| 00:01.29 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 00:02.03 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:03.56 | adamowitz | ping jamincollins |
| 00:05.54 | *** join/#asterisk RoyK (n=roy@cm-84.210.18.089.chello.no) |
| 00:06.10 | niter3 | Anyone here setup a SIP trunk.. |
| 00:06.27 | niter3 | I have registered with a SIP provider and would like to get it so I can place outgoing calls with it in Asterisk. |
| 00:06.30 | *** join/#asterisk saftsack (n=saftsack@p54A7C9D0.dip.t-dialin.net) |
| 00:06.34 | niter3 | http://www.cliconnect.com/br/Asterisk_sample_pt.html Is all the info I can get from the compnay. |
| 00:06.39 | niter3 | any help would be really appreciated. |
| 00:07.37 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 00:12.58 | adamowitz | anyone here use smokeping |
| 00:13.14 | adamowitz | ? |
| 00:13.17 | *** join/#asterisk snappy (i=naveen@zerowing.calpop.com) |
| 00:13.19 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 00:14.38 | snappy | A week ago or so I was advised to move from our existing infrastructure (which is a t1 trunk and several analogue phones connected to a punchblock with an amphenol connector) that to move to Asterisk the minimum equipment I would require is a TDM2400p (for the analgoue lines) and a TE110p for the T1 trunk. |
| 00:15.44 | snappy | What I'm wondering is, how are phones 'addressed' in Asterisk for the lines going into the TDM2400P. Is it actual 'ports' that are addressed? |
| 00:16.38 | mog | inside of asterisk you will adress the phones as zap/XX for the channel it is |
| 00:20.48 | `Tingles` | anyone using gentoo and asterisk? where are the tools such as addmailbox... the scripts are not in the place all the documents say they should... |
| 00:21.30 | wunderkin | `Tingles`, it has not been needed for a long time |
| 00:22.04 | `Tingles` | I am trying to do a test of setting up voicemail.. all documentation refers to that script |
| 00:24.49 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.91) |
| 00:27.19 | jamincollins | adamowitz: pong |
| 00:27.22 | niter3 | can anyone help me setup a SIP trunk |
| 00:27.42 | jamincollins | adamowitz: I use smokeping |
| 00:27.47 | adamowitz | :) |
| 00:27.49 | adamowitz | hey |
| 00:28.00 | adamowitz | smokeping: wow, that's a really impressive tool! |
| 00:28.10 | jamincollins | I like it |
| 00:28.30 | jamincollins | comes in handy for indicating if there's a latency problem or not |
| 00:28.47 | adamowitz | It's something that I've been wanting for a long time and had no idea one existed. |
| 00:28.57 | adamowitz | thanks so much for the reference. |
| 00:29.01 | jamincollins | np |
| 00:29.14 | adamowitz | I have been reading the docs (extensive and detailed). |
| 00:29.14 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:29.20 | jamincollins | between it an munin, most monitoring bases are covered |
| 00:29.35 | adamowitz | but do you know of an "Intro" type document? That gets and the concepts, how it works, etc. |
| 00:29.37 | `Tingles` | wunderkin> what should i be using or doing.. |
| 00:29.41 | EzWayz | do you beleive htb init script is a good way priorizing ? |
| 00:30.09 | jamincollins | adamowitz: only futz'd with the config file for most of what I need |
| 00:30.33 | jamincollins | I haven't even scratched the surface of having it alert me or anything like that |
| 00:31.28 | wunderkin | `Tingles`, well its not difficult and not much has changed, just follow the rest probably, all you need is in voicemail.conf and extensions.conf |
| 00:32.55 | adamowitz | jamincollins: ok. i noticed in the config file with sections and subsections for checking latency of places all over the world, and I like the idea of that, but not knowing exactly what smokeping will do to those places, I'm hesitant to list them in the config file as targets. What hosts do you have as targets. Only service providers for * or others too? |
| 00:33.37 | jamincollins | it sends a smallburst of packets over a 5 minute period as I understand it |
| 00:33.59 | adamowitz | and I guess it varies based upon the probe you use: echoping, fping, other... |
| 00:34.02 | jamincollins | I normally list google.com, yahoo.com, and the points I'm interested in |
| 00:34.40 | jamincollins | the first two acting for me as a bit of a "control group" |
| 00:35.04 | jamincollins | so I can get a sense of whether the problem is with a specific host or more widespread |
| 00:35.24 | adamowitz | good idea. |
| 00:35.33 | jamincollins | we also use it at work to track latency on remote employee WAN connections |
| 00:35.34 | *** join/#asterisk RoyKa (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 00:36.16 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 00:37.13 | kusznir | Hi all: I've been trying to find out how to (if possible) configure a SIP user such that multiple devices can connect symultanously to the same account (ie, I can have 2 different phones both be the same extention, like what FWD does) |
| 00:37.56 | jamincollins | kusznir: it is not necessary to have two devices register under one account for this |
| 00:38.17 | jamincollins | you can have a single extension alert multiple SIP users/devices at one time |
| 00:38.53 | jamincollins | exten => 123,1,Dial(SIP/dev1&SIP/dev2) |
| 00:39.02 | jamincollins | that's off the top of my head |
| 00:39.07 | niter3 | is anyone around.. I need a bit of help. I'm trying to setup a SIP outbound trunk..... |
| 00:39.13 | kusznir | Ok. I think I've done that before, but it seemed kinda messy. |
| 00:39.15 | niter3 | I have the provider and all the stuff, but I cannot figure thsi out. |
| 00:39.35 | jamincollins | kusznir: messy? |
| 00:40.54 | kusznir | The fact I've got to go and edit multiple files to connect a second device. It seems a lot "cleaner" and "simpler" to just connect with the same username and pass. For example, I'm away and want to use my phone. I can download a softphone client, connect with my username & pass, and I'm there...no logging in via ssh and editing multiple files to add a connection. |
| 00:42.15 | jamincollins | kusznir: edit the files one time, if the 2nd device isn't present the 1st is alerted |
| 00:42.28 | jamincollins | if the 2nd is present it's alerted two |
| 00:42.35 | jamincollins | err too |
| 00:43.06 | jamincollins | I've got my household alert set to ring 4 places |
| 00:43.15 | jamincollins | only two of which are normally present |
| 00:43.31 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.17) |
| 00:43.32 | kusznir | ok. |
| 00:43.34 | jamincollins | then I can log in a softphone on my laptop or my wife can |
| 00:46.21 | kusznir | What about dialing extentions. We have a current setup where a phone in one room is "70". I now want 70 to ring another device as well. Can I still remap it like that since 70 is already "owned" by an actual extention? |
| 00:47.00 | jamincollins | that 70 is in extensions.conf with a Dial command, right? |
| 00:47.09 | kusznir | SIP/70 |
| 00:47.22 | jamincollins | just add the other to that dial command |
| 00:48.03 | jamincollins | exten => 70,1,Dial(SIP/70&SIP/dev2) |
| 00:48.10 | kusznir | Ahh..Yes, I just found that. |
| 00:48.13 | kusznir | cool. |
| 00:48.40 | jamincollins | I've taken to using global variables for my alert groupgs |
| 00:48.45 | jamincollins | err groups |
| 00:48.56 | jamincollins | so I can change it in one place |
| 00:55.08 | niter3 | anyone around that can help me with this? |
| 00:59.02 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 00:59.37 | niter3 | I've signed up with a SIP VoIP provider. I would like to have Asterisk function with it so anyone on my PBX can use this provider to make outgoing calls.. How can I do this? |
| 00:59.48 | niter3 | I don't just want one phone that's capable of using it, i want everyone |
| 00:59.56 | niter3 | So it's an SIP outbound trunk |
| 01:03.20 | *** join/#asterisk Here_And_There (n=Here_And@pool-71-246-0-110.phlapa.fios.verizon.net) |
| 01:05.04 | kusznir | niter3: see http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 01:06.14 | kusznir | BTW: many providers will offer asterisk config snippits for using them. They are usually located under the support portion of their site (try searching for asterisk sip on their site) |
| 01:14.05 | *** part/#asterisk voipman (i=distorti@junipero.3sheep.com) |
| 01:15.38 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:19.15 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:19.36 | niter3 | kusznir: got that to work. How can I make just an extension like 222 to dial out to a remote phone? |
| 01:20.06 | kusznir | you mean, from inside you dial "222", and it places a call to 1-234-567-8912? |
| 01:21.01 | niter3 | nevermind I go it |
| 01:21.42 | *** join/#asterisk Luke-Jr_work (n=ubuntu@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 01:21.49 | Luke-Jr_work | Debian + Asterisk = trouble :( |
| 01:21.58 | Luke-Jr_work | will need to switch it to Gentoo I guess |
| 01:28.00 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 01:28.56 | [TK]D-Fender | Luke-Jr_work : Dunno about that.. Debian works great from what I've heard. Whats not going so well for you with it? |
| 01:30.09 | Luke-Jr_work | [TK]D-Fender: broken init scripts, it expected a user to exist (but didn't add it), wrong permissions, etc |
| 01:30.23 | Luke-Jr_work | oh, and drivers |
| 01:31.03 | Qwell | welcome to binary packages |
| 01:31.21 | [TK]D-Fender | Luke-Jr_work : Yes... Binary packages = trouble, not Debian. |
| 01:31.32 | [TK]D-Fender | Use the Source Luke! |
| 01:32.41 | Luke-Jr_work | Debian is binary |
| 01:32.53 | Luke-Jr_work | ... |
| 01:32.59 | Luke-Jr_work | it also installed sounds into the wrong directory |
| 01:33.34 | hacked`` | guys |
| 01:33.36 | hacked`` | you know asterisk |
| 01:33.53 | hacked`` | is there any way to make like a script, to like, get a customer's date of payment and tell it to them |
| 01:33.58 | hacked`` | which is stored in a mysql table |
| 01:34.00 | Luke-Jr_work | ...... |
| 01:34.04 | Luke-Jr_work | yes |
| 01:34.09 | hacked`` | ya, but how |
| 01:36.08 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 01:36.46 | hacked`` | i mean, not how |
| 01:36.55 | hacked`` | but like, with what |
| 01:37.03 | hacked`` | i mean, is there a book that i can buy |
| 01:37.18 | hacked`` | i hate having to learn everything on line |
| 01:37.26 | hacked`` | i would rather just buy 1 book, and learn from that |
| 01:37.32 | hacked`` | but does such literature exist |
| 01:37.38 | hacked`` | if not, why not |
| 01:38.59 | Qwell | ~book |
| 01:39.09 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:39.09 | hacked`` | ya |
| 01:39.46 | hacked`` | will that show me how to write scripts that talk to my mysql table ? |
| 01:39.58 | Qwell | it will teach you asterisk |
| 01:40.04 | hacked`` | i dont want to learn asterisk |
| 01:40.09 | hacked`` | i want to learn how to make asterisk scripts |
| 01:40.14 | hacked`` | what "language" do i have to learn, to do what i want to do |
| 01:40.24 | Qwell | You can do it all in dialplan |
| 01:40.32 | hacked`` | o'reilly ? |
| 01:40.46 | hacked`` | but dont i have to connect to the database ? |
| 01:40.53 | Nugget | there isn't a database. |
| 01:40.59 | Nugget | there can be, but usually there isn't. |
| 01:41.07 | Qwell | hacked``: func_odbc |
| 01:41.09 | hacked`` | what do you mean there isnt a database |
| 01:41.12 | hacked`` | i have my own database |
| 01:41.32 | Nugget | then what do you need asterisk for? :) |
| 01:41.54 | file | Nugget: . |
| 01:41.57 | file | Qwell: .. |
| 01:42.01 | Qwell | file: ... |
| 01:42.07 | wunderkin | file: !!! |
| 01:42.10 | Nugget | I am CURRENT DIRECTORY! |
| 01:42.12 | Qwell | wunderkin: showoff |
| 01:42.23 | wunderkin | <3 |
| 01:42.31 | file | Nugget: hrm! Nugget != Nugget? |
| 01:42.40 | Qwell | mv Nugget . |
| 01:43.02 | hacked`` | k, caller calls, system sees caller's phone number, searches mysql table to find if caller is a customer, if so, asks for his dob, validates dob from mysql table, once authenticated, reads out customer's date of payment |
| 01:43.06 | hacked`` | thats what i want |
| 01:43.13 | hacked`` | is there some sort of asterisk language ? |
| 01:43.20 | Qwell | ...dialplan |
| 01:43.20 | hacked`` | what language is dialplan |
| 01:43.27 | Qwell | It isn't a language |
| 01:43.27 | anonymouz666 | hacked``: ASM |
| 01:43.30 | Qwell | ~book |
| 01:43.33 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:43.33 | Nugget | the dialplan is its own thing. |
| 01:43.51 | hacked`` | Asterisk: The Future of Telephony is an e-book |
| 01:43.56 | hacked`` | i dont do e-books |
| 01:44.00 | Nugget | so print the shit out, jeez. |
| 01:44.06 | Qwell | Or BUY it |
| 01:44.06 | wunderkin | or buy it |
| 01:44.13 | hacked`` | k |
| 01:44.17 | wunderkin | Qwell: !!! |
| 01:44.37 | hacked`` | $55 CAD |
| 01:44.42 | hacked`` | thats crazy, for something that is free |
| 01:44.51 | Qwell | ...it's not free |
| 01:45.00 | Qwell | printing is expensive |
| 01:45.07 | file | good golly, you're awfully picky |
| 01:45.23 | wunderkin | good golly miss molly? |
| 01:45.33 | file | quite |
| 01:45.35 | hacked`` | thats why they make the book available for free via e-book, because they know nobody is gonna print it |
| 01:45.36 | anonymouz666 | hacked``: drink a beer, forget about the book. |
| 01:45.40 | wunderkin | contrary |
| 01:46.06 | Qwell | hacked``: If you want a hard copy - buy it |
| 01:46.15 | Qwell | if you want to read it on your computer - get the pdf |
| 01:46.52 | hacked`` | which route did you take |
| 01:46.57 | hacked`` | the pdf one ? |
| 01:46.58 | Qwell | neither |
| 01:47.03 | anonymouz666 | but If you don't wnat to buy a hard copy because you think it's expensive and don't read it on your computer because you don't like e-books. drink a beer. lol. |
| 01:47.05 | Qwell | I got a free copy - signed :p |
| 01:47.22 | *** join/#asterisk LoneShadow (n=duh@59.92.192.14) |
| 01:47.22 | Nugget | anonymouz666 has the right idea. |
| 01:47.30 | tzanger | heh |
| 01:47.31 | file | Nugget: slacker. |
| 01:47.33 | hacked`` | signed by who |
| 01:47.38 | hacked`` | bill o'reilly himself ? |
| 01:47.43 | Qwell | hacked``: the authors |
| 01:47.44 | LoneShadow | anyone in here who is good with Sipura 3000 PSTN setup ? |
| 01:47.47 | tzanger | you don't want to summon him |
| 01:47.50 | Qwell | o'reilly's signature is fairly worthless |
| 01:47.55 | tzanger | last time I saw him he was going on about the unfairness of wearing pants |
| 01:47.59 | tzanger | and he was going to do something ugly about it |
| 01:48.02 | tzanger | such as taking htem off |
| 01:48.11 | Qwell | tzanger: sounds about right |
| 01:48.25 | Qwell | I'll have to question him about it next week |
| 01:48.41 | hacked`` | i'll just buy a used copy off amazon.ca, hopefuly nobody pissed on it |
| 01:48.52 | file | I have to probably... pack a bit... sometime |
| 01:48.57 | Qwell | file: why? |
| 01:49.06 | Qwell | You don't need to change |
| 01:49.10 | file | Qwell: good point, it won't end up there anyway |
| 01:49.15 | Qwell | heh, that too |
| 01:51.01 | anonymouz666 | a good song? whole lotta love. |
| 01:53.59 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 02:02.48 | [TK]D-Fender | hacked``: E-Book is expensive? Well TFOT is 376 pages and a corporate class laser printer is at MOST 1.2c ($CAD) per page so that = $4.50 plus 75% or a ream of paper ($5 TOPS) for a grand total of 8.25$ CAD to print TFOT for yourself. So STOP WHINING! |
| 02:03.24 | [TK]D-Fender | s/or/of/ |
| 02:03.43 | ATravelingGeek | haha, that's a sweet bot command |
| 02:03.53 | [TK]D-Fender | yeah, it just FAILED... |
| 02:07.27 | EzWayz | i was speaking with some one then the call dropped : Didn't get a frame from channel:ZAP1 , how to solve this problem ???? |
| 02:09.35 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 02:12.59 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 02:13.22 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 02:15.23 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 02:26.47 | snappy | mog: Sorry, I just read the response from 2 hours ago. What do you mean by 'channel'? Is each line on the punchblock (I know it's our own environment) considered a channel? The t1 we have I believe gives us 24 lines. |
| 02:28.57 | kusznir | Quick question: is there a shortcut when making extension dialing rules to basically achieve these two lines in one? exten => 1NXXNXXXXXX; exten => NXXNXXXXXX |
| 02:28.59 | [TK]D-Fender | snappy : Yes each "line" in a digital link is a channel. |
| 02:29.12 | kusznir | (basically, make it match an optional 1 in front)? |
| 02:29.16 | [TK]D-Fender | kusznir :IIRC that is not possible with * dialplan |
| 02:29.24 | [TK]D-Fender | kusznir : You'd have to do it in 2 steps |
| 02:29.45 | kusznir | ok. Wasn't sure; figured I'd ask :) |
| 02:30.00 | [TK]D-Fender | snappy : As for the number of channels in your T1, that depends on signalling. |
| 02:30.15 | kusznir | Oh, and what does the _ in front of a number do again? |
| 02:30.47 | [TK]D-Fender | kusznir : it says that what follows is a pattern match |
| 02:31.03 | [TK]D-Fender | kusznir : Which you NEED for the 2 samples you wrote above |
| 02:31.12 | kusznir | ahh...OK. That explains why some of my extensions wern't working :) |
| 02:31.23 | [TK]D-Fender | kusznir : exten => _1NXXNXXXXXX; exten => _NXXNXXXXXX |
| 02:33.00 | ATravelingGeek | has anyone experienced 'phantom' ringing where asterisk will seemingly ranomdly detect an incomming call, but when one picks up it's a dialtone? |
| 02:33.18 | ATravelingGeek | let me get the details quickly |
| 02:33.41 | snappy | [TK]D-Fender: I see. But these will be analogue lines connected to a TDM2400p. How is it digital at all is what I'm wondering? Since they will be fxs/fxo ports |
| 02:34.24 | ATravelingGeek | We have a TDM400p, and reading the logs it seems that the line's polarity is reversed twice for this phantom call |
| 02:34.34 | [TK]D-Fender | snappy : T1 = Digital. that you plan to use a channel-bank or something else to down-convert them is secordary |
| 02:34.38 | ATravelingGeek | yet when a legit call is detected, the polarity only shows up reversing once |
| 02:35.31 | [TK]D-Fender | snappy : And if you have a T1 coming in you don't WANT a TDM2400. You should take the T1 in directly. |
| 02:36.16 | file | [TK]D-Fender: up to no good? |
| 02:36.20 | snappy | [TK]D-Fender: I apologize, I didn't give all the details. Our phones are analogue, connected by phonelines to a punchblock. |
| 02:37.29 | [TK]D-Fender | snappy : Well you said you had a T1. Where does that come into play? |
| 02:37.31 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 02:37.32 | snappy | [TK]D-Fender: Our current setup is a Adtran CSU/DSU connected to a punchblock, which then connects to a panasonic PBX which has never been (and probably never will be) configured. However each phone has their own extension and such. We want to move the existing infrastructure to asterisk. The T1 card will replace the CSU/DSU. |
| 02:37.45 | snappy | [TK]D-Fender: (the t1 connects to the CSU?DSU) |
| 02:38.00 | [TK]D-Fender | snappy : Which T1 card? |
| 02:38.06 | snappy | [TK]D-Fender: Digium TE110P |
| 02:38.35 | [TK]D-Fender | snappy : Your Adtran will be removed from this equation completely. You would plug the T1 line directly into the TE110P |
| 02:38.36 | snappy | [TK]D-Fender: At least that is what we're considering. But to connect our existing analogue phones, we need to connect using the TDM2400P (with the amphenol connector) |
| 02:39.28 | file | or you can get a dual port card and use a channel bank, up to you |
| 02:39.29 | snappy | [TK]D-Fender: What I'm wondering is, is it possible to logically group phones, and still provide phones with their own internal extensions? |
| 02:39.29 | *** join/#asterisk ATravelingGeek (n=atg@pdpc/supporter/student/ATravelingGeek) |
| 02:39.51 | [TK]D-Fender | snappy : Ok, becoming clear now. Yes the TDM2400P will let you use up to 24 analog phones per card. If you are referring to Panasonc phones belonging to your old PBX make sure they are fully functional as analog phones and not proprietary digital phones. |
| 02:39.59 | [TK]D-Fender | file : Getting by... |
| 02:40.49 | *** join/#asterisk ATravelingGeek (n=atg@pdpc/supporter/student/ATravelingGeek) |
| 02:40.55 | [TK]D-Fender | snappy : Sure. For your T1 though I'd seriously get your telco to set it for PRI signalling. That will allow you to use DIDs to direct incoming calls. |
| 02:41.01 | snappy | [TK]D-Fender: Okay. The panasonic phones in the brochure claim they can double as a regular analogue phone, but can connect to a Panasonic PBX and do some proprietary stuff. This is what I'm fairly confused about. |
| 02:41.08 | [TK]D-Fender | snappy : And * can do whatever you want with any given call. |
| 02:41.33 | [TK]D-Fender | snappy : it could be that they are dual-mode. Take one to a normal analog phone line and try it out. |
| 02:41.37 | snappy | [TK]D-Fender: Excellent. I will check with the telco first. |
| 02:41.49 | snappy | ah, excellent idea, thanks |
| 02:41.51 | [TK]D-Fender | snappy : though mind you that'd the the first time I'd have heard of a company making one like that. |
| 02:44.30 | snappy | Not a common configuration? Would like to get some VOIP/SIP-capable phones, but I think first thing is to get something working with our existing infrastructure if possible. |
| 02:44.31 | file | there are no phones... only magical boxes that make irritating sound |
| 02:44.50 | snappy | too true, too true |
| 02:45.06 | [TK]D-Fender | snappy : How many phones do you have? |
| 02:45.35 | snappy | we have about 15 |
| 02:46.14 | [TK]D-Fender | snappy : What cards have you already purchased? |
| 02:46.40 | snappy | Well none at the moment, but my boss is going to order the te110p & tdm2400p tomorrow |
| 02:47.39 | [TK]D-Fender | snappy : I might strongly recosider the TDM2400P choice to support your old phones. This iwll be a kludgy feeling setup at best. |
| 02:48.01 | [TK]D-Fender | snappy : SIP phones will offer you much better functionality at a price not much higher. |
| 02:48.25 | file | but can you use the SIP phones as a weapon? a TDM2400P with no modules is very handy at that |
| 02:48.30 | [TK]D-Fender | snappy : Figure in $150/set for a nice setup. |
| 02:48.38 | snappy | Hm our idea was to stop buying panasonic phones and slowly migrate to SIP phones, sell off the old phones and the TDM2400p card once done. |
| 02:48.56 | snappy | Yeah, I've definitely seen some interesting phones out there on the cheap. |
| 02:49.12 | [TK]D-Fender | file : No the TDM2400 with a full compliment of modules will weigh more providing a greater source from which to transfer kinetic energy! |
| 02:49.34 | [TK]D-Fender | snappy : Paying to support the old setup will end up costing you in the long run and lose you functionality. |
| 02:49.40 | file | my TDM2400P is very unhappy as I forgot to plug in the power cable |
| 02:50.23 | *** join/#asterisk ATravelingGeek (n=atg@pdpc/supporter/student/ATravelingGeek) |
| 02:51.49 | *** join/#asterisk jdg (n=jdg@155.90-90-202.mana.pf) |
| 02:52.20 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:52.57 | snappy | [TK]D-Fender: I'd love to move, and I agree with you, but I'm pushing my luck as it is getting Asterisk. The only real reason why is we're in dire need of some sort of PBX. One of my bosses in charge of getting the pbx configured hasnt bothered for like 9 months straight. And he claims 'tomorrow' it'll be setup, but of course it never happens. So I spoke to one of my other bosses and he said we can start off the equipment and see how it plays out. |
| 02:53.23 | snappy | So trying to get all the phones replaced may not work. Or if the setup works, we might just order 15 replacement phones. |
| 02:53.34 | snappy | that are sip-enabled |
| 02:53.44 | file | destroy phones! |
| 02:55.36 | [TK]D-Fender | snappy : Not much to say there then.... but do let them know about the tradeoff in spending to try and enable your old phones. Do you have power bricks for them? |
| 02:56.10 | snappy | yep, I made it quite clear that if this works, we'll definitely want SIP phones. |
| 02:57.10 | snappy | Especially since we're expanding to a new office, and I'm hoping to have everything routed to the asterisk machine, but that'll be in the next 3 or so months. |
| 02:57.48 | snappy | It's quite a shame though, I tried to configure the panasonic pbx, but theres windows software only available to panasonic certified repairmen, no documentation, etc. |
| 02:58.02 | [TK]D-Fender | snappy : amke sure you wire your phones for Cat5E, not CAT3 single pair. |
| 02:58.20 | [TK]D-Fender | snappy : Again, do you have power bricks for your old phones? |
| 02:58.30 | snappy | [TK]D-Fender: No |
| 02:59.14 | [TK]D-Fender | snappy : Ok,this is now a real problem. You will not have CallerID, or speakerphone or ANYTHING on those phones in your planned scenario... |
| 02:59.31 | snappy | Eek. |
| 03:00.05 | [TK]D-Fender | snappy : Analog is LIVEABLE if you have at least that, but being in a stand-alone scenario without power your setup will seriously feel like shit. |
| 03:00.26 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 03:00.36 | snappy | I take it our existing PBX provides the power, but with the TDM2400P it doesnt facilitate this? |
| 03:00.39 | [TK]D-Fender | snappy : Under your Panasonic setup they were run as digital phones powered off your main system. |
| 03:01.09 | [TK]D-Fender | snappy : Run as analog phones they are equivalent to sub $10 phones bought in a grocery store. |
| 03:02.24 | snappy | ah great |
| 03:03.03 | [TK]D-Fender | snappy : If you don't want them to feel * is a cheap POS you'd better put a hold on things. |
| 03:03.04 | snappy | There goes that idea, or at least I can try to talk to my boss again. |
| 03:03.50 | kusznir | Grandstream GXP2000's seem to be pretty decent (with current firmware); bought individually they're ~$85 ea. The price drops with quantities (even small quantites like 10-15). |
| 03:03.55 | snappy | Hm yeah, maybe I'll start with the new office being built, and if it appears to be a better phone system than we have, then see if I can sway them to switch the current offices system. |
| 03:04.26 | [TK]D-Fender | snappy : Avoid Grandstream. Flakys junk I'd never let into a company... |
| 03:04.39 | [TK]D-Fender | snappy : whats the ETA for your move? |
| 03:04.53 | [TK]D-Fender | snappy : PM |
| 03:05.47 | kusznir | Ok, I've never noticed issues, but I'm also not doing a commertial deployment. What low-cost phone would you recommend [TK]D-Fender? |
| 03:09.49 | [TK]D-Fender | kusznir : Only phones I suggest are Polycom's except where pricing is prohibitive (outside North America mostly). But thats only to "cheap" customers. |
| 03:10.14 | [TK]D-Fender | kusznir : For those cases I'd suggest Linksys/Aastra. |
| 03:10.34 | kusznir | ok. |
| 03:10.50 | kusznir | What problems do you usually see with Grandstream? |
| 03:13.19 | [TK]D-Fender | kusznir : Actual echo in the handset, general firmware flakeyness, cheap-ass feel, poor use of LCD resources. |
| 03:13.46 | kusznir | ok. |
| 03:14.02 | [TK]D-Fender | kusznir : Only things that seem good about them is the fact its LCD is back-lit, and the price is low. Then again I own a Polycom IP 301, 430, and 501 at home as it is. Quality is worth paying for. |
| 03:17.06 | kusznir | I'm reding up on the polycom 301, and am curious how/what the LCD is used for. |
| 03:17.42 | Nugget | porn. |
| 03:17.59 | [TK]D-Fender | Nugget : Go play with ..... telnet ! |
| 03:18.26 | file | you can't trick Nugget like that, he's too smart |
| 03:18.34 | Strom_C | the polycoms and ciscos are both fantastic phones, but I still prefer the sound quality of the Ciscos. The Polycoms seem to do some DSP to make voices sound more "natural" which only makes them less intelligible to me |
| 03:20.32 | file | Strom_C: Storm! |
| 03:20.53 | Strom_C | balls |
| 03:22.27 | [TK]D-Fender | Strom_C : Never had an issue with them, but have never really used a Cisco. Could never validate their cost given the lower usability vs Polycom. |
| 03:22.53 | Strom_C | [TK]D-Fender: I'll do a direct audio quality comparison within a few days and post it on my site |
| 03:24.29 | [TK]D-Fender | Strom_C : If I could get one at a decent price I wouldn't mind owning one to try if only to learn them so as to better support its users. |
| 03:24.50 | Strom_C | i got my 7960s for $250 each on ebay over a year ago |
| 03:27.02 | [TK]D-Fender | Strom_C : Pricy by my standards when I can get an IP601 for less than that new. |
| 03:28.14 | Strom_C | yeah, but the audio quality difference is night and day |
| 03:28.37 | [TK]D-Fender | Strom_C : Between which 2 models? |
| 03:28.48 | Strom_C | 7960 / ip430 |
| 03:29.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 03:29.22 | [TK]D-Fender | Strom_C : I have heard that the initial firmware load on the 430's might need some tweaking. Have you investigated witht he new 2.0.1 firmware? |
| 03:29.47 | Strom_C | i'm using the 1.6.7somethingorother I think |
| 03:30.11 | [TK]D-Fender | Strom_C : I'd suggest the upgrade. Many new audio quality options... |
| 03:30.11 | Strom_C | 1.6.7.0094 |
| 03:30.31 | [TK]D-Fender | Strom_C : I'm about to upgrade all my home ones, then my work ones following testing here. |
| 03:30.38 | Strom_C | well, i'll fiddle with it...but on default, the polycom still sounds like shit compared to my cisco |
| 03:30.56 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:31.07 | Strom_C | surprising, really, because i would have expected, you know, quality |
| 03:31.52 | [TK]D-Fender | Strom_C : I have head the 430 leaves something to be desired vs the rest of the family though. its is particular to it. |
| 03:34.48 | [TK]D-Fender | Strom_C : I will test vs my 501 once I've gottent hem all upgraded. |
| 03:37.32 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 03:37.35 | Strom_C | well that was fun |
| 03:37.41 | Qwell | w00t |
| 03:37.42 | Strom_C | what did I miss |
| 03:37.51 | Qwell | Strom_C: I found a house :D |
| 03:37.56 | Strom_C | awesome |
| 03:38.09 | Strom_C | is it like three million square feet for thirty dollars? |
| 03:38.15 | Qwell | close |
| 03:38.19 | Strom_C | nice |
| 03:38.24 | Qwell | 2.8 million for fourty-five dollars |
| 03:38.31 | Strom_C | that's forty |
| 03:38.35 | Strom_C | not fourty |
| 03:38.36 | Qwell | die |
| 03:38.41 | mog | shebus |
| 03:39.24 | [TK]D-Fender | [23:32] <[TK]D-Fender> Strom_C : I have head the 430 leaves something to be desired vs the rest of the family though. its is particular to it. |
| 03:39.24 | [TK]D-Fender | [23:34] <[TK]D-Fender> Strom_C : I will test vs my 501 once I've gottent hem all upgraded. |
| 03:39.32 | Strom_M | ah |
| 03:40.00 | Strom_M | but yeah, it just seems brainfucked to me that one should have to adjust how a phone handles audio |
| 03:40.07 | Strom_M | it's a PHONE |
| 03:41.20 | Strom_M | adjusting an analog set for long loops? ok, that makes sense |
| 03:41.24 | Strom_M | but these are digital sets. |
| 03:42.46 | [TK]D-Fender | Strom_C : True, though these things do happen. GXP's have had serious issues an SNOM is known as well. This is just a first for Polycom in my experience. |
| 03:43.13 | *** join/#asterisk ginvent (n=FreePBX5@ip68-111-168-125.sd.sd.cox.net) |
| 03:43.15 | Strom_M | well, grandstream is junk, and snom is on the fence between junk and quality |
| 03:43.22 | Strom_M | so i would expect this from those phones |
| 03:43.46 | ginvent | Anyone know why my remote phone (spa-3000) doesnt hang up the trunks on asterisk? |
| 03:46.39 | *** join/#asterisk ginvent (n=ginvent@ip68-111-168-125.sd.sd.cox.net) |
| 03:46.46 | ginvent | Whoa, got smacked... sheesh |
| 03:47.10 | Strom_M | the joy of Cox |
| 03:47.21 | Strom_M | ginvent, can you be more specific? |
| 03:47.59 | ginvent | more specific? hmm... well, I make a call on a trunk like zap... when I hang up the phone... it keeps the trunk open. |
| 03:48.13 | ginvent | If I call another extension, its fine. |
| 03:48.35 | ginvent | Certain trunks I get a attempting bridge message and nothing. |
| 03:49.01 | Strom_M | are you calling from a phone attached to the fxs port on the spa-3000? |
| 03:49.03 | ginvent | I go SIP from SPA-3000 (at home) to asterisk box at work... then to say sipphone... |
| 03:49.07 | ginvent | Yes. |
| 03:49.18 | tzafrir_laptop | what trunk exactly is it? |
| 03:49.22 | Strom_M | is the asterisk box at work behind NAT? |
| 03:49.37 | Strom_M | is the SPA-3000 also behind NAT? |
| 03:49.38 | ginvent | One is sipphone... one is friend... one is teliax... |
| 03:49.51 | ginvent | Nah, I have them on linksys wrt54gs |
| 03:50.02 | ginvent | With all ports passing. |
| 03:50.06 | ginvent | right now. |
| 03:50.16 | Strom_M | but is the router doing address translation? |
| 03:50.20 | Strom_M | what codec is the spa using? |
| 03:50.24 | ginvent | ulaw |
| 03:50.43 | ginvent | G729 didn't work for DTMF when asterisk picks up. |
| 03:50.52 | ginvent | That is another problem. |
| 03:50.58 | Strom_M | what happens if you put the spa on the same network as the asterisk box? |
| 03:51.07 | ginvent | Not sure... I will have to try that. |
| 03:51.08 | Strom_M | do you encounter the same problems? |
| 03:51.24 | bkw_ | DTMF won't work over g729 unless its out of band |
| 03:51.25 | Strom_M | try that and then come back :) |
| 03:51.27 | ginvent | I have no problem accessing the asterisk box, but I can't seem to get out. |
| 03:52.01 | ginvent | So your best guess is a port problem? |
| 03:52.15 | Strom_M | i want to rule out a network issue first |
| 03:53.13 | ginvent | It's weird cause the trunk works fine... everything works... it just doesn't hang up. |
| 03:53.37 | Strom_M | hm |
| 03:53.46 | Strom_M | is it only the zap trunk that's failing to hang up? |
| 03:53.47 | [TK]D-Fender | ginvent : Go have your telco add disconnect supervision to your line. |
| 03:53.58 | Strom_M | no no no, thats not the problem |
| 03:54.03 | [TK]D-Fender | Strom_C : SPA-3000 is an FXO trunk.... |
| 03:54.05 | Strom_M | at least based on what ginvent is describing |
| 03:54.26 | [TK]D-Fender | Strom_C : And not hanging up has nothing to do with DTMF. |
| 03:54.44 | ginvent | No... any trunk is. |
| 03:54.46 | Strom_M | ginvent is using the fxs port on the spa-3000, and the asterisk box is not hanging up a zap trunk |
| 03:54.57 | ginvent | No trunk hangs up |
| 03:54.59 | Strom_M | and i'm not even attempting to diagnose the dtmf issue |
| 03:55.00 | ginvent | zap, sip, iax2 |
| 03:55.04 | Strom_M | ok |
| 03:55.22 | Strom_M | you're using a phone attached to the spa-3000, right? |
| 03:55.23 | ginvent | I don't even have th spa-3000 using the analog line... |
| 03:55.30 | ginvent | Yes, phone to spa-3000 |
| 03:55.33 | Strom_M | ok |
| 03:55.44 | ginvent | I can call extensions and voicemail, everything good... it hangs up. |
| 03:55.55 | ginvent | if I dial out any trunk, it will not hang up the trunk. |
| 03:55.58 | Strom_M | just for shits and giggles, what happens when you use an iax2 softphone to call the asterisk box |
| 03:56.08 | [TK]D-Fender | ginvent : So you use the FXS port on the SPA-3000 with an alaong phone ; Place a call through * which then uses a Zaptel card ; You hang up from the SPA side of the call and the Zaptel card won't hang up? |
| 03:56.29 | ginvent | TK, yes... zap, sip, and IAX2 trunks. |
| 03:56.49 | ginvent | Hmmm... interesting idea on iax2 phone... |
| 03:57.00 | Strom_M | let's try that and see what happens |
| 03:57.18 | ginvent | Can do... hang on. |
| 03:57.23 | *** join/#asterisk ComputerWarm (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 03:57.41 | [TK]D-Fender | ginvent : Something is SERIOUSLY wrong then. I can understand if it does not detect a remote end disconnect, but * telling the Zap card to hangup failing? Unheard of in my experience. |
| 03:58.20 | ComputerWarm | hello i am trying to use the realtime feature for asterisk, i have every set in extconfig file when i place a inbount call it rings my phone but gives the inbound number a no circuit |
| 03:58.22 | ComputerWarm | anyideas please |
| 03:58.27 | Strom_M | [TK]D-Fender, i'm having him try an iax2 phone before i go proclaiming that things are seriously wrong |
| 03:58.36 | Strom_M | i want to rule out some weirdo sip issue |
| 03:59.20 | ComputerWarm | anyone? |
| 03:59.56 | ginvent | Ok, IAX2 works fine. |
| 04:00.30 | Strom_M | ginvent, ok....the next step is to try the sipura from the same network as the asterisk box |
| 04:01.07 | Strom_M | sip and nat do not historically play well together even with all ports open, and especially not in a double-nat scenario like you'e got |
| 04:01.44 | ginvent | So basically this spa-3000 is a pos. |
| 04:01.45 | ginvent | :D |
| 04:01.54 | [TK]D-Fender | ginvent : Does * acknowledge the SPA hanging up and simply fail to terminate the Zap leg? |
| 04:02.13 | ginvent | TK, yes... or sip or iax trunk. |
| 04:02.15 | [TK]D-Fender | I've never had trouble with double NAT myself... |
| 04:02.39 | Strom_M | ginvent, i'd rule out a networking problem first before deciding that it's the hardware or software |
| 04:03.05 | [TK]D-Fender | ginvent : Well if * sees the SIP end disconnecting then it wouldn't seem to matter WHAT your non-Zap side is doing. |
| 04:03.06 | ginvent | Well, IAX works great... |
| 04:03.22 | Strom_M | what about a sip softphone? |
| 04:03.34 | Strom_M | maybe the spa-3000 is signaling hangup in some oddball way |
| 04:04.05 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 04:04.23 | ginvent | Now your thinking... hang on. |
| 04:04.33 | Strom_M | oh, only now? L) |
| 04:04.35 | Strom_M | er :) |
| 04:05.27 | [TK]D-Fender | the fact that you say * is acknowledging the SPA hangup really leads me away from that... |
| 04:05.44 | Strom_M | well, test first, theorize later |
| 04:06.25 | file | I have a theory! |
| 04:06.31 | Qwell | theorize first, test never |
| 04:06.38 | file | some kid is dreaming and we're all stuck inside his wacky broadway nightmare |
| 04:06.42 | file | I've got a theory we should work this out |
| 04:07.02 | file | it's getting eery what's this cheery singing all about |
| 04:07.18 | file | that is all. |
| 04:07.34 | Strom_M | cock in a box |
| 04:09.14 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.91) |
| 04:11.59 | ginvent | Intersting... the soft sip phone doesn't hang the trunks... but I get the same attempting native bridge and no audio. |
| 04:12.46 | Strom_M | seems like it's either a network issue (likely) or something wrong with asterisk (unlikely) |
| 04:13.04 | ginvent | Weird... cause it doesn't hang the trunks like the spa-3000 |
| 04:13.11 | ginvent | and voicemail works great. |
| 04:13.15 | ginvent | just the trunks are an issue. |
| 04:13.28 | Strom_M | test the spa-3000 and the softphone on the same network as the asterisk box |
| 04:13.35 | Strom_M | i want to rule out a network problem |
| 04:15.25 | ginvent | Ok, so maybe it is a port issue... wonder which one it is. |
| 04:15.43 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:16.46 | ginvent | Lesson here... use iax... :D |
| 04:17.16 | ginvent | This SPA3K has been nothing but trouble... I think it is slated for destruction. |
| 04:17.27 | file | awwwwwwwww be nice to it |
| 04:17.45 | ginvent | Oh... I will be nice... right up until it hits the back of the dumpster. |
| 04:18.10 | ginvent | I just need to buy another Polycom IP600 for home... |
| 04:18.24 | ginvent | Anyone make an IAX phone? hardphone? |
| 04:20.00 | ginvent | found one... wonder if it is crap: http://iaxtalk.com/index.php?main_page=product_info&products_id=3&zenid=c426016690e0a4add13317528f44768c |
| 04:20.31 | Strom_M | wow, a mind-bogglingly pricey $63 |
| 04:20.38 | Strom_M | looks like garbage :) |
| 04:20.45 | ginvent | Yeah, but if it is a pile of crap... |
| 04:21.39 | Strom_M | the lesson here, really, is "put the asterisk box on a world-facing IP if you plan to use SIP devices behind NAT" |
| 04:22.42 | ginvent | I guess I can hamachi it in. |
| 04:22.47 | ginvent | but that adds another layer. |
| 04:23.17 | Strom_M | hamachi? |
| 04:24.32 | [TK]D-Fender | ginvent : IP601 for hom? Do you consult in VoIP and need to register to a lot of servers at once? |
| 04:24.48 | ginvent | Awesome vpn |
| 04:24.51 | ginvent | hamachi. |
| 04:25.09 | ginvent | TK, not really... I have a few at my shop... and they are great. |
| 04:25.17 | ginvent | What I need for home is a good iax2 hard phone. |
| 04:25.29 | ginvent | So I can stop messing with this sip ship. |
| 04:26.50 | Strom_M | ginvent, im telling you - you will have no trouble if you just put your asterisk box on a public world-facing IP |
| 04:27.13 | Strom_M | or set up an asterisk box behind each NAT and trunk via IAX2 |
| 04:27.14 | ginvent | Strom... yeah, but I have no good method to do that right now. |
| 04:27.31 | ginvent | Hmm... asterisk box for home is a possibility. |
| 04:27.44 | ginvent | I have a pc laying here splain open... |
| 04:27.59 | ginvent | Wonder if It would tax my mythtv box too much to add it to that. |
| 04:28.39 | [TK]D-Fender | ginvent : Well my server does *, X (for HTPC use), and is my router/ everything box. Works just fine... |
| 04:32.16 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.202) |
| 04:32.49 | DarKnesS_WolF | where i can read about asterisk codecs ? and dose asterisk convert codecs between both ends ? |
| 04:33.51 | Qwell | CunningPike: should |
| 04:33.58 | Qwell | zaptel might not |
| 04:34.21 | CunningPike | Qwell: Don't need zaptel - SIP here and IAX back to the office |
| 04:34.28 | CunningPike | Hmmm - I should try it |
| 04:34.56 | Qwell | should work fine |
| 04:35.02 | CunningPike | Qwell: Cool :D |
| 04:36.18 | CunningPike | uname -r gives 2.2.20-pmac lol |
| 04:36.19 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
| 04:36.19 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:36.26 | benjk | Zaptel works fine on PPC |
| 04:36.33 | twisted | since when? |
| 04:36.35 | benjk | under LinuxPPC |
| 04:36.38 | twisted | oh |
| 04:36.39 | benjk | since ages ago |
| 04:36.47 | twisted | well yea, linuxppc |
| 04:36.48 | CunningPike | benjk: Really? Interesting |
| 04:36.50 | twisted | but not os x ;) |
| 04:37.03 | benjk | a company called Terrasoft paid Digium to make sure it worked |
| 04:37.08 | CunningPike | twisted: OS X doesn't run on a PPC ;) |
| 04:37.17 | twisted | CunningPike uh, yes it does. |
| 04:37.20 | benjk | Terrasoft are the ones behind Yellow Dog Linux |
| 04:37.40 | benjk | they are now focused on IBM POWER and IBM PPC blades |
| 04:37.44 | CunningPike | twisted: Not the PPC I'm thinking of.......... 603e :D |
| 04:38.01 | twisted | CunningPike no, not on a 603e |
| 04:38.04 | CunningPike | lol |
| 04:38.08 | twisted | but the machines i have here are PPC's |
| 04:38.15 | twisted | and they run osx |
| 04:38.20 | benjk | I have Asterisk and Zaptel running nicely on a PPC 603 under YDL |
| 04:38.24 | twisted | except for my powermac |
| 04:38.50 | twisted | benjk i had asterisk running on my 603 for awhile, but w/o zaptel since mine didn't have pci slots |
| 04:38.55 | CunningPike | Yup - I have a PowerBook G4 and a couple iMacs - they run OS X just fine ;) |
| 04:39.13 | benjk | if it doesn't have PCI slots, well, then most of zaptel doesn't make much sense |
| 04:39.15 | CunningPike | twisted: Using YellowDog or Debian? |
| 04:39.18 | twisted | benjk exactly |
| 04:39.22 | twisted | CunningPike YDL |
| 04:39.24 | benjk | I used YDL |
| 04:39.34 | file | ack it's a twisted |
| 04:39.38 | benjk | but there is no reason why it shouldn't work with Debian PPC |
| 04:39.40 | twisted | oh gnoes, it's a file |
| 04:39.40 | CunningPike | twisted: Hmm - I have Debian - I'll give it a shot |
| 04:39.51 | twisted | CunningPike, linux is linux. |
| 04:40.01 | CunningPike | twisted: Yup |
| 04:40.06 | twisted | should work regardless of distro unless your maintainer did something retarded |
| 04:40.20 | benjk | as far as kernel stuff is concerned its all one kernel, sort of |
| 04:40.29 | benjk | so it shouldn't make a difference |
| 04:40.39 | twisted | posix > * |
| 04:41.19 | twisted | meanwhile, back in the lab... |
| 04:41.45 | twisted | still generating locales.... sometimes i hate prebuilt crap heaps |
| 04:41.48 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 04:41.53 | file | twisted: let's have a sword fight with soldering irons! |
| 04:42.08 | *** join/#asterisk chameleon_95 (n=SteveDuc@cm218-252-160-131.hkcable.com.hk) |
| 04:42.12 | twisted | file, that doesn't sound fun |
| 04:42.14 | wunderkin | uhoh, file's gettin kinky |
| 04:42.18 | twisted | i'd rather shoot fluxballs |
| 04:42.28 | [TK]D-Fender | file : If you want an "edge" I'll lend you my kat ;) |
| 04:42.35 | file | [TK]D-Fender: haha |
| 04:42.42 | twisted | you have a katrina? |
| 04:42.51 | [TK]D-Fender | twisted : Indeed |
| 04:43.17 | twisted | i didn't mean it in the sense of a hurricane |
| 04:43.24 | twisted | although that sounds good right about now |
| 04:43.28 | twisted | too bad i don't have any hurricane mix |
| 04:43.35 | [TK]D-Fender | http://forums.swordforum.com/showthread.php?s=&threadid=67812&highlight=andrew+bushi+pics |
| 04:43.48 | twisted | hrm... too bad i'm not in a gui. |
| 04:44.12 | [TK]D-Fender | twisted : Sorry, I think you'd lose something in ascii-art conversion ;) |
| 04:44.23 | Qwell | twisted: y0 |
| 04:44.36 | twisted | heh |
| 04:44.44 | twisted | nah, i just highlighted it and sent it to my laptop |
| 04:44.47 | twisted | nice sword |
| 04:44.51 | twisted | Qwell, wassabi |
| 04:44.57 | Qwell | I'm here :D |
| 04:44.58 | twisted | i thought you were going to join kp thursdaynight? |
| 04:45.08 | Qwell | twisted: yeah, we got in late |
| 04:45.20 | twisted | you were with him at dinner |
| 04:45.29 | twisted | according to him |
| 04:45.37 | Qwell | huh? |
| 04:46.00 | twisted | he said you were eating with him when I called, then were going to join at bumpers |
| 04:46.13 | file | massive confusion! |
| 04:46.15 | Qwell | I got in at like 9:30 |
| 04:46.18 | twisted | perhaps i misinterpreted |
| 04:46.21 | benjk | lotsa allohol |
| 04:46.32 | file | Qwell... alcohol... |
| 04:46.39 | Qwell | speaking of which! |
| 04:46.44 | twisted | wow i can write that word much faster on a clicker keyboard than I can a laptop one |
| 04:46.52 | twisted | Qwell, yeah, speaking of which... |
| 04:47.23 | Qwell | twisted: tomorrow night? |
| 04:47.26 | Qwell | maybe |
| 04:47.26 | file | ut roh |
| 04:47.28 | twisted | Qwell, perhaps. |
| 04:47.28 | Qwell | ":D |
| 04:49.11 | wunderkin | plonk |
| 04:49.13 | twisted | haha |
| 04:49.19 | twisted | file doesn't fall, he implodes |
| 04:49.50 | twisted | yay! my hard disk decided to play a percussionary piece |
| 04:49.51 | wunderkin | but his inodes fall to the ground |
| 04:50.07 | tzanger | keep your inodes around your waist, good god that's disgusting |
| 04:50.47 | twisted | yeah |
| 04:50.50 | twisted | that's what fanny packs are for |
| 04:50.51 | [TK]D-Fender | Weebles may wobble but they don't fall down... |
| 04:50.52 | wunderkin | look, all of the files are joining together, in some kind of.. organazation.. |
| 04:51.20 | twisted | omfg the world is collapsing |
| 04:51.25 | twisted | the machine finished generating locales |
| 04:51.25 | twisted | brb |
| 04:52.44 | Qwell | generating locales? What are you doing, installing glibc? |
| 04:53.40 | tzafrir_laptop | or is it libc6? |
| 04:53.54 | Qwell | or maybe it's klocaleswitcher |
| 04:54.05 | chameleon_95 | I have a problem with "Failed to authenticate on INVITE" |
| 04:54.48 | chameleon_95 | I have a simple setup.. 1 Asterisk server on public IP, 1 Grandstream IP Handset behind NAT router, trying to make outgoing call using VOIPSTUNT |
| 04:55.00 | chameleon_95 | I have pasted config and settings to http://pastebin.ca/165789 |
| 04:55.15 | chameleon_95 | Also posted SIP DEBUG output... |
| 04:58.32 | `Tingles` | to create a meet me conference do i have do anything other then conf => 600 |
| 04:58.36 | `Tingles` | in meetme.conf |
| 04:58.41 | `Tingles` | and the extensions for it correct? |
| 04:58.51 | tzafrir_laptop | right |
| 04:59.07 | `Tingles` | does it need to be restarted for conferences or jsut reloaded..? |
| 04:59.15 | `Tingles` | anything else to check? |
| 04:59.45 | `Tingles` | exten => 600,1,MeetMe(600,i,54321) |
| 04:59.52 | `Tingles` | is the line in extensions.conf |
| 05:00.30 | CunningPike | Qwell: Should I define LOW_MEMORY in the Makefile for 64MB? |
| 05:02.43 | `Tingles` | are there any other flags that need to be activated for the conf to work? |
| 05:03.17 | CunningPike | `Tingles`: What does your CLI say? |
| 05:05.56 | `Tingles` | no don`t have verbose setup ... |
| 05:06.51 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 05:08.52 | CunningPike | Does anyone know what constitutes 'low memory' for the purposes of setting the LOW_MEMORY flag in the asterisk Makefile? |
| 05:09.44 | `Tingles` | says |
| 05:09.44 | `Tingles` | Sep 10 07:10:12 WARNING[7236]: pbx.c:1700 pbx_extension_helper: No application 'MeetMe' for extension (test, 600, 1) |
| 05:10.20 | CunningPike | `Tingles`: There you go then - 'load app_meetme.so' |
| 05:10.45 | CunningPike | `Tingles`: And add a load statement for it in modules.conf |
| 05:11.10 | chameleon_95 | I have a problem with "Failed to authenticate on INVITE" |
| 05:11.28 | chameleon_95 | Why is my Asterisk server passing the username from the IP Telephone to VoipStunt |
| 05:11.33 | `Tingles` | thanks CunningPike |
| 05:11.36 | chameleon_95 | Why is not passing the Voipstunt username |
| 05:11.42 | CunningPike | `Tingles`: np - get it working? |
| 05:12.38 | `Tingles` | will check in a sec |
| 05:12.42 | `Tingles` | just adding it and reloading |
| 05:13.19 | CunningPike | chameleon_95: Do you have a register statement in your sip.conf? |
| 05:13.31 | `Tingles` | same thing... do i have to restart.. for modules? |
| 05:14.06 | CunningPike | `Tingles`: No - but did you load it from the CLI? Type 'load app_meetme.so' at the CLI prompt |
| 05:14.40 | CunningPike | `Tingles`: The modules.conf entry (load => app_meetme.so) is just so that it gets loaded each time asterisk starts |
| 05:14.54 | `Tingles` | aww perfect.. thats why.. |
| 05:16.00 | `Tingles` | Unable to load module app_meetme.so |
| 05:16.00 | `Tingles` | Sep 10 07:16:41 WARNING[7349]: config.c:525 process_text_line: No '=' (equal sign) in line 49 of modules.conf |
| 05:16.00 | `Tingles` | Sep 10 07:16:41 WARNING[7349]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_meetme.so: cannot open shared object file: No such file or directory |
| 05:16.04 | `Tingles` | hmm... |
| 05:16.53 | CunningPike | `Tingles`: Hmm - that's weird..... |
| 05:17.43 | chameleon_95 | CunningPike: Yes. I have a register in sip.conf... |
| 05:17.55 | chameleon_95 | CunningPike: It registers successfully to VOIPSTUNT.. |
| 05:17.57 | CunningPike | `Tingles`: What does line 49 of modules.conf say (not relevant to this problem, but I'd be interested to see what's causing that error) |
| 05:18.29 | CunningPike | chameleon_95: What does your Dial() statement look like? |
| 05:18.50 | chameleon_95 | CunningPike: When I call from my grandstream ip phone to the Asterisk box with canreinvite=no it passes the username from the grandstream phone over to VoipStunt |
| 05:18.51 | niter3 | hey guys i'm setting up FWD in asterisk and I got my box to dial out no problem and I have my incoming for forward but these entries seem to be messed up.. |
| 05:18.52 | niter3 | exten => ${FWDEXTEN},1,Dial(${PHONE1},30) |
| 05:18.52 | niter3 | exten => ${FWDEXTEN},2,Voicemail(u${PHONE1VM}) |
| 05:18.52 | niter3 | exten => ${FWDEXTEN},3,Hangup |
| 05:18.52 | niter3 | exten => ${FWDEXTEN},102,Voicemail(b${PHONE1VM}) |
| 05:18.53 | niter3 | exten => ${FWDEXTEN},103,Hangup |
| 05:19.03 | niter3 | What should be in place of PHONE1 my extension? |
| 05:19.18 | chameleon_95 | Dial(SIP/900442071009361@voipstunt) |
| 05:20.00 | niter3 | chameleon_95: to take income calls? |
| 05:20.29 | niter3 | Wouldn't it be more like DIal(SIP/222,30) |
| 05:20.35 | niter3 | so it's my asterisk extension? |
| 05:20.44 | `Tingles` | CunningPike: i had the line as load app_meetme.so instead of load => app_meetme.so |
| 05:20.55 | CunningPike | niter3: chameleon_95 isn't talking to you |
| 05:21.05 | niter3 | oh darn |
| 05:21.10 | niter3 | CunningPike: you have any idea? |
| 05:21.10 | CunningPike | `Tingles`: OK - perfect - fixed that? |
| 05:21.19 | `Tingles` | ya |
| 05:21.21 | `Tingles` | fixed |
| 05:21.39 | CunningPike | `Tingles`: Excellent |
| 05:21.56 | `Tingles` | still get the second error thoe |
| 05:22.02 | CunningPike | niter3: I would say it should be SIP/whatever your phone it |
| 05:22.05 | CunningPike | s/it/is/ |
| 05:22.06 | chameleon_95 | CunningPike: I have pasted output to http://pastebin.ca/165789 |
| 05:22.13 | niter3 | CunningPike: yah that's what I thought but it complains |
| 05:22.40 | niter3 | exten => ${FWDEXTEN},1,Dial(${SIP/222},30) |
| 05:22.41 | niter3 | ? |
| 05:22.55 | `Tingles` | ya i checked the file is not there.. in the modules folder... |
| 05:22.58 | chameleon_95 | exten => ${FWDEXTEN},1,Dial(${SIP/222,30) |
| 05:23.08 | chameleon_95 | niter3: exten => ${FWDEXTEN},1,Dial(SIP/222,30) |
| 05:23.32 | chameleon_95 | niter3: remove the ${..} as this is if you have set global variable... |
| 05:24.05 | CunningPike | niter3: Just Dial(SIP/222,30) - no ${} |
| 05:24.10 | `Tingles` | CunningPike: any suggestions? |
| 05:24.20 | CunningPike | `Tingles`: How did you install Asterisk? |
| 05:24.32 | CunningPike | `Tingles`: Bizarre that the module is missing |
| 05:25.17 | `Tingles` | on gentoo |
| 05:25.19 | `Tingles` | asterisk |
| 05:25.22 | `Tingles` | emerge .. |
| 05:25.47 | CunningPike | chameleon_95: I need the various usernames to make sense of what's going on - I presume they are all different? |
| 05:25.54 | `Tingles` | only that one i i have like 40 others in that folder |
| 05:26.01 | CunningPike | chameleon_95: Just obscure the passwords |
| 05:26.10 | CunningPike | `Tingles`: Hmm - weird |
| 05:26.13 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 05:26.19 | CunningPike | `Tingles`: What version? |
| 05:27.04 | `Tingles` | 1.2.9_p1 |
| 05:27.48 | *** join/#asterisk HaMYaI (i=CKGLOB@125.25.130.44) |
| 05:28.09 | niter3 | CunningPike: Ok i found the problem... It seems I can't be on a call when a call is being placed .. However, I have call waiting on this phone |
| 05:28.21 | `Tingles` | i could upgrade to 1.2.11 |
| 05:28.21 | CunningPike | niter3: What's the phone |
| 05:28.26 | niter3 | CunningPike: IT should at least beep on my phone letting me know there is an incoming call instead of bitching that it's busy |
| 05:28.39 | niter3 | I just have a linksys adapter |
| 05:28.44 | *** join/#asterisk chameleon_95 (n=SteveDuc@cm218-252-160-131.hkcable.com.hk) |
| 05:28.47 | HaMYaI | anyone got the h323 working with asterisk 1.2.11 on FC5? |
| 05:28.54 | niter3 | i have 3 way and everything working |
| 05:29.35 | CunningPike | niter3: It sounds like your ATA isn't set up for call waiting...... |
| 05:29.49 | niter3 | like I said it does work if anyone outside FWD calls me |
| 05:29.56 | niter3 | meaning anyone on my box except FWD |
| 05:32.06 | CunningPike | niter3: So, you're on a call, any call, and someone calls from FWD and they get busy? |
| 05:32.12 | niter3 | i lied |
| 05:32.15 | niter3 | 3 way isn't working now |
| 05:32.17 | niter3 | it was |
| 05:32.22 | niter3 | that is weird |
| 05:32.41 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-59-102.nwrknj.east.verizon.net) |
| 05:32.46 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 05:32.50 | `Tingles` | CunningPike: anything else to try? |
| 05:32.53 | Dovid | morning all |
| 05:33.08 | *** join/#asterisk chameleon_95 (n=SteveDuc@cm218-252-160-131.hkcable.com.hk) |
| 05:33.20 | CunningPike | `Tingles`: I would re-install/upgrade from www.asterisk.org if I were you - is this a production system? |
| 05:33.27 | `Tingles` | test |
| 05:33.31 | `Tingles` | upgrade |
| 05:33.34 | CunningPike | `Tingles`: Go for it then |
| 05:33.54 | CunningPike | Bloody packages ;) |
| 05:34.30 | CunningPike | No-one has an answer for my DLOW-MEMORY question? |
| 05:34.44 | Qwell | CunningPike: Are you running out of memory? |
| 05:35.00 | Qwell | It won't help a whole lot afaik, but it does help some, and it's only useful if you're actually running out |
| 05:35.05 | CunningPike | Qwell: No - just about to install on a PPC with only 64MB |
| 05:35.08 | Dovid | i am having a problem with the read command |
| 05:35.14 | Qwell | CunningPike: try it without |
| 05:35.20 | CunningPike | Qwell: OK - will do... |
| 05:35.22 | CunningPike | Thanks |
| 05:35.30 | Qwell | bed |
| 05:35.34 | CunningPike | Qwell: Night |
| 05:35.34 | Dovid | was working with my macro and then stoped when I upgarded asterisk. i downgraded but it still dosent want to work |
| 05:35.38 | Dovid | http://pastebin.ca/165809 |
| 05:36.46 | chameleon_95 | CunningPike: http://pastebin.ca/165808 |
| 05:37.04 | CunningPike | chameleon_95: Ya - looking now |
| 05:38.16 | `Tingles` | compiling now.. |
| 05:38.54 | CunningPike | Dovid: What does your CLI say? |
| 05:40.40 | directory | [TK]D-Fender: eep! |
| 05:41.14 | Dovid | nothing |
| 05:41.21 | Dovid | acts like it didnt get drmf |
| 05:41.27 | Dovid | dtmf* |
| 05:41.28 | niter3 | `Tingles`: Awesome got it fixed. |
| 05:41.31 | niter3 | Ok everythign is working |
| 05:41.33 | Dovid | tryng a diff carrier. one sec |
| 05:41.59 | directory | the user agent from voipstunt amuses me |
| 05:42.10 | niter3 | seems like I had to do a reboot on my adapter. |
| 05:43.01 | `Tingles` | good stuff niter |
| 05:43.29 | CunningPike | Dovid: Is DTMF working at all? (Voicemail, for example) |
| 05:44.15 | Dovid | yes |
| 05:44.41 | Dovid | i just noticed that when i press something it just exits from my macro. let me test again |
| 05:44.41 | Dovid | i tired diff provider just to make sure |
| 05:45.45 | `Tingles` | bad thing is after compiling i still don`t have that file.. |
| 05:46.03 | CunningPike | `Tingles`: Wait - are you installing zaptel? |
| 05:46.03 | `Tingles` | is there a special make flag i am to use to compile modules.. thoe it seems i am only missing that one |
| 05:46.06 | `Tingles` | nope |
| 05:46.15 | CunningPike | `Tingles`: You'll need it........ |
| 05:46.16 | `Tingles` | will |
| 05:46.20 | `Tingles` | ok |
| 05:46.29 | CunningPike | `Tingles`: Install zaptel, then re-install asterisk |
| 05:46.43 | CunningPike | `Tingles`: If you have no zaptel hardware, load ztdummy |
| 05:47.38 | Dovid | CunningPike: it seems to see my dtmf coming in cause as soon as i press something it stops playing the file however it ignores the rest of my macro. it just jumpsout of the macro. this started just now. ARGH !!! |
| 05:47.50 | `Tingles` | 2 secs |
| 05:47.56 | `Tingles` | almost done recompiling with zaptel |
| 05:48.08 | CunningPike | Dovid: Do 'set verbose 3' and then pastebin your CLI output |
| 05:48.11 | `Tingles` | arg |
| 05:48.17 | CunningPike | `Tingles`: arg? |
| 05:48.22 | `Tingles` | need to compile kernel aswell need some module |
| 05:48.37 | Dovid | CunningPike: do that in CLI correct ? |
| 05:48.39 | CunningPike | `Tingles`: You just need the kernel sources, that's all |
| 05:48.46 | `Tingles` | <PROTECTED> |
| 05:48.46 | `Tingles` | <PROTECTED> |
| 05:48.46 | CunningPike | Dovid: Affirmative |
| 05:48.59 | CunningPike | `Tingles`: Ugh |
| 05:49.07 | `Tingles` | ya give me one sec i will recompile that too |
| 05:49.18 | CunningPike | `Tingles`: I am not a gentoo fan. You are doing nothing to change my mind ;) |
| 05:49.33 | Dovid | k |
| 05:49.43 | CunningPike | ~pb |
| 05:49.47 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 05:49.54 | CunningPike | rofl |
| 05:50.38 | Dovid | lol |
| 05:50.43 | Dovid | CunningPike: http://pastebin.ca/165819 |
| 05:51.04 | Dovid | it seems to be just jumping out of the macro as soon as it see's my entry |
| 05:52.01 | CunningPike | Dovid: 'User disconnected' - which key are you pressing? |
| 05:52.15 | Dovid | 1 |
| 05:52.28 | Dovid | that should quit the macro and patch the call |
| 05:53.19 | CunningPike | Dovid: OK - let me read for a bit..... |
| 05:53.32 | Dovid | CunningPike: now it seems to be working |
| 05:53.36 | Dovid | it may be a carrier issue |
| 05:53.41 | Dovid | hold up |
| 05:54.17 | Dovid | ;) |
| 05:54.48 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 05:55.44 | Dovid | CunningPike: it was a provider issue. the reason it didnt work wit provider 2 is because i had a typo ;) |
| 05:55.57 | CunningPike | Dovid: Great :D |
| 05:57.44 | Dovid | CunningPike: do u know of any proxies for asteirsk manager so i dont have a million users connecting in at once ? |
| 05:57.56 | CunningPike | Dovid: AstManProxy |
| 05:58.02 | Dovid | thx |
| 05:58.37 | HaMYaI | anyone had a success compiling openh323_v1_19_0_1 on asterisk 1.2.11? |
| 05:58.43 | CunningPike | This is the most work this old PPC has done in years lol |
| 05:58.48 | CunningPike | There's smoke coming out |
| 05:59.12 | HaMYaI | and with pwlib_v1_11_1 |
| 06:00.03 | Dovid | CunningPike: u use teliax ? |
| 06:00.12 | CunningPike | Dovid: Nope |
| 06:00.47 | Dovid | they are the ones givig me DTMF issue |
| 06:01.01 | Dovid | it seems to be one way dtmf issue's |
| 06:01.01 | CunningPike | Dovid: Not a typo then? |
| 06:01.05 | Dovid | no |
| 06:01.14 | Dovid | the reciever cant send dtmf to asterisk |
| 06:01.21 | CunningPike | Dovid: Hmm - you using rfc2388? |
| 06:01.46 | Dovid | if i use thier 'lines' to call out i can hit dtmf but if asterisk calls some one and ia m uding read it dosent work |
| 06:01.48 | Dovid | leme check |
| 06:02.59 | Dovid | using inbanf |
| 06:03.03 | Dovid | inband* |
| 06:03.10 | Dovid | can relaxeddtmf be the issue ? |
| 06:05.17 | Dovid | i was using inband |
| 06:05.20 | Dovid | swtiched it |
| 06:05.21 | Dovid | one sec |
| 06:06.29 | Dovid | yes. seems to be working :) |
| 06:06.36 | Dovid | thx |
| 06:07.01 | Dovid | also another question |
| 06:07.21 | Dovid | can i have 2 users login to asterisk manager with the same user id and or pass or will it crash asterisk ? |
| 06:07.49 | [TK]D-Fender | Dovid : the 2nd will knock off the first. |
| 06:08.17 | Dovid | ok |
| 06:08.20 | [TK]D-Fender | Dovid : What you're looking for is SIP-B (Shared Line Appearances) which appears to be slated for * 1.4 |
| 06:08.40 | Dovid | :) |
| 06:08.44 | CunningPike | Compiling asterisk - what does this mean I need: configure: error: termcap support not found |
| 06:09.00 | [TK]D-Fender | CunningPike : You need lib-termpcap installed |
| 06:09.15 | Dovid | the issue is i have FOP running under the same user id and apss as a plug in that i have with outlook to send calls to the manager |
| 06:09.20 | Dovid | do u know of any proxies ? |
| 06:09.36 | CunningPike | Dovid: Different users? |
| 06:09.53 | [TK]D-Fender | Dovid : OH... sorry UI misread that. As far as I'm aware you should be able to do taht just fine actually. |
| 06:09.55 | Dovid | CunningPike told me astmanproxy but that seems for diff servers |
| 06:10.10 | Dovid | yes CunningPike: |
| 06:10.35 | Dovid | TK: So i can have multiple boxes logging in with the same user id and pass ? |
| 06:10.45 | [TK]D-Fender | Dovid : I believe so. |
| 06:10.55 | Dovid | seems to be crashing me :( |
| 06:11.03 | [TK]D-Fender | Dovid : Its just a TCP connection, don't see a problem with that since its just used to stream events |
| 06:11.06 | Dovid | do u know of any proxies for the manager ? |
| 06:11.31 | [TK]D-Fender | Only of astmanproxy |
| 06:11.55 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:12.16 | acehunky | anyone had success to use voipstunt with asterisk ? |
| 06:12.57 | chameleon_95 | acehunky: http://www.voip-info.org/wiki/view/VoipStunt |
| 06:13.07 | acehunky | that doesnt help me much chameleon_95 :( |
| 06:13.17 | Dovid | tk: that seems to only be for one box to manage multiple asterisk servers. i am looking more for multiple outlook's to speak to a proxy and then that proxy speaks to asterisk |
| 06:13.29 | Dovid | so that not everyone is speaking to asterisk at once |
| 06:14.25 | acehunky | http://pastebin.ca/165832 chameleon_95 |
| 06:14.29 | [TK]D-Fender | Dovid : Thats what it does... |
| 06:15.06 | acehunky | handle_response_invite ... error ... |
| 06:15.14 | [TK]D-Fender | acehunky : Isn't that error message obvious enough? |
| 06:15.15 | kusznir | Question: how does asterisk voicemail module deal with timezones? I'm trying to get the message invelope to read in local time, and no matter what I do, it always reads it in UTC. I've also found that the voicemail metadata file stores the date in UTC (and labled as such). |
| 06:15.48 | [TK]D-Fender | kusznir : you can configure the timezone of each box. |
| 06:16.13 | kusznir | I'm running asterisk on an embedded device, so its using uclibc rather than glibc; I realize that is likely to be a signifiant factor, if not the main factor. However, I was looking for more understanding of how asterisk does its stuff so I might make it work correctly. |
| 06:16.19 | [TK]D-Fender | BBIAB (maybe) |
| 06:17.28 | kusznir | yea, I have: 70 => XXXX,Jim,me@myaddress.com,tz=pacific in voicemail.conf; this has no effect. |
| 06:21.27 | chameleon_95 | acehunky: paste your sip.conf settings |
| 06:22.18 | `Tingles` | does asterisk have a log anywhere.. looks like the old configs with the new version is a no start.. |
| 06:22.28 | `Tingles` | i want to see why it is crapping out |
| 06:22.45 | docelmo | ./var/log/asterisk |
| 06:23.11 | acehunky | chameleon_95: sip.conf is quite big .. |
| 06:23.23 | acehunky | i shall telll you the scenario .. |
| 06:23.29 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 06:23.37 | acehunky | asterisk server is behind single NAT |
| 06:23.59 | acehunky | but behind static IP .. hence put externip=x.x.x.x |
| 06:24.03 | acehunky | nat=yes |
| 06:24.53 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:24.53 | PakiPenguin | hello :) |
| 06:25.21 | acehunky | chameleon_95 u there ? |
| 06:27.09 | chameleon_95 | acehunky: i am here.. |
| 06:27.09 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 06:27.28 | SomethingISODD | hello all question anyone ever had a problem where outbound calls you can hear but inbound you can`t? |
| 06:27.47 | chameleon_95 | acehunky: just paste me the settings for voipstunt |
| 06:28.19 | SomethingISODD | anyone? |
| 06:28.57 | kusznir | SomethingISODD: sounds like NAT problems. |
| 06:29.01 | `Tingles` | Sep 10 08:29:58 WARNING[20808]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb6f07708 (len 485) to 68.144.4.42:62839 returned -1: Bad file descriptor |
| 06:29.03 | `Tingles` | any ideas? |
| 06:29.12 | SomethingISODD | kusznir even know i can hear outbound? |
| 06:30.09 | kusznir | Yea, because when you call outbound, you iniitate the connection, which gets written in the NAT tables and translated approprately. For inbound, if you arn't forwarding the RTP ports, then you won't hear the audio, because it gets set up in a second session. |
| 06:30.32 | SomethingISODD | ok what ports would i need to open in the router? |
| 06:31.19 | kusznir | for sip, usually 5060 and 8000-8001. Might want to go a bit farther, say 8000-8006. |
| 06:31.35 | kusznir | 5060 is used for the session management, 800x is used for the audio. |
| 06:31.45 | SomethingISODD | kusznir i have my phone on the dmz i think its called |
| 06:31.51 | SomethingISODD | isn`t that like a open port? |
| 06:31.54 | SomethingISODD | open ports. |
| 06:32.10 | kusznir | In theory, yea... |
| 06:32.25 | SomethingISODD | hrm |
| 06:32.40 | kusznir | You can try doing some network debugging/dumping to make sure traffic is actually making it back to your phone, though. Your problem sounds very much like NAT to me... |
| 06:33.37 | SomethingISODD | wait never noticed i have rtp pointed to a different ip in the router from before i switched to dmz would that do it |
| 06:33.49 | kusznir | that would! |
| 06:33.58 | SomethingISODD | testing |
| 06:34.39 | SomethingISODD | hrm that didn`t do it either |
| 06:35.08 | kusznir | I'd reboot your router and the phone to make sure.... |
| 06:35.12 | chameleon_95 | SomethingISODD: Whats the model of your router |
| 06:35.36 | SomethingISODD | k brb |
| 06:36.06 | acehunky | http://pastebin.ca/165837 @ chameleon_95 |
| 06:38.49 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 06:39.03 | SomethingISODD | kusznir that still didn`t resolve it |
| 06:39.13 | SomethingISODD | do you think if i setup a stun server that would help any? |
| 06:39.36 | kusznir | What is your router? Also, what is it plugged into? |
| 06:39.48 | acehunky | chameleon_95 if you see, all the things that i have commented are the ones that i have tried .. and still i get this handle_response_invite error .. |
| 06:40.22 | SomethingISODD | its a linksys what do you mean what is it plugged into?? |
| 06:40.37 | SomethingISODD | you mean the interwise? if so its a Motorola Cable modem |
| 06:40.43 | kusznir | Linksys WAN port connected to cablemodem? |
| 06:40.53 | SomethingISODD | yes |
| 06:41.31 | kusznir | Ok. And it (your linksys) gets a "real" internet IP? (quick test: compare what the linksys says for its WAN IP with what you get when you browse to http://whatismyip.com. If they're not the same, you've still got NAT) |
| 06:41.56 | SomethingISODD | k |
| 06:42.23 | SomethingISODD | ya they match |
| 06:42.28 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 06:43.00 | linagee | Strom_C: i just got food at tommy's. hrm. chili only burger is not that bad. |
| 06:43.10 | PakiPenguin | SomethingISODD which ip phone is this? |
| 06:43.13 | kusznir | what is your phone? |
| 06:43.22 | kusznir | <PROTECTED> |
| 06:43.40 | SomethingISODD | right now i am playing with GS but i tried with Cisco 1750 i believe it is and Xlite |
| 06:43.43 | SomethingISODD | all three with the same issue |
| 06:44.09 | `Tingles` | Sep 10 08:43:58 WARNING[27879]: app_meetme.c:463 build_conf: Unable to open pseudo device |
| 06:44.31 | PakiPenguin | is you codec set right> |
| 06:44.38 | bkw_ | codec isn't the problem |
| 06:44.43 | bkw_ | no zap pseudo device |
| 06:44.50 | bkw_ | load ztdummy or some zaptel hardware |
| 06:44.58 | SomethingISODD | good day bkw_ |
| 06:44.58 | PakiPenguin | oh |
| 06:45.06 | SomethingISODD | kusznir anyideas? |
| 06:45.12 | bkw_ | SomethingISODD, na its not a good day |
| 06:45.23 | SomethingISODD | bkw_ that sucks.. |
| 06:45.33 | bkw_ | it will be in time |
| 06:45.43 | PakiPenguin | <PROTECTED> |
| 06:45.44 | SomethingISODD | here is hoping |
| 06:45.52 | kusznir | SomethingISODD: do you have a network packet capture program on the machine with the xten softphone? |
| 06:46.10 | SomethingISODD | no |
| 06:46.20 | bkw_ | nobody gets it... really they don't. Look beyond today.. look beyond tomorrow... |
| 06:46.20 | kusznir | Umm..the ztdummy comment was to Tingles...that's a meetme problem, not a sip audio problem :) |
| 06:46.21 | SomethingISODD | PakiPenguin i don`t believe bkw_ was talking to me |
| 06:46.36 | PakiPenguin | ah |
| 06:46.38 | PakiPenguin | <PROTECTED> |
| 06:46.41 | PakiPenguin | lol |
| 06:46.43 | bkw_ | then again nobody gets it. |
| 06:46.45 | PakiPenguin | just woke up |
| 06:46.54 | bkw_ | so why bother |
| 06:47.02 | `Tingles` | i have a TDM-400P but not installed yet.. |
| 06:47.09 | bkw_ | `Tingles`, install it |
| 06:47.14 | bkw_ | meetme requires ztdummy or hardware |
| 06:47.18 | `Tingles` | arg one more restart |
| 06:47.29 | SomethingISODD | kusznir if i run a cap on the server would that help?? |
| 06:47.53 | bkw_ | one of many to come if you run asterisk |
| 06:48.16 | coppice | don't install it. all the troubles start once you actially do something :-) |
| 06:48.24 | SomethingISODD | haha |
| 06:48.27 | bkw_ | ain't that the truth |
| 06:49.01 | bkw_ | but don't dare have an opinion about anything... that seems to be an issue around these parts |
| 06:49.05 | kusznir | It could. If you can get on the asterisk console, do a sip debug, then try and place the call. |
| 06:49.28 | SomethingISODD | ok out bound or inbound? |
| 06:49.30 | bkw_ | I think digium makes sip suck on purpose in Asterisk to make IAX look better. |
| 06:49.59 | SomethingISODD | bkw_ :-) recommendations to better? then asterisk that is |
| 06:50.00 | bkw_ | Example: http://bugs.digium.com/view.php?id=5162 <-- This bug is now over a year old... makes the whole thing better. |
| 06:50.17 | bkw_ | SomethingISODD, at times I think tin cans and string would be better |
| 06:50.33 | denon | and yet, you're still here :) |
| 06:50.38 | bkw_ | SomethingISODD, you're catching me in a bad mood. |
| 06:50.46 | bkw_ | denon, Asterisk needs to be re-written to survive. |
| 06:50.47 | kusznir | SomethingISODD: inbound to your troublesome clients |
| 06:51.23 | SomethingISODD | kusznir on i will grep that and then put it in pastebin |
| 06:51.47 | SomethingISODD | this is a odd message |
| 06:51.48 | bkw_ | denon, I have to use this pile of code for a few more months if not more |
| 06:51.48 | SomethingISODD | Sep 10 01:45:28 NOTICE[29442]: chan_sip.c:11595 sip_poke_noanswer: Peer '5025855375' is now UNREACHABLE! Last qualify: 62 |
| 06:51.55 | HaMYaI | I am having /usr/bin/ld: cannot find -lpt |
| 06:52.03 | bkw_ | install libpt then |
| 06:52.17 | HaMYaI | when compiling openh323 |
| 06:52.21 | tzafrir | don't do that !!! |
| 06:52.36 | tzafrir | (install libpt/opnh323, that is ;-) |
| 06:52.51 | denon | h323 always works behind nat? |
| 06:52.51 | HaMYaI | tzafrir, from yum? |
| 06:52.53 | bkw_ | you need pwlib don't ya |
| 06:52.54 | acehunky | chameleon_95 any guess ? |
| 06:52.56 | denon | wow, what planet are you on? |
| 06:52.57 | bkw_ | denon, yes |
| 06:53.02 | bkw_ | yes it always works behind nat |
| 06:53.04 | bkw_ | always HAS |
| 06:53.13 | tzafrir | HaMYaI, are you sure it's the right version? If so: OK |
| 06:53.14 | denon | bkw_: only because it's old enough that most nat understands it |
| 06:53.20 | bkw_ | no |
| 06:53.26 | bkw_ | OpenH323 has some tricks up it sleve |
| 06:53.28 | HaMYaI | bkw_, pwlib is installed |
| 06:53.32 | kusznir | SomethingISODD: Uhh...this can be bad....Now that I think of it, I had the exact same problem....Spent a week tracing it down...it was my ISP who decided that only Vonage VoIP calls were allowed to pass through its network, and it blocked all other VoIP traffic. The only solution I got working was to switch to IAX... |
| 06:53.44 | denon | wait a sec .. we were talking h.323, now we're talking openh323.. |
| 06:54.07 | HaMYaI | bkw_, openh323 also compiles and now I'm compiling asterisk-oh323-0.7.3 |
| 06:54.09 | bkw_ | well OpenH323 runs 90% of the h.323 stuff out there |
| 06:54.26 | denon | you dont remember the days of netmeeting? :) |
| 06:54.38 | bkw_ | I don't care about netmeeting |
| 06:54.40 | tzafrir | HaMYaI, you have libpt from yum but not openh323? |
| 06:54.40 | bkw_ | its a pile of shit |
| 06:54.45 | bkw_ | moving on to something else |
| 06:54.52 | bkw_ | infact |
| 06:55.04 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 06:55.21 | denon | hm, that's mature |
| 06:55.30 | SomethingISODD | kusznir this is a small ass town, i talked to them a few weeks ago they are cool with voip infact they are going to be installing and playing with asterisk |
| 06:55.32 | kusznir | From what I remember of netmeeting and NAT, the two didn't mix. It took years to get a NAT module for the linux kernel that actually let netmeeting/h.323 pass through...I was a net admin then, and h.323 was a pain in the butt. |
| 06:55.42 | SomethingISODD | but with that said is there anyway to back that up? |
| 06:56.07 | denon | kusznir: nod, you're correct - h.323 is certainly not nat-friendly by design, bkw's just in one of his moods |
| 06:56.42 | SomethingISODD | denon i had a h323 phone yesterday here playing with openh323 it worked great behind a nat |
| 06:57.18 | denon | yeah, as we were saying, modern nat deals with it fine |
| 06:57.20 | SomethingISODD | kusznir is there anyway for me to find out if that is the issue |
| 06:57.24 | denon | and some modern h323 stacks work fine |
| 06:57.32 | denon | but its not that it's inherently nat-friendly |
| 06:57.41 | kusznir | Well, first you can try plugging a hardware phone directly into your cablemodem and see if it works. If one assumes the grandstream is configured properly, this should work; if not, the problem is most likely on the 'net outside of your control. (You can verify this by taking the same device to some other internet connection and see if you can make it work there) |
| 06:57.41 | SomethingISODD | denon ya true. |
| 06:58.33 | SomethingISODD | u know what that does make sence kusznir i was using this phone behind nat.. before i moved and it worked perfect |
| 06:58.44 | kusznir | You might also want to do a packet capture at the server to ensure its actually transmitting packets. |
| 06:59.22 | SomethingISODD | kusznir if i changed the ports on everything do you think that would help any? |
| 06:59.26 | HaMYaI | tzafrir, I'm about to install it (libpt) from yum, didn't have it previuosly |
| 06:59.34 | SomethingISODD | "saying its the carrier" |
| 06:59.34 | kusznir | Yea...It took me a little too long to make the connection that everything worked fine, I switched ISPs, and everything broke..... :) (I took the ISP helpdesk at their word: "No, we don't block VoIP. We don't block anything." I've since found about a dozen ports blocked....) |
| 06:59.46 | HaMYaI | tzafrir, I use "make opt" from the openh323 dir |
| 07:00.15 | SomethingISODD | kusznir are you just doing nmap from outside? |
| 07:00.27 | tzafrir | anyway, don't take me too seriously now... |
| 07:00.36 | HaMYaI | tzafrir, and now I am trying to install http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz |
| 07:00.49 | kusznir | SomethingISODD: changing ports could work if you change them all (I'm not sure off the top of my head how to change the rtp ports) |
| 07:00.58 | HaMYaI | tzafrir, or I only need openh323? |
| 07:01.06 | SomethingISODD | wouldn`t that be in rtp.conf? |
| 07:01.09 | kusznir | No, I didn't do nmap; I had to discover it the hard way.... |
| 07:01.54 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 07:01.59 | tzafrir | HaMYaI, libpt (pwlib) is a library needed for openh323. openh323 is a library used by both chan_h323 (channel/h323 in the asterisk tarball) and oh323 |
| 07:02.02 | kusznir | Don't know...never looked at that file :) (I ran asterisk on the inside of my network, and fortunately all my providers offered IAX. Furthermore, they wern't about to change their port settings for me. So I just switched to IAX) |
| 07:02.17 | *** join/#asterisk deanjames (n=godson@202.149.32.31) |
| 07:02.44 | SomethingISODD | kusznir well i am sure its the provider.... not for me to figure out a way of beating them with out buying a iax phone or another system |
| 07:03.08 | SomethingISODD | not for me= now for me |
| 07:03.09 | denon | or tunneling |
| 07:03.22 | SomethingISODD | denon tunneling? |
| 07:03.23 | denon | shoving your sip traffic over a udp tunnel works pretty well |
| 07:03.34 | `Tingles` | CunningPike: Sep 10 09:04:17 WARNING[8285]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': Permission denied |
| 07:03.36 | `Tingles` | <PROTECTED> |
| 07:03.37 | SomethingISODD | could you tell me how to do that? |
| 07:03.46 | denon | you control both sides of where you're going? |
| 07:04.02 | SomethingISODD | all i can control is asterisk and my phone |
| 07:04.28 | denon | well, you'd want to set something like openvpn up on both sides |
| 07:04.31 | SomethingISODD | and the linux server * is on |
| 07:04.32 | denon | on servers on each network |
| 07:04.39 | kusznir | various ideas I have for your situation: 1) try an iax soft client (there are a few free ones out there); 2) get fancy with iptables and the nat table to rewrite the outbound ports of traffic going to you to different port numbers, then add recripical rules on your linksys. |
| 07:04.46 | SomethingISODD | hrm how about vpn? my router supports that |
| 07:05.00 | denon | eh, "vpn" is a pretty broad term |
| 07:05.07 | denon | that's probably an ipsec implementation though |
| 07:05.16 | kusznir | Linksys routers generally set up IPSec VPN, which is kind of a pain to set up. |
| 07:05.20 | denon | which would probably work with some tinkering, those some isps foolishly block ipsec as well |
| 07:05.22 | HaMYaI | tzafrir, and how do you include it to the asterisk Makefile then? |
| 07:05.38 | denon | SomethingISODD: http://openvpn.net/ |
| 07:05.52 | denon | run that on both sides, tunnel over something fun, like udp/53 :) |
| 07:06.00 | SomethingISODD | denon i don`t have a linux server on the inside. do you think the tunnling would work |
| 07:06.07 | tzafrir | `Tingles`, ls -l /dev/zap/pseudo |
| 07:06.20 | denon | SomethingISODD: you cant muster up an old 486 to run on the inside? |
| 07:06.20 | kusznir | If your linksys is a WRT54G/GS, then you can load OpenWRT (linux for routers) on it, and then run openwrt on your router. (for that matter, you could run asterisk on your router -- that's what I do -- and have it talk iax to your main asterisk box. |
| 07:06.29 | SomethingISODD | denon not atm |
| 07:06.41 | denon | SomethingISODD: what kind of router? |
| 07:06.49 | SomethingISODD | let me check the mod moment |
| 07:07.03 | SomethingISODD | WRT54G |
| 07:07.08 | tzafrir | HaMYaI, Don't know. Last time I looked the debian package built it out-of-tree. |
| 07:07.10 | denon | what version? |
| 07:07.16 | denon | its on the bottom |
| 07:07.35 | kusznir | Yea, you should be able to run OpenWRT if its not a late version (I think its V6, they have it linked from their main page, www.opewrt.net) |
| 07:07.48 | SomethingISODD | sec |
| 07:07.54 | kusznir | I think V6 is the cutoff... |
| 07:08.07 | denon | yeah, it gets a little tricky with vxworks models in general |
| 07:08.12 | SomethingISODD | i have v5 |
| 07:08.31 | denon | SomethingISODD: http://wiki.openwrt.org/OpenWrtDocs/Hardware/Linksys/WRT54G |
| 07:08.36 | kusznir | sorry..openwrt.org |
| 07:08.48 | SomethingISODD | is this hard to do |
| 07:09.09 | denon | SomethingISODD: I've gotta stress, though - it's not for the faint of heart, and you'd be much better off with an old junker PC |
| 07:09.48 | denon | in fact, on v5, you probably wont even have enough memory to pull it off |
| 07:09.53 | benjk | you may want to check out the WRAP |
| 07:09.54 | kusznir | Umm..looking at the current stats, it might be difficult....V5's only have 2MB of flash, and their support status is listed as WiP. |
| 07:10.14 | denon | kusznir: yeah, it runs fine - but with 2mb of flash, you need to streamline a little |
| 07:10.17 | benjk | its just over 100 USD |
| 07:10.47 | SomethingISODD | denon what do you mean stream line? |
| 07:10.57 | denon | uh, strip things down |
| 07:11.03 | benjk | http://pcengines.ch/pic/wrap1c1.jpg |
| 07:11.07 | kusznir | You will either have to run the "minimal" image or build your own with as much stuff removed as possible. |
| 07:11.12 | denon | SomethingISODD: if you dont want to find another machine, then your cheapest and by far best route of action, is to call the ISP, and ask for escalation techs until you get someone with a clue. Then ask them to open stuff up for you. |
| 07:11.30 | benjk | that'll run * easily and its way better than a WRT |
| 07:11.39 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 07:11.47 | denon | benjk: yeah, he's already said he doesnt want to buy anything else |
| 07:12.00 | benjk | fair enough |
| 07:12.06 | SomethingISODD | denon well its not i don`t wanna buy anything |
| 07:12.16 | benjk | some people have time to waste |
| 07:12.18 | SomethingISODD | it is in two weeks i am moving out of this hell whole |
| 07:12.23 | benjk | I envy that |
| 07:12.28 | SomethingISODD | but i need to finish off testing |
| 07:12.30 | kusznir | If you want to test / make something work tonnight without new hardware, I see 2 potential options: |
| 07:12.38 | denon | SomethingISODD: well, someone would give you an old pentium box.. |
| 07:12.59 | SomethingISODD | kusznir? |
| 07:13.05 | kusznir | 1) set up the nat/de-nat stuff I metnioned before; 2) set up openvpn on your windows (I assume) box, and use xten through it. |
| 07:13.11 | denon | but running openwrt on that unit most likely wont leave enough room for openvpn |
| 07:13.27 | `Tingles` | localhost ~ # ls -l /dev/zap/pseudo |
| 07:13.27 | `Tingles` | crw-rw---- 1 root root 196, 255 Sep 10 08:56 /dev/zap/pseudo |
| 07:13.34 | `Tingles` | however i did just install that cardc |
| 07:13.35 | `Tingles` | now.. |
| 07:13.45 | `Tingles` | shouldn`t we point it to that somehow? |
| 07:13.52 | kusznir | (note that option 2 will limit you to using your softphone only; you won't (easily) be able to make your hardware phones work) |
| 07:14.19 | kusznir | benjk: do you have a URL with more info on the WRAP-thingie? |
| 07:14.21 | SomethingISODD | ok. i guess i can just use asterisk meetme for a while... |
| 07:14.25 | denon | kusznir: not really - if he were to run openvpn on his workstation, he could simply have it route |
| 07:14.25 | SomethingISODD | well i am playing |
| 07:14.30 | denon | and point the phone at it |
| 07:14.33 | benjk | sure: pcengines.ch |
| 07:14.33 | denon | dfgw |
| 07:14.45 | benjk | there is also this one: www.routerboard.com |
| 07:14.53 | benjk | and there is Soekris |
| 07:14.54 | SomethingISODD | denon even with my phone plugged right in to the router? |
| 07:15.10 | `Tingles` | tzafrir ? |
| 07:15.15 | denon | sure, just disable dhcp and manually point the phone at your workstation |
| 07:15.18 | kusznir | denon: Ok, I've never seen windows with a single NIC actually route intellegently :) |
| 07:15.25 | benjk | soekris is at http://www.soekris.com/ |
| 07:15.28 | SomethingISODD | never thought isn there not a windows verison of asterisk? |
| 07:15.41 | benjk | there are more pricey though but have a full PCI slot |
| 07:15.42 | denon | yeah, there is |
| 07:15.52 | denon | thats another option, using a local asterisk as a router of sorts |
| 07:15.55 | SomethingISODD | denon i wonder if it would kill my cpu |
| 07:15.55 | tzafrir | `Tingles`, the "access denied" problem: the user of asterisk lacks permissins to access /dev/zap/pseudo |
| 07:16.05 | benjk | Routerboard have both miniPCI and one full PCI mobo |
| 07:16.10 | denon | SomethingISODD: routing? takes zero cpu time for a few mbps |
| 07:16.15 | benjk | PCengines only do miniPCI |
| 07:16.31 | SomethingISODD | ok thank you all very much |
| 07:16.36 | SomethingISODD | let me go find winblows asterisk |
| 07:16.46 | benjk | Routerboard have the highest performance |
| 07:16.48 | SomethingISODD | lol i wonder if it blows as bad as windows lol |
| 07:16.57 | LoneShadow | asterisk + openvpn both at the same time on openwrt ? |
| 07:17.10 | coppice | benjk: how much are those routerboards? |
| 07:17.17 | denon | LoneShadow: nah, just tunneling across, asterisk on the remote side |
| 07:17.20 | benjk | the prices are on their website, coppice |
| 07:17.23 | LoneShadow | ok |
| 07:17.24 | benjk | www.routerboard.com |
| 07:17.26 | denon | SomethingISODD: http://www.voip-info.org/tiki-index.php?page=Asterisk%20Windows%20Support |
| 07:17.29 | kusznir | SomethingISODD: Or, if you have a decent machine with ram, install VMWare Server (its free), and install linux, have it do your routing, or just run asterisk on linux..... :) (but you'll need at least 512MB ram on yoru win box, probably a bit more) |
| 07:17.29 | LoneShadow | coz the router processor is pretty weak |
| 07:17.30 | af_ | benjk, what is routerboard? |
| 07:17.36 | benjk | www.routerboard.com |
| 07:17.47 | benjk | its a small embedded mobo |
| 07:17.50 | LoneShadow | I cant get to do transcoding on wrtg54gs |
| 07:17.52 | SomethingISODD | kusznir no this system as it is reboots for something to do |
| 07:17.53 | LoneShadow | wrt54gs |
| 07:18.06 | benjk | you can use it to build a router or an PBX or a combo router=pbx |
| 07:18.11 | benjk | s/=/- |
| 07:18.26 | SomethingISODD | benjk i think i need to get myself one of them |
| 07:18.26 | denon | hmm.. tw is broken |
| 07:18.30 | denon | guess that means it's time for bed |
| 07:18.46 | benjk | they are perfect for small biz and soho pbx'es |
| 07:18.46 | kusznir | LoneShadow: You're probably not going to be able to, either...WRT's don't have much CPU power, and transcoding takes a fair bit. |
| 07:18.48 | SomethingISODD | denon kusznir benjk thanks very much for the help |
| 07:18.55 | kusznir | You're welcome. |
| 07:19.00 | LoneShadow | kusznir: I already said that :P |
| 07:19.10 | LoneShadow | [12:47] <LoneShadow> coz the router processor is pretty weak |
| 07:19.18 | coppice | benjk: I looked a long time ago, and they didn't have prices then. they are a lot more expensive than using a WRT54G, but the hardware looks similar |
| 07:19.19 | SomethingISODD | hey would openvpn run on cygen or whatever its called |
| 07:19.24 | kusznir | Oh, sorry...I just saw the last two lines :) |
| 07:19.34 | denon | SomethingISODD: openvpn runs on windows natively |
| 07:19.35 | denon | very well |
| 07:19.37 | benjk | coppice, how much are the WRT54s? |
| 07:19.39 | LoneShadow | though I still use asterisk on my router for other reasons |
| 07:20.09 | denon | that's what I said before |
| 07:20.09 | SomethingISODD | denon sorry must have missed that |
| 07:20.09 | coppice | benjk: about $50 |
| 07:20.10 | kusznir | btw: I *do* run openvpn and asterisk on my wrt at the same time :) I also do transcode between ulaw and gsm, but that has some glitches. |
| 07:20.10 | benjk | what CPU? |
| 07:20.10 | denon | SomethingISODD: thats what I meant, you could put it at your workstation, openvpn running, tunnel out from there |
| 07:20.22 | denon | it'd not be terribly straight forward from a routing point of view, and a second nic would make it much easier |
| 07:20.25 | coppice | mips 200+, 16 or 32M RAM (depends on the revision) |
| 07:20.53 | benjk | the routerboards would seem to have more oomph then |
| 07:20.55 | SomethingISODD | denohrm second nic i have now if i have a network cable is the next question |
| 07:21.23 | LoneShadow | kusznir: I tried transcoding between gsm and another codec, with dtmfmode=inbound, and the router was struggling pretty bad |
| 07:21.35 | SomethingISODD | eek sorry i mean denon |
| 07:21.36 | denon | coppice: actually, early wrt54g units only had a 125mhz |
| 07:21.39 | SomethingISODD | gotta stop using tab |
| 07:21.45 | benjk | also, PCengines.ch are working on a new model using the AMD Geode GX |
| 07:21.56 | coppice | demon: early ones had too little RAM, too. |
| 07:22.01 | denon | nod |
| 07:22.04 | benjk | which is comparable to a 586/800MHz |
| 07:22.26 | denon | man, everyone's gettin my nick wrong tonight |
| 07:22.27 | denon | hehe |
| 07:22.35 | coppice | the Geode GX is the K7 renamed, isn't it? |
| 07:22.36 | benjk | low RAM is going to be a bigger problem than low clock rate |
| 07:22.48 | LoneShadow | get the right GS version, it has 32MB |
| 07:22.52 | benjk | something like that, with low power tweaks |
| 07:22.53 | kusznir | LoneShadow: yea, dtmf=inband dies reguardless of the codec for me...That's the main issue I have, as I use a DVG-1120S, and its rfc dtmf doesn't work with asterisk, so I can't do dtmf with it... |
| 07:23.00 | denon | actually, all the G models had 16MB |
| 07:23.26 | denon | up until v5.0 |
| 07:23.28 | benjk | the WRAPs go up to 256MB RAM and any amout of Flash available |
| 07:23.28 | coppice | benjk: more than tweaks. its in a much finer geometry, so the power goes waaaayyy down |
| 07:23.35 | LoneShadow | kusznir: get a msntv box or an old pc off ebay |
| 07:23.35 | benjk | ok |
| 07:23.40 | SomethingISODD | I am thinking the easiest way to resolve this issue tonight is install the windows asterisk |
| 07:23.46 | SomethingISODD | for the time being |
| 07:23.56 | kusznir | denon: up until the 54Gv5, which drops down to 8. |
| 07:23.59 | denon | SomethingISODD: probably, and IAX2 out from there |
| 07:24.02 | LoneShadow | I have to see if my msntv box can do mythfrontend + asterisk at the same time |
| 07:24.06 | coppice | benjk: the ITX boards are much more widely available. why not use those? |
| 07:24.08 | denon | kusznir: that's what I said :) |
| 07:24.20 | SomethingISODD | denon ya.. is it on asterisk site? |
| 07:24.32 | denon | SomethingISODD: i gave you the url a half hour ago :) |
| 07:24.52 | SomethingISODD | thank god for scroll lol thanks |
| 07:25.10 | benjk | too big, crappy firmware that expects all sorts of nonsense that a server should never ever even have the ability to be connected to, like keyboard mouse, parallel ports, jousticks and crap like that |
| 07:25.14 | benjk | utter nonsense |
| 07:25.15 | denon | SomethingISODD: http://www.voip-info.org/tiki-index.php?page=Asterisk%20Windows%20Support |
| 07:25.44 | SomethingISODD | denon you seem good with linksys routers, when you disable the wireless part does it actually stop wireless access? |
| 07:25.47 | coppice | benjk: none of that matters. you can just ignore it |
| 07:25.48 | benjk | I want a totally stripped down to the bones mobo that has no resemblance to a desktop |
| 07:25.52 | benjk | it matters |
| 07:25.54 | SomethingISODD | i am being told different from my isp |
| 07:25.56 | denon | SomethingISODD: you can turn off the radio |
| 07:26.13 | benjk | those PCengines and the Soekris have proper embedded firmware, not desktop firmware |
| 07:26.17 | benjk | makes a huge difference |
| 07:26.29 | coppice | what difference? |
| 07:26.29 | SomethingISODD | denon i disabled the wireless all together and my isp says people can still gain access |
| 07:26.43 | denon | SomethingISODD: well, I dont know anything about the linksys stock firmware .. |
| 07:26.44 | kusznir | LoneShadow: I've never heard of msntv boxes before...what is the hardware specs? |
| 07:26.57 | denon | SomethingISODD: but if you disable wireless (not just disable ssid), it shouldnt be able to |
| 07:27.15 | denon | they probably think you've just disabled ssid broadcasting .. in which case, yes, people can still get on |
| 07:27.15 | SomethingISODD | i disabled everything i could find for wireless |
| 07:27.24 | kusznir | SomethingISODD: I'd say the ISP is full if it. If you disable wireless, as denon said. Keep in mind, this is the same ISP that said SIP works fine :) |
| 07:27.25 | SomethingISODD | and really i could care less but just wondering |
| 07:27.29 | docelmo | MSNTV blows |
| 07:27.32 | LoneShadow | I think its celeron 733Mhz equivalent of a P3 |
| 07:27.34 | benjk | reliability difference |
| 07:27.40 | LoneShadow | 128MB ram |
| 07:27.46 | SomethingISODD | kusznir good point |
| 07:27.48 | benjk | and the difference between booting and not booting |
| 07:27.50 | denon | SomethingISODD: if you're worried, set an insanely complicated WPA2 key |
| 07:27.51 | LoneShadow | I am using it as mythfrontend |
| 07:28.14 | SomethingISODD | denon na.. if people wanna use my internet as long as its not kiddy port crap go for it |
| 07:28.14 | kusznir | If the Wireless-G light on the front pannel is out, then you should be safe :) |
| 07:28.16 | benjk | especially when you are not there and the crap box wont reboot |
| 07:28.17 | LoneShadow | I can put a 1GB CF card and load the required software on ubuntu |
| 07:28.33 | kusznir | LoneShadow: when you mentioned, that, it got my attention :) (I also run myth) |
| 07:28.38 | SomethingISODD | face it if i could remember the mark on the wall that Stream savers said a few years ago i might go do that lol |
| 07:29.11 | coppice | benjk: I have used ITX boards. if you don't use a mouse, keyboard, screen, etc, they work fine. In terms of reliability, I see no reason to believe some board from a tiny outfit it likely to match their reliability |
| 07:29.15 | LoneShadow | I have 2 boxes next to my tv, one is a compaq evo p4 2gig pc with 512MB ram, running as mythbackend + asterisk + freepbx |
| 07:29.36 | LoneShadow | the other is a msntv box working as mythfrontend, nfs booting from the compaq evo |
| 07:29.54 | benjk | well, I have had nothing but bad experience with anything that has those non-server ports |
| 07:29.55 | denon | I like those little aopen PCs for small, quiet boxes |
| 07:29.56 | kusznir | LoneShadow: do you have issues with asterisk sound quality when mythbackend gets loaded? (like comm flagging)? |
| 07:29.59 | LoneShadow | both cost me less $200 :D |
| 07:30.07 | LoneShadow | maybe $250 |
| 07:30.08 | benjk | my principle is this: what is not there cannot break |
| 07:30.23 | LoneShadow | na, p4 2.0 gig is pretty strong enough |
| 07:30.36 | LoneShadow | I dont do hdtv, so its fine |
| 07:30.42 | benjk | what is not there cannot cause trouble |
| 07:30.50 | SomethingISODD | hey anyone know of any good free TV channels online in english? |
| 07:31.03 | kusznir | Ahh..true...My myth box is a combo fe/be and 1.2Ghz. |
| 07:31.33 | kusznir | So these msntv boxes are all easily convertable to whatever OS you want to throw on them? |
| 07:31.41 | LoneShadow | yea |
| 07:31.46 | LoneShadow | linux OS |
| 07:31.48 | denon | they're intel? |
| 07:31.59 | kusznir | Are they quiet? |
| 07:32.01 | LoneShadow | Intel Celeron 733 |
| 07:32.03 | LoneShadow | mhz |
| 07:32.23 | LoneShadow | all are passive devices, runs on a 12V, no motors |
| 07:32.24 | LoneShadow | no hdd |
| 07:32.36 | denon | how much? |
| 07:32.43 | LoneShadow | comes with 128MB cf card, using it nfs boot |
| 07:33.04 | LoneShadow | bought a 1GB card, will try to put a read only mythfrontend image on it |
| 07:33.20 | kusznir | ebay has a few with bids in the 20-30 $$ range. |
| 07:33.32 | LoneShadow | it goes on ebay from $0.99 to $50 |
| 07:33.36 | denon | rca/sony/etc are all the same spec? |
| 07:33.40 | kusznir | Do they have ethernet built in? |
| 07:33.48 | LoneShadow | nope, only rca msntv2 works |
| 07:33.59 | LoneShadow | ethernet, wifi |
| 07:34.00 | LoneShadow | oops |
| 07:34.01 | LoneShadow | no wifi |
| 07:34.04 | denon | dialup |
| 07:34.07 | LoneShadow | i meant usb |
| 07:34.21 | denon | need to do anything special to em to load another OS? screwy bios stuff or such? |
| 07:34.30 | LoneShadow | http://www.toc2rta.com |
| 07:34.39 | LoneShadow | screwy bios yes :P |
| 07:34.47 | denon | ah ic |
| 07:35.00 | benjk | BIOS suxx |
| 07:35.01 | LoneShadow | need to open the box, and use a serial to ttl cable to flash override the evil msn bios :D |
| 07:35.33 | LoneShadow | bought 2 of those boxes |
| 07:35.44 | denon | ah .. so probably not something I could do for a business env |
| 07:35.44 | LoneShadow | wanted to make the 2nd one as a SIP video phone device |
| 07:35.46 | denon | (thin clients) |
| 07:35.59 | LoneShadow | connect a USB vga camera to it, and connect it to the TV |
| 07:36.08 | LoneShadow | pretty complicated, working on it still |
| 07:36.37 | LoneShadow | not many avaialble, its more useful for a hobby, not worth for biz |
| 07:37.12 | `Tingles` | why is it trying to use the pseudo device for /dev/zap... when i have a card in the machine.. |
| 07:37.45 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-87.range86-138.btcentralplus.com) |
| 07:43.47 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 07:50.16 | JT | what a weird telephone |
| 07:50.18 | JT | http://cgi.ebay.com.au/Star-Wars-R2-D2-SFX-Fully-Functioning-Telephone_W0QQitemZ190028143478QQihZ009QQcategoryZ73559QQrdZ1QQcmdZViewItem |
| 07:50.43 | eliXier | lol |
| 07:51.56 | LoneShadow | heh |
| 07:52.05 | LoneShadow | cant imagine using that phone :D |
| 07:52.33 | coppice | a friend had a phone that looked like a duck, and quacked for a ring tone. You kinda picked up the top half of the ducked and talked up its ass |
| 07:53.02 | JT | haha |
| 07:53.14 | LoneShadow | lol |
| 07:53.17 | LoneShadow | seen those |
| 07:53.32 | LoneShadow | I remember it being an expensive gadjet when it came out |
| 07:54.16 | coppice | all phones were expensive back then. China didn't start making the $1 phone until the early 90s |
| 07:55.14 | LoneShadow | it was illegal to have more than one phone device in India during early 90s |
| 07:55.27 | LoneShadow | so was using any phone other than the ones provided by the govt |
| 07:55.40 | LoneShadow | now its illegal only to terminate pstn calls =/ |
| 07:55.56 | LoneShadow | but ppl still do voip to pstn and vice versa |
| 07:56.01 | tzafrir | `Tingles`, /dev/zap/pseudo is the device file through which userspace programs access zaptel's timing. Regardless of the real driver |
| 07:56.10 | JT | i'm guessing that's no longer the case given their massive callcentres |
| 07:56.14 | JT | phones everywhere |
| 07:56.18 | `Tingles` | tzafrir got it all working.. |
| 07:56.50 | coppice | JT: the call centres are a special case. they have fairly direct links in and out of India. land lines in india suck badly |
| 07:57.12 | JT | hmm |
| 07:57.21 | coppice | but cellular has finally hit the big time there |
| 07:58.46 | coppice | >100,000 people in bangalore work permanent night shifts in call centres. that's enough line to justify special arrangements. :-) |
| 07:59.14 | SomethingISODD | denon thanks man that fixed the issue |
| 07:59.24 | coppice | but says bad things about US and european companies, if they can generate that many concurrent calls, which are mostly complaints :-) |
| 07:59.50 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 08:00.24 | kusznir | Actually, I kinda know what is happening with that error message, its its cause I don't get.... |
| 08:00.26 | *** join/#asterisk shtoom (n=godson@202.149.32.31) |
| 08:01.09 | kusznir | My iax peers are not showing up properly; I think something is broke with my iax channel stuff now, but I don't know what...I didn't change anything (that I remember, anyway :( ) |
| 08:01.26 | SomethingISODD | and kusznir thank you as well |
| 08:01.39 | kusznir | you're welcome. |
| 08:01.56 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-87.range86-138.btcentralplus.com) |
| 08:03.39 | `Tingles` | exten => 7777,1,Dial(SIP/7777) |
| 08:03.39 | `Tingles` | exten => 7777,2,Voicemail(7777@test) |
| 08:03.54 | `Tingles` | <PROTECTED> |
| 08:04.07 | `Tingles` | along with teh mailbox => entry |
| 08:04.56 | kusznir | You'll need to put a ,<timeout> after the dial command: |
| 08:05.08 | kusznir | exten => 7777,1,Dial(SIP/7777,30) |
| 08:05.27 | kusznir | That will make it give up after about 5 rings and go to voicemail; otherwise it will ring indefinately. |
| 08:06.27 | `Tingles` | i get a busy signal when i do that.. |
| 08:06.57 | kusznir | After 5 rings? |
| 08:06.59 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 08:07.41 | `Tingles` | like 3 ish.. |
| 08:07.48 | `Tingles` | 7777 => 7777,Tristan Marler,bloodyiron@hotmail |
| 08:07.51 | `Tingles` | <PROTECTED> |
| 08:07.53 | LoneShadow | anyone using a sipura3000 ? |
| 08:07.54 | `Tingles` | but verbose said.. |
| 08:08.04 | `Tingles` | Sep 10 10:07:56 WARNING[9190]: app_voicemail.c:2410 leave_voicemail: No entry in voicemail config file for '7777' |
| 08:08.06 | `Tingles` | when i called it |
| 08:08.41 | kusznir | (you still need your second line, as you origionally wrote it, too. You can insert a "u" in front to specifiy unavailable, but it should default to that) |
| 08:09.01 | kusznir | Umm...have you done a full restart of asterisk since editing the voicemail.conf file? |
| 08:09.37 | `Tingles` | yup |
| 08:09.47 | kusznir | (shouldn't be required, but I've run into a couple issues with that tonnight :-/) |
| 08:10.14 | `Tingles` | ya let me just restart now to verify.. |
| 08:12.31 | LoneShadow | anyone have a callback agi script on a plain vanilla asterisk box ? |
| 08:12.50 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 08:12.56 | `Tingles` | kusznir restarted.. |
| 08:13.35 | `Tingles` | busy signal again... |
| 08:13.58 | kusznir | Umm...got me :( |
| 08:14.10 | LoneShadow | need to simulate an intercomm on my openwrt+asterisk+spa3k system |
| 08:14.26 | `Tingles` | ya same error |
| 08:14.28 | `Tingles` | Sep 10 10:15:20 WARNING[9417]: app_voicemail.c:2410 leave_voicemail: No entry in voicemail config file for '7777' |
| 08:16.16 | `Tingles` | kusznir i had it outside the context it should have been in |
| 08:16.20 | `Tingles` | :| |
| 08:16.22 | `Tingles` | oops |
| 08:16.25 | `Tingles` | but it works now |
| 08:16.27 | `Tingles` | :) |
| 08:16.38 | kusznir | :) |
| 08:16.56 | kusznir | I think its my bedtime.... |
| 08:21.42 | docelmo | Anyone in here know the PHP API? |
| 08:24.15 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 08:28.48 | JT | anyone know anything about Telematrix analogue phones? |
| 08:30.06 | Strom_C | what about them? |
| 08:31.59 | JT | how does "5 lines" or whatever work? does it have 5 modular sockets in it? |
| 08:32.48 | Strom_C | if it's a "five-line" analog phone, it probably has either: |
| 08:32.54 | Strom_C | - 5 single-line jacks |
| 08:33.06 | Strom_C | - two two-line jacks and a single-line jack |
| 08:33.15 | Strom_C | - two three-line jacks |
| 08:33.35 | JT | ah right |
| 08:34.09 | JT | because it insists it needs no KSU, so i assumed it must have connections for 5 lines, although there were no pics depicting the underside |
| 08:34.33 | Strom_C | the standard modular plug spec does specify, among others, a six-position six-conductor plug (RJ16?) |
| 08:34.50 | JT | RJ-12? |
| 08:36.05 | Strom_C | oh ok, im remembering now |
| 08:36.12 | Strom_C | RJ-11 is six-position two-conductor |
| 08:36.20 | Strom_C | RJ-12 is six-position four-conductor |
| 08:36.29 | Strom_C | RJ-14 is six-position six-conductor |
| 08:36.34 | JT | ah |
| 08:40.04 | niter3 | holy fuck i'm giong nuts.. |
| 08:40.19 | Strom_C | ok...? |
| 08:40.21 | niter3 | [fwd-outgoing] |
| 08:40.22 | niter3 | type=friend |
| 08:40.22 | niter3 | secret=showboat54 |
| 08:40.22 | niter3 | username=798201 |
| 08:40.22 | niter3 | host=fwd.pulver.com |
| 08:40.22 | niter3 | fromuser=798201 |
| 08:40.24 | niter3 | fromdomain=fwd.pulver.com |
| 08:40.26 | niter3 | nat=yes |
| 08:40.28 | niter3 | context=party1 ; this section will be defined in extensions.conf |
| 08:40.30 | Strom_C | dude, use pastebin |
| 08:40.32 | niter3 | dtmfmode=inband |
| 08:40.35 | Strom_C | ~pb |
| 08:40.36 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 08:41.01 | Strom_C | don't flood the fucking channel |
| 08:41.36 | niter3 | sorry |
| 08:41.46 | niter3 | yah uhh.. i can't place outgoign calls |
| 08:41.49 | niter3 | dont' know why anymore |
| 08:42.01 | Strom_C | let me see if i can place outgoing calls |
| 08:42.54 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-87.range86-138.btcentralplus.com) |
| 08:43.20 | Strom_C | yep |
| 08:43.26 | Strom_C | i can call out just fine |
| 08:43.40 | Strom_C | now, do me a favor and PASTEBIN your config |
| 08:43.48 | Strom_C | and I'll see if it's different from mine |
| 08:44.15 | niter3 | which part |
| 08:44.18 | niter3 | in the sip.conf |
| 08:44.19 | niter3 | ? |
| 08:45.02 | Strom_C | are you using sip or iax2? |
| 08:45.19 | niter3 | sip |
| 08:45.25 | Strom_C | why not iax2? |
| 08:45.38 | niter3 | http://pastebin.ca/165880 |
| 08:45.41 | niter3 | chose to use sip |
| 08:46.51 | Strom_C | change friend to peer and see what happens |
| 08:47.36 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 08:47.47 | JT | you should also censor your sip secrets, niter3 |
| 08:47.55 | niter3 | those arn't valid |
| 08:48.58 | Rahail | any one have link for some nice Moh |
| 08:49.03 | Rahail | mp3 |
| 08:50.27 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 08:55.22 | benjk | Rahail, didn't get your email |
| 08:55.31 | Rahail | ha |
| 08:55.38 | Rahail | i sent it from my hotmail account |
| 08:55.48 | benjk | for music on hold, you should go to www.shockwave-sound.com |
| 08:56.03 | benjk | this is royalty free music |
| 08:56.07 | benjk | and they have good stuff |
| 08:56.30 | benjk | most of their artists/composers are not members of any collecting society |
| 08:56.35 | Rahail | haa its asking money |
| 08:56.36 | Rahail | hahhe |
| 08:57.20 | benjk | if you buy music from a composer who is a member of a collecting society then you MUST pay broadcast licensing fees to your country's collecting society |
| 08:57.25 | benjk | there is no way around it |
| 08:57.34 | benjk | not even the composer himself can waive it |
| 08:57.46 | coppice | and they pocket the money |
| 08:57.54 | benjk | he would have to pay the license even if he was putting his own music on his own pbx |
| 08:58.00 | benjk | the composer, that is |
| 08:58.18 | benjk | so, you should always choose music from either |
| 08:58.57 | benjk | a) a composer who died before the collecting society system was establlished, ca 1920 |
| 08:59.10 | benjk | b) a composer who is not a member of any collecting society |
| 08:59.15 | coppice | this is why the typical mall is playing the nutcracker all day at christmas :-) |
| 08:59.23 | benjk | yep |
| 09:00.00 | benjk | also, you have to pay those license fees even if you record the music yourself |
| 09:00.05 | benjk | with your own guitar |
| 09:00.20 | coppice | it would be interesting to know where the money goes in these countries with the scam tax on blank media |
| 09:00.25 | benjk | unless the song was written by a composer who is not a member of a collecting society |
| 09:01.13 | benjk | so if you record a Beatles tune yourself with your own guitar and recording equipment, you must still pay for a broadcasting license if you want to put this on your pbx |
| 09:02.11 | coppice | so if you record a Barry Manilow tune yourself with your own guitar and recording equipment, you must seek qualified help :-) |
| 09:02.13 | benjk | Rahail, shockwave-sound are charging only for a one time license |
| 09:02.57 | benjk | so you pay like 100 USD for a CD (real one or mp3 download) and you can broadcast it as you please |
| 09:03.13 | benjk | you have paid one time for all eternity |
| 09:03.25 | coppice | there are lots of places selling CDs that can be used as background crud, without further payment. |
| 09:03.32 | benjk | this is very very reasonable |
| 09:04.11 | benjk | if you buy non-royalty free music you pay like 10.000 USD every year |
| 09:05.12 | benjk | and if you want music from a composer that is covered by a collecting society, then the only way you can get the right to put that on to your pbx is by licensing ALL MUSIC |
| 09:05.21 | benjk | its either ALL or NOTHING |
| 09:05.33 | benjk | there is no way to license only a single track |
| 09:05.48 | benjk | also, in most countries you must submit cue sheets |
| 09:05.52 | coppice | some of that background stuff is quite clever. we used to have one that was like an audio illusion. the tune seemed to be endlessly building to something, but never got there. it avoided that endless blandness of most background music, while never needing to end. |
| 09:06.31 | benjk | which means you have to create a log file of your MOH usage and send in those log files so they know what music you were actually playing, when, how often and for how long |
| 09:06.51 | benjk | yeah, those are called loops |
| 09:07.04 | benjk | they sell them in 15, 30, 45 and 60 seconds versions |
| 09:07.29 | coppice | but most just sound like a loop. this one didn't |
| 09:07.35 | Rahail | ops sorry |
| 09:07.42 | benjk | and, NO YOU CANNOT PLAY MUSIC FROM YOUR CDs ON YOUR PBX !!!!!!!!!!!!!!!!!! |
| 09:07.43 | Rahail | i forget about i asked a question in this channel |
| 09:07.59 | Rahail | any had any luck playing live radio |
| 09:08.00 | Rahail | on MOH |
| 09:08.02 | benjk | and NO YOU CANNOT PLAY MUSIC FROM THE RADIO OR MTV OR OTHER SOURCES |
| 09:08.17 | Rahail | :( |
| 09:08.19 | benjk | ITS ILLEGAL BROADCASTING |
| 09:08.26 | benjk | you cannot rebroadcast |
| 09:08.33 | benjk | you need a broadcasting license for that |
| 09:08.33 | Rahail | hehe |
| 09:08.44 | Rahail | what if i do .... free |
| 09:08.45 | coppice | playing MTV would be dangerous - you'd drive away customers |
| 09:08.59 | Rahail | i am trying to brodcast |
| 09:09.02 | Rahail | my college team |
| 09:09.03 | benjk | so go and buy a CD or two from a royalty-free music publishing company |
| 09:09.08 | JT | benjk: yeah you can get collecting society fees for those too |
| 09:09.31 | benjk | yes, but you cannot without paying the license fees |
| 09:09.42 | JT | sure |
| 09:09.57 | benjk | the thing is that the collecting system is an all-or-nothing system |
| 09:09.58 | JT | in .au it's about $100pa for retransmitting radio on MOH |
| 09:10.14 | benjk | you can only get a license for all the music that has ever been recorded |
| 09:10.14 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 09:10.29 | benjk | and they assess you by your volume |
| 09:10.38 | benjk | that's why you need to submit those cue sheets |
| 09:10.52 | coppice | JT: is that for using ordinary radio, or one of those special background music channels? |
| 09:10.52 | benjk | its very expensive and a pain in the butt |
| 09:10.57 | JT | you don't if you use radio :) |
| 09:11.08 | JT | coppice: any ordinary radio station, rebroadcast as MOH |
| 09:11.09 | benjk | it depends on the country you're in |
| 09:11.50 | benjk | in some countries they have special license packages for MOH and they assess you by the number of phone lines and employees |
| 09:11.55 | coppice | i wonder if those background music channels even exist these days |
| 09:12.02 | benjk | but it is still expensive |
| 09:12.15 | JT | yeah |
| 09:12.28 | JT | some chain stores here have their own "radio stations" broadcast over satellite |
| 09:12.38 | benjk | the by far best solution is to buy royalty-free music tracks |
| 09:12.42 | JT | a coupl use FM subcarriers, too |
| 09:13.07 | benjk | if you check those out you will find some have totally boring stuff |
| 09:13.09 | coppice | i was thinking of the FM sub carrier ones |
| 09:13.15 | benjk | but some have a good selection |
| 09:13.38 | tzafrir | benjk, there's plenty of free music. |
| 09:13.40 | benjk | you can get classical music, jazz, pop, techno/electronica |
| 09:14.19 | tzafrir | (free quality music is a slightly different story, though) |
| 09:14.20 | benjk | and man of these companies are small and they guarantee that 50% or more goes to the actual artists |
| 09:14.54 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 09:14.54 | benjk | free as in totally free of any charges whatsoever music is mostly inadequate for moh |
| 09:15.09 | benjk | you have to be extremely lucky to find something syitable |
| 09:15.41 | benjk | I found that shockwave-sound had the best selection of royalty-free music and reasonable pricing |
| 09:15.42 | JT | hire an orchestra to perform some Ludwig Van Bethoven |
| 09:15.43 | coppice | as in "I've suffered for my music. Now its your turn." :-) |
| 09:15.45 | JT | ;) |
| 09:16.15 | coppice | Beethoven's 6th used to be the most popular thing for MOH |
| 09:16.18 | nextime | asmodai : in wich country 6000/768 for 60 eur? |
| 09:16.28 | coppice | I think Nortel might have popularised it |
| 09:16.31 | dorel__ | I'm experiencing this problem where i dial out on an fxo to a number, in CLI i see "Called g0/0123456789", i hear the 'calling tone' then a second after I get a "Zap/1-1 answered ..." and then nothing, no calling tone, and that number I dialed to never rings. This happens randomly, what could be the problem causing this? |
| 09:16.39 | benjk | Well, you don't really have to hire that orchestra yourself, you can buy plenty of royalty-free Beethoven, Mozart, Bach, Vivaldi, etc etc etc from those royalty free publishing companies |
| 09:16.54 | benjk | usually about 100-250 USD per CD or 30+ USD per track |
| 09:16.59 | benjk | and that's a one time off fee |
| 09:17.06 | benjk | cheaper than hiring the orchestra ;) |
| 09:17.16 | JT | heh |
| 09:17.55 | benjk | Vivaldi four seasons is also fairly popular |
| 09:18.06 | benjk | you can get that royalty-free too |
| 09:18.46 | coppice | in a few years, if the copyrights are ever allowed to expire, I expect MOH compilations like "Tired of waiting", "Hanging on the telephone", etc. :-) |
| 09:19.26 | benjk | I found some decent royalty free Jazz too |
| 09:19.43 | benjk | the album once made it to the top ten Jazz albums in the US |
| 09:19.58 | coppice | royalty free jazz tends to be overpriced :-) |
| 09:20.06 | benjk | then I found that the guy who runs the Jazz trio is the owner of the royalty free music company |
| 09:20.53 | coppice | if you consider the number of PBXes and malls, you can probably get a CD into the top ten on those sales alone :-) |
| 09:21.25 | benjk | John Bickerton |
| 09:21.25 | benjk | Creative Director |
| 09:21.25 | benjk | UniqueTracks |
| 09:21.32 | benjk | www.uniquetracks.com |
| 09:21.39 | benjk | John Bickerton Trio |
| 09:22.07 | benjk | some decent Jazz CDs |
| 09:22.39 | benjk | this one made it into the charts before it went royalty-free |
| 09:23.01 | benjk | it only went royalty-free because John Bickerton started his own royalty free music company ;) |
| 09:23.39 | coppice | "Music to irritate customers by" |
| 09:24.08 | benjk | check it out, you can listen to the tracks online, in your web browser |
| 09:24.34 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 09:24.43 | benjk | I also found some nice techno at shockwave-sound.com |
| 09:25.12 | benjk | and their classical portfolio is quite nice too |
| 09:27.22 | *** join/#asterisk charles___ (n=asfdsadf@c-66-176-57-89.hsd1.fl.comcast.net) |
| 09:32.54 | dorel__ | when i dial out through an fxo (pstn) trunk and i'm seeing -- Called g0/0123456789 -- Zap/1-1 answered SIP/20-abcdefg -- at this point it means that the channel has been given control to the pots? |
| 09:33.21 | *** join/#asterisk purserj (n=purserj@29.124.233.220.exetel.com.au) |
| 09:33.34 | *** join/#asterisk GingerDog (n=GingerDo@oak.palepurple.co.uk) |
| 09:33.42 | purserj | ~book |
| 09:33.49 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:33.53 | GingerDog | hi; is it possible to play two sounds at the same time (e.g. music on hold, and a ringing tone) ? |
| 09:34.22 | benjk | :D |
| 09:34.39 | benjk | you sure you really wanna mix moh and ringing? |
| 09:35.14 | charles___ | anyone using ASTPP ? |
| 09:35.23 | GingerDog | well; kind of; as in i have an mp3 playing in the background, but i'd also like some sort of rining noise to take place |
| 09:35.31 | GingerDog | (well, the boss would anyway) |
| 09:35.38 | dorel__ | it sometimes happens that the call is "lost" on a zap channel... what could cause something like that? |
| 09:36.33 | benjk | GingerDog, you might want to mix in a periodic announcement while turning down the voume of the bg music |
| 09:36.51 | benjk | like "please wait .... blah blah" |
| 09:37.08 | benjk | you could do that with soxmix |
| 09:37.57 | GingerDog | ok thanks |
| 09:48.45 | coppice | NO NO NO. no periodic voice. those damed things keep making me think someone answered, and I start talking :-) |
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| 09:53.32 | *** part/#asterisk GingerDog (n=GingerDo@oak.palepurple.co.uk) |
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| 10:00.29 | benjk | coppice, not so |
| 10:00.42 | benjk | the music will still be playing in the bg |
| 10:00.53 | benjk | that's how you know |
| 10:01.08 | coppice | yeah, but the first hint of voice, and you tend to react |
| 10:01.23 | benjk | well, its a lot better than mixing music and ringing |
| 10:01.36 | benjk | and he insisted on mixing music with some kind of feedback |
| 10:04.11 | coppice | I thought MOH was supposed to *be* some kind of feedback. it sure isn't entertainment :-) |
| 10:05.21 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 10:05.40 | Rahail | hey thre I am trying to setup Queue and have option where people can press 1 for John 2 for Mike 3 for my self 4 for operator |
| 10:05.43 | Rahail | how can i do that |
| 10:06.12 | *** join/#asterisk remiss (i=bofh@46.80-203-38.nextgentel.com) |
| 10:06.43 | benjk | coppice, like I said, he (or as he put it, his boss) insisted |
| 10:08.23 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 10:12.13 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 10:15.18 | Dovid | morning all |
| 10:15.42 | Dovid | if i have a zap channel how do i transfer calls with it ? |
| 10:17.35 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 10:22.37 | docelmo | if Zap FXS usually flash button works |
| 10:23.55 | kumamoto | do you guys use scripts to write your extension or manual is the way to go? |
| 10:25.37 | kumamoto | I kinda like manual until it comes to creating dial plans which are a bitch (don't mind me it is the espresso and nicotine) |
| 10:25.47 | benjk | kumamoto, typically you write a dialplan only once |
| 10:25.50 | benjk | and reuse |
| 10:26.17 | benjk | just change some parameters in astdb |
| 10:26.18 | kumamoto | oh yeah that is it |
| 10:26.29 | kumamoto | it is a dialplan that all extensions use to call out |
| 10:26.33 | kumamoto | hmmmm |
| 10:27.10 | benjk | I have a 3500+ lines dialplan that pretty much does everything my customers ever ask for or not even ask for |
| 10:27.25 | benjk | losts of macros |
| 10:27.38 | kumamoto | holy crap where is god's name are they calling |
| 10:27.47 | kumamoto | ooooooooooooh the macros |
| 10:28.25 | benjk | well, the dialpplan knows just about everything |
| 10:28.28 | kumamoto | I will have to sit down and study macros but I will have to get customers first to start putting all my reading to practice |
| 10:28.38 | benjk | what countries have what kind of numbers |
| 10:28.45 | benjk | how to do LCR routing |
| 10:28.55 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 10:29.10 | kumamoto | benjk: u must have been a telco guy |
| 10:29.13 | benjk | features like DND, CFWD, FollowMe etc etc |
| 10:29.39 | kumamoto | LCR routing to me is like way past me like a race bike |
| 10:30.00 | kumamoto | I get DND CFWD and FollowMe |
| 10:30.53 | benjk | and it also remembers for each extension what number you last called so you can ask via IVR, it knows how called you last so you can ask via IVR or return the call, and it will also tell you the last status of incoming and outgoing calls, it will tell you the new number if you dialed a number that changed etc etc etc |
| 10:31.41 | *** join/#asterisk chode (n=chode@p54B023D6.dip0.t-ipconnect.de) |
| 10:31.43 | benjk | of course you don't write a dialplan like that in one go |
| 10:31.47 | benjk | you do it over time |
| 10:32.04 | benjk | but at some point you probably end up with something you can always reuse |
| 10:32.11 | benjk | at least that should be your goal |
| 10:33.43 | dorel__ | is there a need to explicitly enable callwaiting on pri/e1 on zapata.conf? |
| 10:35.14 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 10:37.13 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 10:37.48 | kumamoto | thanks for the advice benjk |
| 10:37.57 | benjk | welcome |
| 10:38.27 | benjk | keep in mind that dialplan execution is tremendously wasteful in Asterisk |
| 10:38.49 | benjk | it does not tokenise the dialplan |
| 10:39.25 | benjk | every time you excute somehting in the dialplan it is evaluated all over again |
| 10:39.42 | benjk | and without any of the usual techniques that have been in use for 50 years |
| 10:40.42 | benjk | it compares every app name it finds in the dialplan against every app loaded character by character, every single time |
| 10:40.51 | benjk | same for vars, contexts, extensions ... |
| 10:41.23 | kumamoto | damn doesn't that cause load on the server itself |
| 10:41.36 | benjk | so if your dialplan grows, the workload increases |
| 10:41.43 | kumamoto | it that way asterisk requires a big boy cpu |
| 10:41.46 | benjk | and it doesn't increase linear |
| 10:42.09 | benjk | or smarter developers who know their tookit of efficient algorithms |
| 10:42.22 | benjk | like hash compares, hashtable lookup etc |
| 10:42.31 | kumamoto | so if your client makes a lot of international call you dialplan will be huge |
| 10:43.00 | benjk | if you have a dial macro that calls on average about 10 applciations |
| 10:43.13 | benjk | and you have 50 concurrent calls and 150 apps loaded |
| 10:43.34 | benjk | then you look at about 50.000+ lookups |
| 10:43.39 | benjk | and compares |
| 10:44.15 | kumamoto | wow |
| 10:44.52 | benjk | asterisk uses linear searches and it compares character by character |
| 10:45.00 | benjk | every time it finds a name in the dialplan |
| 10:45.10 | benjk | that's a serious waste of resources |
| 10:45.30 | benjk | it bites you if you do embedded stuff |
| 10:46.10 | kumamoto | so one has to really try get the thing as simple as humanly possible for better performace |
| 10:46.28 | benjk | I removed all those char by char comparisons from the dialplan engine and replaced them with hash comparisons |
| 10:46.37 | benjk | got me 2.5 times as many concurrent calls |
| 10:47.00 | benjk | and that's still with linear search |
| 10:47.19 | benjk | only replacing the string comparisons with hash code comparisons |
| 10:48.13 | benjk | I am going to replace the linked lists with hashtables and thus the linear searches with hashtable lookup, that will be anoher major boost |
| 10:48.25 | benjk | I need it on those embedded boards |
| 10:48.40 | benjk | like the WRAP (AMD Geode CPU with 266MHz) |
| 10:49.13 | kumamoto | wow this is just blowing and I haven't even scratched the surface |
| 10:49.39 | benjk | but as you are starting, you probably don't worry about performance yet |
| 10:50.14 | kumamoto | yeah what do I start worrying about apart from get shit to talk to each other |
| 10:51.05 | benjk | do your stuff, learn how to write dialplans, and by the time you worry about performance, the asterisk folks will probably have copied some of the tricks they currently refuse to accept are needed to avoid this waste |
| 10:51.19 | kumamoto | lol |
| 10:51.37 | benjk | cause we're doing this in OpenPBX and that is going to be competition for them |
| 10:51.50 | benjk | eventually, they'll follow |
| 10:51.59 | kumamoto | benjk: are u working for someone or are a independent consultant? |
| 10:52.11 | benjk | I have a small consulting firm |
| 10:52.24 | benjk | well, I am part of that firm |
| 10:52.29 | kumamoto | that is in my works too |
| 10:53.02 | benjk | btw, what's with your IRC nick? |
| 10:53.34 | kumamoto | that is the only way to get experience working a home pbx is going to get one do far |
| 10:53.37 | benjk | 熊本 |
| 10:53.44 | benjk | <=kumamoto |
| 10:54.04 | kumamoto | is that what it shows as jibberish |
| 10:54.18 | benjk | that's how you write kumamoto in Japanese |
| 10:54.34 | benjk | and if you see gibberish that means you don't have the right fonts |
| 10:54.59 | kumamoto | I don't know I just pickup a horny swahili nick |
| 10:55.03 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 10:55.18 | kumamoto | lol |
| 10:55.19 | benjk | japanese != swahili |
| 10:55.42 | dorel__ | with a pri/e1 setup, if i call a number which is busy with another call im hearing the same 'calling tone' as if the destination wasn't busy... is there a way to change that? |
| 10:55.58 | benjk | kumamoto is atown in kyushu, southern japan |
| 10:56.31 | kumamoto | damn didn't know that all I know kumamoto is (hot punany) |
| 10:57.12 | kumamoto | Like I said I wasn't thinking straight at the time of choosing the nick |
| 10:57.20 | benjk | np |
| 10:57.37 | benjk | kumamoto is also a Japanese family name |
| 10:57.50 | benjk | cause the Japanese didn't have family names until about 1870 |
| 10:58.05 | kumamoto | Yeah they shouldn't go to east africa with that name I would be funny as hell |
| 10:58.07 | benjk | when the US forced Japan to open its doors |
| 10:58.29 | benjk | resulting in whats called the Meiji restauration period |
| 10:58.38 | benjk | and then folks were told to pick a family name |
| 10:58.46 | benjk | for going international style |
| 10:59.09 | benjk | so, like the Japanese are, 99% of them chose the name of their home town as family name back then |
| 10:59.17 | kumamoto | so how did they differential each other |
| 10:59.36 | kumamoto | was it like daughter or son of kumamoto? |
| 10:59.56 | *** join/#asterisk ferrox (i=ferrox@login-1.eunet.no) |
| 11:00.47 | benjk | no just an additional name as they were told |
| 11:01.44 | kumamoto | aha |
| 11:01.45 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:02.11 | Madkiss | hi all. does somebody have experienjce with configuring a cisco 7960? It's tellinbg me "Phone Unprovisioned" although I configured a line1 |
| 11:02.32 | kumamoto | oooooooooh firmware alert |
| 11:02.42 | Madkiss | what? |
| 11:03.07 | coppice | 熊本 = this bear. odd name for a town |
| 11:03.26 | kumamoto | lol |
| 11:04.10 | Madkiss | kumamoto: what do you mean? |
| 11:05.19 | benjk | origin of the bear, actually |
| 11:05.29 | kumamoto | Madkiss: I apologize I was talking about my BT200 and not your Cisco phone |
| 11:06.07 | coppice | benjk: 本 = something like "this very" in chinese |
| 11:06.18 | kumamoto | You might want to check on the manual and see what you are missing though i bet it is something minute |
| 11:06.19 | benjk | ah |
| 11:06.27 | benjk | its origin or root in Japanese |
| 11:06.32 | benjk | bear root |
| 11:06.50 | kumamoto | am still sticking to the swahili translation |
| 11:06.53 | kumamoto | lol |
| 11:06.57 | Madkiss | ah, I fixed the problem |
| 11:06.59 | Madkiss | most excellent |
| 11:07.09 | kumamoto | Madkiss: way to go |
| 11:07.28 | kumamoto | something u missed ? |
| 11:08.19 | Madkiss | no, i mistyped something |
| 11:08.56 | kumamoto | that is what usually happens missed something typos or adding shit that isn't supposed to be there |
| 11:12.39 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 11:12.39 | *** join/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 11:13.50 | macTijn | hmm |
| 11:13.59 | macTijn | why the forced +e ? |
| 11:15.21 | kumamoto | let me go for a spliff I will brb |
| 11:29.49 | Madkiss | when somebody calls me via asterisk, neither I can understand him nor can he understand me |
| 11:29.56 | Madkiss | what's wrong? |
| 11:30.00 | kumamoto | back |
| 11:30.17 | kumamoto | echos jitters or what? |
| 11:30.29 | Madkiss | no. i don't hear anything at all. |
| 11:32.40 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.183) |
| 11:33.32 | kumamoto | nothing in CLI>? |
| 11:34.30 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 11:35.52 | Madkiss | nothing real obvioius, no |
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| 11:37.19 | *** join/#asterisk Kuto (n=kuto@125.60.241.24) |
| 11:40.25 | Madkiss | bbl |
| 11:40.37 | Kuto | can i integrate this card Digium Wildcard TE412P with my pbx system? |
| 11:42.00 | kumamoto | Kuto: sure but I personally haven't had experience with it. There are some tutorials on it |
| 11:44.11 | Kuto | kumamoto: you tried using that card with xten software? |
| 11:45.10 | kumamoto | nope |
| 11:45.17 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 11:45.49 | Kuto | so what kind of connection you used that card? |
| 11:47.54 | kumamoto | no I haven't used that card yet |
| 11:49.13 | kumamoto | I haven't had the pleasure of using it |
| 11:49.47 | Kuto | im planning on setting up using e1 |
| 11:51.02 | Kuto | but i dont know how to connect it with voip and with my company pbx system |
| 11:57.49 | *** join/#asterisk RoyK (n=roy@55.80-203-66.nextgentel.com) |
| 11:59.44 | kumamoto | have you checked http://www.voip-info.org for more information |
| 12:05.46 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 12:06.37 | Cyt | Hello! Does anybody here has experience using outgoing lines from IXN (internationalnumber.com) on asterisk? I can't make calls. |
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| 12:09.40 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 12:10.49 | Kuto | kumamoto: ok tnx |
| 12:11.51 | *** join/#asterisk RoyK (n=roy@55.80-203-66.nextgentel.com) |
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| 13:11.13 | DarKnesS_WolF | anyone using java applets with asterisk ? |
| 13:20.33 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
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| 13:29.32 | Cyt | Hello! Does anybody here has experience using outgoing lines from IXN (internationalnumber.com) on asterisk? I can't make calls. |
| 13:31.15 | *** join/#asterisk s0lid (n=jlq@210.213.247.180) |
| 13:36.14 | DarKnesS_WolF | can i have a SIP extnesion that many users using it at the same time ? |
| 13:37.58 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 13:41.26 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 13:46.31 | shap | All: is there a forum or ongoing discussion somewhere concerning the configuration language? |
| 13:48.07 | tzafrir | asterisk-dev ? |
| 13:48.22 | s0lid | hi this is a broadvoice specific but how can you config 2 broadvoice account in asterisk? |
| 13:48.33 | tzafrir | Or this channel for usage questions... |
| 13:48.35 | shap | tzafrir: Where to sign up for asterisk-dev? |
| 13:48.37 | s0lid | the context of broadvoice is [sip.broadvoice.com] |
| 13:48.47 | s0lid | can i change it to anything? im not sure with broadvoice |
| 13:49.18 | tzafrir | shap: I referred to channel #asterisk-dev and the mailing list asterisk-dev@lists.digium.com (two separate entities) |
| 13:49.44 | shap | tzafrir: ack. Thanks. |
| 13:49.49 | tzafrir | anyway, try asking your Q here first. |
| 13:50.18 | tzafrir | usage questions go here and/or asterisk-users@lists.digium.com |
| 13:50.49 | *** join/#asterisk potsboy (n=chrisg@c5-168-1.vic.dial.mweb.co.za) |
| 13:50.51 | shap | tzafrir: It's apparent that there is a lot of unhappiness with the config language. Language design is something I actually know about, but before I go sticking my nose in I wanted to look at the archive/wiki/existing discussion to see what has been discussed already. |
| 13:51.36 | tzafrir | shap, what exactly do you call "configuration language"? |
| 13:52.03 | postel | the syntax of the config files most likely |
| 13:52.13 | tzafrir | If the basic-like dialplan language seems poor, take a look at ael[2] |
| 13:52.41 | shap | tzafrir: the language used in the .conf files. It sucks. Everybody seems to agree about that. I have some *simple* concrete suggestions that would help a lot and be very easy, and some serious suggestions that would require more thought. This probably isn't the appropriate forum for either. |
| 13:53.17 | tzafrir | shap, than the asterisk-dev list is probably the right place. However be sure to take some fire |
| 13:53.41 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-88-174.dsl.chcgil.sbcglobal.net) |
| 13:53.42 | tzafrir | Consider the possibility that similar suggestions have been raised before. |
| 13:54.14 | tzafrir | Also note that suggestions are much more welcomed when someone is actually willing to implement them |
| 13:54.16 | shap | tzafrir: yes, that's why I want to look at previous discussion first. But some of this is damned obvious, and I'ld be willing to do the implementation effort. |
| 13:54.53 | Flauto | when i upgrade asterisk, where is the modules that i need to delete |
| 13:55.07 | tzafrir | Can you give a sample suggestion? |
| 13:55.20 | Dovid | Flauto: if u make and then make install it will tell u |
| 13:55.31 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 13:55.38 | Flauto | okay |
| 13:55.55 | postel | shap: Why dont you blog a quick draft with some keypoints of what is wrong and your suggestions on it and then post a link here and in -dev. Then we'll all have something more concrete to comment on. |
| 14:00.04 | shap | postel: That's a good idea. One question that I should ask first: The AEL design is somewhat damaged. Is there a reason that an existing, well-established scripting language was not adopted? I've recently done some pretty exhaustive investigation of scripting tools in connection with the Coyotos project. Didn't find anything that would work for my application in Coyotos, but found several that would work nicely for Asterisk, and at least on |
| 14:00.50 | shap | all: Is there anything comparable to an "include" statement that can be placed into a .conf file? That is: include of another file. |
| 14:01.02 | postel | yes |
| 14:01.17 | Flauto | do i need to upgrade zaptel? |
| 14:01.25 | shap | postel: yes to which question? |
| 14:01.35 | postel | to the one above my answer |
| 14:01.49 | shap | postel: does the include syntax accept wildcards in the filename? |
| 14:02.53 | postel | Dont think so, not that i have tried, usually i declare includes of a couple of files. |
| 14:03.26 | Flauto | if i upgrade zaptel, do i need to delete the old ko files? |
| 14:03.33 | *** join/#asterisk Sir_Diddymus (n=doe@p54B0D021.dip.t-dialin.net) |
| 14:03.48 | postel | Nothing stopping you trying though, its not much to expand the regular expression |
| 14:04.19 | tzafrir | shap, ael is a wrapper to the dialplan language |
| 14:04.35 | shap | postel: Motivation: sticking extensions in extensions.conf (which is poorly named) is not a good idea. extensions.conf should include "extensions/*.conf". This would make edits more independent. It would simplify pbx_config as well. Indeed, expanding the regexp is not a big deal. This, by the way, is the most important change I had in mind in the short term. |
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| 14:05.17 | tzafrir | as for include: there aer two types of them: "#include" is generic and works with all the config files. "include => " is specific to extensions.conf |
| 14:05.43 | shap | tzafrir: Yes. And all the docs for it have a bunch of cautions about how easy it is to screw up AEL syntactically, and the language syntax is, um, obscure |
| 14:06.03 | shap | tzafrir: #include does text substitution and include => does context insertion? |
| 14:06.14 | tzafrir | to do that simply make you extensions.conf a one-line: |
| 14:06.25 | tzafrir | #include "extensions/*.conf" |
| 14:06.48 | tzafrir | I'm not familiar with ael and/or ael2 |
| 14:06.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 14:07.55 | Corydon76-home | tzafrir: ael imposes an additional abstraction layer over extensions.conf |
| 14:07.56 | tzafrir | shap, yes, you're right about #include nad include=> |
| 14:08.10 | shap | tzafrir: actually, defaults probably need to stay in extensions.conf for compatibility reasons. But this is very good. The current config language really has two distinct problems. The one everybody talks about is that the language is not great, but the bigger one is that the "sections", in practice, correspond to distinct items, and it would be cleaner to have one file for each of those. |
| 14:08.27 | tzafrir | Corydon76-home, so I gather you don't recommended it as a magic bullet |
| 14:08.50 | shap | tzafrir: but I am greatly relieved, because this is now a problem of idiom, not language design, and that discussion is easier because it is purely a matter of giving examples. |
| 14:08.53 | Corydon76-home | tzafrir: ael is "compiled" to an extensions.conf, of sorts |
| 14:09.08 | tzafrir | yeah, I know that |
| 14:09.12 | Corydon76-home | shap: the person you should be talking to is murf |
| 14:09.13 | shap | Corydon76-home: so it is not actually an interpreted language? |
| 14:09.31 | Corydon76-home | shap: no, it is not |
| 14:09.47 | shap | Corydon76: how to reach him? I'm not actually that eager to rebuild the language, but rearranging the config organization would make the whole thing a lot easier to manage. |
| 14:10.03 | Corydon76-home | shap: /whois murf |
| 14:10.13 | shap | Corydon76: ack on murf. Why not interpreted? |
| 14:11.07 | Corydon76-home | shap: if you'd like to make a separate pbx core that does interpreted ael, you're more than welcome to |
| 14:11.12 | shap | corydon: /whois murf responds "no nick or channel" |
| 14:11.17 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 14:11.27 | Corydon76-home | shap: yeah, he's offline ATM |
| 14:11.28 | tzafrir | shap, try writing a dialplan with ael, then run 'show dialplan, and you'll see what it is compiled to |
| 14:11.38 | shap | corydon76: yes, that is roughly what I had in mind, but I would like to understand first why interpretation was not selected. |
| 14:11.42 | shap | tzafrir: tnx |
| 14:12.06 | Corydon76-home | shap: That's a great question that I'm pretty sure nobody has an answer to |
| 14:12.33 | Corydon76-home | shap: however, incompatibility with existing functions/apps is probably part of it |
| 14:13.08 | tzafrir | There was also once a res_perl that allowed writing dialplan logic in perl. And so is res_js (Javascript(!)) |
| 14:13.14 | tzafrir | But those are hardly used |
| 14:13.19 | shap | corydon: I can see an argument based on resources, but the overhead wouldn't be a problem. The reason to think about is that there are *other* things one might want to do (in addition to dialing) in a dial plan. For example, reaching out and pushing a new page to a browser or some such. |
| 14:13.26 | Corydon76-home | Making AEL interpreted would pretty much mean remaking the entire system monolithicly |
| 14:13.42 | shap | tzafrir: I shall look (gritting my teeth, but that's religion) at res_js. |
| 14:14.01 | Corydon76-home | shap: you can already do that |
| 14:14.20 | Strom_C | yeah, you could run an AGI script that'll do the push |
| 14:14.41 | shap | corydon: how? (aside: I really was barfing over AEL syntax -- convergence wasn't my motivation for opening the discussion) |
| 14:14.55 | shap | strom: AGI? |
| 14:15.22 | Strom_C | shap: asterisk gateway interface |
| 14:15.24 | Strom_C | ~agi |
| 14:15.25 | jbot | from memory, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 14:15.37 | tzafrir | shap, it's a somple text-based interface that allows you to interact with an external process of your choosing. Not unlike CGI |
| 14:16.00 | shap | strom, tzafrir: I'll look in to that. Let me not ask stupid questions until I have done so. |
| 14:16.18 | *** join/#asterisk foxmjay (i=root@41.248.6.190) |
| 14:16.32 | shap | All: Thanks on the #include thing, though. That makes things *lots* more comfortable. |
| 14:16.37 | tzafrir | shap, msg jbot term before you feel like asking a stupid question... |
| 14:16.58 | foxmjay | hi , someone tried to compile zaptel on a 2.6.17 kernel |
| 14:17.13 | shap | tzafrir: tnx. Guess it's obvious I'm not an IM expert. |
| 14:17.18 | shap | all: offline for a few minutes |
| 14:17.20 | foxmjay | cause me i find errors while compiling , |
| 14:18.24 | foxmjay | with both the zaptel-1.2.5 and 1.2.9 |
| 14:18.47 | tzafrir | foxmjay, could you pastebin those errors? Also: what distro and what kernel? |
| 14:18.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 14:19.23 | foxmjay | ok with the zaptel 1.2.9 : |
| 14:19.28 | tzafrir | ~pb |
| 14:19.30 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:20.02 | foxmjay | make[3]: /usr/src/zaptel-1.2.9/wct4xxp/../oct612x/octasic-helper : commande introuvable |
| 14:20.02 | foxmjay | make[3]: *** Pas de règle pour fabriquer la cible « /usr/src/zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h », nécessaire pour « /usr/src/zaptel-1.2.9/wct4xxp/vpm450m.o ». Arrêt. |
| 14:20.03 | foxmjay | make[2]: *** [/usr/src/zaptel-1.2.9/wct4xxp] Erreur 2 |
| 14:20.03 | foxmjay | make[1]: *** [_module_/usr/src/zaptel-1.2.9] Erreur 2 |
| 14:20.04 | foxmjay | make[1]: Leaving directory `/usr/src/linux-2.6.17' |
| 14:20.06 | foxmjay | make: *** [linux26] Erreur 2 |
| 14:20.24 | *** mode/#asterisk [+b %foxmjay!*@*] by Corydon76-home |
| 14:20.54 | Strom_C | PASTEBIN, stupid |
| 14:21.02 | kFuQ | ROFL |
| 14:21.15 | kFuQ | good ol id10t error lol |
| 14:21.21 | *** part/#asterisk foxmjay (i=root@41.248.6.190) |
| 14:21.26 | *** join/#asterisk foxmjay (i=root@41.248.6.190) |
| 14:21.28 | *** mode/#asterisk [-b %foxmjay!*@*] by Corydon76-home |
| 14:21.35 | tzafrir | Nice demostration of insticts here... |
| 14:21.40 | foxmjay | sory for posting it here |
| 14:21.46 | foxmjay | :) |
| 14:21.49 | Strom_C | ~pb |
| 14:21.50 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:22.58 | *** join/#asterisk RoyK (n=roy@ti211310a080-0143.bb.online.no) |
| 14:25.01 | foxmjay | tzafrir: ok here is http://bzflag.pastebin.ca/166047 |
| 14:25.48 | tzafrir | anyway, that is basically http://www.sineapps.com/news.php?rssid=1496 (yuck) |
| 14:26.20 | tzafrir | (another fix is not to build that subdirectory, by a slight tweak in the makefile |
| 14:26.37 | tzafrir | An even better tweak would be a release of Zaptel that builds |
| 14:27.50 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
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| 14:29.39 | foxmjay | humm, s o r y , but i don't understand what will be the different to compile it from package or from svn |
| 14:30.41 | Corydon76-home | Packages are snapshots of SVN. SVN development continues, even after a snapshot is taken |
| 14:31.00 | foxmjay | ahh ok |
| 14:31.38 | foxmjay | i'll try and see what happen |
| 14:31.50 | tzafrir | actually that points to an svn tag. A tag will not change |
| 14:32.19 | tzafrir | (that is: not supposed to. In practice it is a branch like any other) |
| 14:33.09 | tzafrir | A certain directory was accidentally left out of the tarball, but it will be pulled if you grab the source using subversion |
| 14:33.30 | tzafrir | This is *exactly* the same version (1.2.9) |
| 14:34.28 | foxmjay | ok i understand now :) |
| 14:38.15 | Flauto | i just upgraded to 1.2.12 now |
| 14:38.23 | Flauto | the music on hold stopped working |
| 14:38.54 | Flauto | telling me that Sep 10 09:33:56 WARNING[6426]: file.c:512 ast_openstream_full: File /var/lib/asterisk/moh-native/11 does not exist in any format |
| 14:39.16 | Flauto | and the music files are in mp3 format |
| 14:39.21 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 14:39.29 | Flauto | it seems that the new version does not support it |
| 14:39.31 | [TK]D-Fender | Flauto : Did you recompile asterisk-addons? |
| 14:39.48 | Flauto | no |
| 14:39.55 | [TK]D-Fender | Flauto : Maybe a new modules.conf is not loading the .so even if its there.... |
| 14:39.55 | Flauto | i will do that now |
| 14:40.10 | Flauto | thanks |
| 14:40.54 | shap | flauto: You may want to consider converting those mp3's to WAV instead -- much lower overhead, and then you don't need the addons. |
| 14:41.26 | [TK]D-Fender | Flauto : Not a bad idea either, but I want support, even if I don't NEED it. |
| 14:41.32 | foxmjay | tnx alot the compilation & installation are fine, but there is another problem :(, modprobe zaptel or wctdm give me module not found |
| 14:41.56 | shap | foxmjay: if you upgraded kernel, need to rebuild |
| 14:42.46 | foxmjay | no i make a package with make-kpkg |
| 14:43.04 | foxmjay | i mad :) |
| 14:44.16 | foxmjay | so the kernel it's already build ? |
| 14:44.53 | Corydon76-home | What distro? |
| 14:45.00 | foxmjay | debian |
| 14:45.03 | foxmjay | sarge |
| 14:45.17 | Corydon76-home | Did you 'make install'? |
| 14:45.35 | foxmjay | make install what zaptel ? |
| 14:45.41 | Corydon76-home | yes |
| 14:45.43 | foxmjay | lol |
| 14:46.04 | foxmjay | yes |
| 14:46.09 | [TK]D-Fender | bbiab |
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| 14:46.30 | Corydon76-home | Then your source probably does not match your installed kernel version |
| 14:46.38 | Corydon76-home | rather, kernel source |
| 14:48.04 | foxmjay | i make a *.deb package with the kernel-source-2.6.17 , and it's the same source that i use to compile the zaptel, so ... |
| 14:48.27 | tzafrir | did you build the kernel on your own? |
| 14:48.40 | foxmjay | yep |
| 14:49.09 | foxmjay | mabe i forgot something in the menuconfig ? |
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| 14:49.22 | tzafrir | I figure that /lib/modules/<VERSION>/build points to the kernel source directory, right? |
| 14:49.31 | foxmjay | yes |
| 14:49.43 | tzafrir | If so, just remove the symlink /usr/src/linux |
| 14:49.59 | foxmjay | ok i'll try |
| 14:51.05 | tzafrir | foxmjay, it's a shame you didn't use zaptel debs. m-a is really nice |
| 14:51.56 | foxmjay | humm , i didn't know that there is a deb package for zaptel , what is the last version ? |
| 14:52.47 | tzafrir | Almost. I still have not uploaded anything for 1.2.9, as I'm not sure how it will be resoved |
| 14:53.25 | Flauto | shap, what bit rate i should use if i convert mp3 files to wav? |
| 14:53.33 | Flauto | now, musiconhold is working |
| 14:53.56 | Flauto | thanks to shap and TK |
| 14:54.20 | tzafrir | Flauto, you need 8khz for Asterisk |
| 14:55.12 | tzafrir | (And even if not: it's a waste of space and CPU time) |
| 14:55.24 | Flauto | okay |
| 14:55.30 | Flauto | i will try to do that |
| 14:55.44 | Flauto | thanks tzafrir |
| 14:56.10 | shap | all: in sip.conf, is secret= field optional? I can use a dummy password, but for phones on internal net I see no reason for a secret. |
| 14:57.24 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 14:59.55 | EzWayz | what can cause this problem ??? Dropping voice to exceptionally long queue on IAX2/600- |
| 15:00.04 | shap | Ok. I'm going through the defaults in sip.conf. I'm sure this is all well documented somewhere online. Where to look? |
| 15:00.40 | *** part/#asterisk sergeus (i=guru-dev@195.112.98.13) |
| 15:02.07 | shap | ~sip.conf |
| 15:04.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.226) |
| 15:07.24 | *** join/#asterisk ESCulapio_ (n=ESCulapi@196stb68.codetel.net.do) |
| 15:10.45 | [TK]D-Fender | shap : ... |
| 15:10.47 | [TK]D-Fender | ~wiki |
| 15:11.15 | [TK]D-Fender | shap : And I highly advise against not setting passwords.... |
| 15:12.08 | [TK]D-Fender | shap : http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 15:13.04 | Flauto | tk d-fender, i have a problem with webvmail, is there anything i need to prepare for it? |
| 15:13.05 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-87.range86-138.btcentralplus.com) |
| 15:17.44 | [TK]D-Fender | Flauto : Never touched it. Whats the problem? |
| 15:18.23 | DarKnesS_WolF | [TK]D-Fender: hello guru ;-) |
| 15:18.34 | DarKnesS_WolF | [TK]D-Fender: show translation told me everything i want to know about codecs ;-) |
| 15:18.36 | Flauto | i was using mandriva 2005 with asterisk and i had webvmail worked perfectly |
| 15:18.36 | *** join/#asterisk markQing (n=mark@82-197-202-242.dsl.cambrium.nl) |
| 15:18.57 | Flauto | but after i upgraded to mandriva 2006, it stopped working |
| 15:19.09 | Flauto | when i compile webvmail |
| 15:19.14 | DarKnesS_WolF | Flauto: what is the error message ? |
| 15:19.26 | DarKnesS_WolF | Flauto: it's webvmail issue or apache2 issue ? |
| 15:19.28 | Flauto | it is show in red color in the cgi-bin |
| 15:19.46 | DarKnesS_WolF | Flauto: red color on the terminal ? |
| 15:20.00 | DarKnesS_WolF | means it's a messing link go to cgi-bin and ls -la |
| 15:20.07 | DarKnesS_WolF | see if this file exists or not |
| 15:20.31 | Flauto | darkness, yes in terminal |
| 15:21.21 | Flauto | i think it might be an apache problem |
| 15:21.21 | DarKnesS_WolF | Flauto: if it's read then it's a deleted file . |
| 15:21.21 | DarKnesS_WolF | [TK]D-Fender: i'm going to eat mahshy " an egyptian food " wann join :P? |
| 15:21.21 | Flauto | oh |
| 15:22.00 | DarKnesS_WolF | Flauto: when i upgraded from 10.1 to 2006 apache stoped working .. cuz they changed the /etc/httpd/ files check this and make sure that apache and is fine and the link to the file is correct. |
| 15:22.19 | [TK]D-Fender | DarKnesS_WolF : Probably, but my DSL line doesn't support matter/energy stream conversion w/ Heisenberg compensators yet ;) |
| 15:24.25 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 15:26.14 | DarKnesS_WolF | [TK]D-Fender: oh too bad :-s |
| 15:28.10 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:32.44 | *** join/#asterisk Dotdesigner (n=icechat5@68-184-145-28.dhcp.stbr.ga.charter.com) |
| 15:34.36 | Dotdesigner | is any one familiar with setting up the mysql with asterisk? |
| 15:35.41 | Dotdesigner | i keep getting a connetion failed to asterisk DB in the CLI when a call comes in |
| 15:39.16 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.226) |
| 15:42.59 | niter3 | Hey guys I notice when users are calling it only passes the name of the caller but not the actual phone number... How can I set this? |
| 15:43.34 | *** join/#asterisk zimdog (n=zimdog@c-24-9-69-162.hsd1.co.comcast.net) |
| 15:45.07 | Strom_M | niter3, are they users local to your system? |
| 15:46.44 | zimdog | Hello. I have just purchased some g.729 licenses from the digium website. I am running x86_64 and did not see a link for this architecture is it not supported? |
| 15:47.42 | niter3 | Strom_C: yes |
| 15:47.47 | niter3 | jus the extensions |
| 15:47.54 | niter3 | i want the their extension numbers showing up |
| 15:48.20 | Strom_M | SIP? ZAP? IAX2? |
| 15:48.25 | niter3 | sip |
| 15:48.37 | Strom_M | in each sip.conf entry, you shoud have |
| 15:48.53 | Strom_M | callerid="Name Here"<3115552368> |
| 15:49.31 | niter3 | Strom_M: So i can have callerid="name"<222> |
| 15:49.32 | niter3 | ? |
| 15:49.34 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 15:49.38 | Strom_M | yes |
| 15:51.19 | niter3 | thanks |
| 15:51.39 | `Tingles` | morning.. |
| 15:52.13 | Strom_M | morning |
| 15:53.22 | inv_arp[work] | need an simple outbound only iax2 provider.. any recommendations |
| 15:54.40 | [TK]D-Fender | inv_arp[work] : www.voicepulse.com |
| 15:55.01 | acehunky | voipjet is quite simple ... |
| 15:55.03 | acehunky | no commitment |
| 15:55.16 | zimdog | Anyone know if the digium g.729 supporrts athlon 64 processor |
| 15:55.18 | acehunky | put $5 and start calling worldwide .. |
| 15:55.41 | h3x | i dont think you want to run asterisk in x64 mode anyway |
| 15:55.52 | acehunky | umm zimdog: I have compiled the recent g729 codec posted on the biz list |
| 15:56.10 | acehunky | and it works good on amd64 + opteron both |
| 15:56.24 | inv_arp[work] | acehunky: seems voipjets test server is down at the moment.. |
| 15:56.31 | acehunky | naa its working .. |
| 15:56.39 | acehunky | i am connected to it inv_arp[work] |
| 15:57.08 | acehunky | voicepulse requires you to get a number .. minimum commitment .. etc etc .. |
| 15:57.11 | *** join/#asterisk kph100 (n=kph100@206-248-135-65.dsl.teksavvy.com) |
| 15:57.14 | acehunky | VoipJet is more cool .. |
| 15:58.10 | zimdog | acehunky is this accesible from the digium website or is it a development version |
| 15:59.03 | inv_arp[work] | acehunky: hmm im getting "No one is available to answer at this time (1:0/0/0)" my balance is sufficient.. |
| 15:59.05 | acehunky | zimdog: are you tracking the asterisk-biz list ? |
| 15:59.20 | acehunky | ohh .. inv_arp[work] : east.voipjet.com (66.246.72.34) more reliable but requires a balance > 20 dollars to use |
| 15:59.29 | inv_arp[work] | im using test |
| 15:59.40 | zimdog | no |
| 16:01.29 | zimdog | I am subscribed but have not been following it for a few months |
| 16:01.51 | kph100 | how can i dynamically create dialplan contexts using realtime *? |
| 16:02.21 | acehunky | zimdog: there was some recent posting on G729 and G723 codec thingy |
| 16:02.30 | acehunky | someone posted URL for codec download |
| 16:02.51 | acehunky | which i think is equal to the free implementation .. |
| 16:03.09 | acehunky | it compiled well on my opteron .. |
| 16:03.42 | zimdog | thanks I was hoping that the version I have paid for from digium would work :) |
| 16:04.05 | Strom_M | zimdog, give digium a ring tomorrow and they can help you out |
| 16:04.32 | acehunky | zimdog: i think you need 2 call them .. :) |
| 16:05.21 | kph100 | storm -- do you know how to create dialplan context in * realtime? |
| 16:05.31 | Strom_M | my name is not storm |
| 16:05.33 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:05.48 | zimdog | Will do . Just been missing their office hours so was hoping could get something going today while i had some free time |
| 16:05.59 | kph100 | storm=>strom_M |
| 16:06.24 | Strom_M | and no, i have not worked with the realtime architecture yet |
| 16:08.18 | kph100 | thank you sotrm. |
| 16:08.31 | Strom_M | sotrm? |
| 16:08.44 | Strom_M | are you just trying to piss me off now? |
| 16:10.27 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 16:10.48 | Strom_M | hey linagee :) |
| 16:11.25 | *** join/#asterisk potsboy (n=chrisg@c5-204-2.vic.dial.mweb.co.za) |
| 16:14.26 | niter3 | Strom_M: I see the issue nwo.... |
| 16:14.41 | Strom_M | ? |
| 16:14.46 | niter3 | Strom_M: It works great within SIPs but when using an outbound sip provider it won't place the call out..... |
| 16:15.05 | niter3 | Strom_M: Somehow I have to strip away the callerid info the the outbound calls work on my sip out provider |
| 16:15.21 | Strom_M | or maybe you have to make the caller ID info look like a real phone number |
| 16:16.35 | niter3 | exten => _9.,1,Dial(CALLERID(number)=) |
| 16:16.35 | niter3 | exten => _9.,2,Dial(CALLERID(name)=) |
| 16:16.35 | niter3 | exten => _9.,3,Dial(SIP/${EXTEN:1}@cliconnect,80,r) |
| 16:16.35 | niter3 | exten => _9.,4,Hangup |
| 16:16.38 | niter3 | would this not work |
| 16:16.51 | niter3 | I have nothing set for caller id but calls will not plae with that |
| 16:16.55 | niter3 | huh |
| 16:16.56 | Strom_M | uh |
| 16:17.08 | Strom_M | first off |
| 16:17.09 | tzafrir | It is a free imlementation, as long as you have a license from Intel. That license is basically free for noncommercial (educational?) use. For commecrcial use it costs quite a bit |
| 16:17.14 | Strom_M | use pastebin next time |
| 16:17.31 | Strom_M | next, you can't Dial(CALLERID(num)=) |
| 16:17.35 | Strom_M | you have to use Sey |
| 16:17.37 | Strom_M | er, Set |
| 16:18.26 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 16:18.40 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:18.44 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 16:18.46 | pifiu | hey everyone |
| 16:20.48 | *** join/#asterisk Kuto (n=kuto@125.60.241.24) |
| 16:26.20 | Dotdesigner | any one want to offer a suggestion as to why asterisk fails to connect to MYSQL |
| 16:26.52 | h3x | user error |
| 16:27.05 | Dotdesigner | lol yeah - i've checked so many times |
| 16:27.37 | Strom_M | balls on a stick |
| 16:27.51 | Dotdesigner | the error log says that there is no user, database, and password provided. i've checked and double checked but no go |
| 16:27.54 | nextime | Dotdesigner : you really need to look at log file to see wich error mysql server say when you try to connect |
| 16:28.51 | Dotdesigner | i've made sure the socket that the mysql driver references is located in the correct location |
| 16:29.33 | *** part/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 16:29.57 | Dotdesigner | i can't figure out why it says there is no login info in the error log. i've made sure at is all correct |
| 16:31.15 | Kuto | hi all, is it possible to directly connect the cisco ip phones to asterisk? |
| 16:31.28 | Strom_M | Kuto, yes |
| 16:31.40 | Strom_M | i have cisco ip phones running off my asterisk box |
| 16:32.49 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 16:33.07 | Kuto | Strom_M: you use this type of card? => Wildcard TE412P |
| 16:33.16 | Dotdesigner | res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. |
| 16:33.24 | Dotdesigner | error log contains stuff like this: res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. |
| 16:33.37 | Strom_M | Kuto, what does that have to do with cisco IP phones? |
| 16:34.46 | Kuto | im trying to setup a call center and if possible to use that card and integrated to cisco ip phones |
| 16:35.08 | Kuto | i just need some detail before purchasing a very expensive card |
| 16:35.28 | Strom_M | It's possible to integrate asterisk with cisco ip phones, but you don't need a T1 card to do it. The T1 card has nothing to do with your IP phones. |
| 16:36.36 | Kuto | e1 => T1 card => cisco ip phone ? |
| 16:37.01 | Strom_M | either i'm not understanding you or you're very confused |
| 16:37.54 | Kuto | confused i guess...so whats e1 or t1 card used for? |
| 16:38.36 | Strom_M | connecting to E1 or T1 lines from your telco or legacy PBX equipment |
| 16:43.38 | Kuto | Telco => T1 card = lan? |
| 16:43.46 | *** join/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 16:43.48 | Strom_M | what? |
| 16:43.50 | Strom_M | no |
| 16:43.56 | Strom_M | telco -> t1 card -> asterisk |
| 16:45.15 | Kuto | telco -> t1 card -> asterisk => lan |
| 16:45.54 | Strom_M | yes, asterisk can talk SIP or IAX2 or whatever you want |
| 16:47.40 | Kuto | SIP is using headset connected to pc..right? |
| 16:48.09 | Strom_M | or a SIP hardphone |
| 16:48.17 | Strom_M | like the cisco sets |
| 16:50.00 | Kuto | Strom_M: what is the maximum line the asterisk can handle? |
| 16:50.23 | Strom_M | it depends on how much computer and bandwidth you throw at it |
| 16:50.30 | benjk | 42 |
| 16:50.36 | benjk | its always 42 |
| 16:50.37 | Strom_M | 800km |
| 16:50.45 | directory | 800km |
| 16:51.12 | Kuto | whats 800 km? |
| 16:51.17 | Strom_M | a joke |
| 16:51.19 | benjk | the length of the line |
| 16:51.25 | benjk | 42 is the number of the line |
| 16:51.37 | benjk | the maximum line you asked for |
| 16:53.06 | *** join/#asterisk LoneShadow (n=duh@59.92.204.231) |
| 16:54.23 | Kuto | 21 toll free number + 21 localines => t1 card => asterisk => SIP..possible? |
| 16:54.40 | Strom_M | 42 is not a hard limit, kuto |
| 16:54.44 | Strom_M | 42 is a joke |
| 16:55.11 | Strom_M | but what you're describing is very common |
| 16:55.26 | h3x | switched long distance and toll free is futile |
| 16:55.30 | h3x | its expensive as hel |
| 16:55.31 | h3x | l |
| 16:55.31 | benjk | Douglas Adams, The Hitchhiker's Guide to the Galaxy |
| 16:55.52 | benjk | 42 is the answer to the question about life, the universe and everything |
| 16:58.26 | Kuto | Strom_M: ok, so if i have 250 lans with voice connected to asterisk, how would i compute for my bandwidth requirements? |
| 16:58.45 | Strom_M | 250 separate local area networks? |
| 16:59.06 | benjk | lans don't really count |
| 16:59.11 | benjk | concurrent calls count |
| 16:59.25 | benjk | how many people will talk at the same time? |
| 16:59.27 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 16:59.43 | Kuto | atleast 20 concurrent calls |
| 16:59.44 | h3x | i think he meant to say lines |
| 16:59.49 | Strom_M | easy |
| 16:59.59 | Strom_M | 20 is nothing |
| 17:00.15 | benjk | if you use ulaw (uncompressed voice) then it would be about 20 x 80kbps |
| 17:00.35 | Strom_M | benjk's example assumes you're using SIP, of course |
| 17:00.36 | benjk | that doesn't account for any savings due to bundling the calls |
| 17:00.49 | benjk | not necessarily |
| 17:01.11 | benjk | if you have 20 locations and no call goes between the same locations, then there is no savings in trunking |
| 17:01.24 | Kuto | that means my bandwidth should be approximately running 1.6mbps? |
| 17:01.54 | Strom_M | right, but a ulaw call over iax2 only uses about 76kbps |
| 17:02.09 | Strom_M | iax2 is a more efficient transport framing mechanism than rtp |
| 17:02.19 | benjk | yeah well, that's why I was using that word "about" |
| 17:03.20 | Kuto | if i have a E1 lines, then definitely i need additional bandwidth? |
| 17:03.37 | Strom_M | Kuto, it depends |
| 17:03.48 | Strom_M | Kuto, how soon do you need to have a solution implemented? |
| 17:04.57 | Kuto | i got 2 weeks the least =( |
| 17:04.58 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 17:05.07 | Strom_M | Kuto, here's my advice: |
| 17:05.09 | Strom_M | ~hafc |
| 17:05.16 | jbot | i heard hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 17:05.26 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 17:05.56 | kink0 | hello, I still having problems with deadlock and many asterisk threads, can anybody help me ? |
| 17:07.47 | Kuto | Strom_M: tnx, your info is quite enough |
| 17:09.05 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 17:10.36 | Kuto | last few question, does asterisk eats internet connection? |
| 17:10.50 | Strom_M | it eats them for lunch |
| 17:11.07 | benjk | dpends on the time you start it |
| 17:11.26 | benjk | depending on time of day, it could be for breakfast or dinner |
| 17:11.48 | Strom_M | or even brunch |
| 17:11.55 | Kuto | if my lan runs on 10/100 mbps, it can handle asterisk right? does internet connections/sharing is affected also? |
| 17:13.49 | linagee | Kuto: no. you have to have gigabit ethernet for asterisk. that's all it supports. |
| 17:13.59 | benjk | heh |
| 17:14.05 | Strom_M | no, you have to have FIBER EVERYWHERE |
| 17:14.09 | linagee | :-D |
| 17:14.13 | Strom_M | voip and copper just dont get along |
| 17:14.19 | benjk | that that's only if you need ultrawideband audio |
| 17:14.19 | Strom_M | they're like oil and lions |
| 17:14.28 | h3x | you need a big fat pipe |
| 17:14.32 | h3x | like the alaska pipeline |
| 17:14.37 | Strom_M | especially one made of steel |
| 17:14.42 | Strom_M | none of that PVC pipe stuff |
| 17:14.45 | *** join/#asterisk ropeguru (n=ropeguru@c-68-57-84-56.hsd1.va.comcast.net) |
| 17:15.00 | linagee | PVC for brunch? mmm... |
| 17:15.33 | linagee | Strom_M: gotta have fiber in your diet |
| 17:15.40 | Strom_M | mmmmmm |
| 17:15.44 | Strom_M | Corning-fresh |
| 17:16.45 | linagee | Strom_M: if my internet comes in a 10/100 line but i have no idea how fast it really is, what is the air speed velocity of an unlaiden swallow? |
| 17:17.08 | Strom_M | what do you mean? an African or a European swallow? |
| 17:17.19 | linagee | :-D american swallow. |
| 17:17.35 | Strom_M | catsex! |
| 17:17.37 | linagee | lol |
| 17:17.41 | Strom_M | or, as you'd say in Dutch: |
| 17:17.46 | Strom_M | caatsechs! |
| 17:19.40 | Kuto | this is nuts..lols |
| 17:21.05 | Strom_M | lolphones |
| 17:22.17 | linagee | Strom_M: help! i have two weeks to implement a fully functional tier 1 data center complete with 40 CRACs, 200 cabinets, 50 megawatt feed, security guard, and a pear tree! |
| 17:22.30 | h3x | hahaha |
| 17:22.31 | Strom_M | zong |
| 17:22.36 | h3x | and i need to do it for $300 ! |
| 17:22.37 | Strom_M | you need.... |
| 17:22.41 | linagee | h3x: lol |
| 17:22.45 | Strom_M | LOL-IN-A-BOX |
| 17:22.59 | Strom_M | the only fully functional idiotproof LOL on the market |
| 17:23.04 | linagee | Strom_M: solution in a box. warning: box may cause permanent blindness. |
| 17:23.07 | h3x | i have a fake tree in my data center lobby |
| 17:33.13 | *** join/#asterisk knarfly (n=bdavis@c-65-34-177-3.hsd1.fl.comcast.net) |
| 17:33.54 | QbY | Members/Queues question.. Agent/@ and Agent/: -- what is the difference? Where can I find documentation on it |
| 17:34.58 | kph100 | w |
| 17:35.24 | kph100 | is it possible to run realtime static extensions.conf and realtime extensions.conf at the same time? |
| 17:36.02 | *** join/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 17:36.30 | pabluss | Hi, afternoon where I can find asterisk rpm packages for centos ? |
| 17:36.54 | pabluss | I've tryed to search by google but without success |
| 17:37.07 | Strom_M | just use svn |
| 17:37.14 | Strom_M | the rpms are going to be out of date |
| 17:37.31 | pabluss | hmm |
| 17:38.35 | pabluss | problem , what happens if i need to update my centos server?? i could'nt to do it by yum I need recompile all the asterisk packages also.. |
| 17:39.06 | Strom_M | gasp, you might actually have to write a bash script or something |
| 17:39.10 | Strom_M | the horror |
| 17:39.18 | Strom_M | :) |
| 17:39.35 | benjk | SVN is going to be out of date too |
| 17:39.45 | benjk | probably a few minutes after you downloaded |
| 17:39.57 | *** join/#asterisk Winkie (i=slain@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 17:39.57 | pabluss | oh oh |
| 17:39.57 | pabluss | :D |
| 17:39.58 | benjk | so you may want to spend the entire day upgrading |
| 17:39.58 | knarfly | benjk: use FreeBSD... |
| 17:39.59 | benjk | every day |
| 17:40.05 | benjk | you don't have to do anything else |
| 17:40.23 | benjk | just upgrade all your software all the time from SVN everywhere |
| 17:40.31 | benjk | what else is there to do? |
| 17:40.43 | benjk | production server? whazzat? |
| 17:40.53 | pabluss | hmm from system administration point of view is more work.... |
| 17:41.07 | pabluss | is more easy rpm or deb packages |
| 17:41.15 | pabluss | yum update and ready |
| 17:41.27 | kph100 | is it possible to have dynamic context creation in extensions using asterisk realtime? |
| 17:41.27 | benjk | especially if you run a production system |
| 17:41.30 | pabluss | or apt-get update |
| 17:41.34 | pabluss | and no more |
| 17:41.36 | benjk | then you want it to be out of date |
| 17:41.36 | pabluss | yeah |
| 17:41.43 | benjk | you dont want to upgrade |
| 17:42.00 | benjk | only very occasionally and only if you absolutely have to |
| 17:42.20 | benjk | of course this doesn't apply to you if this is just a hobby |
| 17:42.51 | benjk | nothing wrong with telephony as a hobby, but not everybody does it as a hobby |
| 17:43.30 | kph100 | anyone knows if one can have dynamic context creation in asterisk realtime? |
| 17:43.41 | pabluss | well... I'm new in asterisk so .. I must to compile from source code.. |
| 17:43.53 | pabluss | no more.. |
| 17:43.58 | pabluss | ok there we go |
| 17:45.34 | benjk | its usually pretty straightforward to build from source, but it doesn't mean you have to have the newest version of trunk all the time |
| 17:45.53 | *** join/#asterisk oej (n=oej@12.38.214.2) |
| 17:46.09 | benjk | find out which releases best work for what you want to do and then use what works best |
| 17:46.56 | pabluss | benjk ok , i've just to buy digium tdm 400p |
| 17:47.14 | pabluss | my principal objetive is to learn about asterisk |
| 17:47.25 | benjk | fair enough |
| 17:47.28 | pabluss | which tutorial you can recomend to me |
| 17:47.34 | pabluss | for start? |
| 17:47.41 | benjk | there is a book |
| 17:48.03 | pabluss | o'reilly? |
| 17:48.06 | benjk | does somebody know the command for jbot to provide the link to the asterisk book? |
| 17:48.11 | benjk | yeah that one |
| 17:48.15 | pabluss | o packt? |
| 17:48.15 | tzafrir | benjk, you also partially say that because you're used to software updates breaking systems |
| 17:48.32 | benjk | I think its also available in PDF form |
| 17:48.51 | benjk | and then there is Voip-info.org |
| 17:49.07 | pabluss | do you have URL ? |
| 17:49.19 | benjk | tzafrir, what I am saying is that if you run production servers, you don't unpgrade for the sake of upgrading |
| 17:49.38 | benjk | you upgrade according to a predetermined schedule or when circumstances force you |
| 17:50.02 | tzafrir | If you follow stable distributions (like Debian and Centos) and resist the temptation of backporting, their updates are almost always fixes with minimal changes |
| 17:50.13 | benjk | let's see if we can get the bot to tell us |
| 17:50.16 | benjk | ~book |
| 17:50.18 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:50.24 | tzafrir | Now, an example to the extreme direction is Fedora |
| 17:50.24 | benjk | ah, there you go |
| 17:51.09 | benjk | I am not saying you never upgrade |
| 17:51.35 | benjk | but I am opposed to the idea that you have to use SVN, "because the rpm is going to be out of date" |
| 17:51.38 | tzafrir | I've learned to trust Debian Stable's updates |
| 17:52.18 | pabluss | i'm agree with tzafrir, because as system administrator if I need to start a serius proyect on my job with servers productions and asterisk I wil get a big triuble with updates .. and stability.. |
| 17:54.28 | benjk | yep |
| 17:58.16 | *** join/#asterisk LoneShadow (n=duh@59.92.157.104) |
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| 18:00.29 | *** part/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 18:20.02 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:20.08 | TripleFFFF | hey |
| 18:20.51 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:20.58 | *** part/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:25.09 | *** part/#asterisk ropeguru (n=ropeguru@c-68-57-84-56.hsd1.va.comcast.net) |
| 18:25.19 | linagee | pablus: use two servers. take one down, upgrade the other, cycle back and forth,. |
| 18:25.50 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:29.09 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 18:34.59 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:35.10 | TripleFFFF | anyone have 1.2.11+ workign with app_Rxfax ? |
| 18:35.16 | TripleFFFF | i get core |
| 18:40.14 | kink0 | I need some help with Asterisk deadlock and many threads |
| 18:40.15 | niter3 | Hey guys what exactly is IAXTel? |
| 18:45.09 | Kuto | any idea on cheap ip phone aside from cisco? |
| 18:46.35 | Strom_C | polycom? |
| 18:49.53 | Kuto | Strom_C: im looking at it, seems hard to run as per the voip-info.org |
| 18:51.43 | Strom_C | they're a bitch to configure |
| 18:51.45 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:51.51 | Strom_C | but once you get them going they're fairly good |
| 18:52.04 | Strom_C | speakerphone is lower-quality compared to the cisco though |
| 18:55.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-191-219.red.bezeqint.net) |
| 19:00.32 | *** join/#asterisk jdg (n=jdg@50.93-90-202.mana.pf) |
| 19:04.05 | niter3 | kind of a dumb question but does Asterisk send out the time to the client phones? |
| 19:04.19 | niter3 | I ask becasue of of my phones have a time feature on it but there is no where in the options to set this |
| 19:06.02 | wunderkin | no that is what ntp is for |
| 19:06.45 | niter3 | hrm.. i'm trying to figure out how to set the freekin time on this phone |
| 19:10.31 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 19:12.27 | *** join/#asterisk adorah (n=top4yng@84.94.122.156.cable.012.net.il) |
| 19:15.57 | QbY | is it possible to load values out of the asterisk database into an array, and then loop through each value from a specific family? |
| 19:16.14 | *** join/#asterisk frigidzephyr (n=frigidze@c-71-207-216-231.hsd1.al.comcast.net) |
| 19:18.27 | frigidzephyr | im a linux noob, and i tried using the CVS instructions on this site: http://www.automated.it/guidetoasterisk.htm to download the zaptel libpri and asterisk packages, i got a Unknown host error, is the cvs site no longer cvs.digium.com ? |
| 19:18.40 | Strom_C | cvs has been deprecated forever now |
| 19:19.35 | frigidzephyr | oooh |
| 19:19.35 | Strom_C | www.asterisk.org has updated instructions |
| 19:19.35 | frigidzephyr | sweet ill check it out |
| 19:19.35 | frigidzephyr | thanks =] |
| 19:21.49 | *** join/#asterisk DrData (n=michael@pD95FE7E3.dip.t-dialin.net) |
| 19:22.16 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 19:25.49 | frigidzephyr | on the downloads page of asterisk.org is the "current snapshot" the latest working version? |
| 19:26.41 | wunderkin | use 1.2 |
| 19:26.53 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:27.06 | frigidzephyr | thx |
| 19:27.21 | Strom_C | frigidzephyr: no, by "current" they mean "last updated during the Nixon administration" |
| 19:28.09 | frigidzephyr | rofl |
| 19:28.38 | frigidzephyr | i was thrown off by the word "snapshot" the keyword here is "noob" hehe |
| 19:29.25 | frigidzephyr | everytime i hear the word snapshot, i tend to only think of that old game x-com |
| 19:34.21 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 19:39.02 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 19:39.30 | jhiver | hi all, i was wondering if IAX2 trunking still needs a timing device to operate, all the info i've found on this is really old |
| 19:41.05 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-140-141.dyn.embarqhsd.net) |
| 19:46.07 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:48.27 | docelmo | jhiver yes |
| 19:48.45 | jhiver | dolcemo, thx for the info |
| 19:49.39 | jhiver | BTW, is anybody in the chan using Yate for H323 - SIP translation? I have set up Yate on a box but I have nothing to test it against :) |
| 19:55.37 | *** join/#asterisk alawguy (n=mike@85-124-38-150.dynamic.xdsl-line.inode.at) |
| 20:00.15 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:03.46 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.42) |
| 20:04.18 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 20:04.22 | ziwapandey1980 | hi, any one there using vicidial with asterisk |
| 20:05.01 | TripleFFFF | yes |
| 20:05.02 | TripleFFFF | ;) |
| 20:06.05 | ziwapandey1980 | i m using vicidial for 12 agents, asterisk hangs after 30 min |
| 20:06.50 | ziwapandey1980 | then i have to kill asterisk process as i can not exe stop now or reload |
| 20:07.23 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 20:07.33 | ziwapandey1980 | hi, any one there using vicidial with asterisk |
| 20:08.06 | TripleFFFF | yesi m using vici |
| 20:08.15 | TripleFFFF | wow |
| 20:08.28 | ziwapandey1980 | i m using vicidial for 12 agents, asterisk hangs after 30 min |
| 20:08.36 | ziwapandey1980 | then i have to kill asterisk process as i can not exe stop now or reload |
| 20:09.06 | ziwapandey1980 | could u plz tell me solu? |
| 20:10.23 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 20:12.35 | TripleFFFF | like i know lol |
| 20:12.38 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:12.48 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.42) |
| 20:13.29 | ziwapandey1980 | any bony using vicidial? |
| 20:17.34 | *** join/#asterisk MoutaPT (n=MoutaPT@a213-22-40-120.cpe.netcabo.pt) |
| 20:17.41 | ziwapandey1980 | any bony using vicidial? |
| 20:21.34 | ziwapandey1980 | exit |
| 20:22.30 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.42) |
| 20:22.42 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:23.24 | ziwapandey1980 | any one using vicidial ? |
| 20:26.26 | *** join/#asterisk adorah (n=admin@84.94.122.156.cable.012.net.il) |
| 20:32.03 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:32.12 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:34.17 | teknoprep | does asterisk run well in vmware yet? |
| 20:34.30 | teknoprep | without slowing of voicemail voice .. the timing issues also |
| 20:34.55 | h3x | why the heck would someone want to run their phone system under an emulator |
| 20:37.14 | teknoprep | vmware is an emulator? |
| 20:37.32 | h3x | technically it isnt |
| 20:37.36 | h3x | but you get the point |
| 20:37.50 | teknoprep | fail-over for one |
| 20:37.55 | teknoprep | if a box or hdd fails |
| 20:38.01 | teknoprep | it will fail-over to another working vmware |
| 20:38.07 | h3x | that definately wont work |
| 20:38.13 | h3x | do failover at the application level |
| 20:38.37 | h3x | hint: dundi |
| 20:38.40 | teknoprep | you are telling me that failover for vmware os's doesn't work? |
| 20:38.50 | h3x | not within one 8000th of a second |
| 20:39.01 | h3x | like a zaptel tick |
| 20:39.10 | teknoprep | don't use zaptel.. only voip |
| 20:39.22 | h3x | well regardless with that time lag it would just hangup all the channels anyway |
| 20:39.40 | teknoprep | say a box blows up |
| 20:39.41 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-88-174.dsl.chcgil.sbcglobal.net) |
| 20:39.51 | teknoprep | how do you have active backup of mysql... voicemail.. all settings |
| 20:39.53 | teknoprep | from one box to another? |
| 20:40.19 | h3x | use asterisk realtime to store all the configurations on a database replicated at the application level between two machines |
| 20:40.36 | h3x | mysql and postgres support multiple hostnames for one database |
| 20:40.46 | h3x | and then asterisk... |
| 20:40.55 | h3x | well its not too difficult |
| 20:41.02 | h3x | use dundi to register extensions |
| 20:41.15 | teknoprep | yeah dundi i know about |
| 20:41.15 | teknoprep | i was wondering about the other stuff |
| 20:41.22 | h3x | hopefully all your voip devices support 2 endpoint ip's |
| 20:41.25 | teknoprep | i was looking for pretty much a failover for a box that fails |
| 20:41.29 | h3x | or you can do it on the firewall side |
| 20:41.42 | h3x | vmware just isnt going to work for multiple reasons |
| 20:41.48 | h3x | and arp is too slow to save calls when moving to another box |
| 20:41.50 | teknoprep | and vmware right now is the best solution.. i can store the entire os on an iSCSI san, have it taken over by any vmware front-end, at any time |
| 20:42.15 | teknoprep | i wasn't looking for a way to have the box keep calls active if the box fails |
| 20:42.18 | h3x | that stuff isnt a split second failover like you would need to keep a voice call up |
| 20:42.31 | h3x | well then you definately dont need to worry about it |
| 20:43.03 | teknoprep | but there are ways to setup a cluster of boxes.. and if one box fails it calls will continue through another box? |
| 20:43.18 | h3x | you really should not put asterisk on top of a virtual machine |
| 20:43.22 | h3x | its not an ideal scenario |
| 20:43.33 | *** join/#asterisk xnon (i=xnon@200.82.222.123) |
| 20:43.41 | teknoprep | any reason why? |
| 20:43.44 | h3x | the real faults that would occur is asterisk crashes and dosent come up |
| 20:43.54 | h3x | your fault tolerancy isnt going to keep that from happening |
| 20:43.59 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 20:44.05 | *** part/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 20:44.10 | h3x | setting up 2 or more asterisk boxes and doing this at the application level with dundi and dialplan magic will |
| 20:44.17 | teknoprep | i am looking for hardware backup as of now |
| 20:44.31 | teknoprep | hm |
| 20:44.39 | h3x | your hardware problems are nothing compared to asterisk crashing somewhat often |
| 20:44.40 | teknoprep | thats alot of work lol |
| 20:44.45 | h3x | no its not |
| 20:44.52 | h3x | its easy |
| 20:45.16 | teknoprep | are there any guides on this? |
| 20:45.24 | h3x | its probably on voip-info.org |
| 20:45.50 | h3x | and postgreSQL has many different data replication variations to pick from |
| 20:45.55 | h3x | mysql 5 is really unstable and crashes all the time |
| 20:46.15 | h3x | you might want to use the odbc driver |
| 20:46.21 | h3x | on the asterisk sde |
| 20:46.22 | h3x | side |
| 20:46.37 | teknoprep | odbc for ms sql ? |
| 20:46.43 | h3x | odbc connected to postgres |
| 20:46.47 | h3x | unixodbc |
| 20:47.12 | h3x | anyway, you just point >1 asterisk box at the same database server |
| 20:47.31 | h3x | the only thing that sucks in this scenario to set up is if you need to do channel limits on accounts |
| 20:47.38 | h3x | in which case you need to use radius to do that |
| 20:47.46 | h3x | because channel groups arent gonna work |
| 20:48.02 | h3x | but basically heres the jist of how it works |
| 20:48.04 | *** join/#asterisk fearthepenguin (n=fearthep@ppp-70-129-35-11.dsl.stlsmo.swbell.net) |
| 20:48.21 | h3x | your phone devices SIP register (or iax register i guess) to either of the asterisk boxes |
| 20:48.31 | h3x | you can set up DNS so that theres more than one A record for that hostname |
| 20:48.57 | h3x | the DNS server would hand out alternating IP's as A records each time your phones do queries so it should load balance it a little |
| 20:49.19 | h3x | if one of the boxes goes down your phones , if they arent braindead, should try to dns request again and eventually connect to the other server |
| 20:49.19 | teknoprep | yes i understand that |
| 20:49.40 | teknoprep | thats won't keep up an active connection tho |
| 20:49.41 | h3x | the nice thing is when doing asterisk administration you can "stop when convenient" |
| 20:49.49 | h3x | which will keep all the channels up but wont accept new calls |
| 20:50.15 | teknoprep | now my question |
| 20:50.21 | teknoprep | what is wrong with vmware and asterisk? |
| 20:50.23 | teknoprep | anything? |
| 20:50.50 | teknoprep | i run just about everything inside of vmware boxen |
| 20:51.06 | teknoprep | with an iSCSI san array always |
| 20:51.20 | fearthepenguin | I can't speak for VMWare, but I run asterisk inside of Xen with a large degree of success |
| 20:51.34 | teknoprep | Xen is awesome for nix tho |
| 20:51.57 | teknoprep | it does virtualization higher up in the kernel then vmware does |
| 20:52.02 | Chaz6 | i gotta say i prefer vmware to xen, but that's because of the things xen doesn't have.. |
| 20:52.16 | Chaz6 | eg, vmotion, drs |
| 20:52.36 | fearthepenguin | I use both everyday, and the one thing that vmware has that is really bad ass is the ability to snapshot a VM at any time with the press of a button |
| 20:52.39 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 20:52.57 | fearthepenguin | but I use xen for my production stuff, mostly because of the hardware passthru that vmware doesn't support yet, at least not for anything other then USB devices |
| 20:53.22 | Damin | teknoprep: ESX 3.0? |
| 20:53.25 | fearthepenguin | Im a best tool for the job kind of guy, so I use them both very heavily and they have maed my work life so much easier |
| 20:53.45 | teknoprep | damin no i run 2.5.1 and 2.5.2 |
| 20:53.46 | fearthepenguin | s/maed/made |
| 20:53.49 | Chaz6 | yeah they both ahve theirp lace |
| 20:53.57 | Chaz6 | we run esx 3 and it's brilliant |
| 20:54.04 | teknoprep | its out? |
| 20:54.07 | teknoprep | i didn't even know that |
| 20:54.09 | Damin | We finally went production with our ESX 3.0 cluster using iSCSI.. |
| 20:54.20 | teknoprep | www.openfiler.com |
| 20:54.28 | teknoprep | if you want a free iscsi target that just plain rocks |
| 20:54.37 | Chaz6 | yeah i use openfiler, but we sue fc for our vi setup |
| 20:54.39 | Damin | teknoprep: Yep... We built our own.. but openfiler does rock.. |
| 20:54.44 | Chaz6 | *use |
| 20:55.05 | Damin | teknoprep: If you are planning on deploying ESX 3.0, wait until they get the first patchset out.. |
| 20:55.15 | fearthepenguin | I'm still on the fence with the iscsi. I tried it on a small 1TB machine for awhile but ended up going back to smb and nfs. Admitidly though, my needs are very modest. Just a hobbiest really |
| 20:55.23 | teknoprep | will we be able to upgrade easily from 2.5.2 to 3.0 ? |
| 20:55.35 | Chaz6 | i tthink iscsi will be used when sas takes off |
| 20:55.43 | Damin | teknoprep: If you plan it correctly, yes.. |
| 20:55.45 | Chaz6 | *better used |
| 20:55.49 | teknoprep | Damin of course |
| 20:55.52 | Chaz6 | fc is still the bettre choice for now |
| 20:55.59 | fearthepenguin | so my interconnect was only Gig-E and I only had two block devices exported at 500GB each |
| 20:56.04 | Damin | teknoprep: You can vmotion from 2.5.2 hosts over to 3.0 hosts.. :) |
| 20:56.07 | E-bola | we had iscsi 3 years ago |
| 20:56.08 | Chaz6 | grr keybboard is decidedly odd when typin into a virtual machine through remote deskttop over a vpn... |
| 20:56.18 | teknoprep | we have 4gbit's transfer using combined 4x1gbit nic cards |
| 20:56.19 | E-bola | its already taken off if u ask me |
| 20:56.21 | teknoprep | on each box |
| 20:56.23 | Damin | teknoprep: But I think the machines have to be down.. |
| 20:56.38 | Chaz6 | our esx machines are blades with 2* gigE and 2* fc hba |
| 20:56.49 | teknoprep | yeah i don't have any HBA's |
| 20:56.53 | teknoprep | no need with vmware really |
| 20:57.05 | *** join/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 20:57.06 | teknoprep | i just run the os localy on sata raid1 2x36gb 10k rpm drives |
| 20:57.09 | Damin | teknoprep: We went w/ Qlogic SAN blades for our boxes.. Figured that the cost of licensing made preserving CPU and memory resources one of the most critical factors.. |
| 20:57.09 | fearthepenguin | If they can make AoE more open, I think it has a solid chance. But it needs to be opened up for it to take off |
| 20:57.28 | teknoprep | Damin i always use intel parts on the ESX compatibility list |
| 20:57.34 | teknoprep | usually SuperMicro servers |
| 20:57.37 | Chaz6 | aoe is kind of lame really.. |
| 20:58.06 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.42) |
| 20:58.16 | fearthepenguin | It has a speed advantage of iscsi though, and if it was opened up and didn't have so much weirdness happening in the background, I think it would have a chance. But yes, currently, it is lame |
| 20:58.22 | Damin | teknoprep: Qlogic is on the supported hardware list as well, although "official" full blow support will be forthcoming in 3.0.1 |
| 20:58.24 | fearthepenguin | s/of/over |
| 20:58.25 | wvtrixbox | hey guys/gals I just updated to 1.2.12 and now i get a unknown signaling in my chan_zap.so 'pri_cpe' |
| 20:58.31 | wvtrixbox | any suggestions? |
| 20:58.48 | Damin | teknoprep: We went w/ Tyan based Opteron boxes for the Vmware cluster.. |
| 20:59.01 | ziwapandey1980 | locate |
| 20:59.02 | teknoprep | i use dual core xeon dual cpu boxen |
| 20:59.04 | teknoprep | with HT |
| 20:59.06 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 20:59.07 | Chaz6 | yeah i think if it wasn't tied down to a better it would beu seful |
| 20:59.21 | Damin | teknoprep: Dual Core, Dual Processor boxes w/ LSI Logic Hardware Raid for the boot drives, and a pair of Qlogic SAN blades for the iSCSI... |
| 20:59.24 | Chaz6 | and i dislike how you manually give each shelf an id |
| 20:59.27 | breakdisk | hi guys,i need your help. |
| 20:59.44 | fearthepenguin | agreed. potential, but it will most likely die on the wire |
| 20:59.50 | Chaz6 | our HBAs are qlogic but our vendor have certified the whole paltform to run esx for us.. |
| 21:00.02 | Damin | teknoprep: We nic-teamed 6 GigE's together on the iSCSI server, which is 3.2 terrabytes of SATA / 3Ware / PCI-X goodness.. |
| 21:00.05 | breakdisk | i got this error.http://sneex.pastebin.ca/166350 |
| 21:00.58 | fearthepenguin | breakdisk: are you root and is your filesystem actually mounted ro? |
| 21:01.01 | Damin | Chaz6: I do not like the fact that you cannot use the Qlogics in load-balanced scenario.. |
| 21:01.10 | Damin | Chaz6: But they are freaking awesome cards.. |
| 21:01.18 | teknoprep | damin thats nice |
| 21:01.41 | Chaz6 | Damin, yes kinda sad.. so we do that by balancing the luns across the sps |
| 21:01.53 | ziwapandey1980 | whois Damin |
| 21:01.56 | teknoprep | damin i don't need that much storage tho.. i am good with running a ~500gb with 10x raid 5 76gb 10k rpm sata array on two boxen |
| 21:01.58 | breakdisk | fearthepenguin:yup. |
| 21:02.13 | fearthepenguin | breakdisk: yup to which? |
| 21:02.19 | teknoprep | Damin the speed on that box with the 10k rpm's is insane |
| 21:02.22 | Chaz6 | Damin, i also found out in testing that they don't fail over properly when there's no i/o, though vmware said that could be ane sx problem.. |
| 21:02.24 | Damin | teknoprep: Yeah.. the tyan boards have 2 E1000 gigEs onboard and 1 E100 100 meg on board, so we slap in an additional pair of dual-port Intel nics to provide for HA/Load Balancing on the boxes.. |
| 21:02.49 | fearthepenguin | Damin: What board is that? sounds nice |
| 21:02.54 | breakdisk | fearthepenguin:to asterisk.my box is running using root. |
| 21:03.21 | fearthepenguin | breakdisk: when you do a `mount` is /var readonly or `df -h` is it full? |
| 21:03.35 | breakdisk | not yet. |
| 21:03.47 | breakdisk | it has high available space. |
| 21:03.53 | fearthepenguin | is it ro? |
| 21:04.19 | breakdisk | nope,just know when i issue restart. |
| 21:04.23 | fearthepenguin | breakdisk: what happens when you run `touch /var/lib/asterisk/agi-bin/test` |
| 21:04.31 | ziwapandey1980 | any one using vicidial? |
| 21:04.45 | breakdisk | fearthepenguin:hold ill try. |
| 21:05.06 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 21:05.41 | breakdisk | fearthepenguin:ouch: cannot touch `/var/lib/asterisk/agi-bin/test': Read-only file system |
| 21:05.54 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 21:06.13 | fearthepenguin | breakdisk: you have bigger issues my friend. either your filesystem is jacked, mounted read-only or you have some extended attributes defined that isn't allowing writes |
| 21:06.25 | fearthepenguin | what does the last few lines of `dmesg` say |
| 21:06.44 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 21:06.52 | Damin | teknoprep: We originally had an Intel uniprocessor box doing the iSCSI w/ the 3ware in a PCI slot, but w/ 16 drives on it, we freaking saturated the PCI bus.. |
| 21:06.54 | ziwapandey1980 | any one using vicidial? |
| 21:07.03 | teknoprep | lol |
| 21:07.09 | teknoprep | damin yeah pci-e is really nice |
| 21:07.20 | ziwapandey1980 | any one using vicidial? |
| 21:07.26 | Damin | teknoprep: Plus, we needed a lot more processing power in user-space for the ietd to do it's job.. |
| 21:07.53 | ziwapandey1980 | any one using vicidial? |
| 21:07.55 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 21:07.58 | Damin | teknoprep: So we moved it to a Dual Core Opteron box and it's running in full 64 bit glory.. |
| 21:08.22 | teknoprep | Damin i was thinking of going that route.. i was having a hard time finding hw compat |
| 21:08.23 | Damin | teknoprep: And the 3Ware card is sitting in a PCI 8X slot.. :) |
| 21:08.23 | breakdisk | fearthepenguin:here is it http://sneex.pastebin.ca/166367 |
| 21:08.46 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.42) |
| 21:08.48 | fearthepenguin | EXT3-fs error (device sda2) in start_transaction: Journal has aborted |
| 21:08.51 | fearthepenguin | your filesystem is jacked |
| 21:09.03 | fearthepenguin | boot off a cd and fsck it |
| 21:09.17 | ziwapandey1980 | any one using vicidial? |
| 21:09.21 | teknoprep | Damin does virtual center 2.0 come with vmotion? |
| 21:09.57 | ziwapandey1980 | any one using vicidial? |
| 21:09.57 | Damin | teknoprep: The licensing is all messed up, but we bought the Enterprise bundle and it has all the features.. |
| 21:10.08 | Damin | teknoprep: DRS is freaking AWESOME in VC2 |
| 21:10.11 | breakdisk | fearthepenguin:it will fix?what if fsck only while rebooting. |
| 21:10.12 | fearthepenguin | ziwapandey1980: I dont believe anyone is, otherwise someone would have answered by now |
| 21:10.59 | Damin | teknoprep: We have three ESX hosts in our cluster, so we started up like 34 virtual machines and set the DRS level to "Aggresive" and watched the migration party... |
| 21:11.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:11.13 | fearthepenguin | breakdisk: you can give that a shot. if it doesn't come up with any errors though, you need to make sure the drive is totally unused and fsck it |
| 21:11.33 | Damin | teknoprep: It automatically monitors resource utilization and bounces stuff around (live, w/ no downtime on the guest) between servers.. |
| 21:11.44 | puzzled | evening |
| 21:11.48 | Damin | teknoprep: I've had a boner for three months over it! |
| 21:11.54 | teknoprep | lol |
| 21:12.00 | teknoprep | yeah i know what it is |
| 21:12.07 | teknoprep | i just haven't had time to play lately |
| 21:12.39 | Damin | teknoprep: However.. there are a couple of bugs in the 3.0 release that will be addressed in 3.0.1 that are show stoppers for aggressive DRS.. |
| 21:12.50 | teknoprep | hmm |
| 21:13.24 | teknoprep | ok quick questin |
| 21:13.31 | teknoprep | i hav elicensing for vmware esx 2.5.2 |
| 21:13.34 | Damin | teknoprep: Sometimes during migrations, linux guests will experience a SCSI timeout in the guest adapter, which causes the ext3 journal to abort and the filesystem goes read-only for sanity purposes.. |
| 21:13.34 | breakdisk | fearthepenguin:ok,ill try if i can fix it remotely because the server is too far. |
| 21:13.37 | teknoprep | and virtual center 2 |
| 21:13.44 | teknoprep | damin what else do i need? |
| 21:14.03 | breakdisk | fearthepenguin:thank you so much. |
| 21:14.20 | Chaz6 | worth getting a vmtn subscription too i think |
| 21:14.35 | teknoprep | Chaz6 maby.. i have yet to need one |
| 21:14.37 | fearthepenguin | breakdisk: no problem and good luck! |
| 21:14.38 | Chaz6 | it includes an agent for vc, but damn, the new licensing server really sucks |
| 21:14.54 | Damin | teknoprep: http://kb.vmware.com/vmtnkb/search.do?cmd=displayKC&docType=ex&bbid=bb0&url=http%3A%2F%2Fkb.vmware.com%2Fvmtnkb%2Fsearch.do%3Fcmd%3DdisplayKC%26docType%3Dkc%26externalId%3Dhttp--wwwvmwarecom-pdf-vi3installationguidepdf%26sliceId%3DpdfPage_35%26dialogID%3D40864%26stateId%3D0%25200%252042299%26doctag%3D&dialogID=1740222&stateId=0%200%201742218&doctag= |
| 21:15.04 | Chaz6 | right now the esx servers are running in host-based mode, and yet they get the licenses for vmotion, ha and drs from the server... |
| 21:15.14 | Chaz6 | i simply cannot get them to take the esx licenses off the server |
| 21:15.46 | Damin | There is another link here.. http://kb.vmware.com/vmtnkb/search.do?cmd=displayKC&docType=kc&externalId=http--wwwvmwarecom-pdf-vipricingpdf&sliceId=pdfPage_2&dialogID=1740222&stateId=0 0 1742218&doctag= |
| 21:16.10 | Damin | Chaz6: Yeah.. don't know what to tell you on that.. we run everything from the license server.. |
| 21:16.31 | *** part/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 21:16.38 | Damin | Chaz6: I'm really kind of pissed that they do not have enterprise licenses for development.. |
| 21:16.44 | Chaz6 | Damin, i guess i'll just try to work through it with support, i've pretty much done everything they've said by the book |
| 21:17.15 | Chaz6 | Damin, there's vmtn, i don't know whether or not that only gives you one 2-cpu license though |
| 21:17.20 | Damin | Chaz6: We'd love nothing more than to be able to carve off a bit of our San and plug a pair of light-duty boxes into it so we could test patchsets before going live.. |
| 21:17.32 | Chaz6 | oh yes, the no-hba restriction does suck.. |
| 21:17.37 | Damin | Chaz6: We bought VMTN strictly for the unlimited P2V licesnses.. |
| 21:18.01 | Chaz6 | it does seem rather silly.. but it's their choice.. |
| 21:18.04 | Damin | Chaz6: But the VMTN licenses are stripped down and wouldn't be an accurate replica of our production system.. |
| 21:18.06 | Chaz6 | :( |
| 21:18.16 | Chaz6 | Damin, that's true.. |
| 21:18.33 | Chaz6 | though the licenses aren't mac-locked, don't see why you couldn't run another non-production setup.. |
| 21:18.44 | Damin | Chaz6: They are willing to keep giving us Eval licenses though, so I guess I'll just talk to them every month until they get tired of doing it! ;) |
| 21:18.57 | Chaz6 | hah well that's one way to do it :) |
| 21:19.10 | Damin | Chaz6: Yeah, but then we need to run a separate license and vc server. |
| 21:19.23 | Chaz6 | Damin, well yes, but you can just run it on a vm :) |
| 21:19.28 | Damin | Chaz6: It was their suggestion.. :) |
| 21:19.43 | Damin | Chaz6: Yeah, but that is an expensive waste of resources on the cluster! :) |
| 21:19.55 | Damin | Chaz6: The average cost of a virtual machine is $1,800 after licensing.. |
| 21:20.07 | Chaz6 | you're running 2003 r2 right? |
| 21:20.21 | Chaz6 | you do know you can run up to 4 instances of that with one license..? |
| 21:20.31 | teknoprep | crack it |
| 21:20.32 | teknoprep | lol |
| 21:20.33 | teknoprep | jk |
| 21:21.11 | Damin | Chaz6: Actually.. :) Our VC center is running on Windows XP right now.. ;) We need to P2V a couple of servers into the cluster to free up the hardware to build the production VC box! ;) |
| 21:21.26 | teknoprep | Damin do i need vmotion if i have license for vc2 ? |
| 21:21.35 | Damin | teknoprep: No doubt.. someone will create a license generator sooner or later.. |
| 21:21.52 | Damin | teknoprep: I don't know how that works.. quite frankly, we just bought the farm... |
| 21:22.00 | teknoprep | ahh |
| 21:22.43 | Chaz6 | Damin, ah, right :) |
| 21:22.54 | Chaz6 | i wish they would provide you with cheap virtual licenses for xp.. |
| 21:23.04 | MoutaPT | hi does any one know any GUI to manage CIDnames of incoming calls of asterisk? |
| 21:23.08 | Chaz6 | seems really unfair to have to buy a whole license for each instance |
| 21:23.30 | Damin | Chaz6: Yeah.. it sucks.. |
| 21:23.36 | Chaz6 | (though i use a stripped down version for testing stuff on xp, tinyxp, only uses 55mb ram on boot, no wga...) |
| 21:24.04 | Damin | Chaz6: Can I get an archive of that? |
| 21:24.17 | teknoprep | Damin this vmotion.. do we need it to be able to fail-over boxen from one vmware to another? |
| 21:24.18 | Damin | Chaz6: That sounds like an awesome thing to have for DR.. |
| 21:24.38 | teknoprep | Damin i just want to be able to fail-over from one guest os on one vmware to another vmware box |
| 21:24.42 | Damin | teknoprep: Vmotion lets you migrate servers, in real time, from host to host in your clister.. |
| 21:25.02 | Damin | teknoprep: Vmware HA lets you do automatic failover.. |
| 21:25.02 | teknoprep | Damin what about fail-over? |
| 21:25.06 | teknoprep | ahh |
| 21:25.11 | teknoprep | is that part of vc2 ? |
| 21:25.39 | Damin | teknoprep: Yeah, but you need VC2 + ESX 3... |
| 21:25.51 | teknoprep | you can't do failover in esx 2.5.2 ? |
| 21:25.53 | Damin | teknoprep: It's a licesned add-on.. |
| 21:26.07 | Damin | teknoprep: i don't know.. I've only ever played w/ 2.5.2 one time.. |
| 21:26.24 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 21:26.43 | *** part/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 21:26.46 | Damin | teknoprep: 2.5.2 didn't support iSCSI and it was pretty lame.. :) |
| 21:26.56 | Damin | teknoprep: 3.0 is light years ahead of it.. |
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| 21:39.34 | Damin | dcf |
| 21:41.24 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 21:41.58 | manopulus | hello, i looking for iax/sip web (java?) softphone, what i can look? |
| 21:42.55 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 21:46.30 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 21:47.14 | Damin | I really have to unbind Ctrl-E from BitchX.. |
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| 21:54.18 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:54.41 | Damin | Anyone alive? |
| 21:55.24 | docelmo | no |
| 21:55.38 | docelmo | but after your snoring from last night Im amazed you are.. :) |
| 21:56.48 | jmls | I used to be. |
| 21:59.09 | Strom_C | I'm not alive...I'm reanimated! |
| 21:59.49 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 22:00.32 | jmls | can anyone have a look at #7671 and see if they can replicate ? |
| 22:01.45 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 22:01.48 | kink0 | hello |
| 22:02.14 | kink0 | anyone can help me with deadlock + thread problem ? |
| 22:12.52 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
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| 22:16.57 | kink0 | anyone can help me with deadlock + thread problem ? |
| 22:17.15 | docelmo | hehe.. thats just too easy |
| 22:18.02 | kink0 | too easy ??? |
| 22:18.49 | kink0 | sometimes I get over 2 days uptime, but now, I am unable to get over 30 minutes uptime :( |
| 22:19.17 | kink0 | and I changed nothing in configurations, just I get deadlock and a lot of threads |
| 22:19.56 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 22:20.28 | anonymouz666 | run asterisk with gdb and watch the threads... partition would say |
| 22:20.53 | kink0 | anonymouz666, I did so, as documentation says, but I got nothing ussefull |
| 22:21.54 | kink0 | I typed "info thread" in gdb but returns nothing, for bt, returns no ussefull info |
| 22:22.28 | kink0 | all people suffers of deadlocks ? or may be related to my libpthread and glibc ? |
| 22:24.17 | `Tingles` | anyone have any ideas.. voicemail was working fine.. until one of th eusers set a password now they or anyone else can not log in... i have verified they are entering the right password in the voicemail.conf file... |
| 22:24.25 | `Tingles` | nothing in verbose... |
| 22:26.23 | kink0 | anonymouz666, your asterisk does not crashes with deadlock after some time while you get about 20 channels in use ? |
| 22:32.14 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.55) |
| 22:34.00 | kink0 | another question... something like exten => _349XXXXXXXX,53,CheckGroup(4|j) will execute exten 154 when more than 4 calls ? |
| 22:34.31 | anonymouz666 | kink0: nops. no crashes. |
| 22:35.20 | kink0 | anonymouz666, really mine no crash, just a lot of threads and fails, so I force a kill when a more than 300 threads |
| 22:41.16 | kink0 | anonymouz666, can you tell me what distribution, kernel version, glibc version and asterisk version are you ussing ? |
| 22:42.43 | anonymouz666 | fedora core 4, asterisk 1.2.12 |
| 22:43.57 | *** part/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
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| 22:51.13 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 22:52.31 | THX2000 | I'm trying to set up a call forwarding/hunting feature that will forward an extension to my cell. However, if my cell is dead or does not have service i dont want it to be the device that answers the call |
| 22:53.01 | THX2000 | I want to do something similar to call parking, to where if i answer the cell and press 1 it accepts the call, but can't figure out where i'd put the call so i could retrieve it on the key press |
| 22:53.57 | THX2000 | any ideas? |
| 22:59.32 | h3x | show application ValetParking |
| 22:59.38 | fearthepenguin | THX2000: google for Asterisk FollowMe |
| 23:00.11 | h3x | THX2000: I fixed that particular problem though by permanently no answer call forwarding to a DID that immediately disconnects |
| 23:00.16 | jmls | thx2000: have a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels, for "Answer confirmation" |
| 23:00.17 | h3x | 28 seconds |
| 23:00.31 | h3x | then you can just use plain ass old Dial |
| 23:00.57 | THX2000 | sweet, all kinds of stuff to check out...thanx |
| 23:01.09 | kink0 | anonymouz666, ok , thanks |
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| 23:21.36 | marl | can someone have a look at my conf files and the debug from iax? ive done a fresh install of * (1.2.1/1.2.8) and am trying to get iax to work inbound to the server, in the paste bin : http://pastebin.ca/166496 : the top area is from iax2 debug and then below that is my extensions.conf and iax.conf files |
| 23:22.29 | marl | when i try and dial the iax number, i get the output at hte top of the pastebin, but nothing rings :( has anyone any ideas? PLEASE this is driving me crazy :( |
| 23:23.58 | Strom_C | marl: why not try this |
| 23:24.10 | Strom_C | put this as the sole entry in [gradwell-in] |
| 23:24.25 | Strom_C | exten => _.,1,NoOp(Caller is trying to reach exten ${EXTEN}) |
| 23:24.32 | Strom_C | and see what it pastes to your console |
| 23:27.04 | marl | ok that didnt sow anything on the console :( |
| 23:27.20 | marl | i take it that meens that its not getting from the iax.conf file to the extensinos.conf file? |
| 23:28.50 | marl | i take it that starting asterisk with : asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvv would show anything output via noop rite? |
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| 23:31.06 | kink0 | good night |
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| 23:37.18 | trevarthan | does anyone have recommendations for FXO + FXS ATA devices? |
| 23:37.37 | bkw__ | sipura |
| 23:37.40 | trevarthan | I've seen reviews of the Handytone 488, stating that it locks up a lot - mark that off my list. |
| 23:38.10 | niter3 | trevarthan: Linksys seems to be good |
| 23:38.37 | trevarthan | and the Sipura 3000 reviews state problems with echo. I really don't want to deal with that. The Handytones, in my experience deal with echo really well. I'm fed up with echo on my x100p. |
| 23:38.49 | trevarthan | so what should I get? |
| 23:39.17 | niter3 | I may have heard echos once or twice, but at most part newp. |
| 23:39.20 | Vec | If I use the setCallerId command, will it change the ID in the cdr record ? |
| 23:39.48 | niter3 | Vec: It will over ride anyone that sets their's at the client. |
| 23:40.00 | hads | trevarthan: The SPA3000 has been obeseleted by the SPA3102 by the way. |
| 23:40.09 | orlock | The Handytones.. sounds like an 80's ska band |
| 23:40.23 | niter3 | hrm.... A ride in the town sounds good right now.... bbl |
| 23:40.26 | hads | trevarthan: Why not replace your X100P with a TDM400? |
| 23:40.36 | trevarthan | hads: too damn expensive. |
| 23:41.12 | trevarthan | hads: besides, I've heard that the 400p has reliability issues. |
| 23:41.39 | hads | I've never had any reliability issues with them and I've sold quite a few. |
| 23:43.02 | trevarthan | ok, so the spa3102 is new to me. The price is right. Anyone have one of these? How's the echo? |
| 23:43.06 | trevarthan | Any issues? |
| 23:43.54 | tainted_ | why not replace your corolla with a sl600 |
| 23:43.57 | trevarthan | hads: that's good to know. I might consider a 400p for a commercial install in a nice integrated package. But I'd like to try something less expensive right now. |
| 23:44.08 | tainted_ | why not replace your ugly wife with heidi klum |
| 23:45.55 | hads | trevarthan: I don't have any experiance with the 3102 as it hasn't been released for sale here yet but AFAIK it is basically very similar to the 3000 at present but different hardware with a larger memory for further firmware expansion. |
| 23:47.54 | hads | The SPA's (or any FXOs) ability to handle echo depends on the individual line characteristics - if you have a very problematic line then there isn't much you can do apart from hardware EC. |
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| 23:50.01 | trevarthan | hads: I think everyone says that about the x100p too, but the fact is that the handytone 486 I've used in the past never had any echo issues, even on the same line as the X100p. |
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| 23:50.23 | trevarthan | hads: not everything is created equal. I'm just trying to determine how equal these things are. :) |
| 23:50.52 | trevarthan | nobody here has a 3102? |
| 23:51.01 | Strom_C | i have a 3102 |
| 23:51.07 | Strom_C | but it's not in service |
| 23:51.18 | hads | I wasn't talking about the X100, I was talking about all FXOs. Everyone says that because it is a fact... |
| 23:51.50 | *** part/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 23:53.20 | trevarthan | hads: I know it's a fact, but I could never get rid of that echo on my x100p, whereas the handytone never had the echo, even on the same line. I hear the tdm 400p is better with echo than the x100p, I just don't want to dump $200+ on a pci card. |
| 23:54.11 | trevarthan | Strom_C: did you get to evaluate the 3102? Any impressions you could give me? |
| 23:54.22 | MoutaPT | any one here taking AsteriskBootCamp @ Marbella? |
| 23:54.58 | Strom_C | never really used it, trevarthan |
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