryuser | asterisk user will never read that |
| 01:36.31 | angryuser | ;) the will use asterisk |
| 01:36.59 | nvez | angryuser: so ive looked into "voip service", but wouldnt i need a specfiic sort of sip trunking or something to make it work with asterisk? |
| 01:38.19 | angryuser | nvez yes kind of, http://www.voip-info.org/wiki-SIP |
| 01:38.48 | nvez | cause i saw that not all provider have that |
| 01:38.55 | nvez | like vonage, etc. |
| 01:39.08 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:39.10 | keith4_ | ~itsp |
| 01:39.11 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 01:41.59 | exothermic | angryuser: hmm I added exten => 4880,hint,SIP/peername and put call-limit=1 on the peer still allows more than one call down the pipe. |
| 01:44.59 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 01:45.51 | angryuser | exothermic have you read the wiki ? like 15 lines http://www.voip-info.org/wiki/view/Asterisk+presence |
| 01:46.07 | exothermic | angryuser: ya |
| 01:46.36 | angryuser | exothermic the sip peer is a 'friend' ? |
| 01:46.55 | exothermic | angryuser: yes |
| 01:47.09 | *** join/#asterisk pcrane (n=pcrane@120.89.80.110) |
| 01:47.15 | angryuser | exothermic have you added limitonpeers=yes ? |
| 01:47.25 | exothermic | to sip.conf general? yes |
| 01:47.26 | gramulhaozin | hey guys |
| 01:47.32 | gramulhaozin | anyone used the OPENVOX cards ? |
| 01:48.00 | angryuser | QUOTE "If you are using friend instead of peer, you will need limitonpeers = yes as well as a call-limit statement for each SIP device /QUOTE |
| 01:48.35 | exothermic | angryuser: Ya read that, which is why I implemented both. |
| 01:49.21 | exothermic | angryuser: hmm I guess that could be read to say that both statements need to be in each peer. |
| 01:49.33 | exothermic | angryuser: let me try that. |
| 01:49.45 | nvez | anyone which uses a specific IP phone of preference? |
| 01:50.21 | angryuser | ~phones |
| 01:50.22 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 01:50.41 | nvez | such as.. |
| 01:50.41 | nvez | :p |
| 01:50.54 | exothermic | angryuser: Ya didn't make any difference |
| 01:51.09 | angryuser | exothermic and core show hints ? |
| 01:51.33 | angryuser | exothermic 'under cli' have you got them ? |
| 01:51.46 | exothermic | angryuser: State:Idle Watchers 0 |
| 01:52.04 | gramulhaozin | anyone used the OPENVOX cards ? |
| 01:52.40 | angryuser | exothermic so try let's say cal with that peer, and monitor it's state under 'queue show' |
| 01:52.48 | angryuser | call* |
| 01:53.09 | exothermic | angryuser: Well I think I first need to get the hint to report something other than idle. |
| 01:53.38 | exothermic | angryuser: When the phone is in a call. |
| 01:54.09 | angryuser | exothermic try calling, and type queue show |
| 01:54.14 | angryuser | is it still idle ? |
| 01:54.56 | exothermic | angryuser: ya |
| 01:56.09 | baliktad | my VOIP provider uses the number I specify in the "fromuser" value of the sip.conf account as the Caller ID |
| 01:56.27 | baliktad | is there any way for me to set this number dynamically? |
| 01:56.35 | angryuser | exothermic that was it, you need to find the way to manioulate agent's devstate, and i dont really know how to do it |
| 01:56.51 | angryuser | ask fender if he is still alive |
| 01:57.40 | baliktad | I would like to change the caller ID based on the station originating the call and the number dialed |
| 01:58.57 | nvez | fax from email + asterisk = yay or nay? |
| 01:59.28 | angryuser | nvez hylafax + iaxmodem |
| 02:04.08 | angryuser | nvez consider avantfax it's a nice web interface to hylafax |
| 02:04.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:04.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:08.01 | *** join/#asterisk km2 (n=x@c-98-210-137-171.hsd1.ca.comcast.net) |
| 02:10.47 | angryuser | sleep |
| 02:13.29 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX -=- #asterisk- |
| 02:13.34 | russellb | hrm |
| 02:14.17 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- #asterisknow/#asterisk-gui for AsteriskNOW -=- #switchvox (switchvox.com) -=- #freepbx -=- #asterisk-commits for bugs/commits moni |
| 02:15.13 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- #asterisknow/#asterisk-gui (asterisknow.org) -=- #switchvox (switchvox.com) -=- #freepbx -=- #asterisk-commits for bugs/commits mo |
| 02:15.46 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- #asterisknow/#asterisk-gui -=- #switchvox (switchvox.com) -=- #freepbx -=- #asterisk-commits, bugs/commits monitoring |
| 02:16.54 | d-k-t | you know when there's too much info in a topic line when.... ;) |
| 02:17.48 | russellb | yeah ... |
| 02:17.50 | russellb | it's out of hand. |
| 02:18.03 | russellb | blames CentOS |
| 02:19.21 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- Related channels: #asterisknow, #asterisk-gui, #switchvox, #freepbx, #asterisk-commits, #asterisk-bugs, #asterisk-dev |
| 02:23.26 | Qwell | you could just set all the channels as an onjoin notice |
| 02:23.40 | Qwell | chanserv can send a notice to people when they join |
| 02:24.04 | Qwell | all the GUI ones, anyways |
| 02:25.41 | Strom_C | Qwell: i'm sure people will continue to ignore it just as much as they always do |
| 02:26.17 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 02:27.04 | nvez | i dont think this is related but.. |
| 02:27.25 | nvez | trixbox is an "asterisk-based ip-pbx product" .. what exactly is trixbox? |
| 02:28.42 | russellb | ~trixbox |
| 02:28.42 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 02:28.57 | Qwell | russellb: You took it out of the topic, and now look what happened. :P |
| 02:28.58 | nvez | oh i see. |
| 02:29.04 | nvez | thanks russellb :) |
| 02:29.10 | russellb | i don't recommend it ... |
| 02:29.18 | nvez | yeah, i dont want to have a server run it alone |
| 02:29.27 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 02:29.30 | nvez | i need to use my servers for other stuff, so i dont want that :p |
| 02:29.34 | nvez | but thanks for the info |
| 02:30.05 | gramulhaozin | nvez: server for Phone and other stuff sound very interesting |
| 02:31.03 | gramulhaozin | one day I put a 486DX2 - 66, to do voip coded to G729 and a fileserver, e-mail with spam/antivirus, firewall and also vpn |
| 02:31.24 | nvez | but i dont really want to redo this whole server just for this |
| 02:31.25 | nvez | =P |
| 02:31.37 | nvez | freepbx vs asterisk gui, what would you guys pick? |
| 02:32.38 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 02:33.52 | russellb | check them both out, i guess ... depends what you want ... most people here will tell you to use neither, and just config manually |
| 02:33.58 | russellb | i personally prefer the asterisk-gui |
| 02:34.08 | russellb | as it's easier to use it along side manual config editing |
| 02:34.27 | russellb | and literally has no install dependencies other than asterisk itself |
| 02:34.51 | *** join/#asterisk kmshanah (n=kmshanah@cubit.disenchant.net) |
| 02:35.01 | nvez | ahh |
| 02:35.11 | jaytee | I'm using the asterisk-gui on a test machine and it seems ok |
| 02:35.30 | nvez | i see, well, config manually is what i prefer (because nothing else relies on it and it doesnt mess up), but i havent dugg enough yet to know, maybe ill end up using asterisk gui :p |
| 02:35.46 | jaytee | I just wish there was more documentation on it |
| 02:38.38 | *** join/#asterisk PepOSX (n=angeldav@190.72.147.85) |
| 02:38.56 | russellb | jaytee: look for the AA50 manual online. If it's posted (i'm sure it is), it uses the same GUI ... |
| 02:39.33 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 02:39.35 | LemensTS | freepbx on debian is pain, they need a better writeup. I may send them one took me a few tries to get it working properly |
| 02:39.40 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 02:39.45 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 02:40.23 | gramulhaozin | hey russellb ever tried the openvox boards ? |
| 02:40.32 | gramulhaozin | people are selling those openvox |
| 02:40.42 | Qwell | ~cheap |
| 02:40.43 | jbot | i heard cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:40.47 | gramulhaozin | I'm not sure about the T1/E1 handling but I need only two FXO |
| 02:40.49 | Qwell | gramulhaozin: crap clones |
| 02:41.15 | gramulhaozin | do you think they work ? |
| 02:41.21 | Qwell | they are crap. |
| 02:41.30 | gramulhaozin | fxo is crap already |
| 02:41.31 | LemensTS | they work but depends how dependable machine you are building |
| 02:41.42 | gramulhaozin | what do you mean dependable ? |
| 02:42.36 | LemensTS | if its a home system go with openvox, if not sangoma is really good and i havent had problem with digium... |
| 02:42.39 | gramulhaozin | you mean how much I need to depend of it ? |
| 02:42.53 | gramulhaozin | how much for sangoma ? |
| 02:42.54 | russellb | +1 to Qwell's comments |
| 02:42.55 | LemensTS | some would say sangoma is better then digium |
| 02:43.13 | LemensTS | gramulhaozin: google it |
| 02:43.18 | russellb | support asterisk, buy digium :-D |
| 02:43.23 | gramulhaozin | :P |
| 02:43.37 | LemensTS | yea i buy digium myself |
| 02:43.49 | gramulhaozin | :P |
| 02:43.53 | gramulhaozin | checking the price |
| 02:44.44 | LemensTS | i would not use digium just to support asterisk...id use it because it was good :D |
| 02:45.02 | gramulhaozin | but the price $$$ |
| 02:45.05 | russellb | http://www.russellbryant.net/blog/files/Analog_comp_analysis_whitepaper.pdf |
| 02:45.15 | gramulhaozin | $200 for openvox |
| 02:45.19 | Qwell | ~cheap |
| 02:45.20 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:45.32 | Qwell | and that's pretty expensive for clone crap |
| 02:45.55 | Qwell | gramulhaozin: http://www.telephonydepot.com/product_p/105-050-tdm410p.htm |
| 02:46.06 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 02:46.24 | LemensTS | Heh being expensive on asterisk is CHEAP compared to proprietary systems |
| 02:46.36 | gramulhaozin | complicated |
| 02:46.42 | gramulhaozin | not all the time less expansive means BAD |
| 02:46.53 | Qwell | In this case, it does. |
| 02:47.27 | russellb | Qwell: are there clones of the voicebus cards, too? or just the older models |
| 02:47.33 | Qwell | dunno |
| 02:47.36 | Qwell | probably not |
| 02:47.45 | LemensTS | russell nice pdf |
| 02:47.49 | file | no known clones of the voicebus stuff |
| 02:48.04 | russellb | file: yay |
| 02:48.13 | russellb | LemensTS: yeah, I think Malcolm did a nice job. |
| 02:48.23 | gramulhaozin | hardware echo cancelation needed for 3 channels ? |
| 02:48.37 | gramulhaozin | from $200 for open vox and $276 for Digium I rather get Digium |
| 02:48.45 | russellb | gramulhaozin: not really, no ... |
| 02:49.11 | russellb | unless the CPU is going to be loaded doing other things |
| 02:49.16 | gramulhaozin | nops |
| 02:49.17 | russellb | you can use HPEC on there and it will be ok |
| 02:49.20 | gramulhaozin | it's a QUAD CORE 2.4Ghz |
| 02:50.10 | LemensTS | openSER handles sip calls as just sip signaling. Is this how asterisk works, or does the rtp traffic go into the aterisk server and out to the other client? |
| 02:50.28 | gramulhaozin | only problem is my G723 codec |
| 02:50.30 | russellb | LemensTS: OpenSER is a SIP proxy. Asterisk acts as a B2BUA. ... |
| 02:50.37 | russellb | this means that the audio _may_ be going through Asterisk |
| 02:50.49 | russellb | in fact, that's how Asterisk sets up calls. Then, it will redirect the media to flow directly if possible |
| 02:50.50 | gramulhaozin | but I think that Quad Core 2.4Ghz would handle 20 G723 calls , what do you think Russel ? |
| 02:51.05 | russellb | (if no transcoding is being done, no features in asterisk enabled that require access to the media, etc) |
| 02:51.12 | russellb | gramulhaozin: should be fine |
| 02:51.17 | LemensTS | russel: so once the call is connected, the client traffic goes from the client to client and not thru asterisk? |
| 02:51.23 | Qwell | G723? |
| 02:51.28 | gramulhaozin | Qwell: yes |
| 02:51.33 | Qwell | gramulhaozin: You'll need hardware to transcode that with Asterisk. |
| 02:51.43 | gramulhaozin | gramulhaozin: software on the CPU |
| 02:51.47 | russellb | LemensTS: correct, assuming Asterisk doesn't need access to the media for some reason |
| 02:51.48 | Qwell | Illegal. |
| 02:52.01 | LemensTS | russel: for transcoding right |
| 02:52.06 | tzanger | Qwell: sick bird? |
| 02:52.13 | russellb | transcoding, call recording, a number of things |
| 02:52.16 | Qwell | tzanger: potato |
| 02:52.45 | russellb | Qwell: yeah, forgot about that. was thinking 726, i guess. |
| 02:52.53 | russellb | gramulhaozin: Qwell is correct about 723 |
| 02:52.53 | gramulhaozin | Qwell: do they sell G723 hardware cards? |
| 02:52.56 | LemensTS | russel: ok cool, i was wondering why openSER could handle so many call setups. |
| 02:52.56 | *** part/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 02:52.59 | russellb | gramulhaozin: digium does, yes |
| 02:53.00 | Qwell | gramulhaozin: Digium does, yes |
| 02:53.05 | Qwell | russellb: get out of my head |
| 02:53.10 | russellb | LemensTS: yep, it's a very different animal |
| 02:53.27 | russellb | LemensTS: Asterisk + OpenSER is a very common setup for larger systems |
| 02:53.32 | gramulhaozin | let me check the price |
| 02:53.43 | file | very common and powerful... |
| 02:53.43 | Qwell | waits for it |
| 02:54.11 | russellb | Qwell: heh .. |
| 02:54.24 | russellb | gramulhaozin: do you have a reason that you're stuck with 723? |
| 02:54.30 | gramulhaozin | russellb: not really |
| 02:54.41 | russellb | ok, then I would just avoid it |
| 02:54.48 | gramulhaozin | actually I need G729 for the Cisco 7940's phones |
| 02:54.57 | russellb | unless you're attached to it for some reason, and want to buy the card, heh |
| 02:55.29 | russellb | you can buy licenses for G.729 from Digium for doing it in software |
| 02:55.32 | russellb | (or you can buy the card) |
| 02:55.37 | russellb | but for just 20, i'd use the software one |
| 02:55.39 | russellb | $10 / channel |
| 02:55.45 | LemensTS | so the tc400b does not require g729 license to transcode from what i got out of reading, that right? |
| 02:55.57 | Qwell | LemensTS: It itself is licensed. |
| 02:56.04 | russellb | LemensTS: that is correct. the licensing is included with the purchase of the card |
| 02:56.11 | tzanger | I'm licensed too |
| 02:56.17 | tzanger | K6168somethingsomething760505 |
| 02:57.16 | russellb | also, you should include some money to send my way in your budget |
| 02:57.21 | russellb | just because i'm a nice guy |
| 02:57.26 | file | lies |
| 02:57.28 | Qwell | *cough* |
| 02:57.50 | LemensTS | ok whats your bank account information so we can send it to you |
| 02:58.00 | russellb | heh |
| 02:58.06 | russellb | paypal russell@russellbryant.net |
| 02:58.09 | russellb | there ya go |
| 02:58.14 | gramulhaozin | hey gus |
| 02:58.21 | [TK]D-Fender | LemensTS: Its with Freddie Mac, is that K? |
| 02:58.43 | gramulhaozin | where is the CODEC cards from Digium ? |
| 02:58.47 | gramulhaozin | not on the digium store |
| 02:58.53 | russellb | gramulhaozin: it's the TC400P |
| 02:58.55 | [TK]D-Fender | gramulhaozin: Slightly to the left |
| 02:59.16 | Qwell | http://store.digium.com/productview.php?product_code=1TC400BLF-01 |
| 02:59.17 | russellb | http://store.digium.com/productview.php?product_code=1TC400BLF-01 |
| 02:59.20 | russellb | Qwell: aww |
| 02:59.33 | Qwell | russellb: I'm closer to the colo |
| 02:59.36 | LemensTS | russelb: check your paypal account you have money |
| 02:59.38 | russellb | heh |
| 02:59.42 | russellb | O.O |
| 02:59.46 | LemensTS | tkd: whats freddie mac |
| 03:00.12 | [TK]D-Fender | LemensTS: http://digg.com/2008_us_elections/Bob_Barr_Freddie_Mac_Fannie_Mae_bailout_added_to_U_S_debt |
| 03:00.16 | nvez | rofl |
| 03:00.21 | nvez | russellb: do you really :--o |
| 03:00.30 | russellb | looks |
| 03:00.33 | Qwell | FDMC FTL |
| 03:00.48 | russellb | lols |
| 03:00.51 | russellb | LemensTS: thanks! :-D |
| 03:00.58 | nvez | oooh |
| 03:01.06 | nvez | LemensTS support open source wooyay. |
| 03:01.08 | *** join/#asterisk TedNJ38 (n=HungLad@ool-435336f7.dyn.optonline.net) |
| 03:01.24 | LemensTS | lol |
| 03:02.08 | TedNJ38 | Is anyone around here able to work with the phone Cisco 7961g and Asterisk? |
| 03:02.58 | Qwell | TedNJ38: asking the same question in multiple channels is considered very rude |
| 03:03.51 | nvez | haha, when i do it, i atleast rephrase it and a couple of minutes after i got no answer on the first |
| 03:03.51 | nvez | =P |
| 03:03.51 | LemensTS | that mac and mae is too boring for me to read heh |
| 03:03.51 | [TK]D-Fender | TedNJ38: You mean Your FreePBX can't AutoMagicallyLikeMcConfiguragate them for you? |
| 03:03.51 | TedNJ38 | When someone does not respond, it is also considered rude. But nobody seems to give a damn. |
| 03:03.51 | *** part/#asterisk TedNJ38 (n=HungLad@ool-435336f7.dyn.optonline.net) |
| 03:03.59 | nvez | LOL |
| 03:04.04 | nvez | come back when you pay for it |
| 03:04.05 | Qwell | jackass waited THIRTY SECONDS |
| 03:04.06 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 03:04.17 | jaytee | McConfigurate? LOL!!!! |
| 03:04.18 | nvez | unbelivable these people. |
| 03:04.24 | LemensTS | This is a paid subscription forum |
| 03:04.35 | [TK]D-Fender | Qwell:/me calls up Guiness to report TedNJ38's new record |
| 03:04.37 | nvez | doesnt pay a cent and expects better support than a $6k project |
| 03:04.45 | Qwell | [TK]D-Fender: it's worse than that - it was 3 channels |
| 03:04.55 | [TK]D-Fender | Qwell: He's on my shit -list :) |
| 03:05.13 | [TK]D-Fender | Qwell: I love feeding trolls.... hey... evey play Pandemic? ;) |
| 03:05.20 | Qwell | yes |
| 03:05.22 | jaytee | omg!!! |
| 03:05.22 | Qwell | it pissed me off |
| 03:05.35 | jaytee | addictive |
| 03:05.55 | [TK]D-Fender | Qwell: I just started with Madagascar and went right for Necrosis :D Another world-kill FTW |
| 03:06.15 | Qwell | the buttons do nothing |
| 03:06.27 | jaytee | can you actually pick what country? I thought it was random |
| 03:06.45 | [TK]D-Fender | jaytee: No, I lucked out. Then again I named mine "Madagascar SUX" |
| 03:06.49 | LemensTS | When is 1.6 expected to be out of beta? |
| 03:06.55 | [TK]D-Fender | jaytee: Figure it was karma :) |
| 03:06.55 | Qwell | LemensTS: when it's ready |
| 03:06.58 | [TK]D-Fender | LemensTS: WHEN IT'S DONE |
| 03:07.18 | jaytee | cuz I never end up getting Madagascar and it's been the only holdout the last 3 games :-( |
| 03:07.20 | x86 | http://www.wikiupload.com/images/dollars.php |
| 03:07.21 | x86 | HAHAHAHAHAHAHAHAHAHHAAHHAHAAHAHHAAHAHAHAHAinfinite |
| 03:07.21 | [TK]D-Fender | LemensTS: Odds are... AT NIGHT |
| 03:07.32 | LemensTS | LemensTS: your an idiot |
| 03:07.37 | Qwell | [TK]D-Fender: I'm gonna go for 11:42am |
| 03:07.41 | Qwell | on a Tuesday |
| 03:07.45 | Qwell | russellb: ^^^ |
| 03:07.54 | russellb | 1.6.0-rc1 will come out when DAHDI 2.0 gets released, which I expect in the next couple of weeks |
| 03:07.57 | *** join/#asterisk BBHoss (n=hoss@user-24-214-218-77.knology.net) |
| 03:08.07 | Qwell | x86: nice |
| 03:08.38 | LemensTS | lol i forgot about dahdi, i remember hearing about that in here when it was announced. Isnt that the name of zaptel or zapta in 1.6? |
| 03:08.40 | [TK]D-Fender | Qwell: Silly American.... think outside your limited longitudinal scope! |
| 03:08.50 | [TK]D-Fender | adjusts for UTC |
| 03:08.51 | russellb | LemensTS: yes |
| 03:08.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:10.48 | LemensTS | http://www.youtube.com/watch?v=kBqsZKE0wuk scars on broadway...system of a down co-singer its cool |
| 03:14.36 | *** join/#asterisk Fiapo-CE (i=Fiapo-CE@201.70.137.40) |
| 03:18.12 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 03:19.17 | *** join/#asterisk mmurdock (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 03:23.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:24.44 | jaytee | time for some zzzzz's. nite all |
| 03:24.49 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:33.34 | *** join/#asterisk centralb (n=hkid2@www.unleast.com) |
| 03:33.53 | rabelais | what would cause echo on a pure digital sip channel? a call going from my local voip client (E51) to my asterisk server through the net and hitting a linksys pap2t at a remote phone has some pretty noticeable echo on the E51's side (E51 hears himself) |
| 03:35.35 | MikeJ | echo on the device hooked up to your ata |
| 03:35.57 | LemensTS | Neither of you are on speaker phone either are u |
| 03:36.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:41.39 | *** join/#asterisk centralb (n=hkid2@www.unleast.com) |
| 03:43.09 | x86 | Qwell: http://s271.photobucket.com/albums/jj135/Talamasca124/reMotivationals/?action=view¤t=withfries664ql4.jpg |
| 03:44.49 | centralb | where's most suitable to discuss the recent 1.6 beta quirks? :) |
| 03:46.33 | LemensTS | x86: lmao i wish i had that in my house |
| 04:05.08 | x86 | centralb: asterisk-dev maybe |
| 04:12.30 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 04:13.16 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 04:13.26 | centralb | thanks |
| 04:18.02 | *** join/#asterisk pcrane (n=pcrane@121.90.63.160) |
| 04:18.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:19.20 | gramulhaozin | hey guys |
| 04:19.37 | gramulhaozin | with a resale certificate can I call digium and ask to resell their cards ? |
| 04:20.22 | gramulhaozin | ops |
| 04:20.42 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:25.51 | *** part/#asterisk centralb (n=hkid2@www.unleast.com) |
| 04:34.32 | exothermic | Hi I know that agentcallback() is deprecated, but I'm using 1.4, and I can't seem to get the agent mapped to the queue once they are logged in. I have them listed as a member in the queue context I'm not sure what else to do. |
| 04:57.17 | exothermic | does it matter what context you put hints in? |
| 04:57.31 | exothermic | can they be in any context in the dial plan? |
| 05:08.57 | cplx | hi guys.. anyone here pretty cluey with codecs etc? |
| 05:28.36 | kmshanah | has anyone seen this error before: "Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)" |
| 05:29.39 | kmshanah | this is coming up on our ISDN PRI channels |
| 05:29.56 | kmshanah | restarting asterisk makes it go away, but that's pretty disruptive |
| 05:41.31 | eharris | Under asterisk 1.4.17, is there any built-in method of creating a "phantom" extension that doesn't actually go anywhere, it just records the line? |
| 05:43.55 | cplx | hi guys.. anyone here pretty cluey with codecs etc? |
| 05:51.06 | cplx | anyone know where to get the full blown cisco call manager music? |
| 05:53.04 | phpboy | kmshanah: I've been having a VERY similar problem |
| 05:53.10 | phpboy | not yet found the solutions :( |
| 05:53.21 | phpboy | my channels get clogged up as a result of this :( |
| 05:57.52 | kmshanah | we have an analog card in the same system (AEX2400) and those channels are unaffected |
| 05:58.17 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 05:59.33 | phpboy | Funny enough, I have two TDM 2400's and they seem to be giving trouble to the LCR system |
| 06:02.13 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 06:02.29 | phpboy | anybody ever done fax from ZAP server to IAX2 server? |
| 06:04.21 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 06:13.02 | *** join/#asterisk TheIzkabola (n=TheIzkab@c-67-171-143-153.hsd1.or.comcast.net) |
| 06:14.46 | TheIzkabola | hello, I am running ubuntu and I just ran: apt-get install asterisk I am not sure what to do next, I can't access the web panel on localhost. |
| 06:14.52 | TheIzkabola | any assistance is much appreciated! |
| 06:15.25 | ManxPower | We mostly deal with installing Asterisk from source here. |
| 06:15.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:16.09 | TheIzkabola | is there a way I can uninstall it and do that? I don't even know what version it installed :( **is there a way I can see?** |
| 06:17.15 | ManxPower | It's 2:16am my time, I'm going to bed. Maybe you'll catch me tomorrow |
| 06:17.33 | TheIzkabola | ah ok, well thanks |
| 06:19.42 | linuxmaniac | TheIzkabola: there is no webpanel on Ubuntu asterisk version |
| 06:20.11 | TheIzkabola | does you know how I can uninstall asterisk? I'd love to reinstall it using source |
| 06:20.16 | TheIzkabola | do* |
| 06:20.28 | linuxmaniac | apt-get remove --purge asterisk |
| 06:20.39 | TheIzkabola | thank you very much! |
| 06:20.59 | TheIzkabola | do you happen to have a good tutorial site for installing? sorry, i'm really new to linux |
| 06:21.07 | linuxmaniac | but source or deb doesn't help you to know how to configure it |
| 06:21.23 | linuxmaniac | you must read a lot before installing |
| 06:21.34 | linuxmaniac | search asterisk book |
| 06:22.41 | TheIzkabola | thanks, do you recommend using FreePBX? |
| 06:22.52 | TheIzkabola | (i've downloaded the book) |
| 06:23.26 | gnorbert | Hi all, I have a problem with call files. When I try to call a meetme conference of Asterisk, it calls it twice and plays welcome sounds strange. Here is the Asterisk message with enabled debug and the call file: http://pastebin.com/d15ca2177 |
| 06:23.56 | gnorbert | Can somebody help in this? |
| 06:24.16 | linuxmaniac | and I recommend you asterisk. But with a previous knowledge |
| 06:26.41 | TheIzkabola | great, thanks for your help |
| 06:32.35 | tzafrir | gnorbert, how do you call that? |
| 06:32.39 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:33.20 | tzafrir | gnorbert, dialplan show 111111@default |
| 06:33.47 | tzafrir | Though I have a feeling you originate a call to there twice |
| 06:33.52 | gnorbert | tzafrir: I cut+paste the call file in outgoing directory. |
| 06:34.54 | tzafrir | Try creating the file elsewhere and then moving it into the outgoing directory |
| 06:34.57 | gnorbert | It calls 11111@default. If I lose either Channel: Local/11111@default, or context and extension line, it doesnát make the call. |
| 06:35.03 | tzafrir | (only after it's done) |
| 06:35.09 | gnorbert | *doesn't |
| 06:35.33 | gnorbert | I moved it, didn't copy. |
| 06:36.14 | phpboy | [Jul 28 08:41:47] NOTICE[54526]: channel.c:2270 __ast_read: Dropping incompatible voice frame on IAX2/192.168.40.1:4569-1744 of format slin since our native format has changed to g726 <------ what would this msg generally mean? |
| 06:36.39 | tzafrir | hmm, you put Local/11111@default as the channel, and then originate a call to (context+extension)11111@default? |
| 06:37.19 | gnorbert | Maybe that's the problem, but without either one, it didn't work. |
| 06:37.33 | tzafrir | Sure that's the problem |
| 06:37.43 | tzafrir | What did you want to do? |
| 06:37.57 | tzafrir | you wanted to get something into a meetme room? |
| 06:38.05 | tzafrir | What is that "something"? |
| 06:38.18 | gnorbert | Call a meetme conference on the same server to play a sound file. |
| 06:39.19 | tzafrir | Use as Channel: Meetme/<room-name> |
| 06:39.35 | gnorbert | I wanted to play a wav file for all of the conference members. |
| 06:39.54 | tzafrir | and use: Application: Playback: demo-intro |
| 06:40.19 | gnorbert | I have an extension for that in *. |
| 06:40.26 | tzafrir | err, likme the above, but with correct syntax |
| 06:40.30 | gnorbert | So if it calls it, that plazs the sound and hangs up. |
| 06:40.58 | gnorbert | *plays, sorry. |
| 06:43.56 | tzafrir | right |
| 06:44.17 | gnorbert | And if I kick one of the two calls, it plays the kicked sound well and kicks "both" calls |
| 06:44.58 | gnorbert | If I get out the context line, it makes the same. |
| 06:49.24 | phpboy | [Jul 28 08:41:47] NOTICE[54526]: channel.c:2270 __ast_read: Dropping incompatible voice frame on IAX2/192.168.40.1:4569-1744 of format slin since our native format has changed to g726 <------ what would this msg generally mean? |
| 06:50.21 | *** join/#asterisk [netman] (n=netman@12.Red-88-25-139.staticIP.rima-tde.net) |
| 06:53.29 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
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| 07:11.34 | gnorbert | The other question is how can I call the last 3 digits of extension (For example: call 345.wav if extension is 12345) |
| 07:12.36 | *** join/#asterisk justanotherpaul (n=pwalker@d75-157-235-70.bchsia.telus.net) |
| 07:13.19 | justanotherpaul | Support for Speex 16 kHz in Asterisk. I'm having trouble finding information about this. |
| 07:14.03 | justanotherpaul | Could anyone help me? |
| 07:14.15 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 07:15.29 | gnorbert | justanotherpaul: I don't know it so much, as far as I know, you should edit codecs.conf and sip.conf/aix.conf. |
| 07:16.01 | gnorbert | In sip.conf you should write disallow=all an after that allow=speex. |
| 07:16.49 | gnorbert | I hope I could help, I know it's not so much, sorry for that, I'm not an expert. |
| 07:16.54 | justanotherpaul | gnorbert: Thanks. I am able to use that to get narrowband speex working. |
| 07:17.31 | justanotherpaul | Wideband speex remains a problem. |
| 07:17.45 | gnorbert | justanotherpaul: I think that should work also for wideband just you should set a variable. |
| 07:17.56 | cplx | anyone know where to get the full blown cisco call manager music? |
| 07:18.02 | justanotherpaul | Hrm. |
| 07:18.08 | justanotherpaul | I'll do some more searching. |
| 07:18.20 | phpboy | iax2_read: I should never be called! <--- why would I get such an MSG? |
| 07:18.56 | *** join/#asterisk admin0 (n=admin@bb116-14-251-74.singnet.com.sg) |
| 07:19.53 | gnorbert | justanotherpaul: The problem can be that you don't have enough space to send it so you should set that somehow. It's only one line, but I don't know, sorry. |
| 07:21.11 | justanotherpaul | gnorbert: what makes you think it would be that? |
| 07:22.19 | gnorbert | justanotherpaul: I had a problem like that. :) Of course not sure, that's the fault. That's just my tip. |
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| 07:29.40 | *** join/#asterisk mrnick (n=basement@kulnet-nat-2.kulnet.kuleuven.be) |
| 07:30.34 | cplx | hi guys.. anyone here pretty cluey with codecs etc? |
| 07:33.01 | gnorbert | Does somebody know, what can be the problem if a call file calls twice instead of one? The call file and the * messages are at http://pastebin.com/d15ca2177 |
| 07:33.10 | Strom_M | cplx: what's your question? |
| 07:35.40 | cplx | Strom_M - im using codec preference 1 g729r8 bytes 20 |
| 07:35.54 | cplx | Strom_M - quality isn't that good, what should I try next? bump up the bytes? |
| 07:36.08 | Strom_M | um...use a codec that doesn't suck? :) |
| 07:36.17 | Strom_M | g729 is going to sound horrid no matter how you slice it |
| 07:36.35 | Strom_M | g711 will give you the best quality but use the most bandwidth |
| 07:36.47 | cplx | Strom_M - ahh ok, well my voip provider supports G726, G723, but when i try it it doesn't seem to negotiate |
| 07:37.01 | cplx | voip provider = asterisk, my end = Cisco callmanager |
| 07:37.08 | Strom_M | they dont support g711? |
| 07:37.14 | cplx | doesn't look like it |
| 07:37.20 | Strom_M | that sounds odd |
| 07:37.25 | Strom_M | everyone supports g711 |
| 07:37.29 | cplx | sec |
| 07:37.31 | Strom_M | who's your provider? |
| 07:37.34 | cplx | faktortel |
| 07:37.38 | cplx | australian |
| 07:37.40 | Strom_M | ok |
| 07:38.38 | cplx | hmm |
| 07:38.42 | cplx | ill tell u what it supports |
| 07:38.55 | *** join/#asterisk MaliutaLap (n=nikolai@124-171-128-169.dyn.iinet.net.au) |
| 07:39.03 | Strom_M | u is a letter; i don't think it knows anything about codecs |
| 07:39.50 | cplx | Codec; G729, ILBC, GSM, G726, ULAW, G723, ALAW |
| 07:39.55 | cplx | use ULAW? |
| 07:40.01 | cplx | or will that flood my connection/bandwidth |
| 07:40.10 | cplx | thats why im here.. to ask these stupid q's p |
| 07:40.23 | Strom_M | ulaw and alaw are both variants of G711 |
| 07:40.33 | Strom_M | ulaw is for north america; alaw is for everywhere else |
| 07:40.37 | cplx | ok so if i use alawy |
| 07:40.41 | cplx | alaw* |
| 07:40.47 | Strom_M | alaw will take 64kbps plus overhead |
| 07:40.51 | cplx | not sure what I would set my call manager to use |
| 07:40.54 | cplx | ie. |
| 07:41.01 | cplx | codec preference 1 g729r8 bytes 20 |
| 07:41.14 | Strom_M | i have no idea; i haven't worked with call manager |
| 07:41.14 | cplx | codec preference 1 G711a |
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| 08:06.43 | cplx | codec preference 1 G711alaw ? |
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| 08:09.08 | gnorbert | Does somebody have an idea, what can be the problem, if the call file calls twice the asterisk? I have the asterisk messages with sip debug and the call file at http://pastebin.com/d15ca2177 |
| 08:09.44 | marc7 | there are a couple of things, as far as I'm aware, you currently can't distribute across multiple different asterisk servers working in a clustered environment.... those are conferences, and call queues... are there any other big challenges I'm missing? |
| 08:12.27 | tzafrir | gnorbert, I already told you why: because both legs of the call you generated are the same |
| 08:15.58 | gnorbert | tzafrir: Sorry, then I think I didn't fully understand that you said... I thought you want to play the sound from the call file. |
| 08:17.20 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 08:18.03 | tzafrir | You originate two legs ("sides") of the call |
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| 08:40.20 | kmshanah | hi guys, I need some help debugging an isdn problem where asterisk thinks all the lines are busy/congested: http://pastebin.com/m7b322856 |
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| 08:42.34 | Strom_M | kmshanah: have you tried contacting digium support? |
| 08:44.43 | kmshanah | Strom_M: no, not as yet |
| 08:45.16 | Strom_M | kmshanah: also, there's something fishy; your global constant is set to "Zap/g1" but you're dialing "ZAP/G1" -- are you sure you're running the same code you're asking for help with? |
| 08:46.03 | Strom_M | kmshanah: I would do a quick sanity check and then call digium support |
| 08:46.24 | kmshanah | oh, sorry. I made that little change afterwards to see if it would make a difference. The diaplan had Zap/G1 while I was logging. |
| 08:47.29 | Strom_M | wonders what other "little changes" you're making... |
| 08:48.26 | kmshanah | No others, honest ;) |
| 08:48.47 | kmshanah | yeah, I'll get in touch with digium. |
| 08:48.51 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
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| 09:02.22 | marc7 | is Asterisk well-suited for threading tasks across multiple processors / cores? |
| 09:03.28 | marc7 | eg: which is better, a single processor with a high clock speed, or a multi-processor, multi-core system with decent clock speeds on each core |
| 09:03.50 | marc7 | i understand from TFOT that a good FPU on the chip is just as important as clock speed |
| 09:08.14 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:08.14 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 09:08.19 | SwK | FPU is important for media interaction things such as transcoding... (and media playback for things like hold music or ivr prompts if you dont have them in the native format for the call (eg: g729 for a g729 call and say G711U for a G711U call) |
| 09:09.33 | SwK | As far as multi-core vs single core imho you are better to have faster then more cores due the to the way many data elemements are tracked and require locking in asterisk |
| 09:09.33 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 09:09.54 | SwK | now that being said, more cores doesnt hurt |
| 09:10.51 | l0verb0y | hey there |
| 09:11.03 | l0verb0y | does anyone have any suggestions on any linux clustering software that would work well with asterisk? |
| 09:11.40 | SwK | you might want to be a little more specific with that... as clustering can mean a pile of things |
| 09:14.42 | *** join/#asterisk oej (n=olle@80.251.192.3) |
| 09:15.00 | l0verb0y | yikes |
| 09:15.07 | l0verb0y | I have 10 servers and I need to run commands on all of them |
| 09:15.24 | jm|home | :-S |
| 09:20.33 | l0verb0y | thats for sure |
| 09:22.46 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 09:24.38 | SwK | l0verb0y, when you say you need to run commands on all of them... are you trying to do something like do something like asterisk -rx 'restart now' on all of them at the same time? |
| 09:25.48 | *** join/#asterisk gilli (n=alec@p57A1E5EE.dip.t-dialin.net) |
| 09:26.24 | gnorbert | tzafrir: If I use Channel: Meetme/name of conference, it doesn't even work. |
| 09:27.13 | tzafrir | gnorbert, what do you use for context / extension (or alternatively: application)? |
| 09:27.15 | l0verb0y | SwK: yeah and a few other simple tasks |
| 09:27.54 | gnorbert | tzafrir: I use _11XXX at extensions.conf and call 11111. |
| 09:28.15 | gnorbert | Call file is now the same again to http://pastebin.com/d15ca2177. |
| 09:32.01 | gnorbert | tzafrir: My extensions.conf is at http://pastebin.com/d4b9e1945 maybe it helps, but I think that's not the problem's source. |
| 09:34.13 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-eebdb9e05092049d) |
| 09:34.58 | *** part/#asterisk admin0 (n=admin@bb116-14-251-74.singnet.com.sg) |
| 09:35.32 | tzafrir_laptop | gnorbert, again, what do you expect that call file to do? |
| 09:36.17 | tzafrir_laptop | because what it does in practice is to put both legs of the call you generate into meetme |
| 09:36.43 | gnorbert | To call the extension 11111. That is a meetme conference that plays a sound. |
| 09:37.00 | gnorbert | Hopefully, but I haven't done anything with that. |
| 09:37.03 | gnorbert | Yet. |
| 09:37.30 | tzafrir_laptop | what should call that extension? Is there some device that should call that extension? |
| 09:37.31 | marc7 | SwK: a belated thank you for the background on what processing conditions are more likely to be ideal for asterisk |
| 09:38.12 | tzafrir_laptop | gnorbert, that extensions is not "a meetme conference that plays a sound" |
| 09:38.15 | gnorbert | No, just * should call itself (Local channel). |
| 09:38.43 | marc7 | SwK: I'm designing a fairly large asterisk cluster, multiple media servers handling G.729 to G.711u translation, and different servers entirely handling the applications (eg: voicemail, conference bridges, dialplan logic, etc) |
| 09:38.49 | tzafrir_laptop | gnorbert, that extension puts the channel into meetme. And after its done (if it has not hung up) it will get the beep |
| 09:40.11 | gnorbert | tzafrir_laptop: Hmm, thanks. Really... You must know something. :) And how can I play a sound then? Is that application: Playback: beep the solution? |
| 09:40.39 | tzafrir_laptop | Application: Aplayback |
| 09:40.45 | tzafrir_laptop | Data: beep |
| 09:40.49 | tzafrir_laptop | right? |
| 09:40.56 | tzafrir_laptop | tries to recall correctly |
| 09:41.23 | gnorbert | Thanks, I try it now. |
| 09:41.43 | gnorbert | But then what shall I do with that extension thing? |
| 09:42.08 | gnorbert | So Channel looks like that must be Local/11111@default. |
| 09:44.09 | tzafrir_laptop | You can use that, right |
| 09:44.34 | gnorbert | But then what shall be extension? |
| 09:44.37 | gnorbert | The same? |
| 09:44.46 | gnorbert | Just because then it connects twice. |
| 09:45.05 | tzafrir_laptop | if you use Application: you should not use Context: and Extensions: |
| 09:45.35 | gnorbert | tzafrir_laptop: Well, ok, thanks very much. Then now I try it. :) |
| 09:47.59 | gilli | hello all. I have quite a few novice questions regarding asterisk. Is this the right place for me? |
| 09:49.30 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 09:52.20 | gilli | I managed to install * for a doubleE1 (T1) card. I googled and overlooked the asterisk book but I'm not sure if I could find the answer. |
| 09:52.31 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 09:54.04 | gilli | What I want to try is to do a simple call from an NT port to a TE port, send some text and watch it arriving. Is there any way I could do this? |
| 09:54.35 | tzafrir_laptop | gilli, which card? E1? The name doubleE1 suggests a Junghanns (or maybe Bero?) E1 card. And that one cannot eb a T1, IIRC |
| 09:54.36 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 09:55.15 | tzafrir_laptop | Getting the Asterisk book (at least the PDF) is a good idea anyway :-) |
| 09:56.18 | gilli | tzafrir_laptop, yes you're right. It's a Junghanns doubleE1. Yeah, I got the pdf. Sorry if my question is too trivial. |
| 09:56.28 | tzafrir_laptop | to do that, you should have an ISDN loopback cable (I think E1 and T1 here have the same wiring, so look for T1 loopback cable) |
| 09:57.22 | tzafrir_laptop | gilli, right now, the card's spans (/proc/zaptel/1 and 2) show in the first line that the span is deactivated (IIRC) |
| 09:57.49 | gilli | I did that. One port of the card is jumpered to TE and botch ports are connected with a cable. |
| 09:57.51 | tzafrir_laptop | a good sign that both ends are basically talking would be when you see there "activated" |
| 09:58.06 | tzafrir_laptop | (I don't recall how this shows with the LEDs on the card) |
| 09:58.28 | gilli | I assume they are activated, because the LEDs turn green when I connect the cable (and load the driver) |
| 09:59.15 | tzafrir_laptop | First time I see here someone with that card (whose driver name stands for 'card without an interesting name') |
| 10:00.10 | gilli | exactly that. cwain :) |
| 10:07.19 | tzafrir_laptop | do you have it defined in asterisk? do you see anything in: |
| 10:07.29 | tzafrir_laptop | asterisk -rx 'zap show channels' |
| 10:13.47 | gilli | sorry for the delay...checking. |
| 10:14.24 | *** join/#asterisk bl4q (i=Bl@dslb-088-066-226-133.pools.arcor-ip.net) |
| 10:14.24 | gilli | ah.. yeah. zap show channels..yeah, that always lists both cards. |
| 10:15.01 | gilli | err..wrong.. |
| 10:15.08 | gilli | i confused with zap show status |
| 10:15.42 | gilli | so..actually 'zap show channels' only lists 'pseud, default, default' . |
| 10:17.33 | gilli | Is that a bad sign? |
| 10:17.43 | tzafrir_laptop | so you don't have any zaptel channels defiend |
| 10:18.09 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 10:18.21 | gilli | ah..I think I could have messed with a .conf file.. |
| 10:18.47 | tzafrir_laptop | you can use the sample zapata.conf |
| 10:18.55 | Coder` | is away (^C0,11b^C0,4y^C0,7e ^C0,10A^C0,6L^C0,2L) |
| 10:19.06 | tompaw | Morning. |
| 10:19.25 | tompaw | With phpagi - is it possible to control REMOTE asterisk, or does it only work for the local one? |
| 10:19.53 | tzafrir_laptop | agi runs as a local process |
| 10:20.33 | tzafrir_laptop | maybe you're looking at using the manager interface? |
| 10:20.48 | tzafrir_laptop | tompaw, what do you want to do, exactly? |
| 10:21.45 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 10:22.34 | gilli | the content of my zapata.conf looks like that http://pastebin.com/d2ffef37b and I remember that ztcfg -vvv actually listed the channels(?) |
| 10:23.34 | *** join/#asterisk oej (n=olle@80.251.192.3) |
| 10:24.40 | tompaw | tzafrir_laptop: I want to remotely execute a call with DTMF |
| 10:24.53 | tompaw | tzafrir_laptop: I'd like to execute it from PHP ;-) |
| 10:25.24 | tzafrir_laptop | hands tompaw a phone wire |
| 10:25.30 | tompaw | tzafrir_laptop: it's for automated top-ups. my framwework is in php, and obviously there are no php-sip libraries. |
| 10:26.37 | tompaw | offtopic question: anybody from Croatia here? |
| 10:29.24 | tzafrir_laptop | tompaw, basically look into originate to execute a call. |
| 10:29.51 | tzafrir_laptop | as for DTMFs: just a few at startup, or over the course of the call? |
| 10:32.12 | tompaw | over the course |
| 10:32.16 | tompaw | but |
| 10:32.30 | tompaw | I can bring it down to a long phone number with 'p's and 'P's |
| 10:32.43 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 10:32.54 | tompaw | like 555pppp2ppp1ppp1234123412341234ppp |
| 10:32.55 | tompaw | ;-) |
| 10:38.00 | gilli | tzafrir_laptop, could it be that manipulating my /etc/asterisk/extensions.conf causes an issue? I changed that file while I was trying to follow the asterisk book. |
| 10:39.41 | tzafrir_laptop | gilli, first off, make it so that you have zap channels. |
| 10:39.49 | tzafrir_laptop | That is unrelated to extensions.conf |
| 10:40.07 | tompaw | tzafrir_laptop: so is it possible to make remote call with DTMFs using ASterisk Manager? |
| 10:40.16 | tompaw | (I think it is) |
| 10:40.36 | tzafrir_laptop | tompaw, yes. Using the Originate manager command |
| 10:40.45 | tompaw | tzafrir_laptop: ok, thanks! |
| 10:41.17 | *** join/#asterisk oilinki (n=oil@ppp-124-120-19-19.revip2.asianet.co.th) |
| 10:41.25 | tzafrir_laptop | gilli, pastebin your /etc/zapata.conf and the output of: cat /proc/zaptel/* |
| 10:41.52 | tompaw | http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) < that's cool! |
| 10:42.43 | gilli | my zapata.conf: http://pastebin.com/d2ffef37b |
| 10:45.31 | gilli | and the /proc/zaptel/* http://pastebin.com/d5a5dd308 |
| 10:45.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 11:02.54 | *** join/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 11:03.28 | gilli | huh? now I just stopped asterisk once, did ztcfg -s, then ztcfg and now I can't even access the zap commands anymore.. |
| 11:05.03 | tompaw | tzafrir_laptop: will asterisk recognize those 'p' chars in the phone number? |
| 11:07.26 | gilli | ..and I can't seem to laod chan_zap.so manually although it's there in ./usr/lib/asterisk/modules ... |
| 11:08.12 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 11:11.06 | gilli | hmm..rebooting... |
| 11:13.57 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:14.18 | *** join/#asterisk knight123 (n=king2676@202.21.177.3) |
| 11:16.18 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:16.34 | knight123 | hello guys, i have a question, I have PBX with 4port FXO and i want to have a hotline, so from 4 directline that i have, 1234567 is my hotline, can this be vacant when the calls transfered to extensions? |
| 11:17.03 | knight123 | im using POTS |
| 11:20.03 | *** join/#asterisk gilli (n=gilli@p57A1E5EE.dip.t-dialin.net) |
| 11:24.29 | gilli | i don't get it. why can't asterisk suddenly load chan_zap.so anymore? |
| 11:26.32 | gnorbert | Hi, I have a question. Is * with meetme able to make a video conference? |
| 11:27.19 | knight123 | As multiple "hunting" lines can be created on a single POTS line using TDM40B 4 port FXO? |
| 11:27.20 | *** join/#asterisk oej (n=olle@80.251.192.3) |
| 11:28.41 | Tebi | hi, is there any guide how to use DSCP with asterisk? |
| 11:29.18 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:34.43 | *** join/#asterisk ionix (n=ionix@p1093-ipbf3005marunouchi.tokyo.ocn.ne.jp) |
| 11:34.57 | gilli | ah...seems that aptitude removed some files chan_zap.so was depending on. |
| 11:35.10 | ionix | hey guys, letsay I have a DID and when someone calls and punches in the secret extension, I want to allow them to make outbound calls. How can I do that |
| 11:35.31 | *** join/#asterisk glut (n=glut@lowe.wronka.pl) |
| 11:35.31 | ionix | i.e people calling on a 1-800 number and they put extension 12345. Then they can dial any outside number and the system connects it. |
| 11:36.58 | UD | i would like to get a very basic setup going |
| 11:37.16 | UD | to start working on the logic of my programs |
| 11:37.45 | UD | i have an external modem blaster and telephone line simulator |
| 11:38.15 | UD | is there a mechanism for answering the phone on the modem when i test call it |
| 11:39.17 | UD | and sending some audio back like a hotline, and doing events on a DTMF tone? |
| 11:39.23 | *** join/#asterisk jonavogt (n=jonavogt@u51-229.dsl.vianetworks.de) |
| 11:40.49 | jonavogt | Hi all |
| 11:41.53 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 11:42.18 | jonavogt | I'm using mISDN and have the problem that some call don't get through. Asterisk doesn't even seem to now about them. Can someone tell me how to find out where the problem is? |
| 11:42.24 | gnorbert | Has anybody ever made a meetme video conference? |
| 11:45.06 | tzafrir_laptop | gilli, what distribution? what architecture? |
| 11:46.23 | gilli | tzafrir_laptop ...I'm sorry, I'm not sure..do you mean the linux distribution? it's ubuntu hardy 8.04 . |
| 11:47.22 | gilli | kernel 2.6.24-19-386 |
| 11:47.48 | knight123 | so if i have 4 direct line and i using FXO, 1 line will bw my hotline like 123, if someone called, pass to the extensions, hotline number will wil be available? |
| 11:48.39 | knight123 | does asterisk has the capability of hunting? |
| 11:51.38 | gilli | puh...got my zap commands back. problem were some files that chan_zap.so depended on. I guess my package manager had removed them. |
| 11:52.11 | ionix | i.e people calling on a 1-800 number and they put extension 12345. Then they can dial any outside number and the system connects it. |
| 11:52.13 | ionix | hey guys, letsay I have a DID and when someone calls and punches in the secret extension, I want to allow them to make outbound calls. How can I do that |
| 11:54.26 | gilli | tzafrir_laptop, my asterisk version is 1.4.21.1-BRIstuffed-0.4.0-RC3b if the question was related to that. |
| 11:54.45 | gilli | (thanks a lot for the help by the way) |
| 12:01.39 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 12:01.50 | tzafrir_laptop | is back from a good lunch |
| 12:03.09 | tzafrir_laptop | gilli, that sounds like the kernel from Ubuntu Hardy |
| 12:03.34 | tzafrir_laptop | Actually I believe that their asterisk/libpri/zaptel packages support the card |
| 12:03.49 | UD | ionix you gave me an idea |
| 12:04.23 | UD | i wish i could give you ideas i'm just looking in the /etc/asterisk for the first time :) |
| 12:04.42 | UD | but i did read a blog on a pcs free cell->home calling hack |
| 12:04.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:04.52 | gnorbert | Does somebody know, if Asterisk is able to make a meetme video conference? |
| 12:04.55 | gilli | tzafrir_laptop, you mean I should install the ubuntu package instead of compiling&installing the stuff myself? |
| 12:05.12 | UD | something along the lines of what i want to do, no fancy hardware, reading digits |
| 12:05.17 | [TK]D-Fender | gnorbert: No. |
| 12:05.27 | tzafrir_laptop | gilli, that's for you to decide |
| 12:05.35 | tzafrir_laptop | I generally prefer working with debs |
| 12:05.41 | UD | only question i have though is can i just use a regular creative voice modem? |
| 12:05.48 | [TK]D-Fender | UD: No. |
| 12:05.50 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 12:05.51 | UD | :( |
| 12:05.54 | tzafrir_laptop | (that said, I often repackage my own debs. Asterisk included) |
| 12:05.59 | UD | i think i had an x100p |
| 12:06.03 | gnorbert | [TK]D-Fender: Thanks, nice, short answer, understandable even for me. :) |
| 12:06.12 | UD | storage got robbed :( |
| 12:06.30 | UD | i bet they didnt even bother with the obsolete looking x100p |
| 12:06.53 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 12:07.19 | gnorbert | [TK]D-Fender: And is Asterisk able to make a video conference somehow else? |
| 12:07.30 | UD | [TK]D-Fender: what if I pay for SIP service? |
| 12:07.34 | [TK]D-Fender | gnorbert: No. |
| 12:07.42 | UD | im not sure what i need to do |
| 12:07.45 | gnorbert | [TK]D-FenderÉ Thanks. :) |
| 12:07.47 | gilli | i see..I'm not that experianced with linux though. I have to admit that. thing is i need to compile the code on another system later. |
| 12:07.48 | *** join/#asterisk knight123 (n=king2676@202.21.177.3) |
| 12:08.08 | gilli | that's why I'm triying to go the code-way... |
| 12:08.35 | UD | i wanted to be able to do some testing in my little workshop without using the internet, but anyway |
| 12:09.11 | knight123 | hello guys i got disconnected, my question is im using 4ports FXO and how can i configure call hunt to asterisk PBX? |
| 12:09.35 | tzafrir_laptop | knight123, what 4port FXO do you use? a card? |
| 12:09.42 | gilli | however, is not-using the .deb's the reason why asterisk can't recognize the channels on my computer? |
| 12:09.44 | UD | is it possible to answer a call made by my cell phone to some SIP provider to asterisk to do some common stuff like play an introduction sound clip ad read some digits and then execute my programs for reading sensors? |
| 12:10.09 | UD | and conditionally play sound clips based on the output of my programs? |
| 12:10.19 | UD | i need like a hello world example |
| 12:12.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.29 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:12.55 | knight123 | <tzafrir_laptop> yes TDM40B, i just wanted to have 4 POTS direct line and 1 will be 123 which is my hotline, then if someone call 123 and transfered to extension 101, someone call from my hotline again then it will hunt the 3 remaining direct line available, is this possible to Asterisk PBX? |
| 12:13.19 | tzafrir_laptop | UD, do something different based on specific digits entered by the user: that's an IVR |
| 12:13.44 | tzafrir_laptop | UD, a sample IVR is the context [demo] in the sample extensions.conf |
| 12:13.55 | tzafrir_laptop | There are many other ways to implement that |
| 12:14.00 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:14.28 | tzafrir_laptop | "someone calls 123" - from inside your PBX? |
| 12:15.03 | UD | is this done by specific hardware with hardware based dtmf decoders? |
| 12:15.38 | tzafrir_laptop | UD, Asterisk detects DTMF in software |
| 12:16.04 | UD | awsome |
| 12:16.12 | UD | thats sorta the reason i installed |
| 12:16.30 | UD | homebrewed stuff is ugly and takes forever to order |
| 12:16.47 | knight123 | no i mean example 1234567 is my hotline number, which is 7digits, then since that is a hotline, it will be transfered to extensions, so hotline is already hook and if theirs incoming call again, can asterisk support the "call hunt" to redirect the call to my remaining 4 POTS line? |
| 12:17.23 | knight123 | i mean 3pots number |
| 12:17.56 | tzafrir_laptop | knight123, I don't understand what you try to do. I suspect it is rather simple, unless I completely miss it |
| 12:18.00 | UD | can i do what i said in vice versa? |
| 12:18.14 | UD | like just include the API in my code |
| 12:18.22 | UD | to answer the phone and read digits |
| 12:19.05 | UD | so i just can launch a binary |
| 12:19.15 | UD | like a C or perl program |
| 12:19.32 | [TK]D-Fender | UD: What are you looking to do in the end? |
| 12:19.56 | UD | have remote control equipment |
| 12:20.02 | UD | and statistics reports |
| 12:20.15 | UD | farm controls |
| 12:20.34 | knight123 | <tzafrir_laptop> ok sample i have a small business, i have 4ports FXO card in my PBX and i got 4POTS, 1 direct line is my hotline number example 1234567, then since it is a hotline, can asterisk support call haunt if someone called to my hotline instead of busy it will redirect to my remaining 3 POTS line. is it possible |
| 12:20.57 | UD | turn my lights on and off at home with a phone |
| 12:21.15 | [TK]D-Fender | knight123: Go download the BOOK, and get cracking on it. The chapter on AGI will be of interest in integrating *'s interactivity with your custom scripting |
| 12:21.19 | UD | that might be better just launching a program in the asterisk script |
| 12:21.31 | [TK]D-Fender | UD rather |
| 12:21.39 | [TK]D-Fender | ~book |
| 12:21.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:21.41 | [TK]D-Fender | ^^^^^^^^^^ |
| 12:21.58 | knight123 | <[TK]D-Fender> have you ever tried this? |
| 12:22.01 | UD | but in the end i want a nice GUI program to do these things |
| 12:22.09 | knight123 | im just asking if its possible? |
| 12:22.13 | UD | then maybe like when i leave i can still do a few things on the phone |
| 12:22.27 | [TK]D-Fender | knight123: You are getting 4 LINES in from your telco? |
| 12:23.07 | knight123 | yes 4 direct line, im using TDM40B digium card with 4FXO |
| 12:23.14 | [TK]D-Fender | UD: Fairly easy |
| 12:23.42 | tompaw | do you know any european pstn terminator with the capability to send custom CID? |
| 12:23.46 | [TK]D-Fender | knight123: If you're talking about dialing OUT, then yes * can do this. If you want it to hunt on INCOMING calls then you have to ask your telco to do that for you |
| 12:23.48 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 12:24.00 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 12:24.55 | *** join/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 12:25.39 | knight123 | <[TK]D-Fender> ic, so in order to have hotline which is incoming calls, the telco must grant this, right? |
| 12:26.07 | knight123 | i just want to make sure |
| 12:27.03 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:27.19 | knight123 | <[TK]D-Fender> thanks |
| 12:27.19 | [TK]D-Fender | knight123: yes |
| 12:27.42 | jonavogt | knight123: you are concerned that the next one will get a busy signal? |
| 12:30.40 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 12:30.59 | knight123 | <jonavogt> no my concerned is that since i have my hotline number, when it transfered the the extensions 101 for example, it is still hook up, so if anyone would like to call my hotline it will be busy signal, so i thought theirs a way to configure in asterisk to have a call hunt, meaning it will redirected to my 3 available direct line. as what <[TK]D-Fender> says now i got the picture of the scenario. |
| 12:31.47 | *** part/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 12:32.34 | knight123 | <jonavogt> yes it will get busy signal to the next call if thats the case. |
| 12:33.03 | jonavogt | knight123: just wondered if I got the problem. |
| 12:33.06 | tompaw | wonders if it's possible to send DTMFs '123p123'-way with * Manager's Originate |
| 12:36.40 | jonavogt | knight123: seems like I didn't :-D |
| 12:36.59 | [TK]D-Fender | jonavogt: on what lines your calls come IN on its your TELCO's job. |
| 12:37.18 | [TK]D-Fender | tompaw: Send to what? |
| 12:37.46 | [TK]D-Fender | jonavogt: Is that clear for you? |
| 12:37.59 | knight123 | <jonavogt>its ok i got the picture i just want to double check. thanks guys |
| 12:38.16 | jonavogt | [TK]D-Fender: Yes, now it is... never worked with more than one external isdn line yet |
| 12:38.21 | tompaw | [TK]D-Fender: I want to dial a number followed by the DTMF sequence using Asterisk Manager |
| 12:38.32 | tompaw | as of now I got "MessageOriginate failed" |
| 12:38.46 | tompaw | any idea where Asterisk Manager keeps the log files |
| 12:38.47 | tompaw | ? |
| 12:38.59 | [TK]D-Fender | tompaw: You don't send DTMF as part of AMI. That ALL in your Dial command. Go read its instructions again. |
| 12:39.30 | [TK]D-Fender | tompaw: there is not more detail than what you jsut pasted |
| 12:39.46 | tompaw | [TK]D-Fender: ok, but actually I have to use AMI, cause I want those calls to be placed automatically |
| 12:40.02 | tompaw | [TK]D-Fender: really? how can I debug it, then? |
| 12:40.20 | [TK]D-Fender | tompaw: I mean there is no "magic" about sending DTMF. this is not some separate functionality |
| 12:40.30 | tompaw | [TK]D-Fender: affirmative. |
| 12:40.38 | [TK]D-Fender | tompaw: Debug it by looking at what you send it for your call |
| 12:40.41 | tompaw | [TK]D-Fender: to begin with, I am trying to place a simple call frits. |
| 12:40.43 | tompaw | first. |
| 12:41.28 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:42.36 | tompaw | you know what? that was actually a very useful advice ;-) |
| 12:42.51 | tompaw | wonders when Asterisk will begin to cook dinners and breed babies. |
| 12:43.50 | [TK]D-Fender | tompaw: Mine was doing that years ago when I started (the dinner part anyways... |
| 12:44.43 | *** join/#asterisk dmz (n=dmz@216.16.220.180.dyn-cm-pool48.pool.hargray.net) |
| 12:46.29 | *** join/#asterisk bassboi (n=bassboi@ip68-226-196-25.lf.br.cox.net) |
| 12:48.43 | tompaw | so.. if I get things right, there is some kind of 'local' channel that I need to use with Originate, right? |
| 12:48.58 | tompaw | (if I want the call to go through my dialplan) |
| 12:49.13 | [TK]D-Fender | tompaw: Look at what you are dialing. That is all. |
| 12:49.50 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 12:50.09 | tompaw | that's the thing - it dials To: <sip:sip@83.13... |
| 12:50.18 | tompaw | according to http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 12:50.58 | [TK]D-Fender | tompaw: You can't use that as your channel type if you want to pass extra DTMF. |
| 12:51.23 | [TK]D-Fender | tompaw: "Channel:" is ONLY the first parameter of Dial. So you;d have to use a local channel to do the work for you |
| 12:53.05 | Tebi | Hi, how to setup ToS with asterisk 1.2? |
| 12:54.10 | rwaite | is there a way to limit access to one at a time with asterisk's comedian mail? |
| 12:54.20 | [TK]D-Fender | rwaite: Why? |
| 12:54.22 | rwaite | (sorry, in a shared mbox) |
| 12:54.39 | [TK]D-Fender | rwaite: Yes, if you do it in the dialplan yourself |
| 12:55.01 | rwaite | i see, so before connecting, check if someone is already connected? |
| 12:55.27 | *** join/#asterisk PepOSX (n=angeldav@190.72.144.29) |
| 12:55.36 | [TK]D-Fender | rwaite: Clearly. |
| 12:55.54 | tompaw | [TK]D-Fender: let me get this straight. Do I have to add channel with an IP address of 127.0.0.1? |
| 12:56.21 | [TK]D-Fender | tompaw: No, you have to dial a Local channel |
| 12:56.38 | tompaw | [TK]D-Fender: ok, but I do have to ADD it first, right? |
| 12:56.57 | [TK]D-Fender | tompaw: NO. It is a CHANNEL TYPE. You do this INSTEAD OF SIP. |
| 12:57.13 | tompaw | [TK]D-Fender: OK, so it's local, not `local` |
| 12:57.15 | tompaw | ;-) |
| 12:57.17 | tompaw | [TK]D-Fender: thx. |
| 12:57.55 | tompaw | MessageInvalid channel :P |
| 12:59.51 | tompaw | works! |
| 12:59.54 | tompaw | yeah! |
| 13:01.37 | *** join/#asterisk pbrown985 (n=na@wh-gtw-0001.woolfharris.com) |
| 13:01.59 | tompaw | ok, so if I got things right, this Originate works in 2 steps. It calls the channel, channels responds and it then calls the extension. |
| 13:02.04 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 13:02.04 | *** mode/#asterisk [+o file] by ChanServ |
| 13:02.13 | *** part/#asterisk bassboi (n=bassboi@ip68-226-196-25.lf.br.cox.net) |
| 13:02.21 | [TK]D-Fender | tompaw: not so much "calls" ans "dumps into the dialplan" |
| 13:02.58 | tompaw | [TK]D-Fender: right, the thing is, those 2 ends are connected with asterisk, so it works similar to callback featuers. |
| 13:03.01 | tompaw | features. |
| 13:03.10 | tompaw | (at least for me here) |
| 13:03.45 | *** part/#asterisk ctaloi (n=ctaloi@pool-72-90-82-34.syrcny.fios.verizon.net) |
| 13:04.21 | *** join/#asterisk jpastore (n=jpastore@crlspr-24.233.166.216.myacc.net) |
| 13:04.29 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:05.09 | tompaw | so, in my case, let's say I try to use Channel=local/555ppp1p2pp@my_numberplan as the channel, and exten=my_phone as extension |
| 13:05.24 | [TK]D-Fender | tompaw: Sure |
| 13:05.25 | tompaw | will it place the call and start dtmfing it while I simply listen to the whole process on my side? |
| 13:06.42 | tompaw | [TK]D-Fender: that is amazing, isn't it? |
| 13:07.04 | tompaw | MessageOriginate successfully queued |
| 13:07.05 | tompaw | :-) |
| 13:07.15 | [TK]D-Fender | tompaw: It ain't Raw-Cat Science... |
| 13:09.05 | tompaw | (; |
| 13:09.10 | tompaw | so now I got this: Dial("Local/40902341000pp2p2@numberplan-custom-2-c1bd,1", "SIP/trunk_6/6969") |
| 13:09.18 | tompaw | (the 2nd part is my phone) |
| 13:09.39 | tompaw | it does connect the call, but those 'pp2p2' don't seem do be working. |
| 13:10.19 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:10.45 | [TK]D-Fender | tompaw: ... NOT BRIGHT.. Its not prat fo the friggen EXTEN. its a PARAMETER TO DIAL |
| 13:11.00 | [TK]D-Fender | tompaw: "core show application dial" <- go read the instructions. |
| 13:11.06 | [TK]D-Fender | part* |
| 13:11.16 | tompaw | yes sir |
| 13:11.31 | Katty | good morning sunshines! |
| 13:12.33 | tompaw | D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. |
| 13:12.54 | tompaw | [TK]D-Fender: I assume that means those DTMFs would be sent before my trunk_6/6969 would be called, right? |
| 13:13.42 | [TK]D-Fender | tompaw: ... learn how to read. |
| 13:13.59 | [TK]D-Fender | tompaw: " Send the specified DTMF strings *after*" <---- |
| 13:14.42 | jonavogt | repeating my earlier question, maybe now some can help. I'm using mISDN with a Digium b410p Card but some calls don't even show up in any log so far. Asterisk is the only Device on the ISDN channel. So where can i start to find those missing calls. |
| 13:15.35 | Katty | morning mike! |
| 13:16.15 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 13:17.08 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:19.26 | [TK]D-Fender | Katty: Mew. |
| 13:19.50 | [TK]D-Fender | jonavogt: What kind of "calls"? |
| 13:21.26 | jonavogt | [TK]D-Fender: External Calls. I haven't found what they have in common yet. As far as I know most calls orginate from ISDN phones |
| 13:22.05 | [TK]D-Fender | jonavogt: Ask your telco to look into it. You don't have anything to start from it seems |
| 13:22.17 | jonavogt | Calls from Cellphones seem always to come thru. |
| 13:22.28 | jonavogt | [TK]D-Fender: okay... to bad. |
| 13:22.44 | tompaw | [TK]D-Fender: but *before* the call gets bridged. |
| 13:23.02 | ManxPower | tompaw: "bridged" == connecting audio between the two calls |
| 13:23.13 | [TK]D-Fender | tompaw: that isn't before your oubound call is placed.... |
| 13:23.21 | tompaw | right, and is'nt that what I do with Dial("Local/40902341000pp2p2@numberplan-custom-2-c1bd,1", "SIP/trunk_6/6969") ? |
| 13:23.26 | [TK]D-Fender | tompaw: Both legs have been called AND have answered |
| 13:23.26 | tompaw | (bridging) |
| 13:24.38 | tompaw | ok. I just need to find out how to set up Dial's options from AMI's Originate. |
| 13:24.40 | *** join/#asterisk moy (n=moy@nat/ibm/x-c92b67da5ffd2fc1) |
| 13:24.41 | [TK]D-Fender | tompaw: it sends AFTER the call has been answered. |
| 13:24.46 | [TK]D-Fender | tompaw: YOU DON'T. |
| 13:25.06 | [TK]D-Fender | tompaw: You don't set Dial's options in ORIGINATE |
| 13:25.23 | [TK]D-Fender | tompaw: You do it in the Llocal channel you call FROM originate. |
| 13:25.50 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 13:27.45 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 13:28.22 | ManxPower | Is Llocal anything like a Llama? |
| 13:28.46 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:29.02 | [TK]D-Fender | ManxPower: mmmmm alternative protein :D |
| 13:29.46 | tompaw | my head hurts. |
| 13:29.54 | ManxPower | "If god didn't want us to eat animals, why did he make them out of meat?" --Homer Simpson |
| 13:30.33 | *** join/#asterisk zoid_99 (n=zoid_99@router.asteriasgi.com) |
| 13:30.35 | *** join/#asterisk LemensTS (i=LemensTS@adsl-70-238-134-114.dsl.stlsmo.sbcglobal.net) |
| 13:30.48 | tompaw | [TK]D-Fender: by "You do it in the Llocal channel", do you mean I have to pre-configure it in the .conf? |
| 13:30.57 | ManxPower | tompaw: correct. |
| 13:31.03 | dominic1 | any ideas how I can initiate a attended transfer via management? How can I put a call on hold? |
| 13:31.40 | [TK]D-Fender | tompaw: Chan_local is just a call into the dialplan. Yuo do EXACTLY what you would under any other notmal circumstance in there. |
| 13:32.01 | [TK]D-Fender | normal* |
| 13:32.08 | tompaw | ok then. so it looks like I have to write it to Asterisk's database first (the top up code) and then use DB() in the local channel dialplan to get those DTMFs value for Dial() |
| 13:32.09 | [TK]D-Fender | tompaw: YOU call Dial, and YOU pass it the aprameters. |
| 13:32.12 | tompaw | does it sound reasonable? |
| 13:32.56 | [TK]D-Fender | tompaw: you can pass the DTMF to dial encoded into the exten you dial in chan_local, or you can pass it as a parameter from your Originate |
| 13:33.45 | tompaw | ok, but then my dialplan has to somehow "unpack" it and use as a Dial's parameter, right? |
| 13:34.25 | [TK]D-Fender | tompaw: If you choose that method, yes. |
| 13:34.26 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 13:34.42 | [TK]D-Fender | tompaw: Or you can set a variable in your Originate and use that instead |
| 13:35.09 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:35.38 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 13:36.32 | Katty | [TK]D-Fender: good morning (= |
| 13:42.55 | Tebi | [TK]D-Fender: can you please help me? Is´t possible to use ToS value 0xB8 with asterisk 1.2.24? |
| 13:43.04 | *** join/#asterisk xacatecas (n=jkroon@dsl-241-158-229.telkomadsl.co.za) |
| 13:48.10 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 13:48.21 | [TK]D-Fender | Tebi: Don't target people with questions like that, its rude. |
| 13:48.29 | tompaw | whoa whoa whoa :) |
| 13:48.38 | ManxPower | Tebi: YES! |
| 13:48.57 | xacatecas | hi guys, if astdb got corrupted, is there any way to "recover" it ? |
| 13:49.01 | *** join/#asterisk _sc0tty_ (i=Sc0ttY@217.144.147.41) |
| 13:49.14 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 13:49.15 | keith4_ | does "restore from backup" qualify as "recover"? |
| 13:49.24 | ManxPower | xacatecas: AstDB is a BerkleyDB. So whatever you would do to recover those. |
| 13:49.24 | tompaw | [TK]D-Fender: http://pastebin.com/m6d322e8c |
| 13:49.29 | tompaw | what do you think? |
| 13:50.20 | ManxPower | (an older version of Berkley DB) |
| 13:50.27 | [TK]D-Fender | tompaw: I think thats a mess that doesn't tell me much of anything. |
| 13:50.36 | ManxPower | tompaw: I think your example is 10x more complex than it should be when showing us. |
| 13:51.14 | ManxPower | We are not going to write your dialplan, we will help you with specific issues. |
| 13:51.36 | tompaw | http://pastebin.com/m5cab991f |
| 13:51.47 | tompaw | ManxPower: it's actually working, I was just showin up ;-) |
| 13:52.03 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 13:52.03 | ManxPower | tompaw: good |
| 13:52.11 | tompaw | the last missing bit is: |
| 13:52.15 | [TK]D-Fender | tompaw: Does it WORK? |
| 13:52.22 | tompaw | [TK]D-Fender: yes. |
| 13:52.33 | [TK]D-Fender | tompaw: Fine, then move on. |
| 13:52.39 | tompaw | Illegal DTMF character 'p' in string. (0-9*#aAbBcCdD allowed) |
| 13:52.44 | tompaw | how do I send a pause? |
| 13:52.54 | [TK]D-Fender | tompaw: Who the hell told you that "p" was legal? |
| 13:53.02 | tompaw | erm... my cellphone? :P |
| 13:53.06 | ManxPower | tompaw: you can only send a pause INSIDE the dialstring on ANALOG and it'snot a "p" it's a"w" |
| 13:53.51 | ManxPower | For anything else use a the D() option to dial, not as part of the number |
| 13:54.41 | tompaw | so there's no way to send a 'digital' pause? |
| 13:54.59 | ManxPower | tompaw: Asterisk is not your cell phone. Stop applying rules for other things to Asterisk. Doing that will just cause you pain and misery -- much like a country song. |
| 13:55.12 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 13:55.39 | xacatecas | ManxPower, i can recover a version 4, but, but i'm struggling with version 1, busy googling though. |
| 13:55.49 | Tebi | thank you :) |
| 13:56.11 | [TK]D-Fender | xacatecas: jsut DELETE the file and * will start a NEW ONE |
| 13:56.12 | tompaw | ManxPower: I was told to develop things through experiments. I had to start with something, right? |
| 13:56.19 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 13:56.25 | Tebi | [TK]D-Fender>: sorry |
| 13:56.45 | ManxPower | tompaw: that is true, but usually that means "as documented in Asterisk" |
| 13:58.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:59.06 | [TK]D-Fender | tompaw: Why don't you go invent entirely new Dialplan apps like "exten => _X.,1,AddDTMFAnyWayILike(SIP/provider/number,30,DTMF(12345,wait 10 seconds,make coffee, Accept voicecommands,etc)" and see if they work too? |
| 14:00.25 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:00.32 | xacatecas | [TK]D-Fender, except i think there is information in there that i really want to keep :(. not by my design though. |
| 14:01.13 | tompaw | [TK]D-Fender: because I'm still suffering from the headache caused by those crappy TELES iGATEs. is 8 |
| 14:01.20 | [TK]D-Fender | xacatecas: Can you get a dump of it? |
| 14:01.25 | tompaw | it *seems* to me like they do not recognize anything but inband |
| 14:01.37 | [TK]D-Fender | tompaw: So go use inband |
| 14:02.01 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:02.14 | tompaw | now, the only way they recognize my dtmfs is in the following configuration: spa922--[g729]--asterisk--[g729]--igate |
| 14:02.18 | tompaw | with INBAND |
| 14:02.25 | tompaw | even though theoretically g729 should kill it. |
| 14:02.37 | [TK]D-Fender | Tebi: "tos=0xB8" under [general] in sip.conf |
| 14:02.52 | [TK]D-Fender | tompaw: Yes, that is jsut plain stupid |
| 14:02.59 | xacatecas | [TK]D-Fender, no, I can use strings on it and that is pretty much the only sensible interaction I can get out of the crashed one. |
| 14:03.10 | tompaw | [TK]D-Fender: totally agree. as stupid as it is, that's the only config in which it works. |
| 14:03.24 | [TK]D-Fender | xacatecas: So "database show" in CLI doesn't give you anything usable? |
| 14:03.40 | xacatecas | well, asterisk goes up to 99% usage and locks my CLI. |
| 14:03.51 | *** join/#asterisk mrnick (n=basement@kulnet-nat-2.kulnet.kuleuven.be) |
| 14:03.56 | tompaw | is there a way to force * to use inband with g729? (even know all odds say it's NOT gonna work) |
| 14:04.05 | [TK]D-Fender | xacatecas: atke the file out to another box and do it there |
| 14:04.11 | ManxPower | tompaw: yes |
| 14:04.56 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 14:04.56 | *** mode/#asterisk [+o file] by ChanServ |
| 14:05.12 | tompaw | ManxPower: any other than sources manipulation? |
| 14:05.22 | ManxPower | huh? |
| 14:07.12 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:07.50 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 14:08.18 | xacatecas | [TK]D-Fender, same thing. as soon as I try and load up that db asterisk goes into 100 % usage and database show gives no results. |
| 14:08.36 | ManxPower | tompaw: If you get more than 2 or 3 correct DTMF out of 10 DTMFs when running G729, then you are no t using inband. |
| 14:08.41 | [TK]D-Fender | xacatecas: Ok, high usage is one thing... no results is another... no idea from this point. |
| 14:08.53 | tompaw | ManxPower: actually, I get 100% |
| 14:09.00 | xacatecas | i had similar results when a bdb crashed under openldap, a db4.2_recover usually managed to get me back up but this bdb version == different from the tools. |
| 14:09.26 | ManxPower | then you know what I'm getting at,. Why are you using G729 anyway? |
| 14:10.09 | tzafrir_laptop | xacatecas, asterisk uses bdb c. 1.86 |
| 14:10.13 | tompaw | to preserve the bandwidth |
| 14:10.32 | [TK]D-Fender | xacatecas: Got enough so you export your data? |
| 14:10.34 | tzafrir_laptop | (the last version that had a GPL-compliant license) |
| 14:10.43 | ManxPower | how much bandwidth do you have between the different devices? |
| 14:10.43 | Hertzy3 | Hey all, Ive got a problem with receiving phone calls. Sometimes, not all the time, there is a 1-way audio problem. The call goes through, but the caller can't hear me at all. Does anyone have any suggestions? This happens for each phone on a call ring, and I have repowered the switch already |
| 14:10.54 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 14:10.54 | ManxPower | specifically betewwn Asterisk and the gateway |
| 14:11.08 | xacatecas | [TK]D-Fender, no, 100 % cpu usage == similar to when bdb crashed under openldap. unable to get any data from it. |
| 14:11.20 | xacatecas | looking for a bdb_dump tool of sorts, that might help. |
| 14:11.33 | [TK]D-Fender | xacatecas: db4.2_recover usually managed to get me back up but this bdb version == different <-- from THI I mean |
| 14:11.40 | tompaw | ManxPower: theoretically 1Mbps |
| 14:11.53 | [TK]D-Fender | herzdescribe that call in DETAIL. |
| 14:11.54 | tompaw | ManxPower: I tested various codecs and it really works best with g729 |
| 14:12.01 | xacatecas | oh, no, it basically bombs out and says the version is incorrect. |
| 14:12.06 | ManxPower | You understand that an ulaw or alaw call uses 0.008Mbps, right? |
| 14:12.38 | tompaw | ManxPower: it still uses 3x more than g729, right? |
| 14:13.02 | ManxPower | When you have a million dollars, then $800 means NOTHING. |
| 14:13.10 | xacatecas | looking to see if I can compile the right version of the tools now ... |
| 14:13.12 | rbd | hi guys, I have a situation where I have openser and asterisk running on the same box. I get a SIP call in from asterisk, it does a Dial() and sends the INVITE over to openser, which rewrites the RURI and then sends the call to asterisk. With this, asterisk gets confused and issues a 486 Loop detected error. The callID is the same, however the RURI is different. I did try running asterisk with pedantic=yes but that doesn't seem to ha |
| 14:13.23 | ManxPower | tompaw: and G729 uses 5x the CPU power. |
| 14:14.04 | tompaw | ManxPower: cpu is not an issue at all. (4 xeons on board) It simply works best with g729, tested all of them. |
| 14:14.29 | ManxPower | tompaw: apparently not according to your problem. 8-) |
| 14:15.25 | tompaw | god, I wish I didn't have to work with this crap, but I don't have much choice in here. |
| 14:15.53 | jaytee | I think 800 bucks is alot to someone who has millions because they're the type of people who figured out how to charge us 9/10ths of a penny extra for a gallon of gas. |
| 14:16.05 | tompaw | jaytee: not all of them. |
| 14:17.00 | tompaw | wow |
| 14:17.14 | tompaw | my screen was flooded with Inband DTMF is not supported on codec g729. Use RFC2833 |
| 14:17.17 | tompaw | yet still, it worked |
| 14:17.17 | jaytee | of course not all of them, that would be a flawed generalization. |
| 14:17.19 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 14:17.25 | tompaw | I do *not* understand it at *All*. |
| 14:19.12 | ManxPower | It's a bad idea. Asterisk is telling you this. |
| 14:19.47 | tompaw | ManxPower: so how much are you saying alaw takes? 0.008? or 0.08 maybe? |
| 14:20.36 | ManxPower | SIP alaw and ulaw take 64Kbps + UDP overhead, which works out to be about 80Kbps, so it would be 0.08Mbs |
| 14:20.51 | *** join/#asterisk Illarane (n=heifer@pdpc/supporter/student/Veratien) |
| 14:21.44 | tompaw | And G729 is around 0.02, right? |
| 14:21.45 | Illarane | Hiya, which file controls whether or not the automatic-announcey-person-thing says 'and' in numbers? |
| 14:21.50 | [TK]D-Fender | ManxPower: I think there's a position open in Verizon's accounting dept for you ;) |
| 14:22.01 | Illarane | [TK]D-Fender: Ouch... :p |
| 14:22.20 | Illarane | $.002 == .002c? |
| 14:24.01 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:24.33 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:25.01 | tompaw | I think I found the solution. |
| 14:25.42 | tompaw | I can leave the trunk defined as g729 for ordinary calls. For topups I will make a copy of the same trunk but with alaw. I'm not going to be doing more than 1 topup/time anyway. |
| 14:25.47 | *** join/#asterisk knight123 (n=king2676@202.21.177.3) |
| 14:26.17 | tompaw | I'm pretty sure it would work with g729 anyway, but asterisk has much more experience in this field than me, so if it warns me against it... it must have some reasons. |
| 14:27.42 | knight123 | hello guys, i forgot to ask earlier, from my inbound, after the IVR, when the calls transfered to extensions, how can i change the settings instead of ringing, i want to hear music playing? |
| 14:28.04 | Hertzy3 | [TK]D-Fender: Theres actually 2 problems, I dont know if they are related or not. Sometimes the call will come into the call ring and just ring off the hook. The answer button will not answer the call, it just keeps ringing on all phones. Other times the call is answered but halfway thru the conversation the caller can no longer hear us, but we can hear them just fine. So they hang up |
| 14:28.09 | *** join/#asterisk Hastalavi (n=kumar@mail.netvita.com) |
| 14:28.10 | ManxPower | knight123: Your question indicates that you never read The Asterisk Book |
| 14:28.35 | tompaw | [TK]D-Fender, ManxPower: thank you guys for help, as always. Now I got everything to launch my top-up application. |
| 14:28.36 | ManxPower | knight123: "/etc/asterisk/musiconhold.conf" and "core show application dial" |
| 14:28.49 | [TK]D-Fender | knight123: "core show application dial" <--- |
| 14:28.58 | *** join/#asterisk udzinari (n=david@6-164.cdn.ge) |
| 14:29.07 | knight123 | i read it but it makes me confused |
| 14:29.37 | [TK]D-Fender | knight123: It tells you the parameter to MoH instead of ringing. Keep reading it til your eyes bleed |
| 14:29.53 | ManxPower | then read it again. Playing music instead of ringing is such a basic part of Asterisk....well...if you can't figure it out, maybe you should be trying something else. |
| 14:29.54 | [TK]D-Fender | Hertzy3: Describe the call's origin in DETAIL. |
| 14:30.57 | knight123 | <[TK]D-Fender> ok thanks guys |
| 14:31.07 | *** part/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:31.27 | ManxPower | knight123: Asterisk is complicated, very, very, very technical, and confusing. It will take you a long time to be comfortable with Asterisk |
| 14:31.42 | ManxPower | It's not really Asterisk, but VoIP in general. |
| 14:34.03 | knight123 | <ManxPower> yes your right thats why ehn i tried to build one it works perfectly but a little adjustment about some extras that needs to be change, confusing but its good, also its nice coz theirs support that i can ask when i'm lost like you guys telling those hints will be good to make my PBX more accurate. |
| 14:34.18 | *** join/#asterisk tobias (n=tobias@cpe-069-134-204-037.nc.res.rr.com) |
| 14:34.20 | knight123 | thanks, thats y i love asterisk even its tricky |
| 14:36.53 | Illarane | blinks as his phone tells him his minutes have been carried away by monkies. |
| 14:38.19 | mshades | i've use asterisk as a fecal foreplay telephone party-line for 3 years and i'm still not comfortable with it |
| 14:38.42 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:39.45 | *** join/#asterisk [1]_sc0tty_ (i=Sc0ttY@217.144.147.41) |
| 14:40.17 | Hertzy3 | [TK]D-Fender: Honestly Im not sure how. I am waiting for it to happen again so that I can see what happens in the console. When it does I will show you |
| 14:41.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:41.23 | *** join/#asterisk rukus (n=daniel@dsl-245-125-70.telkomadsl.co.za) |
| 14:41.57 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 14:42.26 | rukus | quick question, is there a monster faq avaiable ? |
| 14:43.01 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:45.09 | tompaw | regarding dialplan variables - BLAH:5 means BLAH starting from 5th char. how can I make 'BLAH starting from 5th char 7 chars ahead' ? |
| 14:45.12 | tompaw | BLAH:5:7? |
| 14:45.57 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 14:46.20 | Katty | so much sleepy. |
| 14:46.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:46.58 | angryuser | good day |
| 14:49.51 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 14:52.40 | DagMoller | ~book |
| 14:52.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:54.06 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 14:54.10 | Xen^ | hello every one |
| 14:54.21 | Xen^ | [TK]D-Fender: arround ? |
| 14:54.43 | [TK]D-Fender | Xen^: Yes |
| 14:54.49 | Xen^ | [TK]D-Fender: how are you doing ? |
| 14:55.02 | [TK]D-Fender | tompaw: Go read "asterisk variables" on the WIKI |
| 14:55.04 | *** join/#asterisk jmacz (n=jmacz@190.26.189.104) |
| 14:55.09 | [TK]D-Fender | Xen^: Still breathing :) |
| 14:55.20 | Xen^ | [TK]D-Fender: you remember you tried to help me with register sip account under asterisk |
| 14:55.26 | Xen^ | [TK]D-Fender: hehe |
| 14:55.34 | [TK]D-Fender | Xen^: Maybe... |
| 14:55.40 | Xen^ | [TK]D-Fender: well i am still facing same issue, this issue is also on FreeSwitch :( |
| 14:56.08 | Xen^ | [TK]D-Fender: can you please take a look into it once please |
| 14:56.09 | Xen^ | :$ |
| 14:56.17 | [TK]D-Fender | Xen^:Right now "maybe" is looking more like "No.". Whats the issue? |
| 14:56.50 | Xen^ | well i have sip user which can be registered on any softphone but i can not able to register it on asterisk ... |
| 14:56.58 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:57.13 | [TK]D-Fender | Xen^: What provider? |
| 14:57.19 | Xen^ | user: user, auth user: user@domain.com, pass: passcode, domain:domain.com, proxy:ip:9060 |
| 14:57.28 | Xen^ | [TK]D-Fender: its www.wateen.com :$ |
| 14:58.39 | [TK]D-Fender | Xen^: pastebin everything again and maybe someone can add something to that.. |
| 14:58.49 | Xen^ | okies |
| 14:58.49 | Xen^ | hold |
| 14:58.58 | gilli | hey guys... if my ztcfg prints "62 channels to configure" can I assume that the channels are configured or is there something wrong? |
| 14:59.40 | angryuser | gilli all is ok |
| 14:59.57 | [TK]D-Fender | gilli: Thats only zaptel.conf. Doesnt' say your Zapata.conf has anything usable in it... |
| 15:00.30 | gilli | thanks ..but how will I know if my zapata.conf is useful? |
| 15:01.16 | [TK]D-Fender | gilli: Gee I don't know... tried placing any CALLS with it yet? |
| 15:01.23 | tompaw | "Changing DTMF duration when sending for ZAP channels" << how can I change the tone duration (inband) when NOT in ZAP channels? |
| 15:01.35 | ManxPower | tompaw: you can't |
| 15:01.41 | Xen^ | http://rafb.net/p/dsKbPV55.html |
| 15:01.53 | *** join/#asterisk hanchi (n=hanchi@24.182.209.194) |
| 15:02.52 | gilli | [TK]D-Fender, no. I'd like to get there but asterisk doesn't seem to see any channels. |
| 15:03.11 | ManxPower | I only read up to THREE HUNDRED pastebin lines. |
| 15:03.32 | Xen^ | ummm |
| 15:03.42 | *** part/#asterisk hanchi (n=hanchi@24.182.209.194) |
| 15:03.49 | [TK]D-Fender | gilli: And how are you looking at them? |
| 15:03.55 | [TK]D-Fender | gilli: PASTEBIN is your friend. |
| 15:03.57 | [TK]D-Fender | ~pb |
| 15:03.58 | 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 |
| 15:03.58 | Xen^ | it includes eyeBeam Debug, SIP Debug and SIP Configuraton |
| 15:03.58 | ManxPower | Xen^: What specific issue are you having? |
| 15:03.59 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:04.00 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:05.01 | Xen^ | ManxPower: i can not able to register this sip user under asterisk |
| 15:05.01 | *** join/#asterisk ToTo (n=ToTo@209.8.41.202) |
| 15:05.17 | ManxPower | There's the one line we care about: [Jul 28 09:10:35] WARNING[19014]: chan_sip.c:12530 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '0218000342' to '58.27.240.22' |
| 15:05.39 | Xen^ | ManxPower: but same password is working on softphone |
| 15:05.46 | ManxPower | so you have a wrong password in [0218000342] section of sip.conf |
| 15:06.09 | Xen^ | well same password i used in [0218000342] and in register => |
| 15:06.34 | ManxPower | You can argue all day, the message is clear. |
| 15:06.48 | ManxPower | what is running on 58.27.240.22 |
| 15:07.09 | Xen^ | well its have Motorola-IMS/3.1 |
| 15:07.19 | Xen^ | it is run by provider |
| 15:07.54 | ManxPower | As I said, the message is clear. |
| 15:08.50 | ManxPower | remember, many providers only allow you to change the web portal password online and do not allow you to change your SIP auth details via their web interface. |
| 15:08.52 | Xen^ | MaxPower: i can share login details with you also give you access to machine to look by your self. I am very carefull about password. same this is not working on FreeSwitch and there is already bug report... |
| 15:09.12 | gilli | [TK]D-Fender, zapata.conf http://gilli.pastebin.com/d768af75a |
| 15:09.16 | ManxPower | Xen^: I can have access to a Motorola-IMS/3.1? |
| 15:09.32 | Xen^ | ManxPower: if my password changed then i could not able to login using softphone |
| 15:09.33 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:09.51 | [TK]D-Fender | gilli: Ok, looks basic enough. Now show me a problem |
| 15:09.56 | ManxPower | Xen^: I cannot help you further. |
| 15:09.59 | Xen^ | ManxPower: if i do have access to Motoroal-IMS/3.1 then i believe i get it resolved by looking into their debug logs :) |
| 15:10.49 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:11.20 | gilli | [TK]D-Fender, my problem is that I'd like to put a simple call, send some text and see it arriving. to do that, tzafrir told me that asterisk should be able to show the channels. but it doesn't. |
| 15:11.51 | gilli | so 'zap show channels' only prints a pseudo device. |
| 15:12.07 | drako | core show channels |
| 15:12.18 | gilli | trying this.. |
| 15:12.19 | ManxPower | the sending of text will be the major problem |
| 15:12.33 | gilli | y? |
| 15:13.04 | fogo | According to the documentation, TE410Ps can only take timing from one of the four spans - could this cause problems with timing errors on the other ports? |
| 15:13.17 | ManxPower | gilli: You place calls between endpoints (usually phones). What protocol does your phone use to receive text? |
| 15:13.23 | tzafrir_laptop | 'core show channels' is something different |
| 15:13.38 | ManxPower | tzafrir_laptop: *nod* It shows ACTIVE channels. |
| 15:13.47 | tzafrir_laptop | 'zap channels' are similar, in a way, to sip peers and such |
| 15:13.55 | ManxPower | fogo: not normally. |
| 15:14.06 | [TK]D-Fender | gilli: try "module reload chan_zap.so |
| 15:14.15 | ManxPower | tzafrir_laptop: *nod* A good way to be sneaky about seeing if zapta is setup correctly. |
| 15:14.18 | gilli | trying this.. |
| 15:14.18 | [TK]D-Fender | gilli: And then recheck "zap show channels" |
| 15:14.42 | Xen^ | any one else can help me out ? |
| 15:15.02 | fogo | ManxPower: I didn't think so .. I would think Digium would have things figured out. At any rate, our provider was seeing timing errors on the line; although I'm inclined to think it's their problem. |
| 15:15.11 | gilli | reloading chan_zap.so gave some warnings and rechecking 'zap show channels' still just prints the pseudo device. |
| 15:15.36 | ManxPower | gilli: then you don't have a valid /etc/asterisk/zapata.conf |
| 15:15.43 | ManxPower | or you don't have any digium cards. |
| 15:15.52 | gilli | no, I have a junghanns card. |
| 15:16.02 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 15:16.02 | tzafrir_laptop | gilli, could you please re-post your /proc/zaptel/* ? It seems beyond my scroll buffer |
| 15:16.15 | gilli | sure, thanks btw...one moment. |
| 15:16.17 | ManxPower | fogo: the only time you would normally have issues with the single timing source would be if you were plugging different telcos into different ports on the same card. |
| 15:16.54 | [TK]D-Fender | gilli: Junghanns has a full E1 card for normal * usage? |
| 15:17.17 | tzafrir_laptop | [TK]D-Fender, yes. It's the driver in bristuff called cwain' |
| 15:17.23 | tzafrir_laptop | cwain |
| 15:17.37 | [TK]D-Fender | tzafrir_laptop: bleh. |
| 15:17.38 | gilli | tzafrir_laptop: http://gilli.pastebin.com/d4697726c |
| 15:18.05 | [TK]D-Fender | I am so very much backing out of this one... |
| 15:18.12 | ManxPower | [TK]D-Fender: good call! |
| 15:18.15 | fogo | ManxPower: hrm.. we are. 3 PRIs from one provider, one from another. However, I have switched the primary timing source to one of their three, and we're still getting drops (all 3 of their PRIs will drop at the same time)- the other PRI from the other provider hasn't dropped the entire time. |
| 15:18.40 | ManxPower | fogo: no errors I assume (or you would have told me the error messages already) |
| 15:19.14 | ManxPower | ..er.. no error MESSAGES, I assume |
| 15:19.21 | tzafrir_laptop | gilli, good. So you have a good zaptel.conf . Now, what do you have in /etc/asterisk/zapata.conf ? |
| 15:19.39 | gilli | copying... :) |
| 15:19.58 | fogo | fogo: not that I can see - all I can catch is the recovering message in 'zap show status' - should I be looking somewhere else as well? |
| 15:20.06 | *** join/#asterisk dikdust (n=dikdust@77.43.42.95) |
| 15:20.14 | dikdust | hi |
| 15:20.20 | ManxPower | alarms should show up in the CLI |
| 15:21.19 | fogo | ManxPower: checking logs... |
| 15:23.00 | gilli | tzafrir_laptop: http://gilli.pastebin.com/d1068ea33 ...looks like a problem? |
| 15:23.38 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
| 15:23.52 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 15:24.06 | tzafrir_laptop | gilli, you didn't actually add anything there (or replace it) |
| 15:24.41 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:24.44 | gilli | i actually thought that asterisk would read the one from /etc/ and not from the subdir.. |
| 15:24.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:24.55 | tzafrir_laptop | Either use the one from the sample , or use genzaptelconf to generate the missing bits for you |
| 15:25.20 | tzafrir_laptop | There's /etc/zaptel.conf directly under /etc/ . But this is for kernel-level configuration |
| 15:25.49 | gilli | and would it be wrong to simply copy the one from /etc/ to /etc/asterisk ? |
| 15:26.52 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 15:28.57 | fogo | ManxPower: looks like I am getting alarms: http://pastebin.ca/1085296 |
| 15:29.04 | gilli | now I ran genzaptelconf but I'm not sure if it changed anything in /etc/asterisk/zapata.conf |
| 15:32.36 | *** part/#asterisk jonavogt (n=jonavogt@u51-229.dsl.vianetworks.de) |
| 15:32.47 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
| 15:34.12 | ManxPower | fogo: NOW you can tell your telco. |
| 15:34.41 | ManxPower | fogo: RED alarms are almost never timing issues. They are hardware or cable issue. |
| 15:35.41 | fogo | ManxPower: I figured it was their problem - they even saw errors coming from their end on a t-bird, and still think it's my issue because the line tests clean |
| 15:35.46 | gilli | tzafrir_laptop: after having run genzaptelconf I still had the same problem in asterisk. but after replacing zapata.conf with the one from /etc/ I get a lot of warnings and 'zap show channels' seems to print out 62 'demo channels'.. |
| 15:36.20 | ManxPower | gilli: /etc/zapata.conf and /etc/asterisk/zaptel.conf are totally different files and do totally different things and cannot be interchanged. |
| 15:37.00 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 15:37.03 | tzafrir_laptop | add '#include zapata-channels.conf' to the end of /etc/asterisk/zapata.conf |
| 15:37.16 | tzafrir_laptop | or just add its content to the end of zapata.conf |
| 15:37.42 | gilli | doing this... |
| 15:38.08 | Alan_Hicks | Howdy! I'm using Polcom Soundpoint IP320 phones and have implimented a macro that pages all of my phones when a certain extension is dialed. The phones auto-answer and immediately two-way communication starts. |
| 15:38.24 | fogo | ManxPower: upon further inspection, I'm also seeing "Unable to disable echo cancellation on channel XX" - I'm assuming this is because the line just went red? |
| 15:38.36 | Qwell | Alan_Hicks: cool |
| 15:38.39 | Alan_Hicks | What I would like to do is impliment one-way paging, essentially mute the paged phones when this macro is dialed. Can anyone point the way out to me? |
| 15:38.57 | Qwell | Alan_Hicks: I think that's just modifying the SIP header you add |
| 15:38.59 | tompaw | anyone in here has any experience with the TELES equipment? |
| 15:39.04 | Alan_Hicks | I've browsed the admin manual for these phones and the example conf files, but nothing's struck me. |
| 15:39.09 | tompaw | bangs his head against the wall. |
| 15:39.38 | Alan_Hicks | I can pastebin any files you feel may be useful. |
| 15:39.53 | *** join/#asterisk dr_gogeta86 (n=dr_goget@81-208-88-100.ip.fastwebnet.it) |
| 15:39.59 | [TK]D-Fender | Alan_Hicks: Should have started with that. |
| 15:40.06 | dr_gogeta86 | hi to all |
| 15:40.44 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 15:41.18 | Alan_Hicks | macro from extensions.conf --> http://pastebin.com/d6eac2b45 |
| 15:41.32 | *** part/#asterisk exothermic (n=miles@74.85.89.236) |
| 15:42.59 | gilli | tzafrir_laptop: I'm not sure if my zapata-channels.conf is looking right, because everything seems commented... http://gilli.pastebin.com/d6ceabadb |
| 15:44.26 | [TK]D-Fender | Alan_Hicks: DIAL is not PAGE. |
| 15:46.14 | *** join/#asterisk jmacz (n=jmacz@201.244.199.90) |
| 15:46.41 | Alan_Hicks | I'm such a dumb-ass! Thanks. |
| 15:47.33 | tzafrir_laptop | gilli, duh. Because cwain does funny games with the /proc/zaptel file :-( |
| 15:48.05 | tzafrir_laptop | gilli, anyway, it should be something in the lines of: |
| 15:48.47 | fogo | ManxPower: sorry to keep going over this, but looking into my log, I'm seeing an alarm on span 1, span 3, an HDLC Abort on span 2, then an alarm on span 2. Aren't HDLC Aborts caused by interrupts; or could this be coming about due to the alarms? |
| 15:48.57 | *** join/#asterisk ToTo (n=ToTo@209.8.233.137) |
| 15:51.12 | gilli | is watching |
| 15:52.40 | tzafrir_laptop | gilli, http://gilli.pastebin.com/m345804ec |
| 15:54.30 | gilli | thank you so much. pasting it to the file.. |
| 15:54.51 | *** join/#asterisk jpastore (n=jpastore@crlspr-24.233.166.216.myacc.net) |
| 15:59.28 | gilli | tzafrir_laptop: hmm...after including the new zapata-channels.conf to /etc/asterisk/zapata.conf I started asterisk again. |
| 15:59.42 | gilli | but it doesn't provide the zap commands anymore. |
| 15:59.53 | gilli | so I tried to reload chan_zap.so |
| 16:00.10 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-f39c0756dd8b0d80) |
| 16:00.15 | tzafrir_laptop | try: module unload chan_zap.so |
| 16:00.21 | tzafrir_laptop | module load chan_zap.so |
| 16:00.30 | gilli | i did exactly that. |
| 16:00.43 | gilli | first i unloaded and the i loaded manually. |
| 16:01.05 | gilli | but on loading it reports: chan_zap.c:12221 build_channels: Unable to register channel '1-15,16-31 |
| 16:02.00 | gilli | and still it doesn't provide the zap commands. |
| 16:02.24 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:02.30 | tzafrir_laptop | silly me. make that: |
| 16:02.42 | tzafrir_laptop | channel => 1-15,17-31 |
| 16:02.50 | tzafrir_laptop | got used to genzaptelconf |
| 16:03.22 | gilli | replacing ..thanks a lot :) |
| 16:06.10 | gilli | tzafrir_laptop: hmm.. now I do have access to the zap commands again but asterisk keeps floading warnings again. |
| 16:06.26 | tzafrir_laptop | What warnings? |
| 16:06.45 | gilli | chan_zap.c:10523 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 16:07.10 | gilli | chan_zap.c:2510 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 16:08.01 | gilli | these lines are being repeated.. |
| 16:08.07 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:08.30 | angryuser | i got a fring client, can someone tell me how is he able to register to port 50653 from outside? only 5060 and 10000-20000 are routed |
| 16:08.55 | ManxPower | I doubt you can |
| 16:09.05 | angryuser | i saw it in cli |
| 16:09.11 | ManxPower | gilli: your carrier has a loopback on the line |
| 16:09.21 | Qwell | angryuser: That's likely the source port. |
| 16:09.40 | ManxPower | angryuser: no, you saw the SOURCE port of the registration, not the DESTINATION. Learn some networking dide. |
| 16:09.42 | ManxPower | dude. |
| 16:10.23 | gilli | ManxPower ..I guess that's right because I connected one port (TE) from the double E1 card to another one (NT) on the same card. Bad idea? |
| 16:10.23 | angryuser | well show me where it is written in docs of * |
| 16:10.47 | ManxPower | angryuser: It's BASIC NETWORKING, nothing to do with Asterisk or VoIP. |
| 16:11.07 | ManxPower | gilli: what made you think it was a good idea. |
| 16:11.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:11.33 | tzafrir_laptop | gilli, you seem to have signalling = pri_cpe on both ports . You should have signalling = pri_net on the NT one |
| 16:11.41 | angryuser | ManxPower whatever |
| 16:12.00 | ManxPower | angryuser: not knowing networking will make it virtually impossible for you to manage Asterisk |
| 16:12.06 | *** join/#asterisk jpcansa (i=jpcansa@201.196.59.84) |
| 16:12.24 | gilli | tzafrir_laptop: is this something I can set via software or rather by setting jumpers? |
| 16:13.27 | tzafrir_laptop | gilli, I don't know that card. The BRI cards of Junghanns have drivers that report if they are TE or NT in /proc |
| 16:13.34 | angryuser | ManxPower i am just saying some info are confusing, like some people think 'sip show peers' show them the registry status |
| 16:13.41 | *** join/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 16:13.47 | tzafrir_laptop | and there the driiver must know |
| 16:13.50 | mrnick | hi |
| 16:13.54 | ManxPower | angryuser: that is a legit asterisk doc issue. |
| 16:13.55 | angryuser | of remote trunk |
| 16:14.39 | Qwell | sip show peer does show registration... |
| 16:15.05 | angryuser | Qwell i mean for remote trunk |
| 16:15.18 | Qwell | it shows whether something is registered to you |
| 16:15.38 | tzafrir_laptop | gilli, any anyway, that error you got was not because of wrong jumper. It was because of wrong definitions in zapata.conf |
| 16:16.03 | gilli | so is there a way I can fix it? |
| 16:16.12 | ManxPower | Qwell: Any chance of having the source port stuff removed from the default "sip show" stuff? |
| 16:16.22 | ManxPower | all it does is cause confusion |
| 16:16.49 | Qwell | it's useful for people |
| 16:17.19 | angryuser | ManxPower no just some line in doc's would be nice |
| 16:17.32 | ManxPower | Qwell: then leave it in the "show consise" or something like that. |
| 16:17.40 | ManxPower | angryuser: it's not a doc issue. |
| 16:18.02 | ManxPower | do a "netstat -an" on your web server sometime, see all the non-port 80 connections. |
| 16:18.20 | ManxPower | same for every single server that supports TCP/IP ON THE PLANET. |
| 16:18.55 | angryuser | ManxPower sure it is, jus a little comment like 'client's source port' will wipe it all |
| 16:19.17 | ManxPower | angryuser: but if you don't know networking you would now know what that means |
| 16:20.34 | angryuser | ManxPower it is better then just 'port' |
| 16:21.13 | ManxPower | anything is better than just "port". I've complained about it multiple times before. |
| 16:21.28 | tzafrir_laptop | gilli, egrep 'signalling|chanel' /etc/asterisk/zapata.conf |
| 16:21.45 | tzafrir_laptop | and again, read what I wrote above regarding signalling |
| 16:22.33 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 16:22.37 | ManxPower | angryuser: you are like the 400th user to be confused about the output. |
| 16:22.49 | ManxPower | Usually it's the poor sods with SIP NAT problems |
| 16:22.56 | angryuser | ManxPower that's coz nobode listen to you ;) |
| 16:23.01 | angryuser | nobody* |
| 16:23.12 | outtolunc | huh what? |
| 16:23.28 | ManxPower | angryuser: I don't help people that don't listen to me. I tell them "Best of luck with that" or "I cannot help you further". |
| 16:23.36 | ManxPower | TK just beats them up until they start listening. 8-) |
| 16:23.51 | *** join/#asterisk dmz (n=dmz@64.203.203.232) |
| 16:24.00 | angryuser | ManxPower what that tk means anyway ? |
| 16:24.14 | ManxPower | TK means [TK]D-Fender |
| 16:24.23 | angryuser | trying to guess for days now |
| 16:24.33 | angryuser | yes i know but tk ? |
| 16:24.45 | ManxPower | [TK]D-Fender: What does TK stand for? |
| 16:24.56 | angryuser | he wont tell me |
| 16:25.26 | [TK]D-Fender | ManxPower: TK doesn't stand for any BS. The world comes to him as he sits! |
| 16:25.35 | [TK]D-Fender | </chucknorriscode> |
| 16:25.36 | ManxPower | Don't worry, [TK]D-Fender. If you tell me in private, I won't tell angryuser |
| 16:25.55 | gilli | tzafrir_laptop: ah! thanks for helping me understand. so, can I split the configuration for the signaling into two parts in zapata.conf ? |
| 16:26.02 | angryuser | mahahaha |
| 16:26.42 | tzafrir_laptop | reminds gilli of http://gilli.pastebin.com/m345804ec |
| 16:26.44 | LemensTS | too kool |
| 16:27.14 | Qwell | ManxPower: |
| 16:27.17 | Qwell | ~wglwat |
| 16:27.18 | jbot | extra, extra, read all about it, wglwat is well, good luck with all that |
| 16:28.24 | gilli | is too noob to know |
| 16:29.08 | ManxPower | gilli: I'm not helping you because you are using what I refer to as an "exotic card". i.e. BRI or non-Digium/Sangoma card. |
| 16:29.29 | pbrown985 | Sangoma is pimp. |
| 16:30.30 | *** join/#asterisk joel_oliveira (n=asdas@estrela-adm.nortenet.pt) |
| 16:30.35 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 16:30.41 | tzafrir_laptop | ManxPower, I'm helping him because he's using an exotic card :-) |
| 16:31.00 | gilli | tzafrir_laptop: thanks a lot! :) |
| 16:31.04 | tzafrir_laptop | jbot must know what TK stands for |
| 16:31.06 | tzafrir_laptop | ~tk |
| 16:31.07 | jbot | ACTION snipes $herlo with a straw and rolled up piece of paper |
| 16:31.21 | tzafrir_laptop | jbot? |
| 16:31.55 | joel_oliveira | hi all |
| 16:32.07 | joel_oliveira | i am having a problem with the queue_log file |
| 16:32.32 | joel_oliveira | the problem is that when a call is completed by the caller (COMPLETECALLER flag) i get an id channel for that operation |
| 16:32.58 | joel_oliveira | but when the call is completed by the agent (COMPLETEAGENT) I dont't get the id channel |
| 16:32.59 | joel_oliveira | :\ |
| 16:33.44 | joel_oliveira | Agent -> SIP/4365|COMPLETEAGENT|35|91 |
| 16:33.58 | joel_oliveira | Caller -> |SIP/4364-082997c8|COMPLETECALLER|31|223 |
| 16:34.36 | *** join/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 16:34.38 | joel_oliveira | is there any problem with a certain version of asterisk for this manner? |
| 16:34.58 | EricL | Can someone send me a link for an FXO card that I can put in a Dell Poweredge 2950? |
| 16:35.59 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:36.46 | [TK]D-Fender | EricL: www.sangoma.com |
| 16:38.38 | *** join/#asterisk zamba (i=marius@sveigde.hih.no) |
| 16:38.50 | zamba | how can i establish a trunk between two asterisks? |
| 16:39.06 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 16:39.15 | angryuser | zamba you can you sip or iax2 |
| 16:40.16 | zamba | i know the register option is used on one of the sides |
| 16:40.22 | n3hxs | which do you think is better? sip or iax2? |
| 16:40.32 | zamba | but what do i have to do on the other side? |
| 16:40.43 | zamba | if i had to choose, sip? |
| 16:41.29 | gilli | tzafrir_laptop: even after setting the signalling = pri_net for both groups asterisk still reported: We think we're the CPE, but they think they're the CPE too. |
| 16:41.38 | angryuser | have no idea, but look it this way, you need to configure 1 sip/iax friend on both servers and register them |
| 16:42.05 | gilli | should I consult Junghanns maybe? After all I didn't even know that this channel is dedicated to Digium-cards only.... |
| 16:42.36 | [TK]D-Fender | zamba: go read "asterisk dual servers" on the WIKI |
| 16:42.38 | [TK]D-Fender | ~wikis |
| 16:42.39 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:42.53 | [TK]D-Fender | zamba: And passing calsl from one to the other is no different that doing so with an ITSP, etc |
| 16:43.05 | tzafrir_laptop | gilli, "signalling" is not applied on reload . You may need to restart asterisk |
| 16:43.05 | angryuser | zamba it's like the provider trunk configuration, but on this time 2 servers are used both as client and servers |
| 16:43.06 | zamba | [TK]D-Fender: ok, thanks |
| 16:43.53 | gilli | tzafrir_laptop: I did restart asterisk. I exited with 'stop now', started asterisk and connected to it with -r . Wrong way? |
| 16:44.43 | tzafrir_laptop | no |
| 16:44.50 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 16:44.57 | tzafrir_laptop | look at the spans with: pri show spans |
| 16:45.01 | *** part/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 16:45.44 | gilli | tzafrir_laptop: reports that both spans are up and active. |
| 16:45.50 | *** part/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 16:46.21 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 16:47.06 | pabelanger | afternoon all, anybody know of a UI for res_config database? |
| 16:48.13 | Qwell | pabelanger: huh? |
| 16:50.04 | pabelanger | I was looking to see if somebody already created an UI (perl, php, etc) for the res_config database? I wanted to get a better understanding of the database structure. |
| 16:50.52 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 16:51.16 | keith4 | how does mass-deployment of linksys SPA940 series compare with polycom SIP? |
| 16:51.27 | Qwell | keith4: not much can touch Polycom |
| 16:51.48 | keith4 | i don't mind hoop-jumping and being aggravated, as long as the end result is usable |
| 16:51.57 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 16:52.02 | _Raptor_ | hey guys, despite this issue should be resolved (http://bugs.digium.com/view.php?id=13088) i still get this output when compiling zaptel (svn checkout zaptel-1.2) on 2.6.26: http://pbot.rmdir.de/420309d7e19419e43f9226b6615d7bac |
| 16:52.21 | keith4 | but, i'm a bit worried by the fact that i can't find much documentation on mass-deploying linksys sip phones |
| 16:52.25 | [TK]D-Fender | keith4 : Linksys will be a cost-effective choice for you |
| 16:52.38 | [TK]D-Fender | keith4 : Your Google-fu is WEAK |
| 16:53.07 | keith4 | other than the bit in the wiki |
| 16:53.39 | [TK]D-Fender | _Raptor_: And that fix was for 1.4, not 1.2 |
| 16:53.44 | keith4 | well, the SPA941 is price-comparable to the SIP320 or 330, so I don't think cost-effectiveness comes into play much |
| 16:53.52 | *** join/#asterisk dwclarkNU (n=dwclarkN@h-74-0-49-242.cmbrmaor.covad.net) |
| 16:53.52 | [TK]D-Fender | _Raptor_: 1.2 is NOT supported except for security bugs. |
| 16:54.21 | [TK]D-Fender | keith4 : guess you found a better retailer where you are then.... |
| 16:54.36 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-1316feed326f038e) |
| 16:54.41 | Dovid | TK: talking about 1.2 and 1.4.X I finally moved all my boxes over to 1.4.X |
| 16:54.42 | [TK]D-Fender | keith4 : last I checked, Polycom came at a hefty premium in the UK |
| 16:55.05 | [TK]D-Fender | Dovid: "Procrastination : The art of keeping up with yesterday" |
| 16:55.11 | Dovid | llol |
| 16:55.25 | dwclarkNU | hello, i am looking to use asterisk for our centrlized call center design using all SIP, we will have a maximum of 450 concurrant SIP connections from agents and will use sip trunks back to our dialer...i am looking for hardware reccomendations to handle this volume |
| 16:55.57 | _Raptor_ | [TK]D-Fender: 1.2 is not supported any more? |
| 16:56.10 | plik | I'd procrastinate more if I wasn't so lethargic |
| 16:56.16 | Dovid | dwclarkNU: Not that you have to but I would split it over 2 boxes |
| 16:56.20 | [TK]D-Fender | _Raptor_: hasn't been for a LONG time now. |
| 16:56.39 | dwclarkNU | AMD vs Intel, slower quad core or fasater dual core, etc... |
| 16:56.44 | keith4 | [TK]D-Fender: well, on telephonydepot... SPA942 is $116, SIP330 is $108 |
| 16:57.04 | Dovid | _Raptor_: Ever since 1.6.X Beta came out 1.2.x has been in "security fix mode" only |
| 16:57.10 | angryuser | Dovid using openser with failover wouls be better maybe.. |
| 16:57.50 | Dovid | angryuser: Although I do work with OpenSER + Asterisk I never set it up from A-Z so I don't talk about what I don't know to well |
| 16:57.56 | [TK]D-Fender | keith4 : Where are you located again? I was still thinking UK... |
| 16:58.11 | keith4 | US |
| 16:58.25 | keith4 | unfortunately, in the same state as telephonydepot, so I have to pay tax |
| 16:58.38 | _Raptor_ | Dovid: ok i see thanks |
| 16:58.39 | [TK]D-Fender | keith4 : Ok, scracth that, fuck Linksys, take Polycom and don't look back :p |
| 16:58.48 | Dovid | keith4: They still seem to be chepar. I orderd from them a few times and I was happy |
| 16:58.55 | Dovid | TK: Amen to that |
| 16:59.06 | keith4 | [TK]D-Fender: heh... ok. |
| 16:59.07 | dwclarkNU | ok, so if i were to break the single box up into two boxes, would i be better with fast (3.0ghz) dual core or slower quad core (2.5ghz) |
| 16:59.21 | angryuser | Dovid me either, but it looks like openser manage well big volume of clients, and failover is a nice feature when you have 450 people not yelling at you |
| 16:59.25 | dwclarkNU | TK: I agree, Polycom is the way to go |
| 16:59.39 | Dovid | angryuser: I agree with that. it saved my ass a few times |
| 16:59.51 | Dovid | OpenSER + Asterisk + Heartbeat |
| 17:00.23 | angryuser | yes heartbeat is a way to go |
| 17:00.34 | Dovid | dwclarkNU: been out of the CPU game for a while. as a guess I would say quad 2.5 |
| 17:00.46 | gilli | well..i have to leave the office for now... tzafrir_laptop: thanks a big bunch for your help so far. I had read about you and xorcom so I appreciate the time you offered even more. good day everyone. |
| 17:01.33 | dwclarkNU | i thought the same thing, but somone mentioned on a forum that asterisk is about IO rather than multiple computations and the CPUs share IO |
| 17:02.12 | angryuser | but i saw a youtube on asterisktag 2008 , * does not really profit of miltucore, one guy said, it that true ? |
| 17:02.23 | Dovid | like I said I dont know much in the "CPU World". maybe TK can help |
| 17:02.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.59 | Coder` | is away (^C0,11b^C0,4y^C0,7e ^C0,10A^C0,6L^C0,2L) |
| 17:03.40 | dwclarkNU | i just dont want to go ask teh board of driector to spend 1/2 a million on a MPLS WAN, new dialer and then have the box conneccting everything together to not work...i'd kinda look like a schmuck |
| 17:03.49 | Dovid | lol |
| 17:04.23 | Dovid | http://www.google.com/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&hs=NbB&q=dual+core+vs+quad+core%2C+asterisk&btnG=Search |
| 17:04.39 | Dovid | dwclarkNU: Google is your friend |
| 17:05.36 | *** join/#asterisk pbrown985 (n=na@wh-gtw-0001.woolfharris.com) |
| 17:05.47 | dwclarkNU | Dovid: i've read all that, but there was no one conclusive thread, which is why i am in here now |
| 17:06.07 | Dovid | dwclarkNU: Don't know the answer |
| 17:06.46 | *** join/#asterisk exothermic (n=miles@74.85.89.236) |
| 17:07.02 | dwclarkNU | hopefully somone else in the channel can shed some light on hardware reccomendations for an all sip setup |
| 17:07.05 | exothermic | I have the first digit getting stripped off my call every time I try to make a call |
| 17:07.08 | *** join/#asterisk Gnutoo (n=gnutoo@host6-25-dynamic.25-79-r.retail.telecomitalia.it) |
| 17:07.12 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 17:07.52 | Gnutoo | hello, is there a howto for making conferences rooms? all my search returns asterix conferences(real life conferences...) |
| 17:07.52 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 17:08.03 | Dovid | exothermic: do you want it ? do you not want it ? problem ? |
| 17:08.04 | exothermic | it comes into the output saying Executing [ext@context:1] where context is the context specified on the peer |
| 17:08.08 | [TK]D-Fender | Gnutoo: "core show application meetme" |
| 17:08.16 | Gnutoo | [TK]D-Fender, thanks |
| 17:08.19 | exothermic | Dovid: I don't want it. |
| 17:08.21 | [TK]D-Fender | Gnutoo: Go lookup "meetme" on the WIKI |
| 17:08.46 | Dovid | exothermic most likely in ur dial plan. please psot your extensions.conf |
| 17:08.47 | *** join/#asterisk unaffiliate (n=un@unaffiliated/unafilliate) |
| 17:08.48 | kensuke_ | qustion, i can send the "iax2 debug" to a text file? |
| 17:09.13 | [TK]D-Fender | exothermic: 'ext" is what you dialed, so thats what you get. |
| 17:09.20 | angryuser | dwclarkNU it depends on various parameters, codecs, transoding, conferences, maybe read this ? http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning |
| 17:09.28 | [TK]D-Fender | exothermic: what you DO with what was dialed is your job in the dialplan. |
| 17:09.40 | exothermic | [TK]D-Fender: No I'm actually getting the first digit stripped of ext |
| 17:09.59 | Gnutoo | [TK]D-Fender, that's what i did after you answered...thanks a lot |
| 17:10.05 | [TK]D-Fender | exothermic: Pastebin a failed call. If its from a SIP device, then enable SIP debug. |
| 17:10.07 | [TK]D-Fender | ~pb |
| 17:10.08 | 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 |
| 17:10.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:10.25 | exothermic | [TK]D-Fender: Executing [5256522@ebi_devices:1] Pickup("SIP/8702-b7d102b8", "256522") in new stack |
| 17:10.41 | dwclarkNU | thanks |
| 17:10.41 | Dovid | exothermic: Most likely dialplan error |
| 17:10.48 | Dovid | post it on PB !! |
| 17:10.52 | [TK]D-Fender | exothermic: it IS a dialplan error |
| 17:10.53 | exothermic | ok |
| 17:11.03 | [TK]D-Fender | exothermic: YOU are stipping the 1st digit off |
| 17:11.13 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 17:11.16 | exothermic | [TK]D-Fender: Ya I can see that, but I can't figure out where. |
| 17:11.25 | [TK]D-Fender | exothermic: paste the LINE from your extensions.conf |
| 17:11.35 | [TK]D-Fender | (just pastebi,m its 1 line) |
| 17:11.43 | [TK]D-Fender | jsut paste* |
| 17:12.41 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 17:14.08 | exothermic | [TK]D-Fender: http://www.pastebin.ca/1085389 |
| 17:14.46 | exothermic | [TK]D-Fender: The devices in this case have context=ebi_devices for the sip peers |
| 17:15.25 | exothermic | I looked for all instances of ":1" which would strip off the first digit, but can't find any that shouldn't be there. |
| 17:15.36 | exothermic | for that previous example I was sending a 7 digit number. |
| 17:15.48 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 17:15.58 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 17:16.13 | [TK]D-Fender | exothermic: "dialplan show" <- |
| 17:16.37 | DSpair | Hey gang, I have an emergency here. I had to move asterisk to a new server and the zaptel drive will not load anymore?!?!?! |
| 17:16.44 | [TK]D-Fender | exothermic: You show "Executing [5256522@ebi_devices:1] Pickup("SIP/8702-b7d102b8", "256522")" |
| 17:17.03 | [TK]D-Fender | exothermic: there is NO PICKUP anywhere in the pastebin you just provided. |
| 17:17.04 | exothermic | [TK]D-Fender: yes |
| 17:17.09 | DSpair | I get a message about zaptel configuration |
| 17:17.17 | [TK]D-Fender | exothermic: You are showing me apples & oranges. |
| 17:17.26 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 17:17.31 | Dovid | DSpair: What was the error ? building it ? loading it ? |
| 17:17.43 | exothermic | [TK]D-Fender: hmm looks like something out of functional.conf or features.conf is being called then, thanks for the vague hint. |
| 17:18.03 | [TK]D-Fender | exothermic: Oh no... its far from vague. |
| 17:18.11 | outtolunc | did scottie add the right amount of plasma to the zaptel generators prior to attempting to deploy it <G> |
| 17:18.34 | outtolunc | says sorry, weird mood today <G> |
| 17:18.59 | DSpair | Dovid, It says that zaptel support is not compiled in, but I know that it is. |
| 17:19.05 | [TK]D-Fender | outtolunc: I'm givin' er all I cahn captin! |
| 17:19.19 | DSpair | Does moving to a new kernel/recompiling zaptel require a recompile of asterisk? |
| 17:19.35 | [TK]D-Fender | exothermic: Go do a complete call with SIP DEBUG this time. |
| 17:19.41 | [TK]D-Fender | DSpair: No, only zaptel. |
| 17:20.04 | DSpair | [TK]D-Fender, That's what I thought... Weird, and my boss is breathing down my neck to get some phones up... |
| 17:20.21 | Dovid | DSpair: What error do u get ? |
| 17:20.24 | DSpair | The zap driver loaded and the entries in /dev/zap/ are there. |
| 17:20.26 | exothermic | [TK]D-Fender: All you needed to say was "Go find where in your dial plan the "pickup" function is being called, and you should see your issue" Anyway, it is working now thanks |
| 17:20.29 | Dovid | oops. just saw that |
| 17:20.47 | exothermic | [TK]D-Fender: Just got a little over zealous with my pattern matching. |
| 17:21.14 | Dovid | did u do a kernel update and then reboot ? just a hunch |
| 17:21.25 | DSpair | asterisk.c:2966 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options: |
| 17:21.52 | DSpair | Dovid, I did a kernel update, rebooted. Realized I needed to recompile zaptel and did so, rebooted again. |
| 17:21.58 | Dovid | DSPair: make sure u have the right versions and try recompling with make menuselect only using what u need |
| 17:22.13 | Dovid | recomplie after the reboot |
| 17:22.39 | Dovid | and make sure u have the kernel sources and or headers for the new kernel version |
| 17:23.02 | DSpair | Dovid, No such luck... I have recompiled asterisk and no joy. |
| 17:23.05 | [TK]D-Fender | exothermic: :) |
| 17:23.23 | Dovid | DSpair: u recomplied zaptel after reboot ? |
| 17:23.33 | Dovid | try to rmmod and then mopdprobe zaptel |
| 17:23.37 | Dovid | and then start asterisk |
| 17:24.01 | Dovid | kernel update + reboot = zaptel rebuild |
| 17:24.31 | DSpair | Dovid, I have done that... removing the module makes no difference either. |
| 17:24.57 | Dovid | and rebuilt zaptel as well after the reboot ? also latest versions of asterisk and zaptel ? |
| 17:25.14 | Dovid | only other thing i can think of is a bug |
| 17:25.15 | tzafrir_laptop | kernel update, zaptel rebuild reboot -> less downtime |
| 17:25.54 | DSpair | Dovid, No, not the latest versions, but ones that worked before the hardware fault this morning. |
| 17:26.06 | DSpair | Asterisk=1.4.20.1 Zaptel=1.4.9.2 |
| 17:26.19 | Dovid | Tzafrir: Is the best when it comes to zaptel issues |
| 17:26.55 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:26.58 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 17:28.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:28.15 | Dovid | Pinging Tzafrir |
| 17:28.20 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 17:28.52 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:29.24 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 17:29.46 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:29.59 | M1s3ry | ping: unknown Tzafrir |
| 17:30.03 | M1s3ry | :/ |
| 17:30.14 | M1s3ry | ping: unknown host Tzafrir* |
| 17:31.06 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 17:31.23 | Qwell | M1s3ry: works here, using 4.2.2.4 |
| 17:32.47 | M1s3ry | :/ |
| 17:37.08 | tzafrir_laptop | DSpair, what's the output of: cat /proc/zaptel/* |
| 17:39.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:39.44 | *** join/#asterisk rethus (n=rethus@xdsl-84-44-235-86.netcologne.de) |
| 17:40.10 | rethus | is this the right irc for questions with asterisk & php (phpagi)? |
| 17:40.51 | [TK]D-Fender | rethus: Pretty mch |
| 17:40.55 | [TK]D-Fender | much* |
| 17:41.54 | DSpair | tzafrir_laptop, There are 4 items under that directory. Catting each in turn returns a list of channels on the spans. |
| 17:42.25 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 17:42.52 | tzafrir_laptop | DSpair, yes, please pastebin that |
| 17:43.31 | tzafrir_laptop | DSpair, see also http://docs.tzafrir.org.il/#_procfs_interface_proc_zaptel |
| 17:43.44 | DSpair | tzafrir_laptop, http://pastebin.com/m13fd28bf |
| 17:44.20 | DSpair | tzafrir_laptop, Although I have not modified it, when I run |
| 17:44.22 | *** part/#asterisk exothermic (n=miles@74.85.89.236) |
| 17:44.22 | tzafrir_laptop | either you didn't run ztcfg, or you have an invalid configuration |
| 17:44.32 | tzafrir_laptop | What do you have in /etc/zaptel.conf ? |
| 17:44.35 | DSpair | tzafrir_laptop, Although I have not modified it, when I run 'ztcfg' it says that there is an error on span1 |
| 17:44.44 | tzafrir_laptop | What happens when you run ztcfg #? |
| 17:46.09 | DSpair | tzafrir_laptop, zaptel.conf = http://pastebin.com/m6dd9dd8e |
| 17:46.14 | Gnutoo | my meetme config doesn't work...why? http://pastebin.com/m6c7fddcb |
| 17:46.25 | DSpair | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 17:46.31 | LemensTS | Anyone know of a china or mexico itsp? |
| 17:46.45 | Qwell | isn't voip "illegal" in Mexico? |
| 17:48.01 | atis_work | hey, does anybody knows on H323/T38 passtrough? |
| 17:48.10 | atis_work | is it working on some version? |
| 17:48.28 | *** join/#asterisk Xaviertoor (n=Xavierto@200-146-243-009.xf-static.ctbcnetsuper.com.br) |
| 17:48.33 | Gnutoo | and i bet the video won't work in meetme... |
| 17:48.46 | ManxPower | atis_work: H323 and T38a re different protocols |
| 17:48.57 | *** part/#asterisk rethus (n=rethus@xdsl-84-44-235-86.netcologne.de) |
| 17:49.01 | atis_work | well, i thought T.38 is a codec |
| 17:49.14 | ManxPower | no, it's not. It's a protocol |
| 17:49.17 | atis_work | i need to connect t38modem to asterisk somehow |
| 17:49.26 | atis_work | and it seems to be working with h323 |
| 17:49.33 | ManxPower | atis_work: you were asking about passhtru, not terminationh |
| 17:49.37 | atis_work | yes |
| 17:49.58 | ManxPower | so you would need TWO T.38 endpoints if you want passthru. |
| 17:50.04 | [TK]D-Fender | Gnutoo: you need to have compiled and configured Zaptel before you have compiled * |
| 17:50.06 | tzafrir_laptop | DSpair, you have E1 spans, but attempt to configure them as T1 |
| 17:50.22 | DSpair | tzafrir_laptop, I do not have E1 spans. |
| 17:50.22 | ManxPower | I've been told 1.4.x supports T.38 passthru, but I've never tested it. |
| 17:50.23 | atis_work | ManxPower: well, one is t38modem, and another is voip provider |
| 17:50.40 | atis_work | well, yes i'm hoping for that |
| 17:50.43 | tzafrir_laptop | DSpair, it is configured as such |
| 17:50.45 | *** join/#asterisk bl4q (i=Bl@dslb-088-066-228-234.pools.arcor-ip.net) |
| 17:50.46 | DSpair | tzafrir_laptop, I have 2 T1s and a Rhino FXO channel bankl |
| 17:50.48 | DSpair | tzafrir_laptop, I have 2 T1s and a Rhino FXO channel bank |
| 17:50.58 | DSpair | tzafrir_laptop, Where do you see that? |
| 17:51.01 | file | it supports it on SIP, but nobody has ventured into adding support in the H323 stuff |
| 17:51.13 | Qwell | atis_work: patches welcome :D |
| 17:51.14 | tzafrir_laptop | DSpair, what card do you use? configure it to be T1 |
| 17:51.29 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 17:51.33 | DSpair | tzafrir_laptop, How do I acomplish that? I have the digium 4 port T1 card. |
| 17:51.43 | Qwell | DSpair: check the dipswitch on the card |
| 17:51.43 | atis_work | huh, damn.. now i'll have to compile t38modem with OPAL, witch supports SIP.. |
| 17:51.56 | tzafrir_laptop | or t1e1override? |
| 17:51.59 | atis_work | however only some old version of OPAL works with T38modem |
| 17:52.01 | DSpair | tzafrir_laptop, d161:0410 Wildcard TE410P (3rd Gen) |
| 17:52.16 | Gnutoo | [TK]D-Fender, ah Zapatel isn't for the hardware cards? i only want a software system(i don't need to be able to call real telephone number or to get calls from them) |
| 17:52.18 | atis_work | thanks, i'll go compiling.. |
| 17:52.20 | Qwell | tzafrir_laptop: to be honest, I don't know how that option works.. if the switch is set, would that override back to T1? |
| 17:52.22 | DSpair | tzafrir_laptop, I'm not familiar with that option.... |
| 17:52.29 | *** join/#asterisk NTJOCK (n=brian@adsl-76-237-16-169.dsl.hstntx.sbcglobal.net) |
| 17:52.48 | tzafrir_laptop | Qwell, works for the single card we have here :-) |
| 17:52.59 | NTJOCK | good morning all |
| 17:53.07 | [TK]D-Fender | Gnutoo: Zaptel is an interface layer. It provides SOFWTWARE timing when you don't have hardware cards (ztdummy). Zaptel is REQUIRED to build MeetMe |
| 17:53.12 | outtolunc | DSpair: you sure you do not have the jumpers 'on' on that card? (putting it in E1 mode) |
| 17:53.14 | NTJOCK | anyone else had trouble with Teliax? moving from their old platform to the new? |
| 17:53.36 | DSpair | outtolunc, It worked this morning and I have not changed anything aside from the server it is in. |
| 17:53.51 | Gnutoo | [TK]D-Fender, ok thanks i'll rebuild asterisk with the zapatel USE flag...but do i need a kernel driver? |
| 17:54.10 | Gnutoo | s/Zapate/Zaptel |
| 17:54.15 | outtolunc | weird, because the http://pastebin.com/m13fd28bf output has 1-13 channels per |
| 17:54.16 | [TK]D-Fender | Gnutoo: Yes, Zaptel prodides Kernel modules |
| 17:54.21 | outtolunc | er 1-31 |
| 17:54.38 | DSpair | outtolunc, So how do I set the t1e1overrride? |
| 17:55.09 | ManxPower | T-1. B-Channels = 1-23, D-Channel = 24 and on E-1, B-channels = 1-15,17-30(or is it 31), D-Channel on 16. |
| 17:55.29 | DSpair | Please, I hate to be a pain, but my entire company is without phones right now. |
| 17:55.57 | outtolunc | DSpair: modprobe wct4xxp t1e1override=-1 (iirc) |
| 17:56.12 | ManxPower | DSpair: How did you get into this situation? |
| 17:56.39 | Gnutoo | [TK]D-Fender, ok thanks i'll look at it |
| 17:56.55 | NTJOCK | ManxPower: hey, what do you know about IAX inbound authentication? I'm getting the run around from Teliax. |
| 17:57.06 | ManxPower | NTJOCK: Best of luck with that. |
| 17:57.29 | DSpair | THANK YOU TAHNK YOU THANK YOU THANK YOU!!!! |
| 17:57.33 | NTJOCK | isn't the connect string supposed to be 7135551212@username |
| 17:57.56 | ManxPower | DSpair: now figure out why you got into this out of service situation so next time it won't happen |
| 17:58.22 | ManxPower | Dial(SIP/destnum@sipconfpeer) and Dial(IAX2/iaxconfpeer/destnum) |
| 17:58.41 | NTJOCK | ManxPower: right that is for outbound... which works. and we are registered |
| 17:58.45 | NTJOCK | it's the inbound that doesn't work. |
| 17:59.02 | ManxPower | As I SAID, best of luck with that. I cannot help you. |
| 17:59.04 | DSpair | ManxPower, The PCI bus on my IBM xSeries server dies. |
| 17:59.10 | DSpair | s/dies/died/ |
| 17:59.24 | Gnutoo | [TK]D-Fender, and is there a way to do conferences with video? |
| 17:59.30 | DSpair | quit |
| 17:59.35 | [TK]D-Fender | Gnutoo: Not with * |
| 17:59.46 | NTJOCK | ManxPower: thanks... .it was working for months and months ..... until they asked me to move to their new platform. ugh! |
| 18:00.07 | Gnutoo | [TK]D-Fender, so is there is a software that is free(as in freedom) that does it? |
| 18:00.25 | Gnutoo | just in case i need it... |
| 18:00.38 | [TK]D-Fender | Gnutoo: Google-able. Get busy :) |
| 18:00.49 | Gnutoo | [TK]D-Fender, ok |
| 18:01.13 | *** join/#asterisk nicox (n=nicox@212-183-37-65.adsl.highway.telekom.at) |
| 18:11.36 | *** join/#asterisk mercera13 (n=thanksan@ip-118-90-39-33.xdsl.xnet.co.nz) |
| 18:11.56 | *** part/#asterisk mercera13 (n=thanksan@ip-118-90-39-33.xdsl.xnet.co.nz) |
| 18:14.12 | *** join/#asterisk gramulhaozin (n=charles@c-76-110-242-178.hsd1.fl.comcast.net) |
| 18:14.29 | gramulhaozin | hey |
| 18:14.46 | gramulhaozin | Any asterisk Distributor there / |
| 18:14.47 | gramulhaozin | ? |
| 18:15.41 | ManxPower | gramulhaozin: Digium has a list of htem |
| 18:16.53 | [TK]D-Fender | gramulhaozin: Depends where "there" is. |
| 18:18.10 | *** join/#asterisk exothermic (n=miles@74.85.89.236) |
| 18:19.30 | gramulhaozin | I'm in Florida |
| 18:20.07 | [TK]D-Fender | gramulhaozin: One of the better choices for retailer there would be www.telephonydepot.com |
| 18:20.41 | exothermic | [TK]D-Fender: from the console is there a way to manipulate the queue? ie move callers into another context/priority? |
| 18:20.54 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:21.16 | [TK]D-Fender | exothermic: "Redirect" |
| 18:21.27 | exothermic | [TK]D-Fender: thanks |
| 18:22.12 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 18:22.37 | exothermic | [TK]D-Fender: is subcommand of something else? |
| 18:24.01 | exothermic | [TK]D-Fender: err rather is that a subcommand of another command? |
| 18:24.19 | exothermic | [TK]D-Fender: because that doesn't seem to be an option for me on the console. |
| 18:24.58 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 18:25.27 | [TK]D-Fender | exothermic: AMI <- |
| 18:25.59 | exothermic | [TK]D-Fender: So nothing can be done from the console? |
| 18:26.25 | [TK]D-Fender | exothermic: Not that I'm aware of |
| 18:26.50 | exothermic | [TK]D-Fender: Is there a way to connect to the AMI like you would the console? |
| 18:27.37 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:28.10 | [TK]D-Fender | exothermic: Telnet |
| 18:28.37 | ManxPower | exothermic: You, as a human, are not equipped to manually use the AMI |
| 18:28.57 | ManxPower | Well, maybe if you have total perfect memory recall and never make a typoe. |
| 18:29.09 | jeev | Fender, i didn't get to impliment the phone system yet cause i'm leaving town on tuesday and they dont wanna deal with problems if i'm not here :/ |
| 18:29.18 | jeev | but i made a script to add phones and shit, it's c00l! |
| 18:29.33 | jeev | takes me 1 min to get a 330 up and running + its stupid long time it takes to boot it up |
| 18:29.35 | Katty | ohai |
| 18:29.44 | Katty | would anyone like some of my headache? |
| 18:29.54 | rob0 | please, mine went away |
| 18:29.55 | jeev | nop |
| 18:30.00 | Katty | *hee* |
| 18:31.39 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
| 18:33.39 | l0verb0y | anyone install zaptel with centos 5.2? |
| 18:34.19 | ManxPower | ~centos52bug |
| 18:34.20 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 18:36.10 | icel | hey, i am trying to convince my boss not to replace our * box with trixbox. Can you guys give me any reasons to help my argument? |
| 18:36.45 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 18:36.59 | _mm_ | whats his reasons for wanting to do so? |
| 18:37.42 | icel | i think he likes the web interface |
| 18:37.56 | icel | but they just use freepbx which can be used with plain old *, right? |
| 18:38.39 | *** join/#asterisk unbkbl (n=unbkbl@mx.humanitas.com.co) |
| 18:38.59 | BCS-Satori | Why is it when I loose all my external trunks, can I not reload asterisk or place internal system phone calls. i.e: if I loose connection to vonage (my only external trunk) local extension dialing and the ability to reload asterisk does not work, the system "halts" thats the best way i can describe it |
| 18:39.07 | Katty | icel: which web interface? |
| 18:39.18 | icel | Katty: I think freepbx |
| 18:39.26 | LemensTS | FreePBX is just a gui to asterisk basicly |
| 18:39.26 | Katty | icel: so the config bits? |
| 18:39.37 | gramulhaozin | [TK]D-Fender: does telephony depot gives discount if I'm reselling ? |
| 18:39.50 | icel | yeah, i can't see why trixbox would be better than * really |
| 18:40.06 | Katty | icel: well, clearly, you have more control over what * is doing if you're not limited to the webbits it gives you. |
| 18:40.11 | LemensTS | If the programmer doesnt know AEL then its probably a good idea |
| 18:40.36 | Katty | icel: troubleshooting is easier, because tribox loves to make macros out of everything. |
| 18:40.58 | icel | Yeah, all I know is that I tried to configure a T1 card in trixbox and couldn't figure it out. Then I broke it by restarting * and it wouldn't start again. |
| 18:41.16 | Katty | well there's a good reason for him |
| 18:41.19 | icel | took me about 3 minutes (with help from D-Fender) to get it working fine in regular * |
| 18:41.33 | Katty | good ole fender. |
| 18:41.39 | icel | yep |
| 18:41.54 | Katty | [TK]D-Fender: you get a cookie. |
| 18:41.59 | icel | heh |
| 18:42.09 | LemensTS | He takes paypal probably |
| 18:42.16 | Katty | but from a "i don't know what the crap i'm doing with a phone system" point of view.... |
| 18:42.27 | Katty | i can see the appeal of dropdowns and things |
| 18:42.39 | icel | Yeah, that is probably what appeals to him |
| 18:42.51 | Katty | is it for in house use? |
| 18:42.55 | icel | no |
| 18:42.58 | icel | err |
| 18:42.58 | icel | yeah |
| 18:43.01 | [TK]D-Fender | gramulhaozin: CALL THEM |
| 18:43.04 | Katty | so just your company? |
| 18:43.04 | icel | ~150 employees |
| 18:43.09 | icel | just ours |
| 18:43.11 | Katty | you don't sell it as a product/service, etc |
| 18:43.11 | [TK]D-Fender | Katty: Mew. |
| 18:43.11 | Katty | k |
| 18:43.16 | Katty | [TK]D-Fender: mew. |
| 18:43.21 | AlexTO | gramulhaozin.. maybe i can help you |
| 18:43.44 | Katty | icel: are you the only one there that knows how to make changes to it? |
| 18:44.00 | [TK]D-Fender | icel: Trixbox is fine.... so long as you never have to do anything more than it offers up front and like being told how to work. |
| 18:44.13 | icel | Katty: yes, but I am trying to change that. I even wrote a web page interface to add/remove users and stuff, they are just being braindead and not using it |
| 18:44.30 | Katty | icel: job security ;) |
| 18:44.34 | icel | lol |
| 18:44.39 | [TK]D-Fender | icel: Then tell them they'll only lose MORE control on something they'll never learn to use anyways |
| 18:44.44 | Katty | icel: i understand tho. the other IT guy is too scared to look at our linux box. |
| 18:44.47 | LemensTS | If you still want customizations isnt Asterisk GUI better for that |
| 18:44.58 | Katty | icel: much less change it. |
| 18:44.58 | AlexTO | I know a reseller in south florida who can gice you discounts |
| 18:44.58 | icel | D-Fender: thx, i will |
| 18:45.07 | Katty | i think conf files and emacs is the best thing for customizations |
| 18:45.11 | Katty | and fender. |
| 18:45.19 | Katty | he's pretty handy. |
| 18:45.23 | Katty | practically my pocket reference. |
| 18:45.39 | unbkbl | hello! i'm dealing with a big big problem right now i need urgent help!. i've a IBM x3200 server, Xeon de 1.8 dual core, 1G de RAM, running Asterisk 1.4.21.2 with two Digium cards. one with 2 PRI and another with 4. After the instalation of a new PRI line i'm getting errors like this: |
| 18:45.46 | Katty | [TK]D-Fender: what would i do without you? (= |
| 18:45.59 | unbkbl | [Jul 28 13:44:44] NOTICE[2797] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
| 18:45.59 | unbkbl | [Jul 28 13:44:48] NOTICE[2796] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 18:45.59 | unbkbl | [Jul 28 13:44:50] NOTICE[2796] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:46.03 | icel | Katty: Where does fender come from, and how does he know all that stuff? ;p |
| 18:46.28 | LemensTS | !pastebin |
| 18:46.41 | [TK]D-Fender | Katty: Dunno... what do you do WITH me? :) |
| 18:46.45 | icel | Hey thx guys for some ammo |
| 18:46.45 | tzanger | [TK]D-Fender is part of the Illuminati |
| 18:46.52 | unbkbl | the queues of my callcenter are restarted after this errors |
| 18:46.53 | icel | heh |
| 18:47.16 | Katty | icel: he's canadian. |
| 18:47.16 | unbkbl | [Jul 28 13:37:27] WARNING[2797] chan_zap.c: No D-channels available! Using Primary channel 47 as D-channel anyway! |
| 18:47.21 | unbkbl | any idea? |
| 18:47.23 | Katty | icel: canadian explains everything. |
| 18:47.28 | tzanger | unbkbl: fix your d channel |
| 18:47.34 | icel | Katty: i am beginning to understand now |
| 18:47.34 | tzanger | yes, us canadians are wicked cool |
| 18:47.42 | Katty | tzanger: <3 |
| 18:47.50 | Katty | tzanger: if this north american union thing goes through... |
| 18:47.52 | Katty | tzanger: i'm moving. |
| 18:47.59 | mshades | i think i'm falling for you |
| 18:48.00 | tzanger | eww, I don't want a NA union |
| 18:48.08 | [TK]D-Fender | tzanger: Yes.... cold & evil muaahahhaahaha *cough* |
| 18:48.11 | Katty | tzanger: none of us do. but i intend to make the best of it. |
| 18:48.18 | tzanger | haha |
| 18:48.37 | tzanger | unbkbl: don't PM please, make use of EVERYONE in here's ability to help |
| 18:48.44 | tzanger | unbkbl: you need to figure out why your D channel isn't happy |
| 18:48.44 | tzanger | now |
| 18:48.47 | Katty | although those fun isymphony people in new mexico are pretty cool too |
| 18:48.52 | tzanger | from what I see, your HDLC controller is unhappy |
| 18:49.03 | tzanger | that could be poor line conditions, or improper settings |
| 18:49.23 | hardwire | cat chewing on t1 cable? |
| 18:49.35 | Katty | probably katori shinto related |
| 18:49.35 | [TK]D-Fender | unbkbl: pastebin your zaptel.conf , zapata.conf , "ztcfg -vvvv" and "cat /proc/interrupts" |
| 18:49.37 | Katty | blames [TK]D-Fender |
| 18:50.01 | [TK]D-Fender | Katty: Not my fault I'm such an adept :) |
| 18:50.09 | AlexTO | Hi, How can i set dialplan to use one provider whom provide me multi-IPs ? any Ideas? |
| 18:50.38 | [TK]D-Fender | AlexTO: bind to ONE. |
| 18:50.41 | Katty | [TK]D-Fender: you are so 70s. |
| 18:50.44 | DSpair | THANK YOU THANK YOU THANK YOU THANK YOU!!!! |
| 18:50.50 | DSpair | Thank you all so much.... |
| 18:50.56 | DSpair | Everything is now completely fixed. |
| 18:51.09 | unbkbl | [TK]D-Fender: im sorry but i dont know how to use ur !pastebin |
| 18:51.15 | [TK]D-Fender | ~pb |
| 18:51.16 | 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 |
| 18:51.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 18:51.43 | unbkbl | ok |
| 18:51.47 | AlexTO | sorry, i didn't get it... can you explain to me how is that? |
| 18:52.54 | [TK]D-Fender | AlexTO: "bin=1.2.3.4" <- pick ONE IP to bind to. |
| 18:52.58 | [TK]D-Fender | bind* |
| 18:54.00 | AlexTO | Oki.. thanks.. |
| 18:58.46 | *** join/#asterisk Itiliti (n=Itiliti@75.150.198.1) |
| 18:59.40 | DSpair | uptime |
| 19:00.11 | MikeJ | downtime |
| 19:00.14 | jeev | FENDER |
| 19:00.42 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 19:00.48 | drfreeze | Hello |
| 19:01.26 | drfreeze | Anyone have a number that when dialed from * just rings and never answers. But if you dial it from (say your) cell phone, it picks up immediately |
| 19:02.01 | [TK]D-Fender | drfreeze: pastebin <- |
| 19:02.24 | drfreeze | [TK]D-Fender: what? |
| 19:02.26 | Itiliti | I have an 800 DID is not generating ringtone when a call comes in. but when calls come in over our normal did's they generate ringtone fine. any ideas? |
| 19:02.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:02.55 | drfreeze | It's a 877 number - we just add the prefix 1 if the user doesn't provide it |
| 19:03.28 | [TK]D-Fender | drfreeze: Show us eht call in detail. |
| 19:05.42 | unbkbl | [TK]D-Fender: http://pastebin.com/d25164791 |
| 19:06.28 | DSpair | tzafrir_laptop and ManxPower, can I PayPal you guys some money for your assistance?!?! |
| 19:06.31 | [TK]D-Fender | unbkbl: 169: 283286 5175221 IO-APIC-level ioc0, uhci_hcd:usb4, wct2xxp <- BAD. Get it on its own IRQ via your BIOS. |
| 19:06.59 | [TK]D-Fender | unbkbl: And go try restarting * & zaptel |
| 19:07.29 | unbkbl | and the error mesages in /var/log/asterisk/full are http://pastebin.com/d3bfbd03e |
| 19:07.37 | unbkbl | i'll try that |
| 19:09.35 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 19:09.52 | Dr-Linux|home | anybody is using A2billing with Asterisk? |
| 19:10.52 | unbkbl | but... [TK]D-Fender, we had the card in a diferent server with it's own IRQ and we had the same error |
| 19:11.31 | [TK]D-Fender | unbkbl: [Jul 28 13:45:20] NOTICE[2796] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 <- normally a sign of lost IRQ's |
| 19:13.05 | unbkbl | [TK]D-Fender: zttool isn't showing IRQ loses |
| 19:13.27 | [TK]D-Fender | unbkbl: Go get it it on its won. |
| 19:13.41 | [TK]D-Fender | Dr-Linux|home: Whats your question with it this time? |
| 19:13.44 | [TK]D-Fender | own* |
| 19:13.54 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 19:14.26 | Katty | [TK]D-Fender: we should abandon asterisk and go get some iced coffee. |
| 19:14.26 | *** join/#asterisk bildo (n=bildoto@bildo.tk) |
| 19:14.52 | [TK]D-Fender | Katty: Sounds good, but they frown on mid-day departures like that :) |
| 19:15.04 | unbkbl | [TK]D-Fender: i've put a hardhdlc and the problem seems to be solved but in a matter of minutes the quality of the voice get f*cked up |
| 19:15.17 | Katty | [TK]D-Fender: i would say it's an emergency. |
| 19:15.28 | *** join/#asterisk bildo (n=tobbe@bildo.tk) |
| 19:16.02 | Katty | [TK]D-Fender: threaten them with some aikido |
| 19:16.06 | Dr-Linux|home | [TK]D-Fender: I've some issue with RateCard, the user balance is $5 and system says that but when user dial desired number then system repplies "sorry but your credit is 0" |
| 19:16.08 | Katty | [TK]D-Fender: even if it is supposed to be self defense. |
| 19:16.29 | [TK]D-Fender | Katty: Katori Shinto is far more threatening :) |
| 19:16.48 | Katty | [TK]D-Fender: hey, whatever works. |
| 19:17.02 | [TK]D-Fender | Dr-Linux|home: They've got their own support resources. |
| 19:17.12 | justanotherpaul | Is there a way to force asterisk to "pass-through" for a specific codec? I'm having a problem where I want speex/16000 on both ends but asterisk insists on sending invites with speex/8000. |
| 19:17.18 | Katty | [TK]D-Fender: my boss is avoiding me today. did i mention that? |
| 19:17.36 | Dr-Linux|home | [TK]D-Fender: already tried alot on the web but no luck so far |
| 19:17.49 | [TK]D-Fender | Dr-Linux|home: if you expect any kind of random hints perhaps you'd at least do yourself (and us) the favor of PASTEBINNING All of the backup for it. |
| 19:18.18 | [TK]D-Fender | Katty: You mean the one who's weeks behind on that evaluation? :) |
| 19:18.24 | PakiPenguin | hey Dr-Linux|home |
| 19:18.24 | Katty | [TK]D-Fender: he did it friday. |
| 19:18.25 | PakiPenguin | :) |
| 19:18.35 | [TK]D-Fender | Katty: and? |
| 19:18.37 | Katty | [TK]D-Fender: and i informed him i was not Pleased |
| 19:18.40 | justanotherpaul | I can post wireshark captures of the SIP traffic if that helps clarify. |
| 19:19.05 | [TK]D-Fender | justanotherpaul: No, you can't specify by codec. |
| 19:19.20 | Katty | [TK]D-Fender: he wanted the weekend to think about what i wanted |
| 19:19.39 | Dr-Linux|home | [TK]D-Fender: actually doing this for the first time, not sure what to pastebin .. since there is lot of stuff, but i guess problem is with RateCard somewhere |
| 19:19.50 | justanotherpaul | [TK]D-Fender: ok, thanks. Do you have any other ideas how to get speex/16000 through Asterisk? |
| 19:19.51 | Dr-Linux|home | PakiPenguin: Hi, How are you today? |
| 19:20.04 | justanotherpaul | [TK]D-Fender: maybe I should just allow reinvites? |
| 19:20.09 | PakiPenguin | I am good, how about you? |
| 19:20.14 | [TK]D-Fender | justanotherpaul: Nope. Why so centered on that codec? |
| 19:20.25 | Katty | seanmh: ohai! |
| 19:20.33 | [TK]D-Fender | Dr-Linux|home: If you have no idea what to look for you are FUBAR'd |
| 19:20.34 | seanmh | Yo! |
| 19:20.45 | Katty | seanmh: how'rechu? |
| 19:20.59 | seanmh | Pretty good.. figure out your MAC address problem? |
| 19:21.12 | Katty | lol, no ;) i'm ignoring it for the moment. |
| 19:21.12 | unbkbl | [TK]D-Fender: http://pastebin.com/m26354be4 no matter what the problem is the same |
| 19:21.25 | Katty | seanmh: good to hear you're doing well. (= |
| 19:21.26 | justanotherpaul | [TK]D-Fender: well, it's the only wideband one available with my softphone (Ekiga) as far as I know. I really just need high-quality audio from Linux to Windows though, so I'm open to suggestions. |
| 19:21.32 | Dr-Linux|home | [TK]D-Fender: :) |
| 19:22.22 | [TK]D-Fender | unbkbl: 1st PB = 169: 283286 5175221 IO-APIC-level ioc0, uhci_hcd:usb4, wct2xxp , 2nd PB = 90: 7723659 4558478 IO-APIC-level wct4xxp. Why the hell is the DRIVER DIFFERENT? |
| 19:22.56 | [TK]D-Fender | justanotherpaul: Whats the need for wideband?: |
| 19:23.01 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:23.15 | *** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:23.45 | *** join/#asterisk minaguib (n=mina@modemcable118.145-203-24.mc.videotron.ca) |
| 19:23.51 | [TK]D-Fender | unbkbl: In fact AACRAID is also not present. You are showing me 2 &%$# different machines. |
| 19:24.03 | *** join/#asterisk oej (n=olle@cust-IP-11.data.tre.se) |
| 19:24.12 | gramulhaozin | HELLO THERE, need a 1TDM403BF |
| 19:24.18 | gramulhaozin | anyone out there can sell to Florida ? |
| 19:24.26 | [TK]D-Fender | gramulhaozin: you have our permission. Go buy one. |
| 19:24.30 | minaguib | Hi. I have a general telephone question. For a local area code (514 - montreal), is it easy to get a vanity phone number from the voip provider I'm ordering from ? Or so I just pick from their pool and/or transfer an external phone and that's it ? |
| 19:24.45 | [TK]D-Fender | gramulhaozin: I already linked you to one of the best retailers for them |
| 19:24.49 | gramulhaozin | [TK]D-Fender: need a discount, need to resell. |
| 19:24.49 | justanotherpaul | [TK]D-Fender: There are transcribers at the other end and their job is easier with higher quality sound. Am I mistaken that wideband is important to sound quality? |
| 19:25.08 | gramulhaozin | [TK]D-Fender: telephony depot is good, but we need a discout to resell. |
| 19:25.10 | Katty | [TK]D-Fender: i'd like to know how you keep all these conversations straight. |
| 19:25.38 | justanotherpaul | Katty: I second that. |
| 19:25.42 | [TK]D-Fender | minaguib: Depends on the ITSP. checkout les.net , and unlimitel.ca |
| 19:25.55 | [TK]D-Fender | minaguib: And nice to see another MLUG-er around.... |
| 19:26.07 | minaguib | [TK]D-Fender: I'm going with unlimitel, but no mention of vanity on their site |
| 19:26.16 | minaguib | [TK]D-Fender: Hmm. Sorry I don't recognize you. You are :) ? |
| 19:26.31 | [TK]D-Fender | gramulhaozin: This channel is not a store. if you want to resell more than any other place you can go through, call Digium direct. |
| 19:27.20 | Katty | it's more like barrens chat. |
| 19:27.47 | [TK]D-Fender | Katty: I'm special.... in more than just "little bus" capacity :p |
| 19:28.03 | Katty | okay. i really don't want to hear about your capacity. |
| 19:28.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:28.36 | Katty | [TK]D-Fender: ;) |
| 19:29.31 | [TK]D-Fender | minaguib: Call them up directly (based out of Ottawa IIRC). They can probably do some special stuff if needed. they will have a DID pool to work with, so don't bet on just a single one being free. |
| 19:29.45 | [TK]D-Fender | minaguib: vanity costs :) |
| 19:31.18 | *** join/#asterisk pjezek (n=pj@193.85.164.154) |
| 19:34.19 | file | how many people in here liked the show 'Xena'? |
| 19:34.41 | Katty | i like gabriel better |
| 19:34.50 | mshades | is Xena that new drag show on queenie.tv? |
| 19:35.05 | Katty | oh. i guess i'm showing my age. |
| 19:35.06 | Katty | nevermind. |
| 19:35.14 | [TK]D-Fender | file: Another Raimi production. Senseless fun I guess. |
| 19:35.30 | Yourname`` | Hi. Is there a way I can login Agent 1 into two queues using AgentCallBack? |
| 19:35.36 | [TK]D-Fender | file: Bruce Campbell > ALL |
| 19:35.51 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 19:35.53 | [TK]D-Fender | Yourname``: Yes... make the AGENT a member of both. |
| 19:35.56 | Qwell | Yourname``: Don't use AgentCallbackLogin |
| 19:36.19 | Yourname`` | [TK]D-Fender: Wouldn't that be statically logged in then? |
| 19:36.35 | Yourname`` | Qwell: No sir, I hate it now. AgentCallBack reduces the caller wait time, lol |
| 19:36.55 | [TK]D-Fender | Yourname``: Do you want the agent to only TEMPORARILY belong to both? |
| 19:38.22 | Yourname`` | [TK]D-Fender: No, it will be a permanent thing. Except, I want them to be able to type, let's say 4 to login to SalesQ and lets say 5 to login to HelpQ. Then, when they're done with any one of those queues they just logout of that queue. |
| 19:38.43 | ManxPower | I thought AgentCallbackLogin was removed from 1.6? |
| 19:39.01 | [TK]D-Fender | Yourname``: then use another login method. "core show applications like queue" |
| 19:39.20 | Qwell | ManxPower: hence the "don't use it" |
| 19:39.48 | ManxPower | Qwell: But you did not give a reason, did you? |
| 19:40.42 | Yourname`` | [TK]D-Fender: I used to like AQM/RQM, but I liked the functionality of the "always-on" that AgentCallBack provided where the call comes in with a beep. |
| 19:40.57 | Qwell | Yourname``: That *isn't* what AgentCallback does |
| 19:41.19 | Qwell | AgentCallbackLogin, that is. There is no such thing as AgentCallback |
| 19:41.42 | errr | hi Katty :) |
| 19:41.43 | [TK]D-Fender | that would be "AgentLogin" |
| 19:42.04 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 19:42.20 | *** join/#asterisk PakiPenguin_ (n=junaid@linuxpakistan/admin/pakipenguin) |
| 19:42.32 | Qwell | You could do that with something like subqueues, but that would be a hack |
| 19:43.46 | *** join/#asterisk ait^ (n=airani@66-146-175-59.skyriver.net) |
| 19:44.08 | ait^ | Major issue with dropped calls. Running Asterisk with FreePBX gui. bandwidth consumption with one call is about 40KB/s or 300-320 kbps...our dedicated T1 drops call with 4 concurrent calls! Using Ulaw |
| 19:44.38 | ait^ | I'm running BWM-NG, a bandwidth tool...and with 4-5 concurrent calls, I'm right around 1500kbps |
| 19:44.46 | Strom_M | 320kbps for a single call? I think you're making a huge mistake somewhere |
| 19:45.04 | ait^ | at 3-5 calls, it blogs the system down |
| 19:45.11 | Strom_M | blogs? |
| 19:45.16 | ait^ | bogs |
| 19:45.28 | ait^ | voice quality and call drops |
| 19:45.40 | ait^ | defiantely a bandwidth issue....but I don't know why |
| 19:45.50 | ait^ | the server is running on its own dedicated T1 |
| 19:45.54 | ManxPower | ait^: no voice protocol that Asterisk supports uses more than about 80Kbpx |
| 19:45.56 | Yourname`` | Sorry [TK]D-Fender Qwell I meant AgentLogin |
| 19:45.58 | ManxPower | Kbps, that is. |
| 19:46.08 | jameswf-home | hmmm should have a module that blogs its own failures... I think with an EMO theme |
| 19:46.08 | ManxPower | So, you screwed up somewhere. |
| 19:46.33 | ait^ | ManxPower, that's right...with G.711, at most it should be 90kbps...so what the hell is going on? |
| 19:46.39 | [TK]D-Fender | Yourname``: You know the pieces. live with it. |
| 19:46.50 | ManxPower | ait^: no idea. |
| 19:47.04 | ait^ | looking at bandwidth usage as we speak and with two calls its at 800kbps |
| 19:47.10 | ManxPower | how many channels does "show channels" (or maybe "core show channes") give you? |
| 19:47.16 | Yourname`` | [TK]D-Fender: So there's no way I can get AQM type functionality with a mix of the beep-you-got-called that AgentLogin gives you? |
| 19:47.21 | ait^ | when the calls end, bandwidth drops to nearly 0 |
| 19:47.39 | ManxPower | 800 kbps is about 80Kbps, right? |
| 19:47.42 | [TK]D-Fender | ait^: Guess there's something else going over your connection. |
| 19:48.06 | ait^ | ManxPower, 13 channels |
| 19:48.15 | ManxPower | ait^: so, 6 calls then |
| 19:48.45 | ait^ | can i pm you? |
| 19:48.59 | *** join/#asterisk unbkbl (n=unbkbl@mx.humanitas.com.co) |
| 19:49.05 | ManxPower | ait^: only if you have a credit card number along with the PM that has a high credit limit. |
| 19:49.06 | ait^ | ManxPower, can i message you |
| 19:49.17 | errr | heh |
| 19:49.19 | ManxPower | PM is WORK and I don't do WORK for free. Heck, I don't even normally accept new clients. |
| 19:49.20 | unbkbl | damin |
| 19:49.22 | unbkbl | damian |
| 19:49.25 | unbkbl | hello? |
| 19:49.29 | ait^ | hehe, i don't want to paste the results here...19 lines ok to paste here? |
| 19:49.36 | jameswf-home | ~pb |
| 19:49.38 | 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 |
| 19:49.38 | unbkbl | r u there? |
| 19:49.56 | ManxPower | ait^: Really, I think you are confusing Kbps with kbps. |
| 19:50.09 | Yourname`` | AOLer alert |
| 19:50.27 | ManxPower | kilobYtes .vs. KilobIts |
| 19:50.44 | ManxPower | Yourname``: We knew you are on AOL. No need to tell us. |
| 19:51.03 | *** join/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 19:51.06 | ait^ | http://paste.debian.net/13258/ |
| 19:51.07 | [TK]D-Fender | ManxPower: that'd be Kbps vs KBps. The "b" is what capitalizes, not the "k" |
| 19:51.25 | ManxPower | [TK]D-Fender: Maybe more coffee is in order for me. |
| 19:51.26 | gramulhaozin | ManxPower: I always thought you get paid by digium to talk here. |
| 19:51.27 | [TK]D-Fender | ManxPower: "k" (kilo) as "thousand" is constant. |
| 19:51.28 | ctooley | Someone know where the setup for Polycom busy lamp field might be documented? |
| 19:51.46 | [TK]D-Fender | ctooley: on the WIKI |
| 19:52.00 | ManxPower | gramulhaozin: I have never in my life received a single penny or payment or other compensation from Digium for helping people on this channel. |
| 19:52.06 | ManxPower | I don't know if [TK]D-Fender has, but I doubt it. |
| 19:52.09 | [TK]D-Fender | ctooley: Make sure you're on SIP 1.6.7 minimum, and go read up on "Presence". |
| 19:52.10 | Yourname`` | haha ManxPower really now.. |
| 19:52.10 | ait^ | ManxPower, sounds odd, I know...but I'm seeming 600-700kilo bits per second, or about 75Kilo Bytes Per Second right now with two calls... |
| 19:52.26 | [TK]D-Fender | ManxPower: Nope, nothing from Digium. |
| 19:52.44 | ManxPower | Which can be a good thing. I can be an asshole to people that deserve it. If I was paid by Digium I'd have to help the idiots. |
| 19:52.49 | [TK]D-Fender | ait^: And where are you "seeing" this? |
| 19:52.56 | ManxPower | Or at least be nice to them |
| 19:53.26 | ManxPower | [TK]D-Fender: I knew the B .vs. b, my brain just crossed wires there for a min. |
| 19:53.38 | ait^ | command line, using Bandwidth Monitor |
| 19:54.06 | *** join/#asterisk robevans (n=robevans@OL6-231.fibertel.com.ar) |
| 19:54.23 | gramulhaozin | wtf |
| 19:54.34 | [TK]D-Fender | ait^: Sorry, but RTP fo ULAW sits at about 85 kiloBIT per second. |
| 19:54.41 | gramulhaozin | ManxPower: Digium should look to give you some award$$$ |
| 19:54.54 | [TK]D-Fender | ait^: Your process is FUBAR'd somehow. |
| 19:55.03 | ManxPower | gramulhaozin: people are always welcome to contribute via Paypal to eric@fnords.org |
| 19:55.12 | Qwell | gramulhaozin: he means you ^^ |
| 19:55.30 | ait^ | [TK]D-Fender, I know that...or about 12KBps, but I'm seeing around 35KBps...that's why with a few calls, I'm seeing drops...so what else could be going on? |
| 19:55.38 | ManxPower | Qwell: Digium too. *stare* |
| 19:56.01 | ManxPower | ait^: get out a raw packet sniffer like wireshark with the rtp addin |
| 19:56.21 | [TK]D-Fender | ait^: If you can't tell, nobody here is going to play psychic on that one. |
| 19:57.00 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:57.12 | ait^ | [TK]D-Fender, hehe...I'm not very technical...It's a fresh install (and my first)...i hope you guys might have some suggestions |
| 19:57.54 | ctooley | We had 2 Grandstream phones and the Executive Assistance's phone lit up when the Executive was on the phone. Now the Executive has switched to a Polycom phone. Other than the device switch, everything is the same. The BLF setup is on the assistant's phone which hasn't changed. Still, the light doesn't light up. |
| 19:58.03 | ManxPower | ait^: VoIP is very, very, very technical. *Most* of the time you won't have to look at packet captures, but you will occasionally |
| 19:58.38 | [TK]D-Fender | ctooley: Presence has to be enabled in your provisioning. It is not by default. |
| 19:59.01 | ctooley | [TK]D-Fender, meaning that the polycom doesn't send the subscribe otherwise? |
| 19:59.19 | [TK]D-Fender | ait^: Either your numbers are fudged by not looking at the right things, or you aren't getting a complete list. |
| 19:59.24 | ManxPower | ctooley: look at http://www.fnords.org/~eric/polycom-config-examples/ Pay special attention to the end of http://www.fnords.org/~eric/polycom-config-examples/0004f203422d-phone.cfg |
| 19:59.36 | ManxPower | You have to enable "buddy watch" or "presence" for BLF on the Polycoms |
| 19:59.43 | [TK]D-Fender | ctooley: PASTEBIN <- |
| 19:59.50 | ait^ | ManxPower, how can i start to diagnosis this? I know it's not a CPU or mem issue..and everything points to bandwidth...plugging a windows based machine instead of the asterisk and running the speakeasy speed test shows a full 1500kbps up and down...so its not the ISP |
| 20:00.14 | ManxPower | (3:56:01 PM) ManxPower: ait^: get out a raw packet sniffer like wireshark with the rtp addin |
| 20:00.51 | *** join/#asterisk ckotticg_ (n=edwin@static-adsl201-232-88-87.epm.net.co) |
| 20:01.13 | ait^ | [TK]D-Fender, everything looks normal here: http://paste.debian.net/13258/ |
| 20:01.17 | ait^ | ? |
| 20:01.31 | unbkbl | hehehe |
| 20:01.32 | ckotticg_ | hi |
| 20:01.34 | unbkbl | at last1\ |
| 20:01.39 | unbkbl | hi damian |
| 20:01.44 | ckotticg_ | hi... |
| 20:02.06 | ManxPower | ait^: That is my suggestion on how to start diagnosis. a bandwidth montor is a very limited tool. You need something that lets you see a bigger picture. |
| 20:02.07 | ckotticg_ | who's the one that was helping you? |
| 20:02.21 | [TK]D-Fender | ait^: 8 SIP channels, and no validation of codecs used, etc |
| 20:02.28 | ckotticg_ | <PROTECTED> |
| 20:02.29 | unbkbl | [TK]D-Fender: |
| 20:03.03 | [TK]D-Fender | ait^: Though your worse case is 85kbps * 8 |
| 20:03.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:03.46 | unbkbl | [TK]D-Fender: ckotticg_ is my boss he can explain our problem better tan me |
| 20:04.02 | unbkbl | better than me |
| 20:04.03 | [TK]D-Fender | unbkbl: You've shown me 2 very different systems. |
| 20:04.22 | [TK]D-Fender | unbkbl: I do not appreciate wasting time running around for nothing |
| 20:04.33 | ManxPower | unbkbl: You should contact Digium Paid Support as your setup seems to be more complex than can easily be handled by the VOLUNTEERS on this channel. |
| 20:05.33 | rob0 | ~did |
| 20:05.34 | jbot | [did] Direct Inward Dialing, or just a phone number |
| 20:05.47 | ckotticg_ | the problem that unbkbl issued happend just a few days ago and started without making changes in the server |
| 20:06.01 | Qwell | ahh, the old "nobody changed anything" |
| 20:06.01 | jameswf-home | there is always a change.. |
| 20:06.14 | ait^ | [TK]D-Fender, core show channels is not giving me any results |
| 20:06.30 | jameswf-home | Network gnomes usualy go in and use ninja monkey tactics |
| 20:06.36 | [TK]D-Fender | ckotticg_: The card drive was different as was a RAID controller. He showed me 2 completely different boxes. |
| 20:06.39 | ManxPower | ait^: what version of Asterisk? |
| 20:06.46 | [TK]D-Fender | ait^: "sip show channels" |
| 20:06.51 | ckotticg_ | really, we tried to make responsible the provider because we didn't make any change |
| 20:07.13 | [TK]D-Fender | ckotticg_: 2 pieces of different hardware sure consitutes a CHANGE to me. |
| 20:07.31 | ManxPower | ctooley: were those links helpful? |
| 20:07.39 | jameswf-home | if your 100% confident there was no change on your end have the provider come out.... if it is their fault its free |
| 20:07.42 | ckotticg_ | we make the change of the server yesterday after 1 week having the problem |
| 20:07.49 | rob0 | well heck, my stanaphone rings again! |
| 20:08.21 | rob0 | I was about to give up on them. |
| 20:08.27 | ctooley | ManxPower, yeah, thanks. I'm still trying to figure out where this phone is at and how our Ops team is trying to configure it. I'm going to guess that they're configuring them all from the web interface on each phone. |
| 20:08.30 | ckotticg_ | and yes, we didn't make any change |
| 20:08.48 | Qwell | ckotticg_: what is the problem? |
| 20:09.14 | ckotticg_ | ok, we started a week ago to lost some calls |
| 20:09.27 | ManxPower | ctooley: yeah, you can't set those functions via the web interface, you might be able to via the phone screen interface, but the CORRECT way to Provision a Polycom is via ftp/tftp/http/etc and a config file. |
| 20:09.36 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 20:09.36 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:09.50 | ctooley | ManxPower, yeah, not my department. :) |
| 20:10.01 | ckotticg_ | the problem was very intermitent, but 1 day later, all calls drops... |
| 20:10.02 | [TK]D-Fender | ctooley: Anybody configuring a Polycom via its web interface should be dragged out and #^$%ing SHOT. |
| 20:10.17 | ManxPower | ctooley: "It's a phone config issue, not an Asterisk issue. Go away before I turn you into a toad!" |
| 20:10.33 | ckotticg_ | the reason is that at some point, the D-channel is lost and then, all B-channels are reset |
| 20:10.34 | ManxPower | ctooley: there's your closing ticket comment for you. |
| 20:10.49 | ctooley | [TK]D-Fender, This is a company that intentionally bought 300 Grandstream 100's |
| 20:11.04 | [TK]D-Fender | ctooley: Make sure to aim for the HEAD then. |
| 20:11.20 | ctooley | And, 5 Polycoms after the executives decided they couldn't use the 100s |
| 20:11.32 | ManxPower | ctooley: I thought only Hell was allowed to buy that many GS phones. |
| 20:11.46 | ckotticg_ | the provider bring some equipment, but they "everything" fine |
| 20:11.48 | ctooley | ManxPower, where do you think I work? |
| 20:11.57 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 20:12.00 | ManxPower | The first Asterisk install at my client was all analog because they thought doing it right was too expensive. |
| 20:12.48 | ManxPower | Now they are PRI, Polycom, and a direct to telco POTS line for the fax. Like I told them to in the first place. |
| 20:13.00 | unbkbl | Qwell: ckotticg_ and i are working in the same project... we got this error message in the /var/log/asterisk/full http://pastebin.com/d3bfbd03e |
| 20:13.26 | ckotticg_ | Qwell: http://pastebin.com/d3bfbd03e |
| 20:13.42 | ManxPower | unbkbl: Most HDLC and D-Channel errors are as a result of some device or driver in the system locking interrupts for so long that data is lost on the T-1/E-1 card. |
| 20:13.45 | Qwell | what hardware is this? |
| 20:13.50 | unbkbl | our configuration files are http://pastebin.com/d25164791 |
| 20:14.05 | jameswf-home | unbkbl: Sananv? |
| 20:14.09 | ManxPower | The most common devices that do his are RAID controllers, onboard Gigabit Ethernet, SATA controllers, and video controllers |
| 20:14.35 | jameswf-home | Newer kernels do better with said devices |
| 20:15.10 | ckotticg_ | <ManxPower: I understand that, but the PBX was working perfectly with the same config |
| 20:15.15 | Qwell | ioc0 is SCSI :D |
| 20:15.15 | ManxPower | the disk and network controllers may not cause a problem until you have a specific amount of usage of those devices (turning on call recording could trigger this issue, as could running the console in non 80x24 mode) |
| 20:15.26 | ckotticg_ | we are working on kernel 2.6.18 |
| 20:15.32 | Qwell | Somebody changed something. |
| 20:15.42 | unbkbl | the hardware is a IBM x3200 server with a 1.8G Xeon dualcore processor with two Digium PRI cards |
| 20:15.47 | Qwell | disable USB, change the IRQ of the RAID controller. |
| 20:16.30 | ManxPower | Qwell: not always. I had a system that worked fine until the CLI started getting more activity because of more complex dialplans, the logging used the disk a little more and HDLC abort errors ensued. |
| 20:16.45 | ManxPower | turn off all the logging and it worked just spiffy. |
| 20:16.59 | ManxPower | well, all logging, voicemail, anything the did disk activity |
| 20:17.12 | ckotticg_ | umm |
| 20:17.19 | Qwell | ManxPower: In this case, complex dialplan is...yeah. |
| 20:17.48 | ckotticg_ | ok, we will disable all stuff that we can, but it must be done at night |
| 20:17.57 | Qwell | why? |
| 20:18.02 | Qwell | nevermind |
| 20:18.03 | ckotticg_ | so, we try it, and tell you |
| 20:18.04 | Qwell | I don't care. |
| 20:18.11 | unbkbl | hehehe |
| 20:18.13 | Strom_C | let me guess -- there's only one production box, and there's no backup for testing on |
| 20:18.17 | ManxPower | ckotticg_: what is the output of "cat /proc/interrupts"? Put the output on pastebin.ca for us to see. |
| 20:18.19 | unbkbl | yeah,,, |
| 20:18.24 | *** join/#asterisk cmantito (n=gphreak@pool-96-248-64-222.cmdnnj.fios.verizon.net) |
| 20:18.28 | Qwell | ManxPower: he did - it's a lot sharing |
| 20:18.43 | Qwell | ioc0, wct2xxp, and uhci_usb |
| 20:18.48 | ctooley | Using Interrupt Coalescence on the network card can reduce the number of interrupts that the NIC generates lowering overall interrupt loads. If you're using SIP/RTP you can drop interrupts by the NIC by almost an order of magnitude without distortion. |
| 20:18.56 | ckotticg_ | in the new server yes, irq sharing, the old one does 1 irq per device |
| 20:19.07 | Qwell | so go back to the old server |
| 20:19.07 | ManxPower | ckotticg_: there is your problem. |
| 20:19.20 | ckotticg_ | that one have the same error |
| 20:19.32 | Qwell | Showing us output from a new server is useless. |
| 20:19.56 | ckotticg_ | I've already show you the old one output |
| 20:20.13 | Qwell | http://pastebin.com/d25164791 |
| 20:20.18 | Qwell | Is that the new server or the old server? |
| 20:20.55 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-76-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:21.03 | ManxPower | ctooley: how well supported is that on Linux? |
| 20:21.09 | ckotticg_ | this is the new one.... allow me to make a list from the old one |
| 20:21.14 | Qwell | ctooley: iirc, it's per driver |
| 20:21.18 | Qwell | erm, ManxPower |
| 20:21.39 | Qwell | maybe not anymore though.. |
| 20:21.46 | ctooley | It's per device, yes, but in the testing I've done, it's non-trivial... I'll have more results up soon. |
| 20:21.57 | ManxPower | Qwell: Anyone checked his Zaptel version? |
| 20:22.01 | Qwell | 1.4.11 |
| 20:22.27 | ManxPower | Qwell: I know Digium has tried to resolve this issue, but I'm someone that will be VERY VERY VERY hard to convince. That issue is why we no longer use Digium cards. |
| 20:22.40 | ctooley | Intel is trying to make it more difficult... they replaced the e1000 driver with the e1000e and setting Interrupt Coalescence settings is now not dynamic but done at module load time. |
| 20:22.43 | ManxPower | That issue almost cost me my *job*. |
| 20:23.31 | ManxPower | Qwell: Card revision? |
| 20:23.39 | ckotticg_ | Qwell: do you think that using sangoma this problem can be resolved? |
| 20:23.49 | Qwell | ckotticg_: no, you should call Digium support. |
| 20:24.04 | ManxPower | ckotticg_: It could be easily resolved by Digium. Give them a chance and work with them. |
| 20:24.56 | Qwell | err |
| 20:24.57 | ckotticg_ | ok, we'll make the call... |
| 20:25.09 | Qwell | why is your cdr database in mysql marked as crashed? |
| 20:26.13 | unbkbl | yes |
| 20:26.15 | ManxPower | Qwell: can you give ckotticg_ a ticket number to keep them from getting the Tech Support Runaround? |
| 20:26.34 | Qwell | no, I don't have access to any of that |
| 20:26.39 | ckotticg_ | ok |
| 20:27.20 | unbkbl | [Jul 28 15:26:38] ERROR[3589] cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (145) Table './asterisk/cdr' is marked as crashed and should be repaired[Jul 28 15:26:38] ERROR[3535] cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (145) Table './asterisk/cdr' is marked as crashed and should be repaired[Jul 28 15:26:38] NOTICE[2797] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
| 20:27.28 | ManxPower | Qwell: "passphrase", what to say, anything? |
| 20:27.31 | jameswf-home | ~pb |
| 20:27.31 | 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 |
| 20:27.49 | Qwell | ManxPower: "I have problems"? |
| 20:28.14 | ManxPower | Qwell: "We don't provide free support for Asterisk. Goodbye." |
| 20:28.33 | jameswf-home | Qwell: we know but we accept you anyway |
| 20:28.35 | Qwell | ManxPower: if it's a hardware issue, it doesn't matter |
| 20:28.45 | ckotticg_ | well, thanks a lot to all of you... i'll make the call and tell you soon have was it.. |
| 20:28.59 | Qwell | ckotticg_: Find out why that server shutdown improperly. |
| 20:29.51 | ckotticg_ | Qwell: is it relevant to the call drops? |
| 20:30.00 | Qwell | if it was hit by lightning...yeah, it might be |
| 20:30.03 | ManxPower | Ya gotta wonder if the extra disk activity possibly cause by MySQL trying to recover something might have maybe something to do with locking interrupts |
| 20:30.30 | Qwell | or if somebody decided to pull the plug out of the back... |
| 20:30.47 | Qwell | or a myriad of other issues, including what ManxPower just said (that's actually a decent theory) |
| 20:31.28 | [TK]D-Fender | ok, heading home, later all |
| 20:32.02 | ckotticg_ | the old server had a lot of usage, I am aware of that... it was because of call recording, but we disable it and disk activity drops |
| 20:32.16 | ckotticg_ | but problem remains |
| 20:35.12 | *** part/#asterisk unbkbl (n=unbkbl@mx.humanitas.com.co) |
| 20:35.39 | *** join/#asterisk rpm (n=rUssell@121.119.46-69.q9.net) |
| 20:37.08 | rpm | since i upgrade to polycom release 3.0.3revb, my phones have been becoming deregistered infrequently.. being that this is a GA release, i would have assumed it would be more stable that it has been. anyone know if there has been a setting change? my sbc's are overriding the expires field in the sip registrations to 30 seconds (hard limit) 15 seconds soft.. |
| 20:38.28 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:38.53 | Strom_C | rpm: hang on, let me find you some contact info for the "firmware of the month" club |
| 20:45.36 | Katty | wibbles. |
| 20:45.39 | Katty | wobbles. |
| 20:47.51 | rob0 | Weebles wobble but they don't faw down |
| 20:50.26 | jaytee | and 'Bumbles bounce |
| 20:50.43 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 20:50.58 | jaytee | but no one wants a "Charlie in the Box" :-( |
| 20:51.38 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:54.43 | Itiliti | I have an 800 DID is not generating ringtone when a call comes in. but when calls come in over our normal did's they generate ringtone fine. any ideas? |
| 21:02.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:04.31 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 21:05.14 | Katty | [TK]D-Fender: ohai |
| 21:05.23 | Katty | [TK]D-Fender: i keep hearing a dog bark. |
| 21:05.34 | Katty | [TK]D-Fender: but i can't find said mcpoocherkins )= |
| 21:05.37 | Katty | [TK]D-Fender: it's driving me mad. |
| 21:05.59 | [TK]D-Fender | Katty: "Bark" is just dog for "mew" :) |
| 21:06.10 | Katty | i know. |
| 21:06.14 | Katty | but i want to go pet the doggy. |
| 21:06.16 | Katty | and hug on it. |
| 21:06.19 | Katty | and call it george. |
| 21:08.01 | Itiliti | I have an 800 DID is not generating ringtone when a call comes in. but when calls come in over our normal did's they generate ringtone fine. any ideas? |
| 21:09.08 | angryuser | i have a bicyle one is riding another is not, any ideas ? give us details please |
| 21:09.41 | bkw_ | Itiliti: try doing a ring_ready in your dialplan before you bridge elsewhere |
| 21:10.08 | bkw_ | angryuser: is this #mybikedontwork |
| 21:10.19 | bkw_ | haha Itiliti ignore me |
| 21:10.41 | Katty | angryuser: bike riding. yer doin it wrong |
| 21:10.42 | bkw_ | is loosing it today |
| 21:10.58 | Katty | returns bkw_'s it for fear it gets lost. |
| 21:11.11 | bkw_ | Katty: whats this I hear you're moving? |
| 21:11.12 | *** part/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 21:11.13 | rob0 | My bike is a maxi-scooter. |
| 21:11.23 | Katty | bkw_: trying. no one wants my ferrets. |
| 21:11.31 | Katty | bkw_: wants them at the new place i mean. |
| 21:11.31 | rob0 | is looking for an excuse to go ride |
| 21:12.07 | Katty | bkw_: news travels fast here. |
| 21:12.40 | errr | rob0: if the day ends in y its normally a good reason to go ride |
| 21:12.46 | Katty | hai errr! |
| 21:12.50 | [TK]D-Fender | Katty: New place won't allow ferrets |
| 21:12.51 | errr | hi Katty =) |
| 21:12.52 | [TK]D-Fender | ? |
| 21:12.52 | rob0 | Good point! |
| 21:12.57 | Katty | [TK]D-Fender: oh they will. |
| 21:13.00 | Katty | [TK]D-Fender: for the right price ;) |
| 21:13.02 | Qwell | [TK]D-Fender: not legal in some states |
| 21:13.07 | Qwell | (California is one) |
| 21:13.16 | angryuser | what is ferrets ? |
| 21:13.28 | errr | angryuser: like a rat only larger |
| 21:13.53 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 21:13.54 | [TK]D-Fender | angryuser: Cousin of the weasel. |
| 21:14.25 | [TK]D-Fender | Qwell: Pathetic. Meanwhile I keep seeing doberman, ronweiller & pitbull attack videos all the time... |
| 21:14.34 | [TK]D-Fender | rotweiller* |
| 21:15.04 | angryuser | hate that dogs, i have used ulrasound box couple times, works well |
| 21:15.11 | angryuser | ultrasound* |
| 21:15.48 | Katty | angryuser: http://sleekgeek.org/gallery/main.php?g2_itemId=17&g2_page=4 <- Ferrets. |
| 21:16.23 | angryuser | Katty i see |
| 21:18.20 | angryuser | in russian Horki/in french furets |
| 21:18.20 | *** join/#asterisk Gnutoo (n=gnutoo@host6-25-dynamic.25-79-r.retail.telecomitalia.it) |
| 21:18.33 | Gnutoo | hello, does zaptel compile on the 2.6.26 kernel? |
| 21:18.57 | Qwell | Gnutoo: Zaptel from svn does. |
| 21:19.02 | Itiliti | we have SIP trunking through Airespring. both the local did's and the 800 number ring on the asterisk box. but when calls come in on the local did's, they generaqte ringing to the calling user. When one of the 800 numbes is called, it connects fin, and someone can answer it fine, but there is no ringing sound that is generated for the calling user |
| 21:19.16 | Gnutoo | Qwell, ah ok...when will a new version be released? |
| 21:19.24 | Qwell | soon I believe |
| 21:19.36 | Gnutoo | Qwell, ok thanks |
| 21:19.58 | Katty | [TK]D-Fender: maybe i should just get a chihuahua |
| 21:20.19 | Gnutoo | Qwell, so i'll wait...i can have no conference for a while...and anyway it doesn't support video so i don't know if i'll use it |
| 21:20.30 | [TK]D-Fender | Katty: No Paris Pocket Dogs! |
| 21:20.49 | Katty | [TK]D-Fender: but think of how fun socker could be. |
| 21:21.02 | angryuser | Itiliti can you supply us a cli output when that call is coming in ? |
| 21:21.05 | [TK]D-Fender | Katty: name him "punt" :p |
| 21:21.20 | Katty | [TK]D-Fender: i hate little yippers. |
| 21:21.41 | Katty | [TK]D-Fender: now a german shepherd is more my style! |
| 21:21.43 | [TK]D-Fender | Katty: And you even thought about a chihuahua? |
| 21:21.57 | Katty | [TK]D-Fender: well it's a dog and it's little :/ |
| 21:22.13 | [TK]D-Fender | Katty: Good = cat. Better = dog. Best = dog that eats cats :D |
| 21:22.19 | Katty | [TK]D-Fender: mrow. |
| 21:26.47 | *** join/#asterisk dlynes (n=daniel@S01060016b68219f1.vs.shawcable.net) |
| 21:27.06 | Katty | [TK]D-Fender: oh boy. yet /another/ person that doesn't know about ferrets. |
| 21:27.08 | Katty | [TK]D-Fender: sigh. |
| 21:27.38 | Katty | [TK]D-Fender: they're all... what's a ferret? |
| 21:27.57 | [TK]D-Fender | Katty: Think your pets are problematic? Go look up SERVALS. My sister had a pair. |
| 21:29.08 | Katty | [TK]D-Fender: my mom has a bengal |
| 21:29.20 | Katty | [TK]D-Fender: i know all about serval kitties. sadly. |
| 21:29.24 | [TK]D-Fender | Katty: .... a Bengal ... what? |
| 21:29.46 | [TK]D-Fender | Katty: not.. TIGER I'm hoping... |
| 21:30.23 | Katty | [TK]D-Fender: heavens no. |
| 21:30.25 | dlynes | Katty: ferrets are cheap pets to keep...just feed them fish heads, and they're quite happy :) |
| 21:30.28 | Katty | [TK]D-Fender: those eat like 30lbs of meat a day |
| 21:30.32 | _ShrikE | <PROTECTED> |
| 21:30.48 | Katty | [TK]D-Fender: it was an F3 bengal tabby |
| 21:31.18 | Katty | [TK]D-Fender: he's from california. |
| 21:32.11 | Katty | [TK]D-Fender: http://webcon.net/~izaah/gallery/d/261-1/chiggersink.jpg <- random picture |
| 21:32.19 | Katty | [TK]D-Fender: well that is him |
| 21:32.24 | Katty | [TK]D-Fender: random picture of him. whatever. |
| 21:33.08 | [TK]D-Fender | Katty: just found them on Wikipedia. Like the colour on this one : http://en.wikipedia.org/wiki/Image:BengalCat_Stella.jpg |
| 21:33.18 | [TK]D-Fender | _ShrikE: Fat-ass lazy cats |
| 21:33.31 | Katty | yeah. chigger's a bit oranger |
| 21:33.40 | [TK]D-Fender | Katty: But thatnks for introducing them to me... interesting. |
| 21:33.44 | Katty | he jumps on everything. and will not stop meowing for anything. |
| 21:33.54 | Katty | tail also sticks straight up in the air at all times. |
| 21:34.14 | k-man | how can i tell if asterisk managed to register with my sip provider? |
| 21:34.36 | angryuser | k-man 'sip show registry' in cli |
| 21:34.39 | Katty | [TK]D-Fender: i'd love an ashera cat |
| 21:34.52 | Yourname`` | Hi. I had agents 10-50, changed to agents 100-150.. however, if the agent accidentally did the AgentLogin as agent 25 instead of agent 125.. it still accepts the agent as 25 instead of saying not such agent exists or something like that. Why? |
| 21:34.59 | k-man | angryuser, thanks |
| 21:35.00 | Katty | [TK]D-Fender: sadly i don't have 22k to blow on a kitten |
| 21:35.05 | esaym | how to kill a channel in asterisk cli? My software caused a couple of sip channels to stay connected... |
| 21:35.25 | *** join/#asterisk serialthrilla (n=noemail@adsl-71-131-145-38.dsl.sntc01.pacbell.net) |
| 21:35.42 | Yourname`` | esaym: Type "core show application soft hangup" in the CLI |
| 21:36.43 | [TK]D-Fender | Katty: The Bengal is a large breed - weighing between 7 to 20 pounds (lb) (3.2 kg to 9.1 kg). Ouch, heavy for a cat. It'd eat a chihuahua :p |
| 21:36.48 | serialthrilla | has anyone been able to get 802.1p priority working with grandstream phones? |
| 21:37.02 | [TK]D-Fender | Yourname``: PASTEBIN. |
| 21:37.06 | Katty | [TK]D-Fender: ashera cats are well... |
| 21:37.09 | Katty | [TK]D-Fender: heavier than that |
| 21:37.14 | *** join/#asterisk ZX81 (n=matt@120.89.80.110) |
| 21:37.48 | Katty | [TK]D-Fender: wiki claims about 30lbs |
| 21:38.00 | Katty | [TK]D-Fender: REF: http://en.wikipedia.org/wiki/Ashera |
| 21:38.09 | ZX81 | hey anyone know a solution to voicemail deletes always saying undelete? |
| 21:38.09 | [TK]D-Fender | Katty: Those remind me of my sister's servals |
| 21:38.26 | Yourname`` | [TK]D-Fender: There's no pastebin... all I'm saying is they are still able to login as agent 10 when agentlogin asks them for a username when there is [100] and member=>Agent/100 and no instance of 10 anywhere |
| 21:38.29 | ZX81 | someone logs in, tries to delete vm and it says "undeleted" |
| 21:38.32 | Katty | [TK]D-Fender: well they're a serval hybrid. (= |
| 21:38.36 | esaym | Yourname``: it prints a bunch of syntax info |
| 21:39.07 | [TK]D-Fender | Katty: Allerca and labeled "Ashera" were actually raised by him as another hybrid, "Savannah F1 <- F1... oh yeah.. REALLY far removed from "wild animal" :) |
| 21:39.11 | Yourname`` | esaym: So read it |
| 21:39.18 | [TK]D-Fender | Katty: Don't get children as visitors :) |
| 21:39.26 | esaym | whatis "causecode" |
| 21:39.29 | Katty | [TK]D-Fender: hehehhe |
| 21:39.31 | ZX81 | have over 100 staff screaming at me that they can't delete their voicemails :( |
| 21:39.40 | ZX81 | asterisk runs as root |
| 21:39.44 | ZX81 | so not permission problem |
| 21:39.50 | Qwell | not true |
| 21:39.51 | [TK]D-Fender | Yourname``:>>>>>> PASTEBIN <<<<<<<< |
| 21:40.00 | Qwell | it could still be a permission problem |
| 21:40.02 | *** join/#asterisk javb (n=javb@190.80.236.32) |
| 21:40.20 | ZX81 | really? |
| 21:40.24 | javb | is it posible for a person to have interference from another call on VoIP Asterisk based telephony system ? |
| 21:40.25 | Qwell | ZX81: check what lsattr has to say about those dirs |
| 21:40.28 | ZX81 | but root must have created the folders no? |
| 21:40.29 | ZX81 | ok |
| 21:40.30 | Katty | [TK]D-Fender: there was a big ordeal about tiger cubs being sold in a walmart parking lot. did you hear about that? |
| 21:40.51 | [TK]D-Fender | Katty: Nope... |
| 21:40.59 | ZX81 | ------------------- |
| 21:41.04 | [TK]D-Fender | javb: makes no sense. |
| 21:41.04 | ZX81 | is what lsattr says |
| 21:41.07 | Katty | [TK]D-Fender: it was last month in texas. |
| 21:41.08 | Yourname`` | [TK]D-Fender: http://pastebin.ca/1085633 |
| 21:41.17 | Qwell | ZX81: and all dirs leading up to one of the individuals who can't delete |
| 21:41.29 | Qwell | if any aren't ------, you may have issues |
| 21:41.34 | ZX81 | ok |
| 21:41.35 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:41.36 | mshades | i like using asterisk for kinky purposes |
| 21:41.36 | Qwell | (specifically, if any are an i) |
| 21:41.42 | [TK]D-Fender | Yourname``: member => Agent/152 does not belong in sip.conf |
| 21:41.51 | *** join/#asterisk pcrane (n=pcrane@120.89.80.110) |
| 21:41.52 | Yourname`` | It's in queues.conf [TK]D-Fender q |
| 21:41.55 | ZX81 | nag all have ------- |
| 21:42.00 | ZX81 | *nah |
| 21:42.12 | [TK]D-Fender | Yourname``: pastebin RELEVANT things please. |
| 21:42.19 | ZX81 | including /var off the root |
| 21:42.21 | Katty | [TK]D-Fender: REF: http://www.themonitor.com/articles/ones_13216___article.html/cubs_selling.html |
| 21:42.40 | ZX81 | should I do a 777 on /var/spool/asterisk/voicemail -r? |
| 21:42.43 | Yourname`` | [TK]D-Fender: It's a theoretical thing I'm talking about!! |
| 21:43.02 | Yourname`` | All the settings in the confs were 10 befiore.. I removed all occurences of 10 and replaced with 100 |
| 21:43.02 | Katty | [TK]D-Fender: too bad i wasn't there when that went down :/ |
| 21:43.13 | Yourname`` | YET, Mister 100 is able to login to a queue as 10 |
| 21:43.25 | Yourname`` | Is it something to do with a dynamic astdb that needs deletion? |
| 21:43.32 | [TK]D-Fender | Yourname``: what "theoretical" You said someone logged in an impossible way |
| 21:43.42 | [TK]D-Fender | Yourname``: pastebin ALL the friggen backup for your setup |
| 21:44.24 | dlynes | ZX81: no...chown -R root:root /var/spool/asterisk/voicemail && chmod -R 755 /var/spool/asterisk/voicemail |
| 21:44.36 | ZX81 | ok cool |
| 21:44.54 | ZX81 | still the same |
| 21:45.02 | ZX81 | says undelete when you try to delete vm |
| 21:45.20 | dlynes | ZX81: then (assumign you're using bash shell) cd /var/spool/asterisk/voicemail ; for file in `find . -type f`; do chmod 644 $file; done |
| 21:45.59 | dlynes | ZX81: and then voicemail should work just fine, assuming asterisk is running as root user |
| 21:46.18 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 21:46.21 | ZX81 | yep |
| 21:46.29 | Katty | [TK]D-Fender: we should adopt a tiger. |
| 21:46.36 | ZX81 | voicemail or voicemail/default? |
| 21:46.52 | [TK]D-Fender | Katty: lol.... |
| 21:46.53 | Katty | [TK]D-Fender: i could totally ride it to work. |
| 21:47.22 | ZX81 | there aren't actually any files in voicemail/default they're all directories |
| 21:47.22 | [TK]D-Fender | Katty: It could totally eat you for luch :) |
| 21:47.24 | serialthrilla | has anyone been able to get layer 2 QoS working on a grandstream phone? |
| 21:47.41 | Katty | [TK]D-Fender: :< |
| 21:47.46 | ZX81 | oh |
| 21:47.47 | ZX81 | ok |
| 21:47.57 | ZX81 | that find thing does subdirs |
| 21:47.58 | ZX81 | :) |
| 21:48.00 | ZX81 | sec will check |
| 21:48.16 | Katty | speaking of lunch. it's just about dinner time. |
| 21:48.38 | serialthrilla | GXP2000 with 1.1.6.16 |
| 21:48.43 | ZX81 | nope |
| 21:48.44 | ZX81 | :) |
| 21:48.47 | ZX81 | still undeleted |
| 21:49.26 | ZX81 | brb |
| 21:49.41 | ZX81 | back |
| 21:49.42 | ZX81 | :) |
| 21:49.43 | ZX81 | heh |
| 21:49.55 | ZX81 | aparently vm is more important than their lack of network |
| 21:49.56 | ZX81 | lol |
| 21:50.25 | ZX81 | these messages are now in the Old folder |
| 21:50.39 | serialthrilla | changes ZX81's nick to Smiles |
| 21:50.50 | ZX81 | :D |
| 21:51.11 | Sir_Smiles_A_Lot | :D |
| 21:51.21 | serialthrilla | lol. |
| 21:51.27 | Katty | [TK]D-Fender: what's for dinner. |
| 21:51.35 | serialthrilla | Grandstream |
| 21:51.57 | ZX81 | is going to walk around the premises and check if all extens are the same |
| 21:52.06 | ZX81 | maybe vm is corrupted for this person |
| 21:52.17 | Yourname`` | [TK]D-Fender: I deleted astdb and its working now! |
| 21:52.51 | Yourname`` | My only question is since I removed astdb and since I don't use ANY db stuff in extensions.conf, will anything be adversely affected? |
| 21:53.13 | [TK]D-Fender | Yourname``: shouldn't |
| 21:53.25 | [TK]D-Fender | Katty: Just had chinese leftovers. |
| 21:55.00 | angryuser | sip registrations storen normally in asdb dundy secret ,and agents/users associations, not sure about last one, or if it used anyway |
| 21:55.36 | angryuser | stored* |
| 21:56.46 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 22:03.35 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 22:05.18 | mshades | bite my carbuncle to release the fluid |
| 22:08.10 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 22:08.47 | hunmonk | is there a better /dev/null kind of option besides Console/dsp? |
| 22:09.09 | Yourname`` | [TK]D-Fender: thanks |
| 22:09.33 | [TK]D-Fender | hunmonk: To do what exactly? |
| 22:11.28 | hunmonk | [TK]D-Fender: for parkandannounce, you have an announce location as an arg. i really don't want it announced at all, so to date i've been putting in the announce location as Console/dsp. that works ok, but throws a lot of warnings and garbage to the console. i was just hoping to be a bit more elegant about it |
| 22:11.54 | [TK]D-Fender | hunmonk: Local <- |
| 22:13.07 | hunmonk | [TK]D-Fender: just 'Local' ? i'm used to seeing that w/ a number or ext |
| 22:13.34 | [TK]D-Fender | hunmonk: Send it to a Local channel. |
| 22:13.58 | [TK]D-Fender | hunmonk: I didn't write a COMPLETE channel sample, I'm sure you can figure out what to do all by yourself. |
| 22:14.21 | jblack | heh |
| 22:14.32 | hunmonk | jblack: well hello there :) |
| 22:14.45 | jblack | hi |
| 22:15.37 | serialthrilla | is there a way to tell a gxp2000 phone to dial a number from the network? |
| 22:16.06 | rob0 | Use a stern, commanding voice. |
| 22:16.14 | jblack | serialthrilla: route it through a voip provider |
| 22:17.39 | Katty | mmm! dinner! |
| 22:18.06 | serialthrilla | i'm talking about like a program where it sends the command to the phone, the phone dials the number, and i pick up the handset |
| 22:18.24 | serialthrilla | instead of having to punch the buttons on the phone manually |
| 22:18.29 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 22:18.35 | jblack | so... You want speed dial? |
| 22:18.58 | jblack | set up a callfile. You can specify which extension to dial, and which number to connect to. |
| 22:19.12 | watchy | whats the best way to get echo out of a sangoma a200? |
| 22:19.24 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 22:19.34 | jblack | bounce it off the walls of the grand canyon! |
| 22:19.41 | Katty | serialthrilla: isymphony might be something for you |
| 22:19.43 | jblack | <bidi-dish> |
| 22:20.53 | jblack | watchy: Seriously, though, sip phones aren't supposed to suffer from echo. |
| 22:21.12 | jblack | watchy: Perhaps you can find something on here that'll lend you a hand: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 22:21.15 | watchy | hmm i think its the lines |
| 22:21.19 | watchy | the pots lines |
| 22:21.22 | watchy | not the phones them selves |
| 22:21.30 | serialthrilla | jblack: yea callfiles, that's it. if i create the call file with my phone's extension, will it call my phone and then i pick it up and i'm connected? |
| 22:21.33 | jblack | Oh, that's not the sangomas. That's the card. |
| 22:21.42 | jblack | serialthrilla: That's the idea. |
| 22:21.46 | watchy | the card is a sangoma |
| 22:21.48 | serialthrilla | jblack: thank you |
| 22:21.55 | ManxPower | watchy: the best way is to use an echo canceler. There are many of them |
| 22:21.58 | watchy | its a 4 line analog pots card |
| 22:22.05 | ManxPower | I like the HPEC for small number of channels. |
| 22:22.15 | ManxPower | But there is also something called the OSLEC. |
| 22:22.18 | watchy | manx: its like 4 channels |
| 22:22.28 | jblack | watchy: Ok, then yeah, look at enabling the built in EC if it exists, otherwise, those software ones are what you need |
| 22:22.31 | ManxPower | Oh! If your Digium card is under warrenty, you can get the HPEC for FREE from Digium! |
| 22:22.42 | watchy | its a sangoma manx |
| 22:22.57 | watchy | and the guy i work for didnt wanna get one with built in HW echo cancelation |
| 22:23.12 | [TK]D-Fender | watchy: OSLEC <- |
| 22:23.22 | watchy | thanks tk |
| 22:23.27 | watchy | imma check it out right now |
| 22:24.32 | watchy | i love u all |
| 22:24.40 | watchy | if u didnt exist i woulda shot myself years ago |
| 22:24.50 | jblack | heh |
| 22:24.52 | *** join/#asterisk tobias (n=tobias@cpe-069-134-204-037.nc.res.rr.com) |
| 22:24.53 | ManxPower | Oh, sorry. Well you can use the HPEC with non-Digium cards for $10/channel |
| 22:24.53 | Katty | we love you too! |
| 22:25.09 | serialthrilla | jblack: do you know of an already programmed way to create an callfile? such as through the manager API? |
| 22:25.26 | Katty | serialthrilla: i know how to do it. |
| 22:25.28 | Katty | serialthrilla: and i blogged it. |
| 22:25.30 | jblack | serialthrilla: they're just text files. don't be scared. |
| 22:25.41 | Katty | serialthrilla: they're super duper easy. want my post? |
| 22:25.57 | serialthrilla | Katty: yes please |
| 22:26.08 | jblack | on his behalf, please. I don't want to reach for the docs. |
| 22:26.27 | serialthrilla | jblack: i know, but i didn't want to write a network listener to place the call from the phone book app if i can just use the manager API -__- |
| 22:26.29 | Katty | serialthrilla: http://angela.sleekgeek.org/2008/03/13/e911/ |
| 22:26.41 | jblack | Oh, is that your goal? |
| 22:26.42 | Katty | serialthrilla: it's pretty much a straight up example. |
| 22:26.58 | jblack | Make the callfile, and scp it over, then ssh an mv, and profit |
| 22:27.13 | ZX81 | man, I deleted everything in the vm folder for that user, and then left a message and then the same thing happened again |
| 22:27.27 | ZX81 | I checked with someone else and they have no problem |
| 22:27.37 | jblack | or have a template somewhere, do a sed over ssh and copy. spend 20 minutes on it, go out to the bar, and tell your boss it kept you up all night. |
| 22:27.39 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 22:27.48 | Katty | this calls for a quesadilla |
| 22:27.54 | watchy | i want one |
| 22:27.59 | ManxPower | ZX81: delete the entire mailbox, it will be created next time voicemail accesses it |
| 22:28.08 | ZX81 | ok |
| 22:28.14 | ZX81 | as in the folder |
| 22:28.25 | ManxPower | as in the entire mailbox, as I just said. |
| 22:28.36 | ManxPower | OTHER things not in the folder could easily be causing the problem |
| 22:28.42 | ManxPower | like maybe a stale lock file, etc |
| 22:28.48 | ZX81 | hmmm ok |
| 22:29.00 | ZX81 | so remove from vmconf and then readd |
| 22:29.04 | ManxPower | if the person needs to save the messages, then just move the mailbox somewhere else. |
| 22:29.13 | ZX81 | with a reload app_voicemail.so inbetween |
| 22:29.19 | ManxPower | No! rm -rf /path/to/user/mailbox |
| 22:29.29 | ZX81 | ah ok |
| 22:30.08 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 22:30.34 | _Sam-- | hey is there any way when 'extensions reload' is issued to not have the console scroll all the extensions and info? |
| 22:30.46 | ManxPower | _Sam--: tried "set verbose 0" |
| 22:30.48 | ManxPower | ? |
| 22:30.49 | jblack | sure. turn verbose and debug down |
| 22:31.00 | _Sam-- | thanks. |
| 22:31.18 | jblack | ManxPower: No fair. Gimme a chance to put down the potato chips. =) |
| 22:31.38 | ZX81 | wow |
| 22:31.41 | _Sam-- | lol |
| 22:31.50 | ZX81 | removed /var/spool/asterisk/default/8874 -rf |
| 22:31.55 | ZX81 | left a message |
| 22:31.58 | _Sam-- | what other messages will be gone if verbose = 0? |
| 22:31.59 | ZX81 | it recreated the folder |
| 22:32.05 | ZX81 | she went to delete the vm |
| 22:32.05 | _Sam-- | i wont show incoming calls? |
| 22:32.07 | florz | Katty: that code has at least one race condition that could make things fail miserably if multiple people try to reach 911 |
| 22:32.09 | ZX81 | and it said undeleted |
| 22:32.31 | jblack | sam: so... turn down verbose, reload extensions, turn verbose back up. |
| 22:32.35 | *** part/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 22:32.58 | ZX81 | wondering if its doubled dtmf |
| 22:33.04 | jblack | perhaps watch calls differently, by grepping a tail of the log. |
| 22:33.05 | _Sam-- | yeah i guess i could do that within my script...thanks! |
| 22:33.06 | ZX81 | trying to delete from another phone |
| 22:33.29 | jblack | watch out. verbose is global. |
| 22:33.37 | jblack | Are you trying to watch events in a script? |
| 22:34.02 | _Sam-- | no, im running a script every minute from cron, and tired of sseeing the output on my console. |
| 22:34.05 | jblack | if so, that's yer prahblem right thar! |
| 22:34.13 | _Sam-- | i will tell it set verbose 0 before it reloads the extensions |
| 22:34.15 | k-man | is there an html version of the asterisk book? |
| 22:34.16 | _Sam-- | then set it back after. |
| 22:34.16 | jblack | Oh, well then, why not use the cdr ? |
| 22:34.34 | jblack | We need to stop and take a step back. |
| 22:34.38 | jblack | What are you doing, exactly? |
| 22:34.47 | Katty | florz: it's just an example, i don't use that. |
| 22:34.48 | _Sam-- | you dont want to know. |
| 22:34.55 | Katty | florz: it's the concept of how to do an e911 |
| 22:34.59 | Katty | florz: but thank you for your concern. |
| 22:35.03 | jblack | I torture kitty cats for fun. I want to know about suffering. |
| 22:35.09 | _Sam-- | we are overloaded with incoming sales calls, and im randonly generating phone error messages every minute based on a php random number. |
| 22:35.16 | _Sam-- | to try to help our sales guys maintain sanity. |
| 22:35.19 | jblack | lol |
| 22:35.24 | jblack | sweet! |
| 22:35.39 | _Sam-- | so if the rand is greater than 75, it cp's something to extensions.conf |
| 22:35.44 | _Sam-- | and then reloads them. |
| 22:35.50 | jblack | What if... |
| 22:36.07 | jblack | You overflow people to a message with your mailing adress, and hang up on 'em? |
| 22:36.21 | _Sam-- | the way it stands now, they think its an error on their own callind side. |
| 22:36.28 | jblack | what kind of business hangs up on sales? |
| 22:36.28 | _Sam-- | i have some really nifty messages and tactics :) |
| 22:36.38 | _Sam-- | one that is so busy with online orders that they dont need to service phones :) |
| 22:36.40 | jblack | These aren't sales. These are complaints about sales. |
| 22:37.26 | _Sam-- | mostly customer service calls. we are an internet retailer top 500 company.....and receive alot of calls, especially on mondays. |
| 22:37.27 | jblack | So, give the mailing address, and hang up. Or just let them sit in the queue until they give up. Random errors won't really solve the problem, because they'll just call back. |
| 22:37.37 | florz | Katty: well, I'd rather argue that it's _not_ a concept of how to do it =:-) - after all, people could think that that's the way to go ... |
| 22:37.41 | _Sam-- | listen, i didnt come here for your advice. but thank you for offering it for free....you get what you pay for . |
| 22:37.58 | jblack | oh, I'm sorry. I thought you were looking for help. My mistake. |
| 22:38.19 | _Sam-- | i was looking for help, and you solved the problem like 20 lines ago....THANK YOU |
| 22:38.25 | jblack | fuck off |
| 22:38.25 | *** join/#asterisk TheIzkabola (n=TheIzkab@c-67-171-143-153.hsd1.or.comcast.net) |
| 22:38.45 | _Sam-- | ok scrantonian. |
| 22:38.58 | Strom_C | ok, both of you |
| 22:39.00 | Strom_C | knock it off |
| 22:39.08 | jblack | Heh. sorry for the french. I'm done |
| 22:39.21 | _Sam-- | scrantonian isnt a deragatory term. i have a place north of there is the only rason i said it. |
| 22:39.27 | _Sam-- | but im done too. thanks / sorry to all. |
| 22:39.41 | ZX81 | HAH!!!! |
| 22:39.42 | ZX81 | Yay! |
| 22:39.47 | ZX81 | it was doubled dtmf |
| 22:39.49 | ZX81 | :) |
| 22:39.57 | ZX81 | was doing delete then undelete |
| 22:39.57 | jblack | ZX81: Grats. |
| 22:39.57 | ZX81 | :D |
| 22:40.01 | TheIzkabola | Hello, i'm doing a clean install of CentOS for asterisk, and I was wondering which would work well. Desktop Gnome, KDE or Server, Server GUI? If it makes a difference, I'm a noobie :) |
| 22:40.05 | ZX81 | ty |
| 22:40.20 | jblack | TheIzkabola: I'd skip a gui entirely. |
| 22:40.47 | angryuser | TheIzkabola you dont need it anyway |
| 22:40.48 | [TK]D-Fender | TheIzkabola: "EVERYTHING" <- |
| 22:40.57 | TheIzkabola | ok, so when installing I shouldn't select anythign for "desktop or server"? |
| 22:40.59 | jblack | If you understand a reasonable editor and the shell, then you're just throwing away the memory on a gui. |
| 22:41.16 | TheIzkabola | ok |
| 22:41.17 | _Sam-- | in the old days, people used to say that a GUI would just waste unneccessary resources...with resources being so readily available and cheap anymore, i dont know why a person cant use a GUI. |
| 22:41.51 | TheIzkabola | so should I select "server"? or do I even need to bother? |
| 22:41.55 | jblack | says the guy looking for new ways to hang up on his customers.. |
| 22:41.57 | serialthrilla | oh yea, that call file is sweeeeet |
| 22:41.57 | [TK]D-Fender | TheIzkabola: INSTALL everying, then set your runlevel to 5 so it doesn't start until you call it. And then use whicher desktop you want |
| 22:42.07 | jblack | TheIzkabola: Yeah. |
| 22:42.39 | TheIzkabola | Yeah to what? lol btw, thanks for the help |
| 22:42.46 | _Sam-- | jblack : if you have an intelligent retort to my statement, fine...otherwise, save it. |
| 22:42.56 | _Sam-- | resources are cheap and available, why cant a person use a GUI? |
| 22:43.11 | serialthrilla | jblack: just fyi, there's a manager way to do it: http://www.voip-info.org/wiki/view/Asterisk+manager+dialout |
| 22:43.47 | TheIzkabola | fender: what would you recommend I do? |
| 22:43.51 | *** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 22:44.15 | angryuser | _Sam-- question of preference, i just dont see how gui will help me, i prefer to ssh to it from remote |
| 22:44.32 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 22:44.33 | [TK]D-Fender | TheIzkabola: I've already answered that rather precisely. |
| 22:44.36 | _Sam-- | angryuser : im the same way, no gui, but if a person wanted one, i dont see how it is a deal breaker. |
| 22:44.38 | ManxPower | ~zeeek |
| 22:44.41 | jbot | rumour has it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 22:45.11 | mshades | sounds sensual |
| 22:46.06 | TheIzkabola | fender: thanks, I don't know how to set the runlevel at 5 though |
| 22:46.31 | jblack | serialthrilla: Yes, there's the AMI. That's more difficult than you're looking for. |
| 22:46.57 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:47.05 | Hydrant | hey all... I want to setup something with asterisk to prompt for a phone number to call, then call that number using asterisk... so I basically want a call bridge... I call that bridge, then enter the number I really want to call over voip |
| 22:47.08 | Hydrant | any ideas / suggestions ? |
| 22:47.09 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 22:47.18 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:48.44 | [TK]D-Fender | TheIzkabola: /etc/inttab <--- |
| 22:48.45 | jameswf-home | jbot: tell Hydrant about book |
| 22:49.01 | jameswf-home | ~ch4 |
| 22:49.06 | Hydrant | is this book all I need for such a thing ? |
| 22:49.06 | TheIzkabola | k thanks |
| 22:49.07 | *** join/#asterisk vxworks (n=vxworks@189.81.176.64) |
| 22:49.22 | jameswf-home | the book knows all |
| 22:49.24 | jameswf-home | ~ch5 |
| 22:49.25 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 22:49.26 | TheIzkabola | What are people's opinion of trixbox? |
| 22:49.29 | jblack | hydrant: Yeah, just play a message, do a read, then dial what was read. |
| 22:49.29 | jameswf-home | tada |
| 22:49.37 | ManxPower | ~trixbox |
| 22:49.38 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:49.41 | [TK]D-Fender | Hydrant: You need to understand the dialplan and how to setup each interface * will get/pace calls from/to |
| 22:50.05 | jblack | TheIzkabola: Here, it's generally seen as the short bus of phone systems. |
| 22:50.13 | Hydrant | alright, was just hoping there was an idiot proof module |
| 22:50.30 | TheIzkabola | lol |
| 22:50.53 | jameswf-home | Hydrant: asterisk is a get your hands dirty sorta thing |
| 22:51.05 | jblack | Hydrant: It's all of 3 lines in a dialplan. You can do it. |
| 22:51.23 | Hydrant | any tutorials / blogs that have such a thing? |
| 22:51.26 | vxworks | is it easy to write an addon to asterisk ? |
| 22:51.30 | jameswf-home | oi |
| 22:51.30 | Hydrant | I have a million 3-line things to do |
| 22:51.30 | jblack | ~book |
| 22:51.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:51.44 | jblack | Hydrant: There's the whole book for you, for download at no cost. |
| 22:51.47 | angryuser | Hydrant do you know tha basics how to setup dialpans? i can give you the code and all you need is to adapt |
| 22:51.58 | jameswf-home | Hydrant: you could hire a consultant |
| 22:52.03 | ManxPower | Hydrant: now you have have 3.1 million things to do. Callbacks are not easy for newbies, not easy at all |
| 22:52.18 | Qwell | 3.000001 million |
| 22:52.34 | jameswf-home | ManxPower: he doesnt want callback he wants passthrough |
| 22:52.35 | jblack | gives jameswf-home a nasty look that is completely unrelated to hydrant. |
| 22:52.39 | Hydrant | hard to hire a consultant that I wouldn't write off as an idiot |
| 22:52.42 | jameswf-home | yay |
| 22:52.48 | angryuser | ManxPower i consider callback as easy really, if i understood ho to generete a call file, everybody can ;) |
| 22:52.51 | Hydrant | besides, I'm not a business just a guy trying to rip off his cell phone company |
| 22:53.21 | Hydrant | ... buy plan to call / receive calls from certain numbers.... one of those numbers is the voip bridge, so low-cost calls :-) |
| 22:54.40 | jblack | Speaking of getting ripped... verizon bumped my DSL from $35 a month to $42 |
| 22:55.05 | angryuser | jblack what speed ? |
| 22:55.07 | _Sam-- | the ripped part...is that you cant get fios :) |
| 22:55.22 | jblack | Ohhh. 3.0/768 or some such. |
| 22:55.23 | ManxPower | angryuser: for an experienced user -- 30 mins at the most. |
| 22:56.02 | _Sam-- | up there in beautiful lackawanna county |
| 22:56.03 | jblack | Not in Wilkes-Barre, Mr. "I have too much trust in dns" |
| 22:56.23 | _Sam-- | like you aer spoofing a verizon ip on irc. |
| 22:56.27 | _Sam-- | maybe 10 years ago. |
| 22:56.32 | _Sam-- | or maybe you are proxied through it. |
| 22:56.45 | _Sam-- | but you or someone else you know are connected from there. |
| 22:57.08 | jblack | No, they feed Wilkes-Barre through the Scranton DSLAM, you dork. |
| 22:57.34 | _Sam-- | sorry, i live 4 hours from there...where i live ...scranton and W.B. are one metro area. |
| 22:57.46 | Qwell | Scranton doesn't have a DSLAM. |
| 22:57.57 | angryuser | 42$ that would be 28E hm it depends where you are i suppose , same price in europe, but a bit faster |
| 22:58.14 | jblack | fair enough, qwell. |
| 22:58.23 | Qwell | I have no idea what Scranton is. |
| 22:58.31 | ManxPower | angryuser: it is a very good price for that speed in the USA |
| 22:58.35 | jblack | It's a city about 15 miles from where I live. |
| 22:58.41 | ManxPower | In much of the USA it can be double that. |
| 22:59.09 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 22:59.26 | jblack | I miss San diego. I could have fios by now. |
| 22:59.42 | JT | usa, europe, all very cheap Internet |
| 22:59.55 | _Sam-- | that is a bad deal there -- from SD to WB!! |
| 22:59.58 | jblack | i hear Japan is cheaper still. |
| 23:00.03 | _Sam-- | at least if you are making the same money, you are doing ok :) |
| 23:00.13 | jblack | As it happens, I am. |
| 23:00.20 | angryuser | ManxPower major people get 8mbit/1mbit and for the cityes more than 20k 25mbit/2mbit in major cityes 500k 70mbit/5mbit , and all the same price 30E |
| 23:00.31 | jblack | Don't have have some paper to rumple in a handset or something? |
| 23:00.32 | angryuser | it's for france |
| 23:00.32 | JT | japan is faster as VDSL2+ is widely deployed |
| 23:00.34 | _Sam-- | you would have to make double for the quality of live to be the same :) |
| 23:00.47 | _Sam-- | s/v/f |
| 23:01.10 | _Sam-- | i finished rumpling all my papers a few minutes ago |
| 23:01.16 | _Sam-- | fax me some more |
| 23:02.11 | mshades | is your name rust limbaugh |
| 23:02.24 | _Sam-- | no wonder you are a miserable bastard...i would be to if i lived in WB after moving there from la jolla. |
| 23:02.47 | jblack | Now, why would you think I lived in La Jolla ? |
| 23:03.22 | _Sam-- | because you are trying to portray yourself as smart and a man of means...where else in SD would you live if that were the case? |
| 23:03.37 | jblack | i don't claim the former, and I'm not the latter. |
| 23:04.07 | _Sam-- | either way, im just making small talk, killing time, waiting for people to pack up the shit and hit the road over here so i can go home...sorry to be offensive, its natural. |
| 23:05.08 | watchy | kisses everyone |
| 23:05.08 | jblack | I'm a borderline psychotic that moved from San Diego to Wilkes-Barre to stretch my disability. I sit here on irc and try to not decide who i want to kill next. |
| 23:05.14 | jblack | SPeaking of which, wanna go out for some coffee? |
| 23:05.16 | jaytee | La Jolla is in South Dakota? shit! I gotta get me a new map, this one's screwed up. it says it's in California. |
| 23:05.17 | watchy | jblack you can kill me |
| 23:05.31 | _Sam-- | SD = san diego |
| 23:05.32 | watchy | anyone here near shreveport louisiana |
| 23:05.58 | _Sam-- | jblack : sure , but you have to bring copa lua |
| 23:06.03 | Katty | eww. |
| 23:06.04 | jaytee | I don't want to kill anyone but I'd like to see my Republican senator die of terminal diarrhea. |
| 23:06.07 | Katty | watchy germs. |
| 23:06.48 | _Sam-- | kopi luwak, sorry, bad spelling |
| 23:06.55 | _Sam-- | bring some of that, we'll meet for coffee next time im at elk. |
| 23:07.12 | jaytee | I love coffee but I'd never drink anything that went through a civet's intestines |
| 23:07.30 | _Sam-- | i like to eat the beans raw. |
| 23:07.39 | watchy | yea im pretty germy katty, but i have on cheery chapstick |
| 23:07.54 | jaytee | why not just feed your cat some beans and save the money? |
| 23:08.18 | _Sam-- | because my cat isnt an indonesian monkey! |
| 23:08.32 | jaytee | civets aren't monkeys |
| 23:08.50 | _Sam-- | not a cat either. but yo're right it is cat-like. |
| 23:08.52 | _Sam-- | The animal is a palm civet, a dark brown tree-dwelling cat-like creature found throughout Southeast Asia. The scientific name is paradoxurus hermaphroditus. |
| 23:09.23 | _Sam-- | this one says they are monkeys |
| 23:09.24 | _Sam-- | Brits are flooding exclusive US stockists with orders for the brew, called |
| 23:09.24 | _Sam-- | Kopi Luwak, made from berries that have passed through the digestive |
| 23:09.25 | _Sam-- | system of Indonesian monkeys. |
| 23:10.29 | jaytee | I'm a coffee nut and I've known about kopi luwak for years. The brits are idiots and need to check their facts, the only thing civets have in common with monkeys is they're both mammals and thats about it. |
| 23:11.21 | watchy | man polycom 650 with backlit displays are so fin nice |
| 23:12.30 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 23:12.34 | watchy | it makes me want to diddle myself at my clients office |
| 23:13.09 | jaytee | now there's an image I didn't need |
| 23:13.14 | Kobaz | how |
| 23:13.17 | Kobaz | er |
| 23:13.25 | Kobaz | how would i do a split into multiple variables |
| 23:13.41 | *** join/#asterisk xiando (n=xiando@2a01:48:219:b:0:0:0:1) |
| 23:13.44 | Kobaz | like, i have ivrMain,s,1 stored in a var... i want to split on "," and pass them in as seperate parameters |
| 23:13.56 | _Sam-- | jblack : if you want my script to randomly turn off your phones, just let me know, i offer it to you in good faith! |
| 23:14.08 | [TK]D-Fender | Kobaz: "core show function CUT" |
| 23:14.13 | ManxPower | Kobaz: "core show function CUT" caps are important |
| 23:14.16 | Kobaz | k |
| 23:14.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:14.29 | watchy | wow |
| 23:14.39 | watchy | i think OSLEC fixed my echo on this tdm400p |
| 23:14.51 | watchy | i owe tk yet another make out session and a steak |
| 23:15.57 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 23:15.59 | ManxPower | watchy: you don't have a TDM400P |
| 23:16.18 | ManxPower | If you did you could get the free HPEC. |
| 23:16.30 | [netman] | where? |
| 23:16.31 | grandpapadot | Hey all. In Asterisk 1.2, does app_voicemail.so ignore attach=no in voicemail.conf? I ask because we have it set to no but asterisk continues to send email notifications if the mailbox has an email address defined. |
| 23:16.40 | watchy | of fcourse i have a tdm400p |
| 23:16.45 | watchy | why would i lie |
| 23:16.54 | watchy | i maybe fat but im no lier |
| 23:17.07 | ManxPower | (6:22:42 PM) watchy: its a sangoma manx |
| 23:17.12 | watchy | differnet box |
| 23:17.17 | watchy | tdm400p is here at work |
| 23:17.21 | watchy | sangoma is at clients |
| 23:17.26 | watchy | Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) |
| 23:17.26 | watchy | Registered tone zone 0 (United States / North America) |
| 23:17.34 | watchy | see fat people arent always liers |
| 23:17.37 | watchy | just most of the time |
| 23:17.44 | ManxPower | [netman]: Digium analog cards under warrenty get the HPEC for free. |
| 23:17.45 | jaytee | but they take up more room |
| 23:17.46 | _Sam-- | liar is how you spell it, for the future. |
| 23:17.55 | watchy | thanks sam |
| 23:18.06 | _Sam-- | plier is your way |
| 23:18.07 | watchy | i graduated from arkansas, so i hope you understand |
| 23:18.24 | _Sam-- | i work with a PHP programmer who graduated from UA little rock -- i understand totally. |
| 23:18.30 | jaytee | what do girls from Arkansas and bear cubs have in common? |
| 23:18.37 | watchy | no idea |
| 23:18.45 | jaytee | they both like to suck their paws |
| 23:18.53 | watchy | haha damn thats bad |
| 23:19.24 | watchy | anyways. manx i installed hpec before but it was annoyign registering the cards etc. i just wanted to try oslec |
| 23:19.44 | jaytee | what's a tornado in Oklahoma and a couple getting a divorce in Arkansas have in common? Either way, someone's gonna lose a trailer! |
| 23:19.49 | [netman] | ManxPower: thx |
| 23:19.59 | watchy | i dont live in a trailer jaytee:( |
| 23:20.28 | watchy | lastnight at 3am the tornado sirens went off for no reason, it sure psised me off |
| 23:20.30 | jaytee | that's good! cuz you know they magnetically attract tornados. |
| 23:21.09 | grandpapadot | It even ignores it if I put attach=no in the mailbox options ... argh |
| 23:21.12 | [netman] | ManxPower: but HPEC is only suitable to analog cards or also to T1/E1 cards? |
| 23:21.38 | watchy | i wouldnt personally buy a new card without hw echo cancellation |
| 23:21.46 | watchy | but the guy who sold this phone system decided to |
| 23:22.02 | jaytee | Digium's cards with HW EC rock |
| 23:22.21 | watchy | i like my sangomas with HE EC |
| 23:22.23 | watchy | HW |
| 23:22.34 | watchy | anyone here play with a polycom 650? |
| 23:22.56 | jaytee | I tried playing Gears of War on it but the graphics sucked |
| 23:23.05 | watchy | haha |
| 23:23.08 | jaytee | and the game kept crashing |
| 23:23.25 | watchy | im gonna try to write a site i can use on the phone to vote for lunch |
| 23:23.33 | jaytee | but I've programmed and used a 550 and it performs nice as a phone. |
| 23:24.01 | watchy | we sell 330s and 650s now |
| 23:24.24 | watchy | i think the backlit displays of the 650s make a world of difference |
| 23:25.05 | jaytee | I love Polycoms but their website is retarded and their documentation is crap. |
| 23:26.24 | watchy | there documantion is pretty good in the admin guides for provisioning |
| 23:27.15 | jaytee | I beg to differ. I've seen better docs written by Microsoft |
| 23:27.26 | watchy | well i mean not on what to do to provision |
| 23:27.35 | watchy | but on what needs to go into sip.cfg and mac.cfg |
| 23:27.39 | watchy | that stuffs pretty good |
| 23:28.40 | jaytee | just on the configurable parameters they give one basic template with no other examples and no clear explanation. |
| 23:28.47 | watchy | Singer Winehouse admitted to hospital |
| 23:28.51 | watchy | i hope that bitch dies |
| 23:29.00 | watchy | i seriously hate her more then anything |
| 23:29.14 | watchy | it just proves you can literally smoke crack and be given awards |
| 23:29.20 | jaytee | hate is a pretty strong emotion to waste on someone you've probably never met. |
| 23:29.31 | watchy | oh dude trust me i hate her |
| 23:29.47 | watchy | shes just another wasted celebrity that thinks they can get away with anything |
| 23:29.50 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples |
| 23:30.12 | jaytee | trust is something earned, not freely given except by fools and hate usually does more to harm the hater than it does the hated. |
| 23:30.23 | watchy | probably |
| 23:30.38 | watchy | but ive put $50 down she dies in the next 2 years at a betting site |
| 23:30.42 | watchy | im just hoping i hit |
| 23:31.04 | watchy | as many drugs and shes doing currently and as bad of health as shes in im sure ill hit |
| 23:31.44 | watchy | whats funny is her famous song is called Rehab |
| 23:33.13 | watchy | ok im closing my office before someone else walks in and bothers me |
| 23:33.16 | watchy | good day folks |
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| 23:40.57 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:55.42 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 23:55.56 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 23:56.20 | TheIzkabola | Does anyone know how to install kernel-smp-devel on CentOS? I tried "yum install -y kernel-smp-devel" but it could not find the package... |
| 23:58.26 | outtolunc | you always need to determine what is in the repo you are using.. yum list kernel* |
| 23:59.40 | outtolunc | repo(s) |
