| 00:02.21 | shmaltz | bkw_, when a calls b, and b blind xfers (using cisco blindxfer), the parked spot gets announced to the a |
| 00:02.45 | bkw_ | you got a broken valetparking |
| 00:02.50 | bkw_ | or the wrong asterisk |
| 00:02.54 | bkw_ | the best way to do a blind park |
| 00:03.05 | shmaltz | asterisk is from today. |
| 00:03.07 | bkw_ | is to do a SetVar(BLINDTRANSFER=yes) |
| 00:03.18 | shmaltz | and valet is the one from the link that anthm gave me |
| 00:03.29 | bkw_ | cvs-head? |
| 00:03.32 | shmaltz | thanks |
| 00:03.33 | shmaltz | yep |
| 00:03.41 | bkw_ | just set the var then |
| 00:03.50 | bkw_ | something else is wrong we'll look at it tommorow maybe |
| 00:03.59 | shmaltz | thanks |
| 00:04.07 | bkw_ | we gotta get ClueCON set this week |
| 00:04.22 | bkw_ | we are kinda busy with the details on that ;) |
| 00:04.43 | shmaltz | ~ClueCON |
| 00:04.55 | *** join/#asterisk xcoyote (~coyote@dsl-200-95-78-238.prod-infinitum.com.mx) |
| 00:05.10 | bkw_ | its a conf in Chicago for beginner devs, and hardcore devs |
| 00:05.28 | bkw_ | we will be going over some pretty hard core stuff there |
| 00:05.35 | bkw_ | along with a dev round table |
| 00:05.40 | shmaltz | when? |
| 00:05.47 | bkw_ | thats what we are gonna get set in stone this week |
| 00:05.57 | shmaltz | hm, i c |
| 00:06.00 | bkw_ | June I think is the target |
| 00:06.02 | denon | do it in vegas .. in a couple weeks, when im gonna be there :) |
| 00:06.09 | buddah | lol |
| 00:06.11 | bkw_ | haha |
| 00:06.13 | buddah | fuck vegas |
| 00:06.23 | denon | oh yeah, this coming from a guy named buddah |
| 00:06.24 | shmaltz | vegas is too far 4 me |
| 00:06.26 | bkw_ | I have to find people to speak there too |
| 00:06.37 | buddah | ok mr. denon |
| 00:06.42 | buddah | vegas is played out |
| 00:06.52 | file | bkw_: hmm? |
| 00:06.57 | xcoyote | i have a question about dtmf collection for asterisk, i am trying to request a password to login, so it must be a collection of digits typed by the user. and then compare it agains another number and finaly apply a gotoif function. which function allows to collect DTMF input from caller? |
| 00:07.14 | bkw_ | show application read |
| 00:07.18 | denon | bkw: you had way too much fun recording mark's ogm |
| 00:07.19 | buddah | chicago sounds like a fun time |
| 00:07.31 | bkw_ | really? |
| 00:07.34 | denon | sounded like it |
| 00:07.43 | bkw_ | twisted helped |
| 00:07.45 | bkw_ | so did allison |
| 00:07.47 | denon | yeah, I heard |
| 00:07.50 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 00:07.52 | file | bkw_: what's this about dev thing? |
| 00:08.01 | implicit | you in file? |
| 00:08.06 | file | I'm here. |
| 00:08.14 | bkw_ | file its going to be in early june in Chicago |
| 00:08.26 | bkw_ | if we work this out right Mark will be there too |
| 00:08.26 | file | hrm could I do early june.... |
| 00:08.27 | h3x | bkw_: have you ever got SendURL to work right |
| 00:08.28 | bkw_ | thast what we are working on |
| 00:08.33 | bkw_ | h3x, no |
| 00:08.36 | file | I think that's when my graduation happens |
| 00:08.37 | implicit | did u get my aim message? |
| 00:08.43 | h3x | im convinced it isnt supposed to |
| 00:08.43 | h3x | heh |
| 00:08.49 | file | implicit: no, not really at my computer where AIM is |
| 00:08.50 | yashax | In IP500, when you press the [Voice Mail] button, what do you press to get into VM? I tried *, #..... |
| 00:08.54 | bkw_ | h3x, I think gnophone worked with it at one time |
| 00:09.01 | bkw_ | Only IAX |
| 00:09.02 | h3x | well diax phone says it wors |
| 00:09.04 | implicit | ok ill msg u here |
| 00:09.05 | h3x | in the new version |
| 00:09.09 | h3x | but it dosent do anything |
| 00:09.35 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 00:10.11 | *** join/#asterisk nitram (martin@superblob.com) |
| 00:10.19 | xcoyote | which application allows me to know the dtmf digits that callers pressed? |
| 00:10.55 | bkw_ | xcoyote, Read |
| 00:10.57 | bkw_ | PAY ATTENTION |
| 00:10.58 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
| 00:10.59 | bkw_ | I told you |
| 00:11.01 | bkw_ | damn |
| 00:11.15 | bkw_ | the application is called "Read" |
| 00:11.19 | bkw_ | show application read |
| 00:11.57 | snewpy | yashax: when you've got bypassInstantMessage="1" set, pressing the messages button should dial straight away |
| 00:12.01 | pdracevich | i need some major help, please with IAX, can any one Private Message me please |
| 00:12.07 | bkw_ | no |
| 00:12.08 | tzanger | BrianR___: norstar-asterisk works well enough |
| 00:12.09 | bkw_ | you ask here in the channel |
| 00:12.13 | xcoyote | ok |
| 00:12.19 | hardwire | norstar-asterisk? |
| 00:12.38 | denon | ~jbot is it true that bkw will do 1-on-1 privmsg for $1.99/minute? |
| 00:12.57 | BrianR___ | yashax: The voicemail button on the soundpoint phones rings the extension defined in the phone's config file. You should put the extension for the voicemail there and the voicemail button will work just as if you had called the voicemail extension manually. |
| 00:13.00 | Sedorox | what are the advantages of trunk in iax2? |
| 00:13.11 | tzanger | pdracevich: have you got basic iax connectivity working? |
| 00:13.14 | bkw_ | i'l do anythign 1 on 1 for 1.99 a min |
| 00:13.16 | denon | Sedorox: less bandwidth with multiple calls .. think of it like a D-channel for a T1 |
| 00:13.20 | hardwire | to get around all the disadvanteges of not having one :) |
| 00:13.23 | algorithmn | lol |
| 00:13.37 | pdracevich | tzanger: I have the one of the two boxes registering |
| 00:13.43 | Sedorox | so I guess I should enable it between my three * boxes then... |
| 00:13.47 | BrianR___ | tzanger: Are you referring to any special software, or just using VMI's and/or PRI to interface? |
| 00:13.56 | denon | yes, if you have more than one simultaneous call |
| 00:14.01 | shmaltz | bkw_ setvar helped. how big is the app_valetparking.c file suppose to be? |
| 00:14.05 | tzanger | BrianR___: PRI between them, no qsig |
| 00:14.17 | BrianR___ | we're lucky - our norstar has two PRi cards in it. |
| 00:14.22 | tzanger | pdracevich: so what, precisely, is the problem |
| 00:14.26 | tzanger | BrianR___: very lucky |
| 00:14.27 | bkw_ | shmaltz, no clue i'll check later |
| 00:14.28 | tzanger | that ain't cheap |
| 00:14.33 | BrianR___ | tzanger: How do you get voicemail to work properly for station-to-station calls on the norstar? |
| 00:14.40 | shmaltz | b/c my older version is 30+ k and this from anthms' link is round 24 |
| 00:14.41 | tzanger | BrianR___: you don't |
| 00:14.49 | tzanger | BrianR___: you need SL1 to have voicemail on * |
| 00:14.56 | tzanger | and * doesn't talk SL1 |
| 00:14.57 | BrianR___ | SL1? |
| 00:14.59 | tzanger | since it's proprietary |
| 00:15.06 | tzanger | a.k.a MCDN, NAPN |
| 00:15.07 | pdracevich | tzanger: the dial rules, when I try to write one, it does not work. :-( |
| 00:15.12 | BrianR___ | MCDN? NAPN? |
| 00:15.30 | tzanger | BrianR___: you need licence keys on the norstar, and the protocol in * |
| 00:15.43 | BrianR___ | tzanger: What is the protocol? |
| 00:15.57 | tzanger | SL1 |
| 00:16.11 | BrianR___ | The norstar talks SL1 over what kind of interface? |
| 00:16.16 | tzanger | PRI |
| 00:16.17 | yashax | snewpy: I do have that in the phone1.cfg, but when I press the VM button, I still see the 2 options (Message Center and Instant Messages) |
| 00:16.23 | BrianR___ | Aah. |
| 00:16.40 | BrianR___ | What about using a VMI? |
| 00:16.47 | Sedorox | what codec do must people use over iax2? |
| 00:16.50 | tzanger | BrianR___: like what |
| 00:16.55 | tzanger | Sedorox: I use ulaw and gsm |
| 00:17.07 | *** join/#asterisk rett (~rett@c-67-171-236-169.client.comcast.net) |
| 00:17.12 | Sedorox | ulaw uses a lot od BW tho, don't it? |
| 00:17.16 | BrianR___ | tzanger: There's a semi-obsolescent device for using non-nortel voicemails on a norstar. I bought 3 of em on eBay. |
| 00:17.41 | BrianR___ | each one turns two norstar station ports into two fxs ports with disconnect supervision and dtmf signalling of extension dialed. |
| 00:17.49 | pdracevich | tzanger: I am at Point "B" the server says "Registered to '218.101.54.x', who sees us as 210.54.249.x:50017" (210.54.2489.x) being point "B" i want to place a call from point "B" and have it come out point "A" |
| 00:17.53 | tzanger | BrianR___: well then they'd likely work :-) |
| 00:18.11 | tzanger | pdracevich: smells like NAT |
| 00:18.35 | tzanger | I've connected * to analogue trunk lines, CT1, PRI and through norstar ATAs |
| 00:18.44 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 00:18.59 | snewpy | yashax: hmm... must be something else in the configuration, I don't get that effect |
| 00:19.01 | Sedorox | another stupid question.. in iax.conf.. on host= |
| 00:19.04 | Sedorox | can you use a DNS name? |
| 00:19.23 | snewpy | yashax: in the phone's configuration, that is... but hitting message center dials thru to voicemail, right? |
| 00:19.23 | BrianR___ | tzanger: I know the norstar we have can do callforward no answer to an outside number, but I'm not sure if it puts in enough info to the final destination to tell which extension was originally called. |
| 00:19.25 | tzanger | I can redirect voicemail (fwd on busy/no answer) to an extension on asterisk, but you can't get asterisk to indicate MWI |
| 00:19.37 | pdracevich | tzanger: any ideas?, and is there a web site that will explain, in detail with config file that does not confuse the hell out of me |
| 00:19.57 | tzanger | BrianR___: with PRI it does, but as I just said, there's no way for asteirsk to notify the DN that there's messages |
| 00:20.11 | BrianR___ | tzanger: That's the easy part of the problem. |
| 00:20.28 | snewpy | a lot of PBXs have either standard or as some kind of add-in a serial port for controlling MWI, or let you set up extensions to trigger MWI on/off |
| 00:20.30 | tzanger | pdracevich: as I said, it smells like you've got NAT in the middle. have you got udp/4569 being forwarded to each * box |
| 00:20.31 | BrianR___ | tzanger: You can use an ATA or a port on on an analog station module fo rthat. |
| 00:20.45 | BrianR___ | tzanger: You just need to do <hookflash> *1 extension. |
| 00:20.54 | tzanger | BrianR___: how do you notify DN 243 that there's voicemail ofr it? |
| 00:20.58 | tzanger | really |
| 00:21.07 | tzanger | *1243 then? |
| 00:21.10 | tzanger | from an ATA or ATA2? |
| 00:21.18 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
| 00:21.20 | yashax | snewpy: Right now when I select [MC] choice and press Connect, it says "Person at extension X is on the phone" |
| 00:21.22 | tzanger | I'll have to try that tomorrow |
| 00:21.30 | BrianR___ | tzanger: Dial *1 1234. No joke. The extension from which you dial that from needs to be forwaded to commedian mail so the callback feature works right. |
| 00:21.43 | tzanger | BrianR___: *very* interesting |
| 00:21.46 | tzanger | I wonder if it works over PRI |
| 00:21.50 | BrianR___ | tzanger: There's an upper bound on how many pending outbound messages a given extension can have. |
| 00:21.52 | doughecka | anyone ever play with it? |
| 00:21.56 | tzanger | I know I can't hookflash but I wonder |
| 00:21.58 | tzanger | brb |
| 00:22.00 | pdracevich | tzanger: hummm, at point "B" a new router has been put in place i dont think udp/4569 has been opened |
| 00:22.13 | BrianR___ | tzanger: If I recall correctly, I couldn't make it work over DISA. |
| 00:22.43 | *** part/#asterisk xcoyote (~coyote@dsl-200-95-78-238.prod-infinitum.com.mx) |
| 00:23.14 | tzanger | BrianR___: not trying it through DISA |
| 00:23.31 | BrianR___ | tzanger: How does one access internal features over a norstar PRI? |
| 00:23.37 | DonX | How can I find out what timer asterisk is using? |
| 00:24.11 | BrianR___ | The norstar PRI cards don't even support NET mode, do they? |
| 00:24.30 | DonX | If it's using one at all. I got zaprtc to load btu my inbound IAX calls are still choppy so I'm suspecting that something is wrong |
| 00:26.39 | bkw_ | haha my isp is on crack I tell ya |
| 00:26.41 | bkw_ | C R A C K |
| 00:26.47 | doughecka | ahahah |
| 00:26.55 | bkw_ | they confused me for a few |
| 00:26.59 | bkw_ | but I caught myself |
| 00:27.38 | yashax | snewpy: ? |
| 00:27.43 | tzanger | BrianR___: you don't |
| 00:28.13 | tzanger | I have a route which sends any 9 traffic over the PRI with an unlimited length |
| 00:28.22 | tzanger | and another which routes 8 traffic ot the PRI with a 3-digit length |
| 00:28.34 | bkw_ | I want my freakin reverse DNS back |
| 00:28.37 | tzanger | * sees "call from "224 to 844" or whatever |
| 00:29.23 | BrianR___ | tzanger: Ok. So all of your asterisk extensions start with an 8? |
| 00:29.39 | doughecka | haha |
| 00:29.40 | syslod | bkw_: Is this related to why I can't get to you? |
| 00:30.32 | bkw_ | nope |
| 00:30.32 | DonX | Ok, so there is no way to show what timer, if any, that asterisk is using? |
| 00:30.45 | bkw_ | its my /28 here at home |
| 00:30.51 | syslod | :) |
| 00:31.09 | doughecka | bkw_: you have nufone setup? |
| 00:31.16 | bkw_ | no |
| 00:31.17 | bkw_ | ? |
| 00:31.22 | doughecka | exten => nufonenumber,1,Dial(SIP/2001,60,tr) |
| 00:31.24 | bkw_ | why? |
| 00:31.26 | doughecka | do I put my whole number in therE? |
| 00:31.32 | doughecka | or last four digits? |
| 00:31.36 | bkw_ | try it with last 4 or full |
| 00:31.38 | BrianR___ | tzanger: If extension 201 on the norstar is forwarded to 801, and extension 202 calls 201, what does asterisk see? |
| 00:31.42 | bkw_ | aasdfiwefad |
| 00:31.52 | doughecka | full being with or without the 1 |
| 00:32.01 | bkw_ | ya |
| 00:32.14 | qwerp | harlo... |
| 00:32.19 | DonX | If there is no way then I'll just use the force and pray |
| 00:32.22 | doughecka | bkw_: ya.... |
| 00:32.29 | doughecka | with, or without :P |
| 00:32.30 | qwerp | is there anyway i can block only 15 incoming and 15 outgoing line on a PRI line? |
| 00:32.34 | bkw_ | without |
| 00:32.37 | doughecka | ah, k |
| 00:32.42 | bkw_ | try it all three ways |
| 00:32.45 | Uajal | bkw: I corrected dtmf by unbelivable way |
| 00:32.53 | bkw_ | Uajal, what was it? |
| 00:32.54 | doughecka | crap, I could even get the call to reach my pbx last time |
| 00:32.55 | bkw_ | or how did you do it |
| 00:33.09 | doughecka | couldn;t |
| 00:34.17 | PyroSteve | i though Broadvoice allowed sip users to set caller id ? |
| 00:34.31 | bkw_ | riiight |
| 00:34.31 | tzanger | BrianR___: uhm... 201 I think |
| 00:34.33 | Uajal | host=proxy.dca.broadvoice.com --- didn't call at all; host=sip.broadvoice.com -- called but no dtmf; host=proxy.chi.broadvoice.com - works! |
| 00:34.35 | tzanger | yes |
| 00:34.36 | syslod | qwerp: Do what? |
| 00:34.40 | Damin | Hmm.. |
| 00:34.42 | tzanger | it does because I get the right voicemail I think |
| 00:34.43 | PyroSteve | I tried setting it like i do with VoicepPulse |
| 00:34.45 | tzanger | I'll have to check again |
| 00:34.47 | Damin | Has the Dundi patch been updated for 1.0.5 yet? |
| 00:34.48 | tzanger | I'm at home right now |
| 00:34.56 | bkw_ | Damin, don't think |
| 00:34.58 | qwerp | is there anyway i can block only 15 incoming and 15 outgoing line on a PRI line? |
| 00:35.03 | Damin | I've getting conflicts when I Cvsup |
| 00:35.09 | doughecka | bkw_: hah, when I call, its silent for about 20 seconds, and rings once, and hangs up |
| 00:35.12 | qwerp | syslod: got a pri with 40 channels.. |
| 00:35.15 | doughecka | and I dont see a bloomin thing on my console |
| 00:35.19 | qwerp | syslod: got a pri with 30 channels.. |
| 00:35.20 | BrianR___ | tzanger: Aah... Even if it doesn't it's still possible to get the right voicemail by assigning each user a direct-to-voicemail extension on asterisk. |
| 00:35.25 | Damin | bkw_: Who do I have to pay to fix the patch? :) |
| 00:35.26 | tzanger | yes |
| 00:35.35 | syslod | qwerp: U want to block 15 incoming and 15 our or you want to set one way trunks in those qua? |
| 00:35.36 | tzanger | BrianR___: I also have DIDs assigned for several extensions |
| 00:35.36 | qwerp | syslod: but wanna make it in such a way that 15 incoming and 15 out going calls.. |
| 00:35.37 | Damin | bkw_: I've got cash in the PayPal account! ;) |
| 00:35.40 | tzanger | but there's a limit of 30 I think |
| 00:35.43 | tzanger | so 0000243 goes to my extension |
| 00:35.46 | bkw_ | Damin, anthm? |
| 00:35.52 | qwerp | syslod: it sort of like a quota.. |
| 00:35.54 | Damin | bkw_: Tell him to get on it.. |
| 00:35.56 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 00:35.58 | bkw_ | I will |
| 00:36.03 | Damin | bkw_: $100? |
| 00:36.07 | bkw_ | maybe |
| 00:36.14 | bkw_ | put in the topic in #asterisk-dev |
| 00:36.17 | BrianR___ | tzanger: We have a norstar that's about 10 stations away from being max'd out. |
| 00:36.26 | syslod | qwerp: You can do that but you'll need the co-op of your carrier to setup some of the Bs for 1 way trunking. |
| 00:36.26 | *** join/#asterisk DaLion (~Miranda@HSE-QuebecCity-ppp3497400.sympatico.ca) |
| 00:36.39 | BrianR___ | The norstar has been a really awesome system since the phones are only about $30 on eBay. |
| 00:36.41 | DaLion | hi |
| 00:36.50 | DaLion | got a prob |
| 00:36.50 | DaLion | i rebooted and now RH says |
| 00:36.51 | DaLion | Feb 21 19:34:27 pobox kernel: PCI: Sharing IRQ 10 with 00:02.7 |
| 00:36.52 | DaLion | Feb 21 19:34:27 pobox kernel: wcfxo: Out of space to write register 06 with e0 |
| 00:36.53 | tzanger | BrianR___: maxed out in the fully extended sense or maxed out in in teh 30 or whatever station ports a standard MICS has |
| 00:36.54 | BrianR___ | A 16 port station module typically cost us under $200 used. |
| 00:37.04 | DaLion | cand init DAA giving up |
| 00:37.07 | DaLion | any ifdea ? |
| 00:37.13 | qwerp | syslod: i did dome reading, initially i tot that i can use CheckGroup(max) to limit calls.. |
| 00:37.16 | BrianR___ | tzanger: All of the internal ports and all of the station module ports are filled. :( |
| 00:37.17 | PyroSteve | does broadvoice allow thier users to specify caller id ? I tried and isn't working |
| 00:37.19 | tzanger | I have a 32x0 and a 0x16 in addition to my standard set |
| 00:37.21 | pcm | DaLion: is that a real X100P ? |
| 00:37.39 | DaLion | it worked like 2 days ago |
| 00:37.39 | tzanger | PyroSteve: why don't you call the service you're paying and ask them? |
| 00:37.41 | DaLion | why |
| 00:37.41 | qwerp | syslod: but CheckGroup onli applies on a single channel, not a whole PRI card.. |
| 00:37.46 | DaLion | think so |
| 00:37.51 | syslod | qwerp: Well I guess you could on one side but the telco is gonna send you more calls than you want. |
| 00:37.52 | bkw_ | WRONG |
| 00:37.53 | tzanger | DaLion: did you buy it from digium or ebay |
| 00:37.57 | BrianR___ | tzanger: Is there any way to mamke the last 2 station module ports not be reserved for those lame wireles station modules? |
| 00:37.57 | pcm | DaLion: sounds like it's not taking IRQs |
| 00:38.03 | bkw_ | qwerp, define what you mean |
| 00:38.06 | tzanger | BrianR___: not sure |
| 00:38.07 | bkw_ | a whole PRI card? |
| 00:38.12 | tzanger | BrianR___: we have two of those, btw |
| 00:38.13 | PyroSteve | because this channel is for asterisk help and I am an asterisk users |
| 00:38.14 | bkw_ | check group will work on ANY and all channels |
| 00:38.16 | qwerp | bkw_: pri with 30 channels.. |
| 00:38.21 | bkw_ | it will work |
| 00:38.22 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
| 00:38.25 | bkw_ | if you set the groups |
| 00:38.25 | DaLion | pcm.. yeah... but .. onlything changed i removed an old modem card.. and my wirless pci from board |
| 00:38.26 | BrianR___ | tzanger: How well do those wireless sets work anyway? |
| 00:38.26 | tzanger | PyroSteve: but you're asking a BROADVOICE question |
| 00:38.30 | tzanger | BrianR___: shittily |
| 00:38.34 | PyroSteve | tzanger: so next time you have questions about a provider ... dont ask here |
| 00:38.35 | DaLion | also i made sure bios plug n play os is to NO |
| 00:38.36 | bkw_ | setgroup sets a channel var |
| 00:38.37 | bkw_ | thats all |
| 00:38.40 | tzanger | BrianR___: I mean they work but they sound poor, go screwy sometimes, etc. |
| 00:38.41 | tzanger | PyroSteve: exactly |
| 00:38.42 | qwerp | bkw_: wanna make i such a way that onli accept 15 incoming calls so that there is 15 reserved for outgoing calls. |
| 00:38.43 | DaLion | so bios assigns irq;s etc |
| 00:38.44 | bkw_ | checkgroup counts how many channels in tht group |
| 00:38.52 | tzanger | PyroSteve: you ask the provider of the service you're paying for |
| 00:38.52 | bkw_ | qwerp, that will do it |
| 00:38.56 | pcm | DaLion: is there anything else assigned to the same IRQ ? |
| 00:38.58 | bkw_ | thats what its written for |
| 00:39.03 | syslod | Umm what u gonna do just let the trunk ring on inbound calls? |
| 00:39.06 | BrianR___ | tzanger: Aah. I haven't been able to find them on eBay and I can't see payign full price for something my gut told me was likely to be lame. |
| 00:39.16 | DaLion | yes |
| 00:39.16 | PyroSteve | tzanger: so you really have never asked anyone about something they may have experienced ? |
| 00:39.19 | DaLion | its sharing |
| 00:39.26 | DaLion | Feb 21 19:34:27 pobox kernel: PCI: Sharing IRQ 10 with 00:02.7 |
| 00:39.28 | pcm | DaLion: it doesn't like to share hehe |
| 00:39.30 | DaLion | but with what |
| 00:39.32 | tzanger | PyroSteve: it stands to reason that since you're giving them money and have a question about their service that they'd have the answer |
| 00:39.32 | DaLion | i dont know |
| 00:39.35 | qwerp | bkw_: i read doc on checkgroup. it onli checks on a channel, but pri line got 30 channels.. |
| 00:39.41 | bkw_ | so |
| 00:39.45 | bkw_ | you don't understand how it works then |
| 00:39.46 | bkw_ | ok |
| 00:39.49 | bkw_ | inbound calls |
| 00:39.55 | bkw_ | you do a SetGroup(INBOUND) |
| 00:39.59 | qwerp | bkw_: how can i make it in such away so that 30 channel is treated as a channel? |
| 00:40.02 | PyroSteve | tzanger: have you bought digim hardware before ? |
| 00:40.06 | bkw_ | you don't |
| 00:40.10 | tzanger | PyroSteve: of course I have... but I haven't asked such a blatantly provider-specific question in a general forum before |
| 00:40.11 | bkw_ | you're thinking in 2D |
| 00:40.12 | bkw_ | stop it |
| 00:40.14 | bkw_ | think in 3d please |
| 00:40.17 | bkw_ | ok |
| 00:40.21 | bkw_ | check group will walk the channel list |
| 00:40.21 | qwerp | k |
| 00:40.25 | DaLion | PCI: Using IRQ router SIS [1039/0008] at 00:02.0 |
| 00:40.29 | bkw_ | and count the number of channels in use in GROUP X |
| 00:40.32 | bkw_ | if its over that |
| 00:40.33 | tzanger | PyroSteve: of course. Two TE405Ps, two T100Ps and a half dozen TDM4xxPs |
| 00:40.33 | bkw_ | it goes n+101 |
| 00:40.44 | bkw_ | thus you can give a busy signal when its over the group count limit |
| 00:40.45 | tzanger | PyroSteve: but you can bet your ass any specific questions I had about the hardware went to Digium first. |
| 00:41.06 | bkw_ | groupcount has nothing to do with the group= line in zapata.conf |
| 00:41.12 | bkw_ | they are totally diffrent things |
| 00:41.12 | qwerp | to use that, initially we need to SetGroup(inbound), right? |
| 00:41.17 | bkw_ | yes |
| 00:41.34 | DaLion | pcm how cna i know with what its sharing |
| 00:41.38 | Moc____ | Anyone here Sell Polycom IP Phone ? |
| 00:41.49 | qwerp | when i run this line in my dial plan, it show on my CLI something like "SetGroup("Zap/1-1", "inbound") |
| 00:41.50 | tzanger | PyroSteve: don't get your back up -- I am just saying that your question is FAR better asked of the people you're trying to work with rather than a bunch of people who may or may not use the service from one of the only providers which is regularly criticized in -users |
| 00:41.50 | pcm | dalion: after you load a module do 'cat /proc/interrupts' |
| 00:41.53 | bkw_ | the checkgroup(15) |
| 00:41.53 | doughecka | anyone in here use Nufone with a DID that can give me a working config? |
| 00:42.01 | syslod | Moc: To who? Customers or installers? |
| 00:42.04 | bkw_ | if its over 15 it will go n+101 |
| 00:42.06 | bkw_ | get it |
| 00:42.07 | tzanger | doughecka: have you emailed support@ |
| 00:42.07 | Moc____ | to customers |
| 00:42.15 | Moc____ | well installers I mean.. |
| 00:42.17 | *** join/#asterisk drastixnz (~paul@smtp.aucklandtax.co.nz) |
| 00:42.18 | Moc____ | Im selling to customers |
| 00:42.19 | DaLion | i dont seep 10 |
| 00:42.22 | doughecka | tzanger: nah :P |
| 00:42.24 | DaLion | cat /proc/interrupts' |
| 00:42.35 | syslod | Moc: U need distrib then. Graybar is one. |
| 00:42.35 | BrianR___ | tzanger: I will be doing some serious norstar <-> asterisk hacking this week. We should keep in touch. |
| 00:42.38 | DaLion | firs col is IRQ ? |
| 00:42.41 | doughecka | I thought I'd ask, since it was working before I sorta lost the config files |
| 00:42.42 | doughecka | :P |
| 00:42.47 | tzanger | doughecka: :-) email 'em, you'll get a ticket back and if you don't have an answer in a few hours bug jerjer, he'll smack the responsible party upside the head and get your answer quickly |
| 00:42.50 | qwerp | bkw_: so it doesn't care on which channel? as long as the SetGroup(name) name is the same? |
| 00:42.55 | tzanger | BrianR___: akohlsmith@mixdown.ca |
| 00:42.59 | bkw_ | qwerp, right |
| 00:43.03 | bkw_ | doughecka you register? |
| 00:43.05 | BrianR___ | tzanger: bristuccia@starentnetworks.com |
| 00:43.05 | pcm | Daion: yeah |
| 00:43.08 | bkw_ | did you add the register line boi? |
| 00:43.10 | doughecka | bkw_: it sats it registered |
| 00:43.13 | doughecka | says* |
| 00:43.14 | qwerp | bkw_: i will give it a try now.. ;P |
| 00:43.21 | doughecka | I Tried both switch-1 and switch-2 |
| 00:43.22 | PyroSteve | tzanger: well there are several people in here that represent some company or server like VoicePulse and Nufone, and probably Broadvoice |
| 00:43.27 | DaLion | eb 21 19:34:27 pobox kernel: PCI: Found IRQ 10 for device 00:0b.0 |
| 00:43.38 | DaLion | hmm no irq 10 on interupts.. but |
| 00:43.46 | PyroSteve | and I have asked similar questions in the past |
| 00:43.53 | syslod | bkw: Won't that still "use" a trunk to give the busy on inbound? |
| 00:44.02 | drastixnz | tzanger: pdracevich can you please have a look at my dial rule and tell me what i have done wrong "exten => _[4]X.,1,Dial(IAX/Whangarei/0${EXTEN})" Whangarei is point "A" and the rule is in the extinson.conf file at point "B" |
| 00:44.12 | tzanger | PyroSteve: this is true -- have you got an answer from anyone though yet? :-) |
| 00:44.19 | bkw_ | drastixnz, who told you to dial like that |
| 00:44.25 | tzanger | wtf is _[4]X. ?? |
| 00:44.25 | bkw_ | you need IAX2/user@peer/exten |
| 00:44.34 | bkw_ | _4X is the same |
| 00:44.38 | bkw_ | as _[4]X |
| 00:44.51 | tzanger | bkw_: ahh ok caracter alternatives |
| 00:44.55 | tzanger | (can't think of the acutal regex name0 |
| 00:45.07 | bkw_ | you can do _[123]X |
| 00:45.07 | *** join/#asterisk w0w0 (~w0w0@80-28-171-26.adsl.nuria.telefonica-data.net) |
| 00:45.13 | tzanger | bkw_: right I understand |
| 00:45.16 | DaLion | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 00:45.17 | bkw_ | wook |
| 00:45.21 | doughecka | bkw_: btw outgoing calls work :P |
| 00:45.23 | bkw_ | DaLion, call digium support |
| 00:45.26 | PyroSteve | tzanger: not yet, but there are tons of questions that are asked that may or may not catch somones attention |
| 00:45.34 | PyroSteve | but I have in the past |
| 00:45.37 | DaLion | oh well |
| 00:45.37 | bkw_ | doughecka sounds l ike you need to email support |
| 00:45.44 | tzanger | drastixnz: simplify it and see if it works. try "exten => 400,1,Dial(IAX2/Whangerei/0${EXTEN})" and reload and see if dialing 400 gets you there |
| 00:45.53 | DaLion | can i force this ? |
| 00:45.57 | bkw_ | well IAX2/peer/exten is wrong |
| 00:45.58 | DaLion | irq ? |
| 00:45.58 | PyroSteve | tzanger: i have questioned others in the past about my voicepulse trouble |
| 00:46.01 | drastixnz | ok bbs |
| 00:46.03 | tzanger | PyroSteve: again correct -- I was just trying to help you get your answer more quickly |
| 00:46.05 | bkw_ | what username are you gonna use on the other end |
| 00:46.07 | tzanger | bkw_: it is?? |
| 00:46.17 | bkw_ | well its gonna pick guest@ or what ever it can find |
| 00:46.22 | bkw_ | which you have no control over it |
| 00:46.22 | PyroSteve | tzanger: i have gotten help from voicepulse as well as other voicepulse reps |
| 00:46.31 | tzanger | bkw_: it'll pick whatever the usrename in [peer] is set to |
| 00:46.37 | bkw_ | no it wont |
| 00:46.41 | qwerp | bkw_: i did the test, not working.. |
| 00:46.51 | bkw_ | qwerp, you're doing something wrong |
| 00:46.54 | *** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 00:46.57 | drastixnz | tzanger: nothing |
| 00:46.58 | Sedorox | you can't do <user>@<hostinsip.conf> to dial a SIP person? |
| 00:47.08 | qwerp | first call comes in, in CLI SetGroup("Zap/2-1", "INBOUND") |
| 00:47.09 | bkw_ | this is IAX |
| 00:47.30 | Sedorox | yea.. but I'm trying it with sip and it doesn't wanna work... |
| 00:47.30 | qwerp | CLI CheckGroup("Zap/2-1", "1") |
| 00:47.36 | bkw_ | that lets one thru |
| 00:47.41 | tzanger | drastixnz: well there's a problem then innit |
| 00:47.43 | bkw_ | then do you have N+101 on checkgroup |
| 00:47.45 | qwerp | then the second call comes in, |
| 00:47.50 | tzanger | drastixnz: are you paying attention to the output of the commands? |
| 00:47.58 | qwerp | SetGroup("Zap/3-1", "INBOUND") |
| 00:48.00 | tzanger | drastixnz: :-) |
| 00:48.01 | trelane | bkw_, iaxy's are sick, I'm eternally awed |
| 00:48.01 | qwerp | CLI CheckGroup("Zap/3-1", "1") |
| 00:48.02 | tzanger | er doughecka :-) |
| 00:48.09 | doughecka | lol |
| 00:48.15 | doughecka | 7:48EST, lets see |
| 00:48.22 | qwerp | i think check group is checking on 2 different groups. |
| 00:48.44 | bkw_ | qwerp, The checkgroup has NOTHING to do with the group= stuff in zapata.conf |
| 00:48.48 | bkw_ | stop trying to mix the two please |
| 00:48.51 | drastixnz | tzanger: *blush* err is there a way of logging the iax2 commands? |
| 00:48.52 | *** join/#asterisk cool_blade (~johnhewit@mail.lanskey.com.au) |
| 00:49.01 | tzanger | drastixnz: no need for that |
| 00:49.01 | qwerp | i do have exten => s,103,hangup |
| 00:49.03 | tzanger | set verbose 10 |
| 00:49.08 | tzanger | you should see stuff showing up on the console |
| 00:49.21 | bkw_ | qwerp, works here for me unless the syntax has changed |
| 00:49.25 | bkw_ | try 15@INBOUND |
| 00:49.40 | bkw_ | I would have to looka t the code |
| 00:49.41 | DaLion | pcm how can i know hwat its sharing with ? |
| 00:49.41 | DaLion | pcm how can i know what its sharing with ? |
| 00:49.41 | tzanger | drastixnz: you should see stuff like this |
| 00:49.42 | qwerp | SetGroup(15@INBOUND) <-- |
| 00:49.45 | tzanger | <PROTECTED> |
| 00:49.48 | tzanger | <PROTECTED> |
| 00:49.51 | tzanger | <PROTECTED> |
| 00:49.53 | tzanger | <PROTECTED> |
| 00:49.56 | tzanger | <PROTECTED> |
| 00:50.00 | bkw_ | qwerp, sure |
| 00:50.02 | tzanger | it usually gives you a good idea of what's going on |
| 00:50.02 | DaLion | apata Telephony Interface Registered on major 196 |
| 00:50.02 | DaLion | PCI: Found IRQ 10 for device 00:0b.0 |
| 00:50.02 | DaLion | PCI: Sharing IRQ 10 with 00:02.7 |
| 00:50.10 | bkw_ | DaLion, Dude call tech support |
| 00:50.17 | DaLion | ? |
| 00:50.19 | tzanger | tech support? |
| 00:50.19 | tzanger | for what |
| 00:50.27 | tzanger | a motherboard that won't NOT share IRQs? |
| 00:50.28 | bkw_ | his card install |
| 00:50.29 | DaLion | lol |
| 00:50.35 | qwerp | bkw_: done. |
| 00:50.40 | DaLion | it worked yesterday |
| 00:50.48 | tzanger | DaLion: what did you change from yesterday |
| 00:50.50 | qwerp | then i should put CheckGroup(15@INBOUND) too? |
| 00:50.51 | tzanger | and don't say nothing |
| 00:50.51 | DaLion | so if config i changed somewhere |
| 00:50.59 | DaLion | not sure... alot |
| 00:51.01 | doughecka | tzanger: I havnt recieved a ticket number yet!!!! its been 2 min!!! :P |
| 00:51.03 | bkw_ | no ninny |
| 00:51.05 | tzanger | DaLion: well there's problem #1 |
| 00:51.06 | bkw_ | Setgroup INBOUND |
| 00:51.09 | bkw_ | checkgroup 15@inbound |
| 00:51.09 | DaLion | removed 2 useless cards.. one modem + 1 pci wireless |
| 00:51.11 | tzanger | doughecka: hmm it should be htere soon :-) |
| 00:51.16 | qwerp | oh.. okie. |
| 00:51.23 | tzanger | DaLion: did you move the X101P around? |
| 00:51.27 | bkw_ | DaLion, let me guess 50 dollar motherboard? |
| 00:51.29 | DaLion | well its REMOVED i sad.. so should be more irq to play with not less |
| 00:51.44 | DaLion | bkw_ whats your problem today |
| 00:51.46 | qwerp | bkw_ : off trying.. |
| 00:51.51 | doughecka | DaLion: lol |
| 00:51.54 | bkw_ | DaLion, I don't have a problem.. this is me |
| 00:52.01 | DaLion | hehe yeah |
| 00:52.06 | doughecka | bkw_: what motherboard ISNT 50 bucks? :P |
| 00:52.11 | tzanger | DaLion: agreed, did you go int o the bios and make sure all the slots aren't set to the same IRQ (if you can set it) and also did you reset the ECSD data |
| 00:52.14 | DaLion | PCI: Found IRQ 10 for device 00:0b.0 |
| 00:52.14 | DaLion | PCI: Sharing IRQ 10 with 00:02.7 |
| 00:52.21 | DaLion | yes and yes |
| 00:52.23 | tzanger | DaLion: did you change kernels |
| 00:52.30 | bkw_ | also enable APIC |
| 00:52.30 | DaLion | im tring to know what is using 00:0b |
| 00:52.36 | DaLion | ah ... |
| 00:52.38 | DaLion | i diabled |
| 00:52.39 | tzanger | lspci -v | grep 00:0b |
| 00:52.41 | yashax | (IP500) Guys, if the phone1.conf is setup correctly to bypass the IM, when I click on VM button, should I still see a menu, or should it go right to my VM? |
| 00:52.47 | bkw_ | APIC will help |
| 00:52.49 | tzanger | yes |
| 00:52.51 | tzanger | it will |
| 00:53.06 | tzanger | unless you have a Shuttle S51, in which case it will cause its own special brand of problems :-) |
| 00:53.08 | Damin | bkw_: patching file pbx.c |
| 00:53.08 | Damin | Hunk #1 succeeded at 820 with fuzz 3 (offset 4 lines). |
| 00:53.08 | Damin | Hunk #2 succeeded at 829 (offset 4 lines). |
| 00:53.08 | Damin | Hunk #3 succeeded at 853 with fuzz 3 (offset 4 lines). |
| 00:53.22 | DaLion | ok see |
| 00:53.24 | Damin | bkw_: 2 line mismatch.. ;) |
| 00:53.28 | DaLion | only needed to know lspci command |
| 00:53.30 | DaLion | 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] SiS7012 PCI Audio Accelerator (rev a0) |
| 00:53.33 | *** join/#asterisk mixi (~mixi@pD9E592CC.dip.t-dialin.net) |
| 00:53.40 | DaLion | its confilting with on board audio |
| 00:53.48 | tzanger | DaLion: bingo |
| 00:53.49 | DaLion | if i disable i guess i cant use moh no more ? |
| 00:53.55 | doughecka | <Peewee> HA ha |
| 00:53.59 | yashax | tzanger: was that to me "yes it will" ? |
| 00:54.03 | drastixnz | tzanger: in my iax.conf at Point "B" should the type be set too peer? for the information about point "A" |
| 00:54.04 | tzanger | hmm I can get .in domains for US$28 |
| 00:54.05 | tzanger | er $18 |
| 00:54.15 | doughecka | .in being state of indiana? |
| 00:54.17 | doughecka | sweet |
| 00:54.17 | DaLion | any way to force RH to make it elsewhere ? or make zap elsewhere ? |
| 00:54.18 | tzanger | yashax: no that was for DaLion |
| 00:54.19 | doughecka | where from? |
| 00:54.29 | tzanger | yashax: I don't know what you're doing :-) |
| 00:54.31 | doughecka | DaLion: MoH shouldnt need a sound card |
| 00:54.39 | tzanger | doughecka: haha how about .india |
| 00:54.41 | yashax | k |
| 00:54.47 | doughecka | tzanger: ah, right |
| 00:54.50 | yashax | (IP500) Guys, if the phone1.conf is setup correctly to bypass the IM, when I click on VM button, should I still see a menu, or should it go right to my VM? |
| 00:54.56 | drastixnz | tzanger: set it to 10 and the phone does not even get the the asterisk box, while on sip calls it works. |
| 00:54.57 | doughecka | still that would be cool |
| 00:54.59 | yashax | that's what I am doing |
| 00:55.06 | tzanger | drastixnz: huh? |
| 00:55.06 | DaLion | ok let me try thanks |
| 00:55.27 | drastixnz | [Whangarei] |
| 00:55.27 | drastixnz | type=peer |
| 00:55.27 | drastixnz | secret=paswyas1 |
| 00:55.27 | drastixnz | host=dynamic |
| 00:55.27 | drastixnz | ;context=inbound |
| 00:55.27 | drastixnz | defaultip=218.101.54.x |
| 00:55.35 | doughecka | sugar in the bloodstream, yay |
| 00:55.46 | qwerp | bkw_: not working.. |
| 00:55.49 | doughecka | and NO email back yet, tzanger =D |
| 00:55.53 | bkw_ | qwerp, check the wiki |
| 00:55.55 | bkw_ | because I knwo it does |
| 00:56.10 | Sedorox | anyone know if I can user user@userinsip for a sip call? |
| 00:56.18 | qwerp | bkw_: i know it does, i remember trying it once.. |
| 00:56.20 | Sedorox | ererrrr bm |
| 00:56.23 | Sedorox | errrr nm |
| 00:56.24 | Sedorox | I'm stupid |
| 00:56.29 | qwerp | bkw_: but it just doesn't seems to work now.. |
| 00:56.46 | tzanger | set what to 10 |
| 00:56.48 | tzanger | drastixnz: |
| 00:56.49 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 00:56.55 | dsmouse | stop when convenient |
| 00:57.00 | dsmouse | er, wrong window |
| 00:57.02 | tzanger | haha |
| 00:57.02 | doughecka | Defraz: sonicwall.dcdi.net? |
| 00:57.04 | ACiDV | I have few agents/queues defined, does it's possible to put an agent in "break time" to avoid multiple login/logout in the same day ? |
| 00:57.13 | tzanger | what's the difference between restart gracefully and restart when convenient |
| 00:57.14 | Defraz | hey. |
| 00:57.22 | doughecka | tzanger: more letters |
| 00:57.23 | bkw_ | yes it works |
| 00:57.26 | tzanger | doughecka: heh |
| 00:57.40 | drastixnz | tzanger: Verbosity |
| 00:57.43 | DaLion | trying to reboot box |
| 00:57.57 | bkw_ | qwerp, check your msgs |
| 00:57.59 | bkw_ | I just tested it |
| 00:57.59 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-10.sbi.ct.charter.com) |
| 00:58.02 | bkw_ | what I pasted works |
| 00:58.04 | tzanger | drastixnz: and what do you mean by the phone doesn't get to the asterisk box |
| 00:58.11 | Defraz | when do I need a g729 license? |
| 00:58.13 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 00:58.18 | MrEntropy | yo |
| 00:58.20 | tzanger | Defraz: when you're converting from g729 to anything else |
| 00:58.25 | *** join/#asterisk mrproper_ (~psynode@61.95.55.242) |
| 00:58.45 | drastixnz | tzanger: I ring, 400 and it does not even log anything at all |
| 00:58.50 | mrproper_ | hey all, whats the recommended gui configuration manager for asterisk? (bit daunted with the 50 options to choose from) |
| 00:58.58 | DaLion | now sharing with 00:02.6 Modem: Silicon Integrated Systems [SiS] Intel 537 [56k Winmodem] (rev a0) |
| 00:59.00 | Defraz | so if I have a sip phone and I dial and it pass it on to my ld provider Cloud voice then it should work. |
| 00:59.03 | DaLion | god !!! |
| 00:59.06 | tzanger | drastixnz: well... is your sip phone going to the right context? |
| 00:59.08 | DaLion | what the hell... |
| 00:59.10 | mishehu | mrproper_: there is a recommended gui configurator? |
| 00:59.14 | tzanger | the context wtih the exten => 400,1,Dial... |
| 00:59.18 | bkw_ | qwerp, did you see the example I pasted to you |
| 00:59.20 | doughecka | mrproper_: get asterisk@home if you just want to start from scract |
| 00:59.21 | tzanger | mishehu: vim, of course |
| 00:59.24 | tzanger | oh you meant GUI |
| 00:59.25 | tzanger | gvim |
| 00:59.29 | mishehu | tzanger: hahaaha |
| 00:59.31 | doughecka | tzanger: funny... =) |
| 00:59.32 | mrproper_ | lol |
| 00:59.44 | DaLion | ok il try to disable modem .. |
| 00:59.55 | bkw_ | oh DaLion didn't buy from digium |
| 01:00.02 | DaLion | might as well diable all motherboard with a hammer or 8 pounder |
| 01:00.07 | bkw_ | haha |
| 01:00.13 | tzanger | DaLion: I bet that'll get rid of the IRQ sharing |
| 01:00.15 | bkw_ | when you disable it in the bios it doesn't really disable it |
| 01:00.22 | bkw_ | shuttle boards are nasty anywy |
| 01:00.22 | tzanger | btw |
| 01:00.24 | bkw_ | er anyway |
| 01:00.26 | tzanger | just to make you guys envious |
| 01:00.41 | tzanger | <PROTECTED> |
| 01:00.49 | drastixnz | tzanger: please forgive me I am very very tired, I have been up all night. The context part is confusing me. :-( and thanks for you help |
| 01:00.50 | tzanger | this box is an NFS and samba server |
| 01:00.54 | bkw_ | tzanger, if you have alot of network activity you'll sure hear it |
| 01:00.58 | tzanger | bkw_: nope |
| 01:01.02 | doughecka | tzanger: holy crap |
| 01:01.05 | bkw_ | you're lucky as hell then |
| 01:01.07 | tzanger | I *hammered* this thing just to test |
| 01:01.09 | tzanger | bkw_: I agree :-) |
| 01:01.13 | tzanger | it's an old P3/700 |
| 01:01.13 | DaLion | hey.. bkw that on board modem |
| 01:01.15 | bkw_ | beacuse thats not normal |
| 01:01.19 | doughecka | well, heck, my server is a samba server... nobody ever connects =D |
| 01:01.28 | tzanger | P3/733, via chipset |
| 01:01.31 | bkw_ | hehe |
| 01:01.41 | tzanger | I know it's not normal :-) |
| 01:01.45 | tzanger | I'm just amazed |
| 01:01.49 | mishehu | I use either samba or nfs |
| 01:01.49 | bkw_ | lucky bastard |
| 01:01.50 | mishehu | not both |
| 01:01.52 | bkw_ | one day it will just stop working |
| 01:01.55 | bkw_ | you wait and see |
| 01:01.55 | bkw_ | haha |
| 01:01.58 | doughecka | hah |
| 01:02.07 | bkw_ | it will go WTF I shouldn't be working |
| 01:02.08 | doughecka | no, it start sounding like crap |
| 01:02.11 | mishehu | and then bkw_ will be there tell you he told you so |
| 01:02.13 | tzanger | there are 9 IDE HDDs in this thing too |
| 01:02.17 | bkw_ | ouch |
| 01:02.23 | tzanger | bkw_: shush |
| 01:02.31 | mishehu | ide? blechs |
| 01:02.37 | *** part/#asterisk klasstek (~peracles@sta-206-168-231-55.rockynet.com) |
| 01:02.41 | drastixnz | tzanger: he is my sip rule "exten => _[0]X.,1,Dial(SIP/${EXTEN:1}@10.10.x.x)" |
| 01:02.41 | doughecka | btw dont use SATA with asterisk |
| 01:02.41 | bkw_ | IDE isn't bad |
| 01:02.44 | mishehu | sata or scsi for me... |
| 01:02.45 | doughecka | =D |
| 01:02.47 | drastixnz | here sorry |
| 01:02.51 | bkw_ | drastixnz, stop the _[0] |
| 01:02.58 | bkw_ | its not needed if you only have 1 digit to match |
| 01:03.05 | mishehu | doughecka: and why no sata with asterisk? |
| 01:03.16 | mishehu | especially since I *am* using asterisk with sata |
| 01:03.24 | tzanger | oh..my..god |
| 01:03.29 | doughecka | mishehu: I had problems |
| 01:03.38 | mishehu | 3ware 9500 raid controller |
| 01:03.42 | *** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:03.43 | tzanger | I just got a telemarketer call but the call was automated |
| 01:03.45 | doughecka | the gods must smile apon thee |
| 01:03.48 | tzanger | like a reverse answering machine |
| 01:03.54 | doughecka | mishehu: ah, thats diferent |
| 01:04.04 | doughecka | I was using onboard SATA crap |
| 01:04.12 | tzanger | autodialer... "thank you for taking our call... got the winter blahs? You've been selected to win a vacation for 3 days and 3 nights...<click>" |
| 01:04.17 | doughecka | kept interupting my voice traffic |
| 01:04.24 | mishehu | doughecka: I've used it on on asus which has a crappy SIIG sata controller, and no problems either. |
| 01:04.28 | bkw_ | tzanger, isn't that illegal? |
| 01:04.36 | tzanger | I dunno |
| 01:04.40 | bkw_ | I think it is |
| 01:04.42 | doughecka | SIIG aint bad, but this was onboard Dell crap |
| 01:04.51 | mishehu | dell is from hell |
| 01:04.52 | doughecka | bkw_: SUE THEM |
| 01:04.53 | doughecka | NEXT!!! |
| 01:05.03 | doughecka | now SCSI on the other hand |
| 01:05.09 | doughecka | that works ok |
| 01:05.40 | drastixnz | tzanger: YAAAAAAA i am now getting somewhere |
| 01:05.49 | tzanger | drastixnz: good :-) |
| 01:05.53 | tzanger | and nary a PM required |
| 01:06.23 | drastixnz | tzanger: but I am now gettiing this :-( Executing Dial("SIP/4701705-1b9a", "IAX2/Whangerei/00") in new stack |
| 01:06.25 | drastixnz | chan_iax2.c:2212 create_addr: No such host: Whangerei |
| 01:06.25 | drastixnz | No such host: Whangerei |
| 01:06.46 | tzanger | drastixnz: it doesn't see a [Whangerei] |
| 01:06.48 | bkw_ | adfajjliksdjfa |
| 01:06.48 | tzanger | peer |
| 01:06.54 | bkw_ | crazy track pad input |
| 01:07.03 | bkw_ | hate that |
| 01:07.10 | bkw_ | bbl |
| 01:07.13 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 01:07.49 | drastixnz | tzanger: i did a iax2 show peers and i get "Whangarei 218.101.54.x (D) 255.255.255.255 4569 Unmonitored" |
| 01:07.59 | tzanger | IAX2/Whangerei |
| 01:08.02 | tzanger | Whangarei |
| 01:08.09 | tzanger | I see a difference, do you see a difference? |
| 01:08.14 | drastixnz | "DOH!" |
| 01:08.23 | drastixnz | tzanger: and i live here to!! |
| 01:08.33 | mrproper_ | has anyone pushed sip calls to MS live communications server clients? |
| 01:08.36 | tzanger | for some reason "Whangarei" makes me think of a waving penis |
| 01:08.41 | tzanger | not exactly sure why |
| 01:08.43 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 01:08.43 | Defraz | can someone explain what a pass thru is concidered when using g729? |
| 01:08.45 | doughecka | HAHAha |
| 01:08.58 | tzanger | Defraz: when asterisk does not have to listen to the audio stream |
| 01:09.08 | doughecka | tzanger: that actully made me laugh outloud |
| 01:09.30 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 01:09.32 | tzanger | doughecka: yeah me too when I first saw the word... I was like "Wtf that guy is either from India or he's got the same sense of humour that I do" |
| 01:09.46 | marc32344 | ne1 knows about digium te110? |
| 01:09.58 | tzanger | marc32344: it's a single span T1 card, similar to the T100P |
| 01:10.18 | doughecka | tzanger: and still no email from nufone, not even an automated reply... you sure they have a mail server? :P |
| 01:10.26 | Defraz | So if I am on my sip phone, and I connect using g729 and then my asterisk box passes it to and NexTone Iserver then would that be a pass thru? |
| 01:10.51 | doughecka | Defraz: correct IF nextone accepts g729 |
| 01:10.51 | tzanger | doughecka: it should have got a response to you by now -- my last test was 5min. bug jerjer directly -- he will want to know about this |
| 01:11.24 | shmaltz | how would one dial multiple multiline sip phones (cisco 7960) and making sure that all the phones ring on the next available line appearance? |
| 01:11.25 | doughecka | lol |
| 01:11.32 | Defraz | okay so in my sip.conf can I only allow g729? |
| 01:11.42 | mrproper_ | Defraz, yes |
| 01:11.45 | tzanger | Defraz: of course |
| 01:11.53 | tzanger | Defraz: that won't protect you though |
| 01:11.55 | *** part/#asterisk doughecka (~Doug@doughecka.user) |
| 01:12.00 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
| 01:12.05 | doughecka | crappy irc client |
| 01:12.06 | marc32344 | what is connected to the port of te110? |
| 01:12.07 | tzanger | Defraz: if you use 't' or 'T' or use Read or anything in the dialplan |
| 01:12.07 | Defraz | hmm I guess I don't quite undersand. |
| 01:12.13 | tzanger | marc32344: a T1 or PRI |
| 01:12.21 | doughecka | why put the stupid X to close the tab ON THE FRICKIN TAB ITSELF |
| 01:12.24 | doughecka | :P |
| 01:12.31 | tzanger | doughecka: :-) |
| 01:12.36 | Sedorox | you can't do a switch with a SIP? |
| 01:12.47 | shmaltz | how would one dial multiple multiline sip phones (cisco 7960) and making sure that all the phones ring on the next available line appearance? |
| 01:12.58 | marc32344 | whats the difference between T1 and PRI? |
| 01:13.00 | mikegrb | doughecka: use a real irc client and you won't have that problem |
| 01:13.08 | tzanger | shmaltz: jeez man wait a few minutes |
| 01:13.10 | mrproper_ | shmaltz, you mean something like you have an incoming number and you want it to call a range of extensions? |
| 01:13.11 | pcm | PRI is a signalling on T1 |
| 01:13.13 | doughecka | ping |
| 01:13.15 | tzanger | marc32344: physically nothing |
| 01:13.17 | doughecka | yea, gaim |
| 01:13.34 | doughecka | I rather would have 1 app covering all IM and IRC than 6 of them |
| 01:13.36 | tzanger | marc32344: a T1 is a physical and data spec. PRI is just a data spec, it rides on top of a T1 or E1 (or J1 even I think) |
| 01:13.45 | tzanger | doughecka: irssi |
| 01:13.45 | shmaltz | mrproper_, as well as rollover to the next available line appearance if the first one is busy |
| 01:13.53 | mikegrb | doughecka: I have one app covering im and 4 irc servers |
| 01:14.04 | mikegrb | doughecka: it is called irssi with bitlbee |
| 01:14.08 | doughecka | oh |
| 01:14.10 | doughecka | windows? |
| 01:14.15 | mikegrb | it can |
| 01:14.21 | doughecka | supported? |
| 01:14.22 | doughecka | :P |
| 01:14.24 | tzanger | doughecka: no... irssi is on linux |
| 01:14.28 | shmaltz | sorry tzanger, my IRC client acted up |
| 01:14.28 | mikegrb | it's text |
| 01:14.33 | tzanger | Psi is my jabber client, it's available for windows and mac and linux |
| 01:14.41 | mrproper_ | shmaltz, hang on do you want a round robin (ie call the first phone if its busy, call the second phone) or do you want to call ALL extensions at once (so the only phones it wont ring are ones that are busy) |
| 01:14.49 | doughecka | ah |
| 01:14.49 | mikegrb | tzanger: irssi can do jabber with bitlbee |
| 01:15.04 | tzanger | oh? |
| 01:15.18 | doughecka | lol |
| 01:15.28 | doughecka | CRAP |
| 01:15.30 | tzanger | "Your laptop is downloading pr0n and warez from the internet, and is unresponsive when you try to get it to do something useful. I'd put it's age at about 14." |
| 01:15.33 | doughecka | jerjer replyed |
| 01:15.33 | tzanger | hahahaha |
| 01:15.37 | doughecka | he doesnt see an account |
| 01:15.38 | mikegrb | tzanger: ja, connect to thegrebs.com for example, there are lots of public bitlbee servers or you can grab it yourself from bitlbee.org |
| 01:15.41 | doughecka | under my home email |
| 01:15.49 | shmaltz | mrproper_, I want that it should call all at once, and when ringing each phone (together with the others) if the first line appearance is busy it should rollover to the next one. |
| 01:16.15 | tzanger | does anyone else see "ars technica" as "arse technica" ? |
| 01:16.21 | tzanger | I have always, always seen that |
| 01:16.22 | doughecka | tzanger: :) |
| 01:16.40 | tzanger | similarly there's a road in Stratford called Embro Road... I always see it as "Embryo Road" |
| 01:16.47 | doughecka | LOL |
| 01:17.00 | doughecka | tzanger: trying to keep my yogurt in my mouth |
| 01:17.07 | tzanger | "I learned a thing or two from Charlie don't ya know, you better stay a way from embryo road..." |
| 01:17.17 | tzanger | doughecka: mmm yougurt |
| 01:17.20 | tzanger | er yogurt |
| 01:17.21 | mrproper_ | shmaltz, hang on just looking at my config |
| 01:17.29 | tzanger | I made tacos tonight wiht the kids |
| 01:17.32 | doughecka | lol |
| 01:17.34 | doughecka | fun fun fun |
| 01:17.39 | tzanger | my 4yo is hilarious... taCO! |
| 01:17.42 | jayden | hey tzanger, how's jitterbuffer going? |
| 01:17.44 | doughecka | hah |
| 01:17.51 | tzanger | of course he just takes the fucking thing apart and eats the cheese only, but oh well |
| 01:18.00 | tzanger | jayden: VERY well |
| 01:18.00 | doughecka | jayden: I dont think he has a kid called jitterbuffer... |
| 01:18.02 | doughecka | =D |
| 01:18.03 | tzanger | hahaha |
| 01:18.09 | doughecka | tzanger: hah |
| 01:18.11 | jayden | :) |
| 01:18.24 | doughecka | change your name, its too simaler to tzanger |
| 01:18.25 | doughecka | :P |
| 01:18.38 | tzanger | yeah really... I've been on IRC for the better part of a decade now |
| 01:18.42 | doughecka | silly tab completion |
| 01:18.51 | doughecka | lol |
| 01:18.52 | tzanger | actually probably over a decade now if you include my mIRC days |
| 01:18.58 | tzanger | efnet and mIRC just didn't mix |
| 01:19.15 | doughecka | funny |
| 01:19.25 | tzanger | doughecka: I am ashamed of some of the stuff that turns up in usenet searches for my name |
| 01:19.41 | Beirdo | hehehe |
| 01:19.48 | doughecka | I was 14 back then |
| 01:19.48 | BrianR___ | tzanger: Get my reply ping? |
| 01:19.48 | tzanger | although I had a guy who wrote a quote book in Holland request to use a quote of mine from years ago |
| 01:19.50 | mrproper_ | shmaltz, check http://www.pastebin.com/244752 theres an example of how to do it |
| 01:19.51 | Beirdo | we all likely have that to say |
| 01:20.05 | iceyp | anyone know if there is a web based system for asterisk to allow users to signup simular to pulver? I want to start a local NZ sip / iax server |
| 01:20.09 | tzanger | BrianR___: yup |
| 01:20.18 | tzanger | X-Greylist: delayed 399 seconds by postgrey-1.17 at mail; Mon, 21 Feb 2005 20:24:53 EST |
| 01:20.20 | Beirdo | I had a PhD candidate ask to use one of my flames in her thesis as an example of internet flaming :) |
| 01:20.33 | tzanger | Beirdo: hahaha |
| 01:20.45 | shmaltz | mrproper_, what happens if I want VM after 2 minutes of ringing? |
| 01:20.48 | Beirdo | so one of my flames is published officially somewhere |
| 01:20.48 | tzanger | what the fuck do you get a Ph.D writing about internet flame wars?? |
| 01:20.50 | yashax | Off the topic: Did anyone hear on the news about this guy who recorded his kid's voice as a phone ringer and now it is very popular? I am trying to find it. Anyone? Thanks... |
| 01:20.59 | tzanger | shmaltz: Dial(,120) |
| 01:21.01 | Beirdo | she sent me a copy, I don't know offhand where it is |
| 01:21.12 | tzanger | yashax: ugh |
| 01:21.21 | Beirdo | it was about the culture of the internet or something |
| 01:21.23 | tzanger | gimme a set of regular electronic ringy noises not fucking mucic |
| 01:21.25 | tzanger | er music |
| 01:21.29 | tzanger | or kids hollering |
| 01:21.32 | tzanger | I get enough of that |
| 01:21.38 | yashax | hahaha |
| 01:21.45 | shmaltz | this wouldn't work for me |
| 01:21.54 | doughecka | not this little song crap |
| 01:21.54 | yashax | it was really cute.... seriously, does anyoine know? |
| 01:21.55 | jayden | MAAMMAAA... ANSWER THE DAMN PHONE!! |
| 01:22.06 | tzanger | doughecka: yeah but battery life sucks with those old bell ringers |
| 01:22.06 | mrproper_ | shmaltz, then in your extensions.conf on the extension line change the 25 to what ever amount of seconds you want then add a second line with priority 2 which moves the call to voicemail |
| 01:22.21 | doughecka | tzanger: I just want a simple ringer |
| 01:22.25 | tzanger | DAAAAAAAAADDYYY... CAMERON HIT MEEEEEEEEEEEE!!! |
| 01:22.27 | doughecka | like what old cordless phones had or whatever |
| 01:22.37 | tzanger | and then the voicemail beep "NOOOOOOOOOOOOOO I DIDN'TTTT!!!!" |
| 01:22.37 | yashax | HHHHHHHHhhaaaaaaaaaaaaaaaaaaaaa. I love it.......... |
| 01:22.42 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 01:22.50 | ta[i]nted | where does asterisk store CLI logs? |
| 01:22.56 | tzanger | ta[i]nted: /var/log/asterisk |
| 01:23.00 | shmaltz | mrproper_, this will not rollover and ring on the second line appearance if the first is busy (i'm using a cisco 7960, which has 6 line appearances) |
| 01:23.02 | tzanger | or wherever you put it in logger.conf |
| 01:23.15 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 01:23.26 | mrproper_ | shmaltz, http://www.pastebin.com/244755 for the voicemail stuff |
| 01:23.45 | jpablo | hi, im having a problem with asterisk reciving sip from a cisco server. |
| 01:23.56 | mrproper_ | shmaltz, what do you mean second line, that queue will just ring any available extensions you specify in the queues.conf |
| 01:24.01 | jpablo | it's a sip call, but the cisco doesnot register with the asterisk. |
| 01:24.18 | *** join/#asterisk _daver_ (~daver@ns1.tmok.com) |
| 01:24.29 | shmaltz | mrproper thsi is how I do it nomraly: |
| 01:24.31 | shmaltz | http://www.pastebin.com/244756 |
| 01:24.32 | jpablo | the call is arriving to the 5060 port (i see it with tcpdump) but asterisk just don't do anything, even with sip debug on i don't see nothing. |
| 01:24.34 | jpablo | any idea ? |
| 01:25.09 | drastixnz | tzanger: Now the next big question is incoming dial rules, can you give us a hint |
| 01:25.22 | mrproper_ | shmaltz, then implement that into the queues.conf if its a problem |
| 01:25.37 | shmaltz | mrproper_, my cisco 7960 register with 6 sip accounts (b/c they have six line appearances) |
| 01:25.42 | yashax | Guys: I am trying to disable the "Instant Messages" from the SIP500 menu to go straight to the VM and did the following, but it still shows up, any ideas? |
| 01:25.42 | yashax | <PROTECTED> |
| 01:25.42 | yashax | <PROTECTED> |
| 01:25.42 | yashax | <PROTECTED> |
| 01:25.47 | shmaltz | how? mrproper_ |
| 01:25.49 | *** join/#asterisk rtomsonII (~rtomson@ip70-181-140-181.sd.sd.cox.net) |
| 01:25.58 | tzanger | drastixnz: "incoming dial rules" ?? |
| 01:26.09 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 01:26.32 | drastixnz | tzanger: *blush* again, at point "A" when I call from point "B" |
| 01:26.42 | tzanger | drastixnz: right |
| 01:26.51 | mrproper_ | shmaltz: with your cisco phones i presume you have an extension configured for each phone in sip.conf right? |
| 01:26.54 | jayden | tzanger, are you |
| 01:27.01 | tzanger | jayden: am I? |
| 01:27.08 | mrproper_ | shmaltz, ie sip.conf----> [101] [102] etc? |
| 01:27.10 | shmaltz | mrproper_, yes |
| 01:27.12 | trelane | is there any more verbosity on startup than -vvv? |
| 01:27.14 | shmaltz | I do |
| 01:27.16 | tzanger | trelane: yup |
| 01:27.20 | drastixnz | it comes up withRejected connect attempt from 210.54.x.x, request '70300@default' does not exist |
| 01:27.21 | jayden | "the guy who knows everything on #asterisk" tongiht (sorry, hit enter too soon) |
| 01:27.23 | mrproper_ | shmaltz, ok whats your main number? |
| 01:27.27 | rtomsonII | Hello all. |
| 01:27.30 | doughecka | tzanger: I thought of so many things in response to that :) |
| 01:27.33 | mrproper_ | shmaltz, your main extension |
| 01:27.52 | doughecka | tzanger: are you... |
| 01:27.58 | tzanger | drastixnz: well that just means that your [user] section a) isn't specifying a context= (NEVER use default) or b) 703000 doesn't exist in the [default] context, or a context include='d from it |
| 01:28.03 | tzanger | doughecka: heh |
| 01:28.06 | shmaltz | I do 101 for dialing and the sip accounts are 1011 thru 1016 |
| 01:28.30 | shmaltz | http://www.pastebin.com/244756 |
| 01:28.30 | rtomsonII | Anyone familiar with running Asterisk and Faxing (Hylafax) off the same PRI with DID for both? |
| 01:28.34 | mrproper_ | shmaltz, so you want exten 101 to call all phones 1011 through to 1016? |
| 01:29.00 | shmaltz | nope |
| 01:29.48 | mrproper_ | shmaltz, so what extension do you want to call 1011 through to 1016? |
| 01:29.51 | Defraz | I have one line with a question on it can I paste it here or should I use pastebin.com? |
| 01:30.03 | tzanger | one line goes here |
| 01:30.04 | shmaltz | I want extension 160 to call all extensions 101 thru 110, and each of those (101-110) when dialed directly usualy call 1XX1 thru 1XX10 if the prevoius one is busy |
| 01:30.24 | Defraz | exten => _1208NXXXXXX,1,Dial(SIP/65.101.69.113/2${EXTEN}:1) |
| 01:30.34 | Defraz | will that hack the 1 off the front and add a two? |
| 01:30.41 | tzanger | no |
| 01:30.46 | tzanger | 2${EXTEN:1} |
| 01:30.48 | mrproper_ | shmaltz, so how are your extensions configured in sip.conf 101,102 etc or 1001 etc? |
| 01:30.52 | tzanger | Defraz: and why didn't you just try it? |
| 01:31.21 | Defraz | oh I thought it was wrong cuz it didn't work |
| 01:31.30 | shmaltz | mrproper_ |
| 01:31.31 | shmaltz | for 101, 1011 thru 1016 |
| 01:31.33 | shmaltz | for 102, 1021 thru 1026 |
| 01:31.35 | shmaltz | and so on |
| 01:31.49 | tzanger | Defraz: that's fine, but why didn't you tell us what it was doing instead? :-) |
| 01:32.02 | ScythelX | anyone know of a good datacenter colo in the US that provides t1 access - looking to setup my asterisk box for my small office |
| 01:32.06 | mrproper_ | shmaltz, ok so 101 is a number you have made up for shortning purposes |
| 01:32.29 | shmaltz | mrproper_, and for an easy way to dial 6 sip devices |
| 01:33.11 | ScythelX | i only need roughly 15 lines so even a fraction t1 would be sufficent |
| 01:34.11 | tzanger | I saw spiderman 2 yesterdya with the kids |
| 01:34.14 | *** join/#asterisk BBRodriguez (~alex@pD95631BB.dip.t-dialin.net) |
| 01:34.17 | tzanger | that elevator scene is so fucking hilairous |
| 01:34.27 | jayden | rtomsonII, we use spandsp.... |
| 01:34.27 | mrproper_ | shmaltz, http://www.pastebin.com/244758 |
| 01:34.37 | tzanger | they didn't wreck it either, it was done very very well |
| 01:34.44 | doughecka | tzanger: and whenever I see spandsp, I see spandex |
| 01:34.49 | tzanger | haha |
| 01:35.01 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) |
| 01:35.08 | tzanger | I see spand sp -- for the longest time I was trying to figure out wtf he was refering to |
| 01:35.11 | shmaltz | mrproper_, this is no good since that wil make all my line appearances show an incming call |
| 01:35.15 | doughecka | haha |
| 01:35.18 | rtomsonII | jayden: Forgive me but what is that? |
| 01:35.19 | tzanger | now I actually prounounce it span dsp and it makes sense |
| 01:35.41 | tzanger | not spanned espee |
| 01:35.50 | jayden | what kind of implementaiton are you looking for... |
| 01:35.51 | yashax | Good night guys.... THANK YOU SO MUCH for all the help today everyone... |
| 01:36.07 | doughecka | ahah |
| 01:36.12 | *** join/#asterisk hellop (~hellop@cpe-70-93-41-67.hawaii.rr.com) |
| 01:36.13 | BrianR___ | Looking forward to using spandsp with a few hundred did's to provide a dedicated fax number for every employee. |
| 01:36.23 | rtomsonII | I need to run my phones and inbound fax routing off one pri. I just need to know if it will work and what hardware to get. |
| 01:36.26 | jayden | http://www.voip-info.org/tiki-index.php?page=Asterisk%20spandsp |
| 01:36.27 | mrproper_ | shmaltz, then setup multiple queues: call queue 1 which has 1011, 1021, 1031 etc then calll queue2 which has 1012,1022,1032? |
| 01:36.29 | Defraz | what does this mean Feb 21 18:35:47 WARNING[25747]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x4240049c (len 434) to 65.101.69.113 returned -1: Invalid argument |
| 01:37.10 | jpablo | hi, has anyone connected asterisk cisco 5350 |
| 01:37.19 | rtomsonII | brianR: what hardware are you using for that? |
| 01:37.33 | BrianR___ | rtomsonII: Not sure. Hasn't been bought yet. |
| 01:37.46 | shmaltz | mrproper_, and how will it go from queue one to queue 2? I want it to do it only to device 1011-1016 if the low numbered one is busy, but for 1021-1026 it should ring on 1021 if it's not busy |
| 01:38.01 | BrianR___ | rtomsonII: But unless spandsp really sucks, I can't see it having trouble doing fax for a handful of PRI channels... |
| 01:38.22 | rtomsonII | Thanks: This might just save me some money. |
| 01:38.22 | drastixnz | tzanger: Once again thanks for the help, at Point "B" i have in the iaz.conf this [Whangarei] |
| 01:38.22 | drastixnz | type=peer |
| 01:38.22 | drastixnz | secret=paswyas1 |
| 01:38.23 | drastixnz | host=dynamic |
| 01:38.23 | drastixnz | context= |
| 01:38.25 | drastixnz | defaultip=218.101.x.x what is wrong with this? |
| 01:38.32 | BrianR___ | Sorry folks. |
| 01:38.48 | BrianR___ | On any reasonable PC. |
| 01:38.58 | tzanger | drastixnz: change context= to context=incoming or something |
| 01:39.09 | BrianR___ | rtomsonII: Prolly setting it up this week. Will let you know how it goes. |
| 01:39.59 | doughecka | crap |
| 01:40.00 | drastixnz | tzanger: Whangarei being point "A" do i have to incudle anything in that IAX.CONF? |
| 01:40.19 | tzanger | drastixnz: huh? |
| 01:41.12 | drastixnz | tzanger: I have point "incoming" in and at the other end it still comes up with request '703@default' does not exist |
| 01:41.27 | tzanger | drastixnz: have you reloaded the box? |
| 01:41.36 | drastixnz | both? |
| 01:41.38 | drastixnz | yes i have |
| 01:41.40 | tzanger | hmm |
| 01:41.45 | tzanger | oh |
| 01:41.54 | tzanger | do you have a context=default on the other side's [peer] config? |
| 01:42.05 | drastixnz | hummm bbs |
| 01:42.11 | tzanger | remember when you call from A to B, B sees A as a "user" and A sees B as a "peer" |
| 01:43.09 | rtomsonII | brianr___: Thanks again. If I get anything working I will let you know. |
| 01:43.19 | VoIPMasta | how do you instruct Festival to use a different voice? |
| 01:43.51 | *** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net) |
| 01:44.12 | drastixnz | tzanger: I found that there was a guest account, that had context=default, i have removed that and now i am getting "Call rejected by 218.101.54.x: No authority found" |
| 01:44.25 | tzanger | drastixnz: because you are not providing a user@ |
| 01:44.33 | drastixnz | tzanger: so so so so so close |
| 01:44.34 | tzanger | IAX2/user@peer/exten |
| 01:44.50 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 01:44.51 | tzanger | then in your iax.conf |
| 01:44.52 | tzanger | have |
| 01:44.53 | tzanger | [peer] |
| 01:44.55 | tzanger | type=peer |
| 01:44.59 | tzanger | secret=somesecret |
| 01:45.04 | tzanger | and on the other side |
| 01:45.11 | tzanger | [blah] |
| 01:45.21 | tzanger | username=user_given_in_the_incomign_dial |
| 01:45.24 | tzanger | secret=somesecret |
| 01:45.31 | tzanger | context=where_to_dump_this_call |
| 01:45.53 | tzanger | besmirch |
| 01:45.58 | tzanger | there's a word that should be used more often |
| 01:46.10 | tzanger | it behooves us to utilize the word 'besmirch' more often |
| 01:46.26 | jayden | tell your kids that |
| 01:46.36 | drastixnz | in Point "B" Extession.conf "exten => 400,1,Dial(IAX2/Whangarei:????????/Whangarei/703)" |
| 01:46.38 | tzanger | unwittingly exacerbating the wear on dictionaries across the land |
| 01:46.40 | shmaltz | anybody using the Local channel |
| 01:46.48 | tzanger | drastixnz: you don't need to provide :secret |
| 01:46.49 | jayden | yes |
| 01:47.05 | tzanger | if you provide secret= in B's iax.conf for [Whangerei] |
| 01:47.15 | shmaltz | jayden, u using the local channel? |
| 01:47.30 | marc32344 | whats a good billing soft? |
| 01:47.32 | VoIPMasta | Does anyone know how to select a different voice for use with the Fesival cmd? |
| 01:48.13 | jayden | no |
| 01:49.06 | tzanger | could a bug marshall PLEASE erase all the attachments from 2532 EXCEPT the last 3? |
| 01:49.54 | tzanger | http://toronto.cbc.ca/regional/servlet/View?filename=to-scout20050221 |
| 01:49.57 | tzanger | that is so fucking sick |
| 01:50.02 | jayden | ~M2532 |
| 01:50.18 | marc32344 | whats the cpu requirement of the te410 at full load? |
| 01:50.37 | tzanger | marc32344: define full load |
| 01:50.51 | marc32344 | full capacity on all 4 ports |
| 01:50.54 | tzanger | marc32344: basically it si recommended P4 minimum |
| 01:50.58 | tzanger | marc32344: again, define full load |
| 01:51.07 | tzanger | 96 channels doesn't mean much |
| 01:51.23 | jayden | of the card, not much, of the card making a bunch of to voip requireing transcoding... see tranzgers messages above |
| 01:51.25 | marc32344 | 96 simul calls |
| 01:51.31 | tzanger | transcoding? if so, to what? any IVRs or echo cancellation? Where are these calls going? SIP? IAX? other TDM? |
| 01:51.50 | tzanger | see what I'm getting at? :-) |
| 01:52.16 | jayden | 96 simul calls in and out the pri's you could do on a p3.... when you start adding other stuff... considering pc's are so cheap these days, you wil lwant more |
| 01:53.00 | jayden | :( no more goatmilk... and I was just thinking how much I wanted some goatmilk |
| 01:53.17 | Sedorox | Can switch statements only be used with IAX.. or can they be used with SIP? |
| 01:53.37 | pfn | only with iax |
| 01:53.45 | Sedorox | damn |
| 01:53.47 | Sedorox | ok |
| 01:55.15 | tzanger | jayden: yeah but remember that echo cancellation takes its toll too |
| 01:55.34 | jayden | echo........ |
| 01:55.37 | tzanger | also if the calls are all terminating to IVR or are being bridged |
| 01:55.47 | tzanger | and if IVR, are the prompts in ulaw or some other codec |
| 01:56.05 | jayden | y, like I said, the card does not require a lot of cpu... But :) |
| 01:58.04 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 01:58.20 | doughecka | hmms |
| 01:59.32 | BrianR___ | Can one preconvert the asterisk prompts to reduce cpu load if the connected phone's codec is not gsm? |
| 02:00.26 | jayden | BrianR, native MOH can. |
| 02:00.37 | jayden | what "prompts" are you talking about |
| 02:00.48 | BrianR___ | jayden: ivr prompts |
| 02:01.23 | BrianR___ | Like the stuff in /usr/share/asterisk/sounds |
| 02:01.25 | *** join/#asterisk OzJames79 (~James@203.208.64.29) |
| 02:01.38 | doughecka | huh, its cheaper to call russia through nufone than to call the US48 |
| 02:01.57 | jpablo | hi, why is asterisk don't showing anything in sip debug? |
| 02:02.03 | jpablo | i can see the invites with tcpdump |
| 02:02.07 | jpablo | but it is ignoring them. |
| 02:02.11 | doughecka | firewall |
| 02:02.11 | jayden | chekck logger.conf |
| 02:02.14 | doughecka | port is set wrong |
| 02:02.23 | doughecka | solar flares |
| 02:02.56 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-22-254.d4.club-internet.fr) |
| 02:03.10 | jpablo | i can see them with tcpdump in the machine i see them .. |
| 02:03.25 | jayden | what is in your logger.conf |
| 02:03.57 | tzanger | doughecka: yup |
| 02:03.58 | jpablo | the default. |
| 02:04.01 | tzanger | just how the things terminate |
| 02:04.24 | *** join/#asterisk carsim (~carsim@203.160.176.202) |
| 02:04.27 | doughecka | tzanger: jerjer replied! |
| 02:04.32 | doughecka | again! |
| 02:04.32 | doughecka | :) |
| 02:04.41 | tzanger | doughecka: hahaha |
| 02:04.42 | doughecka | still nothing though... |
| 02:04.43 | tzanger | you sound surprised |
| 02:04.58 | doughecka | yea |
| 02:05.05 | doughecka | although he still hasnt actully given me the config |
| 02:05.21 | doughecka | maybe its because I only have $1 in my account |
| 02:05.42 | doughecka | thats almost an hours worth of talk time |
| 02:05.50 | jayden | jpablo.. console => notice,warning,error,debug |
| 02:07.28 | jayden | or, starting asterisk console w/ more v's |
| 02:07.35 | jayden | changes verbosity level |
| 02:07.37 | drastixnz | tzanger: you are a god!!!!! i got it i got it YES YES YES YES YES YES YES YES YES there was a line that was left out type=user thanks for your help! |
| 02:07.45 | doughecka | HAHAHAHAHAHAHAHA |
| 02:07.46 | doughecka | wOOO |
| 02:07.51 | doughecka | tzanger SCORES |
| 02:08.05 | tzanger | heh |
| 02:08.12 | doughecka | touch down for the asterisk DOTS!!! |
| 02:08.23 | tzanger | n oworries drastixnz just remember that if you accept calls you are a peer to those calling, whom you refer to as users. |
| 02:08.26 | jayden | I thought you didn't score after you had kids :) |
| 02:08.28 | tzanger | DOTS? |
| 02:08.59 | jayden | can't we all just be friends :) |
| 02:09.05 | doughecka | hmm |
| 02:09.09 | doughecka | find me something better |
| 02:09.44 | tzanger | the asterisk stars? |
| 02:10.06 | doughecka | ah |
| 02:10.13 | tzanger | the asterisk VOIPERS |
| 02:10.13 | tzanger | hahaha |
| 02:10.33 | jayden | logo : http://www.digium.com/images/asterisk_sticker.gif |
| 02:10.46 | tzanger | I don't like those |
| 02:11.07 | tzanger | I like the asterisk logo, with the chat bubble with the slightly jaggy edges |
| 02:11.08 | jayden | logo's? or stickers? |
| 02:11.10 | tzanger | very subtle and very cool |
| 02:11.55 | jayden | this one : http://store1.yimg.com/I/asteriskpbx_1815_17631 ? |
| 02:12.22 | doughecka | I have that shirt |
| 02:12.23 | doughecka | =D |
| 02:13.03 | jayden | o.. this one : http://www.asterisk.org/images/asterisk.gif |
| 02:15.59 | tzanger | eww |
| 02:16.08 | tzanger | yes that one jayden |
| 02:16.33 | tzanger | I'd like a hat with just the callout |
| 02:16.34 | tzanger | not the text |
| 02:16.49 | tzanger | black hat with the orange |
| 02:16.52 | hellop | hello |
| 02:16.52 | tzanger | I think it'd look sharp |
| 02:16.56 | jayden | hey, if you include multiple files in extensions.conf, and have multiple global sections with the same options, who wins? |
| 02:17.05 | tzanger | with orange around the edge of the beak |
| 02:17.12 | jayden | see, I think orange should be banneds |
| 02:17.15 | jayden | banned |
| 02:17.24 | hellop | So I setup the voicemail. I can go to advanced options to leave a message... or I can change to different folders to listen. But, how do I get the voice mail working like normal voicemail? |
| 02:17.27 | tzanger | I have a hat like that now.. the yellow's around the "front" of the beak" not on it |
| 02:17.43 | tzanger | hellop: huh? |
| 02:17.53 | pcm | hellop: use voicemail not voicemailmain app |
| 02:17.54 | hellop | Maybe I need Voicemail, instead of VoicemailMain? |
| 02:18.02 | hellop | in my extensions.conf? |
| 02:18.08 | jayden | echo... |
| 02:18.11 | pcm | hellop: i already said |
| 02:18.15 | pcm | that |
| 02:18.24 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 02:18.42 | iceyp | how easy would it be to add a signup gui to asterisk? |
| 02:19.07 | hellop | ok tks |
| 02:19.11 | jayden | that depends, how much time do you have on your hands, and do you have any skills |
| 02:19.38 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 02:19.59 | *** join/#asterisk MiXi^ (~mixi@pD9E592CC.dip.t-dialin.net) |
| 02:20.01 | hellop | you know like kung-fu skills, hacking skills, skateboarding skills.. girls like guys with skills |
| 02:20.02 | iceyp | errm no :P |
| 02:20.33 | iceyp | girls like guys that can give good licky |
| 02:20.49 | MrEntropy | isn't that a skill? |
| 02:21.01 | jayden | I suppose |
| 02:21.07 | iceyp | mmm, it is indeede |
| 02:21.37 | jayden | the skills I was refering to was more in the realm of db\web coding skills |
| 02:22.00 | *** join/#asterisk a1fa (a1fa@2001:618:400:ab67:0:0:0:200) |
| 02:22.05 | a1fa | hi |
| 02:22.14 | jayden | if you setup * with realtime, it should be fairly trivial to add some db entries from a web interface |
| 02:22.17 | a1fa | anybody got a sip phone that i can dial for ;) |
| 02:22.23 | a1fa | i need to test my sip phone |
| 02:22.27 | jayden | but what else do you want it to do |
| 02:22.32 | *** join/#asterisk didz_ (~omg@200.218.193.30) |
| 02:23.28 | hellop | hmm It didn't play my unavailable message... |
| 02:23.28 | jayden | if you want it to do kung-fu or skateboarding, that would be more difficult |
| 02:23.28 | DaLion | ok tyhen ill read the normal modem card and try to reboot maybe it will take the irq |
| 02:23.28 | a1fa | somebody give me your SIP IP, so i can call you and test my SIP |
| 02:23.42 | a1fa | s/your// ;) |
| 02:23.48 | hellop | Can anyone suggest a serarch term or something for a better Voicemail Howto, Docs, etc..? |
| 02:24.06 | hellop | I haven't found much... |
| 02:24.09 | *** join/#asterisk jsolares (~jsolares@200.12.44.18) |
| 02:24.10 | jayden | setup docs or user docs? |
| 02:24.27 | hellop | So.. 2nd part of the new docs? |
| 02:24.28 | *** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net) |
| 02:24.36 | a1fa | ok :) |
| 02:24.43 | hellop | the site was giving me 404 before... |
| 02:26.31 | a1fa | peeeopleeeeeeeeeeeeeeeeeeeeee |
| 02:26.52 | hellop | hello |
| 02:26.56 | a1fa | hey |
| 02:27.01 | jayden | try http://www.automated.it/guidetoasterisk.htm#_Toc49248768 |
| 02:27.07 | a1fa | do you have a sip phone i can call ? |
| 02:27.16 | drastixnz | tzanger: just one more, if this is ok iax2 softphone setup in iax.conf |
| 02:27.52 | jayden | ^^^ thats for vm hellop |
| 02:27.56 | hellop | my network is so intermittent all-the-sudden.. I have to resinstall my router |
| 02:28.07 | hellop | jayden, tks... Mozilla wheel spinning.... |
| 02:28.14 | hellop | and spinning |
| 02:28.49 | *** part/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 02:28.54 | hellop | jayden, can you get to that site? |
| 02:29.26 | DaLion | wcfxo: DAA mode is 'FCC' |
| 02:29.26 | DaLion | ;) |
| 02:29.26 | DaLion | finally |
| 02:29.26 | DaLion | was easy |
| 02:29.26 | tzanger | drastixnz: it's the same thing |
| 02:29.26 | jayden | y, was just there |
| 02:29.28 | drastixnz | thaks |
| 02:29.31 | tzanger | your iax.conf will have a [username] entry with type=user |
| 02:29.31 | a1fa | jayden |
| 02:29.48 | jayden | a1fa |
| 02:29.56 | *** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
| 02:30.06 | a1fa | sorry to bother you |
| 02:30.07 | jayden | SuperMan is dead. |
| 02:30.15 | jayden | bastard |
| 02:30.26 | a1fa | do you have a sip phone |
| 02:30.29 | SuperMMan | jayden: lol |
| 02:30.37 | jayden | several |
| 02:30.40 | hellop | jayden, I cannot get to that site. maybe its down? |
| 02:30.43 | ScythelX | anyone know of a good telco hotel to colo my PBX for my small bussiness looking for a t1 connection |
| 02:30.53 | jayden | not for me.. I was just there... |
| 02:30.56 | a1fa | can i have your ip |
| 02:31.05 | a1fa | i need to test this sip |
| 02:31.06 | jayden | I don't let sip in, sorry. |
| 02:31.09 | SuperMMan | Anyone here used the Prepaid app thats on the wikkie? if so have you tried using it for us/canada calling. once again if so do you get a core dump everytime trying to use it |
| 02:31.15 | jayden | i am a xenophobe. |
| 02:31.25 | a1fa | lol |
| 02:31.25 | jayden | o.. and my wife and kid are asleep |
| 02:31.27 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 02:31.30 | a1fa | ok |
| 02:31.31 | a1fa | sorry man |
| 02:31.38 | a1fa | anybody else? |
| 02:31.39 | jayden | IAX only in. |
| 02:31.44 | a1fa | dang |
| 02:31.55 | a1fa | how come iax, its not itef yet ;) |
| 02:31.57 | jayden | I don't need sip telemarketers and bill coleectors |
| 02:32.25 | jayden | trunking between the office and house, between offices, and such... |
| 02:32.41 | jayden | and works better for me behind NAT. |
| 02:32.57 | jayden | all my sip phones are internal |
| 02:33.02 | DaLion | lol |
| 02:33.08 | iceyp | how does one add mysql support for both (extensions / sip / iax2).conf, i'd also like to add mysql support for cdr, problem being, i run bsd so cant use any scripts from websites |
| 02:33.11 | DaLion | yeah i seen lall my phones to filters now |
| 02:33.20 | DaLion | and i got a telmartketer script instaled |
| 02:33.53 | a1fa | so i made a mistake of getting a sip phone |
| 02:33.59 | jayden | working on filtering my kids real dad, but alas, that would be wrong... |
| 02:34.02 | a1fa | i cant find no sip softphones forfree |
| 02:34.12 | jayden | x-lite |
| 02:34.21 | a1fa | hm |
| 02:34.27 | jayden | you didn't look very hard |
| 02:34.27 | a1fa | it needs regitration |
| 02:34.28 | ta[i]nted | a1fa what platform |
| 02:34.33 | a1fa | windows |
| 02:34.37 | ta[i]nted | use firefly |
| 02:34.44 | a1fa | ok |
| 02:34.47 | jayden | there are lots... |
| 02:34.55 | ta[i]nted | but use firefly |
| 02:35.02 | jpablo | his there anyway to specify the fromuser and fromdomain from the Dial command ? |
| 02:35.03 | a1fa | ok |
| 02:35.10 | iceyp | a1fa x-lite from xten.com i think |
| 02:35.11 | jpablo | without creating a sip.conf entry .. |
| 02:35.12 | iceyp | woops |
| 02:35.12 | a1fa | i dont want to register to a sip server |
| 02:35.14 | jayden | WWW.VOIP-INFO.ORG... if you havn't been there, why not |
| 02:35.21 | a1fa | i've been there man |
| 02:35.24 | a1fa | :) |
| 02:35.43 | jayden | www.asteriskdocs.org... if you havn't been there, why not. |
| 02:35.46 | a1fa | i just want to register to sip servers for no reason |
| 02:35.48 | jayden | :) |
| 02:36.00 | jayden | you just want to.... |
| 02:36.11 | ta[i]nted | use firefly |
| 02:36.11 | jayden | what are you trying to do ? |
| 02:36.15 | ta[i]nted | do not use x-lite |
| 02:36.21 | a1fa | ta[i]nted |
| 02:36.25 | a1fa | i am downloading it |
| 02:36.41 | ta[i]nted | don't use SIP either |
| 02:36.42 | drastixnz | tzanger: exten => _[4]X.,1,Dial(IAX2/test@Whangarei/${EXTEN:1}) |
| 02:36.42 | drastixnz | <PROTECTED> |
| 02:36.44 | ta[i]nted | use IAX |
| 02:36.47 | *** part/#asterisk didz_ (~omg@200.218.193.30) |
| 02:36.48 | jayden | I am a big fan of testcall myself. |
| 02:36.51 | tzanger | drastixnz: stop that |
| 02:36.58 | jayden | simple is good :) |
| 02:36.59 | tzanger | you don't need [] unless htere's more than one digit in there |
| 02:37.01 | ta[i]nted | a1fa unless u want NAT headaches |
| 02:37.20 | jayden | did somone say use IAX instead? |
| 02:37.25 | drastixnz | *blush* there is, that will cover all of the area i want |
| 02:37.33 | a1fa | ;) |
| 02:37.40 | a1fa | everybody keeps telling me IAX is better |
| 02:37.43 | tzanger | drastixnz: and second is there a reason you're using '.' |
| 02:37.43 | a1fa | but too late now |
| 02:37.46 | a1fa | i got a SIP phone |
| 02:37.56 | MrEntropy | SIP rules |
| 02:38.02 | jayden | sip is fine. |
| 02:38.03 | BrianR___ | Planning a dialplan for my company is going to be a hassle. There was historically no coordination between the offices so they have overlapping extensions and DID's.. :( |
| 02:38.22 | jsolares | a1fa: if you're going to have the asterisk and sip phones on the same network, then sip is great |
| 02:38.36 | jayden | I managed it by moving an office and "not being able too" port some did's |
| 02:38.42 | a1fa | ta[i]nted : this shit wants me to register |
| 02:39.03 | jayden | a1fa, what do you want to do? |
| 02:39.04 | a1fa | i just want to make a simple SIP phonecall ip2ip |
| 02:39.20 | ta[i]nted | a1fa then register your shit |
| 02:39.21 | jsolares | but for nat, iax does not beat it, it kicks it ass and then some, and iax also has trunking so you can save some bw if you have more than one call between 2 servers |
| 02:39.33 | jsolares | a1fa: x-ten lite is free and needs no registration. just settin gup |
| 02:39.55 | ta[i]nted | hes probably talking about registering to provider |
| 02:39.56 | jsolares | firefly is free and needs no registration, it might ask you if you want to register to their network, but you dont have to |
| 02:39.56 | a1fa | x-linte? |
| 02:39.56 | jayden | are you talking about registering to get the product, or sip registration |
| 02:40.05 | *** join/#asterisk MiXi^ (~mixi@pD9E592CC.dip.t-dialin.net) |
| 02:40.11 | a1fa | i dont want to use the sip registration |
| 02:40.14 | marc32344 | how long does it takes to setup asterisk to work with digium cards? |
| 02:40.18 | a1fa | pointless atm |
| 02:40.21 | jayden | what he said above :) |
| 02:40.36 | marc32344 | I have zero knowledge right now. |
| 02:40.42 | jsolares | how do you plan to call the other sip phone a1fa? |
| 02:40.43 | jayden | marc... |
| 02:40.48 | jayden | ummm, |
| 02:40.50 | a1fa | ip2ip |
| 02:40.58 | jayden | not long.... read, then play. |
| 02:41.02 | a1fa | jsolares : i dont see why not |
| 02:41.16 | jayden | www.voip-info.org, www.asteriskdocs.org |
| 02:41.22 | jsolares | well ip2ip is also sip -> asterisk -> sip, it tells both who they are and what their ip is and lets them talk on with each other |
| 02:41.29 | JunK-Y | debian:/usr/src# ./irqmiss2 |
| 02:41.29 | JunK-Y | Wildcard X100P Board 1 ST:(OK ) irq:( 0) bpv:( 0) crc4:( 0) ebit:( 0) fas:( 0) |
| 02:41.29 | JunK-Y | debian:/usr/src# |
| 02:41.31 | JunK-Y | oups |
| 02:41.39 | jsolares | a1fa: i havent seen an sip phone that lets you dial an ip address |
| 02:41.58 | a1fa | jsolares http://www.grandstream.com |
| 02:42.00 | jayden | a1fa, IM.... |
| 02:42.12 | jsolares | yeah, i have grandstream sip phones |
| 02:42.19 | hellop | Unknown host automated.it |
| 02:42.36 | marc32344 | how much mem is required for the te410 boards? |
| 02:43.05 | shmaltz | anybody seen this problem, or know the solution? |
| 02:43.07 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-February/090771.html |
| 02:44.00 | SuperMMan | jsolares: i dial ip 2 ip all the time with my GS |
| 02:44.37 | jayden | marc, as we said earlier.. the boards, next to nothing... depends onwhat you want to do with them |
| 02:44.40 | jsolares | ip 2 ip can mean many things, are you dialing the ip address of a phone? |
| 02:44.49 | a1fa | yes |
| 02:44.52 | a1fa | that will work |
| 02:45.07 | SuperMMan | jsolares: yes i am, all you do is dial it like 192.168.001.001 but the real ip address and away you go |
| 02:45.25 | *** join/#asterisk Ayano (~erik_leee@66.51.208.150) |
| 02:45.56 | jsolares | i guess i'll have to try it tomorrow |
| 02:46.13 | jsolares | and see if the avaya can as well |
| 02:46.13 | Ayano | on the ip 500 it comes with skinny, where can I get the provision files to make it sip? |
| 02:47.01 | Ayano | anyone? |
| 02:47.09 | ScythelX | anyone know of a good telco hotel to colo my PBX for my small bussiness looking for a t1 connection |
| 02:47.28 | jsolares | that makes no sense |
| 02:47.41 | jsolares | atleast not to me |
| 02:47.44 | Ayano | jsolares: me? |
| 02:47.55 | jsolares | no, the telco hotel dude |
| 02:48.07 | jsolares | well you dont make sense to me either hehehe |
| 02:48.14 | Ayano | oops |
| 02:48.15 | jsolares | as i have no ip 500 phone |
| 02:48.21 | ScythelX | looking to put my asterisk box at a place like telx |
| 02:48.30 | jsolares | ah |
| 02:48.41 | *** join/#asterisk a1fa (a1fa@syru144-032.resnet.syr.edu) |
| 02:48.43 | a1fa | ok |
| 02:48.44 | a1fa | it worked |
| 02:48.51 | a1fa | i just dialed myself.. and it was busy... |
| 02:48.52 | jsolares | good |
| 02:48.54 | Ayano | They come with cisco's skinny protocol. I need to upgrade it to sip. I can't find the files. |
| 02:49.00 | a1fa | it would be pointless not to be able to dial direct IP |
| 02:49.06 | *** part/#asterisk DaLion (~Miranda@HSE-QuebecCity-ppp3497400.sympatico.ca) |
| 02:49.11 | a1fa | voip would suck |
| 02:49.28 | jsolares | depends on one's point of view |
| 02:49.36 | jsolares | to me it's pointless to be able to dial direct ip |
| 02:49.40 | jayden | good thing Da Lion is gone... now Da dear and other woodland creatures can roam free without fear |
| 02:49.57 | a1fa | how do you plan to make a phone call to a company on a different sip registrar? |
| 02:49.59 | mikegrb | They so can. |
| 02:50.07 | SuperMMan | jsolares: saves a shit load of money to be able to dial ip 2 ip for my needs anyway |
| 02:50.17 | a1fa | exactly |
| 02:50.17 | mikegrb | a1fa: e164 |
| 02:50.26 | a1fa | ? |
| 02:50.41 | a1fa | mikegrb : do you have a sip phone? |
| 02:50.48 | mikegrb | I have several |
| 02:50.52 | a1fa | ip? |
| 02:51.02 | mikegrb | nat |
| 02:51.09 | jsolares | i have no way of having a phone with a public ip, public ip are too expensive here, so the only use i have out of voip is voip-pstn and since i have an asterisk box i have several sip phones that way |
| 02:51.12 | a1fa | can you do a passthroguh? |
| 02:51.21 | mikegrb | they connect to an asterisk box which talks iax to a box in colo |
| 02:51.34 | mikegrb | not worth the effort |
| 02:51.37 | a1fa | tru |
| 02:51.42 | SuperMMan | jsolares: put your asterisk system as a gateway share the ip address and away you go |
| 02:51.44 | mikegrb | just call one of the fwd test numbers |
| 02:52.18 | a1fa | how do you guys plan to call different people |
| 02:52.19 | a1fa | ? |
| 02:52.23 | jsolares | SuperMMan: if by share you mean letting ppl other than me dial out to pstn then no thanks, it's cheaper for me to dial nufone to the us than to call locally |
| 02:52.26 | mikegrb | pick up the phone and dial |
| 02:52.28 | jayden | who do you want to call? |
| 02:52.30 | jsolares | indeed |
| 02:52.37 | a1fa | on different sip registrars? |
| 02:52.41 | mikegrb | a1fa: yes |
| 02:52.45 | mikegrb | e164 |
| 02:52.46 | SuperMMan | lol no |
| 02:52.53 | mikegrb | add them to your extentions.conf |
| 02:52.55 | mikegrb | whatever |
| 02:53.11 | jayden | a1fa, do you know their #? |
| 02:53.21 | jsolares | SuperMMan : ah i already have it setup that way, and away i'm going hehehe |
| 02:53.22 | mikegrb | I have a friend in germany I dial 1302 and it dials his sipgate.de number |
| 02:53.24 | bjohnson | jayden: ghostbusters !! |
| 02:53.30 | jsolares | my brother is calling canada via nufone's out |
| 02:53.35 | a1fa | mikegrb : lol.. t;) |
| 02:53.47 | jsolares | i'm planning on getting another voip provider to test drive tho |
| 02:53.47 | a1fa | i think it is cheaper just to dial an IP |
| 02:53.49 | mikegrb | wtf is t;) |
| 02:53.54 | mikegrb | it isn't |
| 02:53.56 | a1fa | and eaier |
| 02:53.58 | mikegrb | it is the same |
| 02:53.58 | a1fa | a typo |
| 02:53.59 | mikegrb | free |
| 02:54.05 | mikegrb | no, it is so not easier |
| 02:54.08 | jsolares | what if your friend has dynamic ip? |
| 02:54.13 | mikegrb | it is much easier to type 1302 |
| 02:54.18 | a1fa | dyndns |
| 02:54.21 | a1fa | tru |
| 02:54.23 | SuperMMan | Anyone know who is the maker of asterisk-prepaid? |
| 02:54.26 | mikegrb | jsolares: then that would be stupid * 2 |
| 02:54.36 | bjohnson | registers? |
| 02:54.42 | jsolares | SuperMMan: god |
| 02:54.50 | a1fa | i see your point guys |
| 02:55.04 | mikegrb | bjohnson: :D |
| 02:55.08 | a1fa | and its good, but you still need ip to ip phone calls ;) |
| 02:55.12 | jayden | a1fa, http://www.voip-info.org/tiki-index.php?page=E164.org, dundi.org |
| 02:55.15 | SuperMMan | jsolares: well then god needs pointers in programming |
| 02:55.33 | bjohnson | you don't need ip to ip calls |
| 02:55.49 | a1fa | sometimes you do |
| 02:55.56 | bjohnson | get people to sign up to iaxtel or fwd or sipphone |
| 02:55.57 | a1fa | for example.. i can call your ass right now |
| 02:56.00 | jsolares | if by ip you mean the protocol and not ipaddress then yes you do |
| 02:56.05 | bjohnson | then use those services as register |
| 02:56.08 | bjohnson | then use those services as registers |
| 02:56.11 | a1fa | ;) |
| 02:56.15 | mikegrb | a1fa: I make lots of ip to ip calls without dialing ip addresses |
| 02:56.23 | a1fa | yea? |
| 02:56.25 | mikegrb | a1fa: you obviously don't understand how this stuff works |
| 02:56.32 | bjohnson | why would you bother? |
| 02:56.45 | bjohnson | I mean .. why would you bother calling an ip address? |
| 02:56.49 | mikegrb | yes for the 50,000th time |
| 02:56.55 | mikegrb | bjohnson: exactly |
| 02:56.58 | mikegrb | bjohnson: it's sooo stupid |
| 02:57.20 | a1fa | i do |
| 02:57.29 | marc32344 | how much mem does the digium cards need? |
| 02:57.30 | bjohnson | why? |
| 02:57.34 | a1fa | brb |
| 02:57.38 | bjohnson | marc32344: none |
| 02:57.46 | marc32344 | sure |
| 02:58.02 | jsolares | the card in itself needs none |
| 02:58.12 | bjohnson | marc32344: how much mem does a modem need |
| 02:58.13 | jayden | marc, I have answered you 3 times now, I'm sorry you didn't like my answer, |
| 02:58.22 | marc32344 | whats the system load at full load.... ie 96 simul calls |
| 02:58.25 | bjohnson | how about a serial port |
| 02:58.42 | jayden | DID YOU READ WHAT WAS RESPONDED TO YOU BEFORE |
| 02:58.50 | jsolares | i think he didnt |
| 02:59.13 | jayden | I sent you good links that would have led you to sizing info and everything... |
| 02:59.18 | bjohnson | 96 simul calls may be marc's full expected load .. but systems can handle much, mcuh mpre |
| 02:59.19 | bjohnson | more |
| 02:59.20 | jayden | RTFM |
| 02:59.34 | bjohnson | few pages on the wiki about sizing |
| 02:59.48 | jayden | y, funny, I think I mentioned that earlier..... |
| 02:59.49 | jayden | twice |
| 02:59.52 | bjohnson | I rmember someone's running * on a P100 with 16M RAM |
| 03:00.08 | bjohnson | guess where I heard about that? |
| 03:00.11 | syslod | Any * QOS experts? |
| 03:00.26 | bjohnson | syslod: every one of my users |
| 03:00.51 | syslod | :). I've having a time getting VOIPJET to work. |
| 03:01.17 | syslod | QOS seems to be working with latency but it still skips and chirps. |
| 03:01.24 | jayden | ok.. I definately need a cigarette now... marc, while I'm gone, look at www.voip-info.org... |
| 03:01.44 | jsolares | hehehe |
| 03:02.01 | syslod | Could it be VOIPJET? |
| 03:02.23 | jayden | sorry to snap, but fuck, ask a question 3 times in a row, then ignore quesions....if you want peoples help, you first need to help yourself |
| 03:02.30 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 03:02.58 | file | jayden: welcome to #asterisk. |
| 03:03.06 | Defraz | Has anyone gotten a NexTone and Asterisk server talking? |
| 03:03.13 | file | yes. |
| 03:03.25 | bjohnson | syslod: could be |
| 03:03.35 | file | I just can't run, it's killing me! |
| 03:03.37 | bjohnson | syslod: but unlikely |
| 03:03.37 | Defraz | you mind letting me see a snippet of your extension.conf and sip conf |
| 03:03.38 | file | and taking control... |
| 03:03.55 | shido6 | pastebin.ca |
| 03:04.06 | file | Defraz: it's not like it's rocket science... just another SIP device, and I didn't do it like that |
| 03:04.15 | bjohnson | Defraz: sorry, don't know what a NexTone is |
| 03:04.39 | syslod | bjohnson: Is there a troubleshooting method other than whats on WIKI? |
| 03:04.55 | Defraz | Well, I was trying to figure it out before I asked. |
| 03:05.04 | Defraz | I can't seem to get it to talk right. |
| 03:05.08 | *** join/#asterisk juice (~juice@mo-65-41-197-194.dyn.sprint-hsd.net) |
| 03:05.09 | *** join/#asterisk voiper (~none@pcp09278118pcs.eatntn01.nj.comcast.net) |
| 03:05.10 | *** join/#asterisk pointer-gaim (~pointer@router.cathey.us) |
| 03:05.10 | Defraz | I get like 3 seconds out. |
| 03:05.42 | bjohnson | syslod: I do iax2 show registry to see if I'm registered. Then I try iax2 show peers to see if I'm finding them properly |
| 03:06.11 | voiper | Hi |
| 03:06.28 | voiper | Is there any way that I could increase the transmit volume on SIP channels |
| 03:06.50 | bjohnson | look for gain settings on the sip device |
| 03:07.41 | voiper | i am connecting from asterisk server to one of the sip providers |
| 03:07.53 | Defraz | how did you do it File? |
| 03:08.23 | file | well if you gave a specific reference to your problem, I may be able to help but I can't say exactly how I did it |
| 03:08.46 | Defraz | haha got a NexTone Iserver and asterisk to talk. |
| 03:08.57 | bjohnson | file: <- doesn't know. just randomly punched letters untilit worked |
| 03:09.02 | ariel_ | good evening folks |
| 03:09.04 | Defraz | I see I see. |
| 03:09.06 | file | no it was quite easy... |
| 03:09.20 | voiper | bjohnson, this is what i get while the call is on progress as TX/RX: 246c227f195 00102/00000 |
| 03:09.21 | file | used ip based matching for inbound from it, and used a simple peer entry for outbound |
| 03:10.30 | Defraz | so you registered it with the Iserver and then a type=peer for outbound. |
| 03:10.44 | file | no I didn't register it |
| 03:10.54 | file | it was specifically set. |
| 03:11.03 | Defraz | were? |
| 03:11.06 | Defraz | where? |
| 03:11.29 | file | how the heck do I know? I didn't set the darn Nextone up |
| 03:11.33 | file | you asked about asterisk :p |
| 03:11.51 | Defraz | oh my |
| 03:11.54 | Defraz | haha |
| 03:12.02 | Jayden | ~newbie |
| 03:12.03 | jbot | rumour has it, newbie is someone who is new to linux or debian, and should read the docs (/usr/share/doc/) |
| 03:12.04 | file | the asterisk side is not rocket science |
| 03:12.18 | file | just standard SIP. |
| 03:12.18 | Defraz | can I take a look at what you used in your sip.conf |
| 03:12.21 | marc32344 | how many # can be run over a t1? |
| 03:12.28 | file | it's just standard SIP, I can whip up an example in 10 seconds |
| 03:12.34 | mikegrb | marc32344: unlimited |
| 03:12.37 | file | lemme do that. |
| 03:12.41 | Defraz | jujust the optionst |
| 03:12.43 | dsmouse | IHNJ, IJLS "linux or debian" |
| 03:12.54 | Jayden | jbot: ast-newbiw is somone who needs to read www.voip-info.org and asteriskdocs.org |
| 03:12.55 | jbot | Jayden: okay |
| 03:12.57 | Defraz | Okay thanks a bunch |
| 03:12.59 | paulc | LOL.. "asterisk" and "standard SIP" and "user/peer matching in sip.conf".. LOL.. |
| 03:13.03 | marc32344 | mikegrb-- before running into busy? |
| 03:13.10 | mikegrb | marc32344: 23 |
| 03:13.23 | ariel_ | marc32344, t1 24 voice channels pri 23 |
| 03:13.40 | marc32344 | overselling??? |
| 03:13.42 | file | Defraz: http://pastebin.ca/6271 |
| 03:13.57 | dsmouse | ~rtfw |
| 03:13.58 | jbot | rtfw is probably Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 03:14.10 | Jayden | :) |
| 03:14.15 | Jayden | nice |
| 03:14.17 | file | paulc: what should be today's magical advice? |
| 03:14.21 | Jayden | that is what i was looking for |
| 03:14.22 | Defraz | lets give her a try. Thanks file. |
| 03:14.30 | ScythelX | anyone know a good colocation center in the us that offers TDM |
| 03:14.34 | marc32344 | 150-200 is my target |
| 03:14.58 | Jayden | 150-200 what is your target? |
| 03:15.15 | marc32344 | 150-200 # over a t1 |
| 03:15.22 | ta[i]nted | sure |
| 03:15.23 | ariel_ | ScythelX, I use www.race.com |
| 03:15.31 | mikegrb | you can have as many numbers as you want |
| 03:15.32 | paulc | file: don't run with scissors? no, wait, that's MS Word's tip of the day.. |
| 03:15.38 | mikegrb | but only 24 calls at the same time |
| 03:15.39 | file | You can get 150-200 DIDs to a single PRI, but you can only have 23 concurrent channels |
| 03:15.39 | ta[i]nted | if only 23 are being used at any given time |
| 03:15.52 | file | well, depends... channelized T1 is 24... PRI is 23 + data... |
| 03:16.22 | paulc | spot the brit - he's the one that can't spell ;-) |
| 03:16.28 | JunK-Y | PRI rocks! :) |
| 03:16.29 | file | hehe |
| 03:16.33 | Jayden | hehe |
| 03:16.42 | ScythelX | ariel_: would you mind telling me how much you pay - i dont need more then a t1 its just my small office - we are switching to * slowly by surely |
| 03:17.00 | marc32344 | erlang link??? |
| 03:17.21 | ariel_ | ScythelX, what I pay is different then what you would pay. I do part time work for them. |
| 03:17.25 | Jayden | local voice PRI in the US from $500-$1000/mo, depending on what bell you have |
| 03:17.33 | paulc | hehe.. ariel_ gets "mates rates" :-) |
| 03:17.42 | ta[i]nted | why would u want PRI when T1 is cheaper |
| 03:18.01 | syslod | Caller name maybe? |
| 03:18.08 | ariel_ | Pri you can set your own CallerID and get better information form it |
| 03:18.12 | syslod | CLIP, CLIR. QSIG> |
| 03:18.21 | file | PRI is just all around nicer |
| 03:18.22 | paulc | COLP |
| 03:18.29 | paulc | JOSHUA COLP? ;-) |
| 03:18.33 | file | yesssssssssss |
| 03:18.42 | file | PAUL |
| 03:18.44 | file | PAUL CRICK? |
| 03:18.54 | paulc | COLP = Connected Line Presentation.. |
| 03:18.57 | BrianR___ | file: As best I can tell, there's no upper limit on the number of DID's for a pri. |
| 03:18.58 | paulc | PAUL = uh.. yeah.. |
| 03:19.00 | file | ;) |
| 03:19.05 | file | BrianR___: I said that. |
| 03:19.12 | ScythelX | so as I understand basically all around telco colocation for TDM access is gonna be around 500-1000 a month plus cents per min? |
| 03:19.21 | file | well |
| 03:19.26 | file | I said it in the amount he wanted... |
| 03:19.37 | DJ-Pyro | ScythelX: what kind of line? T1 or T3? |
| 03:19.38 | file | as in a, 'yes this can be done but you can only have 23 calls up at a time' |
| 03:19.39 | Jayden | not colo... that would be for a pri |
| 03:19.53 | DJ-Pyro | we're getting a T3 in at $1200/mo + $293/mo cross connect from TWTC to GC |
| 03:19.54 | ScythelX | DJ-Pyro: I only need 23 lines or less - not 672 |
| 03:19.56 | Jayden | I do not know colo cost |
| 03:20.03 | syslod | ScythelX: What rate centers you looking for? |
| 03:20.08 | BrianR___ | PLanning to get another 200 or so DID's added at my office to give everyone perosnal fax. |
| 03:20.37 | ScythelX | basically its too expensive to get a t1 at my location so im looking into this option |
| 03:20.39 | bjohnson | marc32344: do you mean 150-200 phone numbers (ie DIDs) or 150-200 concurrent calls .. cause their 2 different things |
| 03:20.44 | Jayden | file is dacing, somone turn up the bas |
| 03:20.46 | Jayden | base |
| 03:20.54 | ta[i]nted | DJ-Pyro where are you located? |
| 03:21.00 | DJ-Pyro | ta[i]nted: milwaukee |
| 03:21.00 | marc32344 | bjohnson--phone numbers... |
| 03:21.03 | paulc | or the bass even |
| 03:21.05 | ScythelX | syslod: I looked at a company called www.telx.com I have a feeling there $$$ |
| 03:21.10 | paulc | Go File Go! DDR! |
| 03:21.14 | Jayden | y, that too |
| 03:21.23 | file | paulc: I can't till my new workstation comes ;( |
| 03:21.23 | ta[i]nted | what features are good in a colo? |
| 03:21.29 | file | and Dell keeps delaying it |
| 03:21.30 | ta[i]nted | for hosting asterisk |
| 03:21.32 | DJ-Pyro | ta[i]nted: of course I should add that TWTC is delivering over the SONET network that they're extending into our building for 100/1000mbit network |
| 03:21.41 | Jayden | jbot: mbtt is Mavis Beacons Teaches Typing |
| 03:21.43 | jbot | Jayden: okay |
| 03:21.43 | marc32344 | how many did / t1 is possible.... before busy... |
| 03:21.52 | syslod | SONET 1000mbit? |
| 03:21.55 | ta[i]nted | marc32344 how much money do u have |
| 03:21.56 | mikegrb | marc32344: unlimited did / 21 |
| 03:21.59 | mikegrb | er |
| 03:22.00 | mikegrb | per t1 |
| 03:22.15 | bjohnson | marc32344: they are unrelated |
| 03:22.16 | DJ-Pyro | syslod: they're running metro ethernet over their network |
| 03:22.17 | mikegrb | marc32344: don't ask any more questions, you don't listen to people's answers |
| 03:22.23 | syslod | DWDM? |
| 03:22.25 | *** join/#asterisk telme (~teliax@c-67-166-37-218.client.comcast.net) |
| 03:22.35 | algorithmn | dense wave divison multiplexing |
| 03:22.38 | algorithmn | fddi |
| 03:22.40 | Jayden | 21? 24 channels on T1, 23 on PRI |
| 03:23.03 | marc32344 | unlimited did/t1 makes no sense.... busy signal maybe??? |
| 03:23.06 | DJ-Pyro | syslod: I don't know the specifics of their network, we're handed off a standard ethernet connection and a termination for a T3 |
| 03:23.09 | syslod | me looks at all the colors. |
| 03:23.11 | bjohnson | typo .. I'm sure he meant 23 |
| 03:23.17 | file | a did and a channel are two different things |
| 03:23.19 | mikegrb | Jayden: yes, it was type, I meant t1 |
| 03:23.24 | bjohnson | marc32344: listen |
| 03:23.27 | mikegrb | unlimited did's per t1 |
| 03:23.27 | Defraz | okay file that worked well it connected to the NexTone so I guess I am confused on the entry in the extensions.conf. what entry do you have in there. |
| 03:23.28 | ScythelX | marc32344: a DID is just a # 555-5555 the amount of channels you have is 24 with 1 used for caller id info |
| 03:23.33 | syslod | Yea. Its likely a SONET mux with 1000mb interface. |
| 03:23.37 | file | Defraz: depends what you want to do... call out? |
| 03:23.42 | Jayden | you can have as many #'s as you want pointing at a PRI, but only 23 calls at a time. |
| 03:23.52 | file | hey wait a second, we had this conversation about DIDs/channels a few days ago |
| 03:24.01 | bjohnson | and a few seconds ago |
| 03:24.02 | DJ-Pyro | syslod: we have a conference call tomorrow to finalize and contract is signed thursday |
| 03:24.03 | marc32344 | yes, but realistically how many did will you allocate for a single t1? |
| 03:24.07 | syslod | NFAS will get you past 23. |
| 03:24.14 | mikegrb | and a few minutes ago |
| 03:24.17 | file | Defraz: exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@nextone) |
| 03:24.21 | Defraz | File: yes call out. |
| 03:24.29 | Jayden | marc, it depends on how many calls you get |
| 03:24.34 | syslod | DJ-Pyro: We offer a similar service. |
| 03:24.35 | Jayden | at a time |
| 03:24.37 | *** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net) |
| 03:25.37 | bjohnson | marc32344: AS MANY AS YOU WANT |
| 03:25.48 | Jayden | o, and btw, contrary to popular beleif, asterisk does run on windows |
| 03:25.54 | bjohnson | ewwwwwww |
| 03:25.59 | Jayden | :) |
| 03:26.04 | file | yeah, it's on the Digium webpage... |
| 03:26.05 | marc32344 | my question is not getting answered. |
| 03:26.11 | Jayden | no, real windows... |
| 03:26.20 | PatrickDK | marc, don't ask the wrong question than |
| 03:26.23 | bjohnson | try asking it again |
| 03:26.23 | ScythelX | well all im looking for is a place where i can host my asterisk box with a TDM and inet connection to my local office / if anyone could point me to a decent price I would greatly appreciate it |
| 03:26.23 | Jayden | not linux virtual machine |
| 03:26.27 | bjohnson | hehe |
| 03:26.48 | Jayden | marc, your question is getting answered. |
| 03:26.55 | bjohnson | repeatedly |
| 03:27.05 | file | by different people at that |
| 03:27.13 | ScythelX | marc32344: you messaged me 30 min ago and I explained it to you |
| 03:27.49 | marc32344 | there is a formula for this.... but I have forgot.... |
| 03:27.50 | bjohnson | and even once in caps |
| 03:28.13 | Beirdo | get out some pencil crayons' |
| 03:28.20 | Jayden | the formula just says, how many calls do you get at the same time at peak, and that is your answer |
| 03:28.36 | Jayden | peak calls / 23 and round up. |
| 03:28.44 | bjohnson | and you can have 23 concurrent calls on a PRI |
| 03:28.59 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
| 03:29.02 | Jayden | marc, lets try this a diff way |
| 03:29.09 | Jayden | what are you trying to do |
| 03:29.17 | ScythelX | marc32344 wants to squeeze 150-200 calls incoming and outgoing into a single t1 |
| 03:29.24 | _daver_ | we have over 80 extensions at my work, and rarely are more than 5 channels actually active simultaneously. very low usage. |
| 03:29.46 | *** join/#asterisk juice (~juice@mo-69-68-106-7.dyn.sprint-hsd.net) |
| 03:29.47 | marc32344 | no. there is formula that calculates the number of concurrent calls based on an acceptable busy rate |
| 03:29.55 | BrianR___ | _daver_: My office is the same way... |
| 03:29.55 | Jayden | y, I know. |
| 03:29.58 | bjohnson | and we have 9 extensions and at least once a day 3 channels re in use |
| 03:30.00 | Jayden | so back to my question |
| 03:30.05 | Jayden | what are you trying to do |
| 03:30.33 | bjohnson | marc32344: tell that to 911 |
| 03:30.44 | bjohnson | or my pizza place |
| 03:30.52 | bjohnson | it all depends on NEED!!! |
| 03:31.00 | Jayden | at our call center we have 40 agents making easily using 4 full pri's outbound... |
| 03:31.02 | marc32344 | if you are going to run 24 did/t1 GOODLUCK!!!! |
| 03:31.27 | Defraz | what the heck does this mean, does that mean codec? app_dial.c:1007 dial_exec: Had to drop call because I couldn't make SIP/2203-af82 compatible with SIP/65.101.69.113-2e6c |
| 03:31.29 | Jayden | ok, so, once more... what are you trying to do |
| 03:31.45 | tzanger | bjohnson: unless he's planning on supplying you with pizza or emergency assistance the forumla applies :-) |
| 03:31.45 | file | Defraz: codec incompatibilities |
| 03:31.50 | Defraz | bummer |
| 03:31.57 | Defraz | they told me g729 so that is all i allow |
| 03:32.15 | ScythelX | marc32344: I thought you wanted to sell termination like nufone |
| 03:32.18 | file | well if you're transcoding, then you gotta buy a license |
| 03:32.33 | ScythelX | marc32344: atleast thats what you explained ot me |
| 03:32.35 | bjohnson | tzanger: point was .. I don't think pizza joint needs as many lines available as 911 .. even it was the same number of employees. The needs are different |
| 03:32.41 | ScythelX | ot/to |
| 03:33.05 | Jayden | ~mbcc |
| 03:33.14 | Jayden | ~mbtt |
| 03:33.15 | jbot | rumour has it, mbtt is Mavis Beacons Teaches Typing |
| 03:33.15 | tzanger | bjohnson: and you are correct |
| 03:33.15 | marc32344 | i think 8did/channels should be ok. |
| 03:33.21 | Defraz | hmm so if my phone connects using ulaw then that would require a lic. |
| 03:33.42 | Defraz | but if my phone connects to the asterisk using 729 and the use 729 then we are good to go. |
| 03:33.48 | *** part/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 03:33.53 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 03:33.54 | Jayden | marc, do you know what you are trying to do? |
| 03:33.58 | file | Defraz: yes. |
| 03:34.03 | bjohnson | tzanger: so no formula can be derived .. since "acceptable" busy rate will is not a constant |
| 03:34.03 | Jayden | Junk-Y..... |
| 03:34.12 | JunK-Y | yes? |
| 03:34.22 | Nivex | Jayden: I think he's ignoring you. |
| 03:34.30 | Jayden | o.. I ment, JUNK-Y!!!!!!! |
| 03:34.35 | JunK-Y | im there! |
| 03:34.42 | JunK-Y | :) |
| 03:34.44 | Jayden | file.. your gunna hurt yourself |
| 03:34.48 | marc32344 | jayden-- thank you |
| 03:34.57 | file | helping in this channel brings far more pain |
| 03:35.21 | Defraz | haha |
| 03:35.23 | Defraz | Sorry File |
| 03:35.27 | verge | is there any problems running two sip devices from the same IP behind NAT? |
| 03:35.28 | Defraz | Just trying to understand it all' |
| 03:35.32 | Jayden | is that like when you look someone in the eye and say thank you, like in that commercial after you ask that fat lady when she is due? |
| 03:35.37 | file | Defraz: you're not bad, it's just your stuff was pretty standard |
| 03:35.46 | |Vulture| | Hey guys I have installed a few TDMs but I am looking at installing my first PRI and I have a few questions... |
| 03:35.52 | file | and covered in examples from here to Guatemala |
| 03:36.03 | tzanger | bjohnson: for most businesses the "acceptable" is almost universal |
| 03:36.04 | *** part/#asterisk Defraz (~t0tal@65.103.222.4) |
| 03:36.09 | Jayden | there are examples in Gautamala? nice |
| 03:36.11 | file | wow, I scared him away |
| 03:36.19 | file | Jayden: yeah I found a site being hosted there with examples |
| 03:36.19 | Jayden | ~Guatamala |
| 03:36.26 | |Vulture| | with a PRI you have 23 lines, then do you buy seperate DIDs for each line, or how does that work? |
| 03:36.46 | JunK-Y | vulture: like many extens in ur dialplan |
| 03:36.48 | Jayden | jbot: Guatamala is where the examples end |
| 03:36.49 | jbot | Jayden: okay |
| 03:36.53 | marc32344 | pri is the interface.... not the actual line!!! |
| 03:37.01 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
| 03:37.01 | |Vulture| | marc32344: T1 PRI |
| 03:37.04 | Jayden | tecnically |
| 03:37.06 | Jayden | y |
| 03:37.10 | file | |Vulture|: you buy it for your PRI, channel independent usually... so it comes in over a free channel |
| 03:37.15 | pdracevich | IIIIIIIIIIIIMMMMMMMMMMMMMMMMMM BACCCCKKK!!! |
| 03:37.23 | bjohnson | |Vulture|: yes. you typically buy them in bulk |
| 03:37.27 | JunK-Y | vulture: DID is like exten in ur dialplan |
| 03:37.30 | file | mmm channels |
| 03:37.39 | |Vulture| | JunK-Y: gotchya |
| 03:37.42 | tzanger | |Vulture|: no you just buy DIDs |
| 03:37.47 | tzanger | you can have 1000 DIDs for 23 lines |
| 03:37.48 | tzanger | doesn't matter |
| 03:37.53 | tzanger | you can only take 23 calls at once |
| 03:37.55 | file | tzanger stole the DID from the telco! |
| 03:37.57 | file | who me? |
| 03:37.57 | file | yes you! |
| 03:37.59 | file | not true! |
| 03:37.59 | JunK-Y | vulture: just create 1000 extens if ya want. |
| 03:38.01 | pdracevich | help "Host 210.54.249.228 failed MD5 authentication for 'user1' " I have a IAX softphone, and trying to connect to a asterisk server |
| 03:38.03 | |Vulture| | and can you control the hunt groups, or does the telco still have to do that? |
| 03:38.04 | bjohnson | couldn't b |
| 03:38.07 | implicit | tzanger: you did |
| 03:38.12 | bjohnson | then who? |
| 03:38.13 | tzanger | pdracevich: try auth=plain |
| 03:38.16 | tzanger | implicit: heh |
| 03:38.25 | pdracevich | thanks again |
| 03:38.30 | Jayden | ~Guatamala |
| 03:38.31 | jbot | somebody said guatamala was where the examples end |
| 03:38.40 | bjohnson | file stole the DID from the telco! |
| 03:38.47 | file | who me? |
| 03:38.52 | bjohnson | yeah you |
| 03:38.57 | file | couldn't be! |
| 03:38.59 | Jayden | not true? |
| 03:39.03 | bjohnson | then who? |
| 03:39.10 | file | Jayden stole the DID from the telco! |
| 03:39.16 | *** join/#asterisk Cresl1n (~matt@user-24-236-124-147.knology.net) |
| 03:39.19 | marc32344 | is $700/t1 good price? |
| 03:39.19 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
| 03:39.30 | Jayden | not me |
| 03:39.46 | Jayden | ~mbtt |
| 03:39.47 | jbot | somebody said mbtt was Mavis Beacons Teaches Typing |
| 03:39.50 | pdracevich | thanks it comes up now Registered 'user1' (AUTHENTICATED) at 210.54.249.x |
| 03:39.59 | tzanger | marc32344: sounds about right |
| 03:40.07 | tzanger | can't get much better unless you buy in bulk (i.e. DS3) |
| 03:40.09 | Jayden | marc, depends where, but yeah, that is about right |
| 03:40.16 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 03:40.39 | Jayden | o great.. we need a techie in here.. you answer the questions for a while |
| 03:40.45 | ta[i]nted | marc32344 what area are u in |
| 03:40.45 | Jayden | :) |
| 03:40.46 | *** join/#asterisk file (~file@mctn1-1987.nb.aliant.net) |
| 03:40.55 | ta[i]nted | i can get t1 for 500 |
| 03:41.10 | Jayden | tainted..not in atlanta... those bastards |
| 03:41.31 | ta[i]nted | have u guys colo-ed asterisk before? |
| 03:41.33 | Jayden | don't get me going about SWB today. |
| 03:41.37 | Jayden | no |
| 03:41.42 | ta[i]nted | i'm trying to figure out whether this ISP is good for colo |
| 03:41.51 | ta[i]nted | marc32344 that's USD |
| 03:41.57 | file | gotta be down because I want it all |
| 03:42.00 | ta[i]nted | marc32344 so in CDN should be around 24,000 |
| 03:42.03 | pdracevich | tzanger: still keeps on coming up with the MD5 thing |
| 03:42.07 | Jayden | :) |
| 03:42.18 | file | I should go to sleep |
| 03:42.20 | tzanger | hmm |
| 03:42.24 | tzanger | remove the auth entirely |
| 03:42.39 | Jayden | file... NO.. .please no |
| 03:42.48 | verge | I am trying to add a new extension... |
| 03:42.50 | Jayden | :) |
| 03:43.04 | Jayden | verge: go for it... |
| 03:43.56 | Jayden | hey, who here uses windows? |
| 03:44.21 | file | hrm someone was playing with my digital camera when I was eating my pasta salad |
| 03:44.23 | file | didn't even notice |
| 03:44.43 | Damin | Did they take naughty pictures with it? |
| 03:44.47 | Jayden | was it the pasta salad? |
| 03:44.57 | file | nope, but I'm in it |
| 03:44.58 | Jayden | maybe it was mad |
| 03:45.14 | verge | and it wasn't working |
| 03:45.18 | Jayden | the pasta salad or the naughty pictures? |
| 03:45.27 | file | lol |
| 03:45.40 | Jayden | verge: there is the rest of the thought |
| 03:45.47 | Jayden | why not? |
| 03:46.04 | pdracevich | i did nothing |
| 03:46.17 | verge | but now it is working fine... |
| 03:46.18 | Jayden | congrats |
| 03:46.23 | verge | so I am just a dumb ass... |
| 03:46.30 | file | http://www.file-radio.com/pics3/stolen.jpg |
| 03:46.37 | file | there's me eating my pasta salad on the side |
| 03:46.42 | Jayden | hey... you are a rocket scientist tongiht |
| 03:46.47 | file | I wonder who it was... |
| 03:47.18 | file | the pasta salad was delicious though |
| 03:47.49 | marc32344 | ok. |
| 03:47.53 | marc32344 | signing out |
| 03:48.15 | *** part/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 03:49.03 | Jayden | file, nice glass btw |
| 03:49.28 | Jayden | wtf |
| 03:49.30 | file | where? what? when |
| 03:49.35 | Jayden | pics |
| 03:49.48 | file | long ago, long long ago |
| 03:49.51 | file | it's plexiglass |
| 03:49.53 | Jayden | :) |
| 03:50.03 | Jayden | I thought.. and very nice it is |
| 03:50.14 | Jayden | marc left? |
| 03:50.20 | Jayden | damn.... |
| 03:50.31 | file | now we should talk amongst ourselves |
| 03:50.48 | Jayden | wow... I think people must think I am a total ass hole tonight... |
| 03:50.58 | file | you turn into one |
| 03:50.59 | Jayden | does anyone think I am an ass? |
| 03:51.05 | PatrickDK | your not? |
| 03:51.14 | Jayden | ~SER |
| 03:51.15 | jbot | hmm... ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 03:51.17 | paulc | Someone said you were a cunt.. but they were British, drunk, and joking |
| 03:51.23 | file | and they were named Paul |
| 03:51.24 | paulc | I'm British, sober, and NotVeryFunny(tm) |
| 03:51.33 | file | rrrrrrrright |
| 03:51.35 | paulc | hehe.. kidding.. I don't know Jayden from adam.. |
| 03:51.36 | paulc | but hi :-) |
| 03:51.44 | ScythelX | would anyone have an idea how much a place like http://www.telx.com/carriers.cfm would cost to host an asterisk box at for t1 and inet access |
| 03:52.04 | paulc | I didn't like the telx.com website - too flash heavy for their index page.. |
| 03:52.14 | Jayden | no.. do they have a phone number? |
| 03:52.16 | Jayden | :) |
| 03:52.27 | ScythelX | i emailed them waiting for a response |
| 03:52.32 | ScythelX | just trying to get an idea of what im looking at |
| 03:52.37 | Jayden | i read that at first as file tickles paulc.... |
| 03:52.47 | ScythelX | because where I live ot install a t1 is around 1500 a month |
| 03:52.47 | paulc | he does that too |
| 03:52.49 | ScythelX | whihc is insane |
| 03:52.49 | Jayden | sorry, no idea on colo |
| 03:52.49 | paulc | but not that often |
| 03:52.57 | Jayden | where do you live |
| 03:53.01 | ScythelX | in CT |
| 03:53.05 | Jayden | and voice or data\inet t1 |
| 03:53.06 | ScythelX | in a small town |
| 03:53.11 | ScythelX | voice t1 |
| 03:53.11 | Jayden | verizon? |
| 03:53.19 | Jayden | or smallbell |
| 03:53.21 | ScythelX | SBC i believe |
| 03:53.25 | Beirdo | can you hear me now? |
| 03:53.26 | Damin | SBC sucks.. |
| 03:53.28 | Damin | Ass.. |
| 03:53.34 | Jayden | there is another bunch of fuckers |
| 03:53.39 | techie | yeah but they bought AT&T |
| 03:53.46 | file | Damin: ya know you are so predictable when you come on IRC |
| 03:53.49 | Jayden | ok.. I just hate bell, but especially SWB this week |
| 03:53.53 | ScythelX | so I'm just gonna colo it - problem is finding a place |
| 03:53.59 | file | it always end up you saying that atleast two times, then mysteriously you stop responding |
| 03:54.35 | Jayden | I need to be good w/ SBC for a few more weeks cuz we are hotcutting 14 t1's in an hour on the 4th and I probably should not piss them off too much before then |
| 03:55.08 | Jayden | file: who stops responding? |
| 03:55.14 | file | Damin |
| 03:55.19 | ScythelX | www.telx.com is the only place I can find sort of near me |
| 03:55.33 | paulc | Does it have to be near you? |
| 03:55.49 | file | go paul go paul go paul |
| 03:56.21 | ScythelX | paulc: i guess not |
| 03:56.32 | Jayden | WAKE UP! |
| 03:57.04 | ScythelX | paulc: im just trying to get an idea on price at the moment, but im SURE it will be less |
| 03:57.07 | paulc | ScythelX: So I missed half of this.. are you looking for colo with TDM PRI, or was that Mark138743183? |
| 03:57.13 | Jayden | scyth, what do you want to do, data T or inet to colo place, then voice term it there? |
| 03:57.13 | file | gooooooooooooooooooodnight everyone |
| 03:57.22 | paulc | night file |
| 03:57.29 | Jayden | :( |
| 03:57.30 | ScythelX | I want to have TDM PRI with inet to stream to my office |
| 03:57.32 | Jayden | night |
| 03:57.40 | Jayden | got it... |
| 03:58.02 | Jayden | why not just use one of the voip providers then? |
| 03:58.03 | ScythelX | installed a T1 onsite isnt really an option |
| 03:58.22 | Jayden | broadvoice, nuphone, whoever... |
| 03:58.30 | Jayden | why do you need to colo your own box |
| 03:58.44 | ScythelX | well I will need a few dids |
| 03:58.50 | Jayden | ok |
| 03:58.51 | ScythelX | I thought overall if I did it myself it would be cheaper |
| 03:58.53 | Jayden | how many |
| 03:59.08 | Jayden | 10 or 400 |
| 03:59.14 | implicit | 500 |
| 03:59.29 | ScythelX | it would be less then 100 right now |
| 03:59.33 | Jayden | no, that wasn't one of the options :) |
| 03:59.36 | ScythelX | heh |
| 03:59.40 | Jayden | ummm |
| 04:00.03 | Jayden | hmmmmm |
| 04:00.20 | ScythelX | if I were to use an upstream provider I would want to host in their datacenter location |
| 04:00.20 | Jayden | I honestly don't know how those guys charge for stuff like that... |
| 04:00.33 | Jayden | host what |
| 04:00.48 | ScythelX | our pbx |
| 04:01.00 | ScythelX | for inet connection |
| 04:01.01 | mikegrb | Beirdo: http://thegrebs.com/~michael/mail/ha-ha-Dongs |
| 04:01.04 | Jayden | you can use them to provide dialtone and DID's over voip, then keep your pbx at the office |
| 04:02.09 | ScythelX | yes |
| 04:02.09 | Jayden | problem is, the way that most of them charge per # |
| 04:02.25 | Jayden | I wonder if you can work somthing out... how many minutes are you talking about? |
| 04:02.48 | paulc | That's always the tricky thing innit.. not everyone knows how many minutes they use for inbound stuff, just what they get charged for their outbound traffic |
| 04:02.50 | ScythelX | I would have to figure that out |
| 04:02.54 | Jayden | so voice pri's are 1500 but inet t1's are afordable? |
| 04:03.01 | _daver_ | Scyth: How are you going to connect to the data center? a t1? |
| 04:03.16 | ScythelX | _daver_: ahem... cable modem |
| 04:03.20 | Jayden | yeah, what he said :) |
| 04:03.34 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 04:03.44 | Jayden | cable modem with how many calls on it.. |
| 04:03.46 | _daver_ | why don't you just get a voip service or something? how many simultaneous calls? |
| 04:03.56 | Jayden | goatmilk is back... |
| 04:04.03 | ScythelX | well we use cisco phones with the g729 license |
| 04:04.08 | _daver_ | what is the upstream/downstream of the cable modem? |
| 04:04.08 | Jayden | dude, you need to change that nick |
| 04:04.18 | Beirdo | mikegrb: OMG. |
| 04:04.37 | ScythelX | so I can squeeze in good amount of connections |
| 04:04.46 | mikegrb | Beirdo: it so is |
| 04:05.02 | Beirdo | some people are soooo stupid |
| 04:05.30 | mikegrb | they so are |
| 04:05.43 | mikegrb | haha the closed captions on this video say (SOUND) |
| 04:05.48 | mikegrb | a lot of good that does me |
| 04:05.52 | Jayden | mike\beirdo... nice |
| 04:05.53 | Beirdo | heh |
| 04:05.58 | mikegrb | :< |
| 04:05.58 | *** join/#asterisk Inv_arp (junya@adsl-8-230-122.mia.bellsouth.net) |
| 04:06.05 | mikegrb | i was interested in the news story too :/ |
| 04:06.11 | _daver_ | scythelx: so you'll be doing voip over the cable modem to a colocated asterisk box? why not just bring the box in locally, and get another cable modem or 2. |
| 04:06.15 | ScythelX | well depending on cents per min and cost of DID's I will have to weigh the cost of getting a TDM colo or using an upstream |
| 04:06.16 | _daver_ | it would probably be cheaper. |
| 04:06.29 | Beirdo | (SOUND) |
| 04:06.30 | Beirdo | hehe |
| 04:06.52 | Inv_arp | woah talking to my friend on aol works pretty well from dial-up in domican repuplic to usa dsl.... |
| 04:07.28 | mikegrb | _daver_: my home voip calls go to my colo box |
| 04:07.36 | Inv_arp | and it seems to be using tcp |
| 04:07.49 | mikegrb | Inv_arp: welcome to the inatarweb |
| 04:07.58 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 04:08.15 | Jayden | goatmilk.. you still really need to change that nick |
| 04:08.23 | Jayden | REALLY! |
| 04:08.54 | Beirdo | to "toadlick"? |
| 04:09.11 | mikegrb | he so shoud |
| 04:09.13 | Jayden | somthing... I dunno. |
| 04:09.15 | mikegrb | should even |
| 04:09.26 | Jayden | jbot: what should goatmilk change his nick too? |
| 04:09.28 | Beirdo | what's wrong with goatmilk? |
| 04:09.45 | Jayden | as somthing to drink? |
| 04:09.48 | goatmilk | why do you think i should change it? |
| 04:10.01 | Jayden | ummmm, cuz your goat milk |
| 04:10.08 | *** join/#asterisk Guest^DJ (some@211.24.146.10) |
| 04:10.13 | Beirdo | and? |
| 04:10.31 | mikegrb | Beirdo: Jayden is one of those bigot people |
| 04:10.43 | Jayden | yes, i hate non cow milk |
| 04:10.44 | mikegrb | Beirdo: he hate's milk and proper spelling |
| 04:10.51 | Jayden | ~mbtt |
| 04:10.52 | jbot | methinks mbtt is Mavis Beacons Teaches Typing |
| 04:11.18 | mikegrb | Jayden: mavis beacon tought you cuz? |
| 04:11.25 | Jayden | yes. |
| 04:11.42 | mikegrb | I would ask for a refund |
| 04:11.44 | Jayden | he was a very bad teacher, so don't read into that too much |
| 04:11.46 | Jayden | :) |
| 04:11.51 | mikegrb | and then cut her throat |
| 04:12.11 | mikegrb | Jayden: um, you weren't paying attention much in class, mavis beacon is a woman |
| 04:12.18 | Jayden | really |
| 04:12.20 | Jayden | shit. |
| 04:12.31 | Jayden | was she ugly? |
| 04:12.34 | Beirdo | hehe |
| 04:12.47 | mikegrb | http://images.amazon.com/images/P/B0001GU7DI.01.LZZZZZZZ.jpg |
| 04:12.47 | Beirdo | Mavis is a woman's name after all |
| 04:12.55 | Jayden | who the hell has a name like mavis... |
| 04:13.12 | Beirdo | women do |
| 04:13.14 | Jayden | shit... I'm so going to hell now |
| 04:13.24 | Beirdo | now who would call their kid that |
| 04:13.25 | mikegrb | http://www.whatisthe2gs.apple2.org.za/slam_dunk/educational_pages/educational_boxes_large/mavis_beacon.jpg |
| 04:13.29 | goatmilk | Jayden: stop complaining about everyone's name. |
| 04:13.40 | Jayden | hehe |
| 04:13.46 | Sedorox | looks like a Jackson |
| 04:13.58 | *** join/#asterisk choward (~choward@user-69-1-15-110.knology.net) |
| 04:14.01 | Jayden | your just upset "cuz" I don't like to drink you |
| 04:14.06 | Jayden | sorry |
| 04:14.16 | Beirdo | Mavis Beacon ain't a real person anyways |
| 04:14.23 | goatmilk | i have no idea what you're trying to say... |
| 04:14.33 | Jayden | ain't ain't a real word anyways :) |
| 04:14.33 | mikegrb | Beirdo: she so is, I met her |
| 04:15.00 | mikegrb | Jayden: he is canookia, he is excused |
| 04:15.01 | Jayden | where did you meet her? |
| 04:15.01 | |Vulture| | Beirdo: dont crush my dreems of sumday meating my master typing teecher! |
| 04:15.13 | Jayden | canookia? |
| 04:15.24 | mikegrb | I met her in 1988 |
| 04:15.27 | Beirdo | silly people |
| 04:15.32 | mikegrb | on an Apple II |
| 04:15.33 | mikegrb | in class |
| 04:15.35 | |Vulture| | lol |
| 04:15.38 | Jayden | nice |
| 04:15.43 | Jayden | canookia? |
| 04:15.55 | mikegrb | it is north of the Unfree States of America |
| 04:16.10 | Beirdo | I didn't aay it |
| 04:16.15 | Beirdo | say even |
| 04:16.18 | goatmilk | mikegrb: it stands for United |
| 04:16.34 | Beirdo | no it doesn't :) |
| 04:16.35 | mikegrb | goatmilk: tell that to the leader |
| 04:16.44 | Beirdo | tell that to the voting public |
| 04:16.50 | Beirdo | not terribly united |
| 04:16.52 | goatmilk | mikegrb: i'm pretty sure he already knows. |
| 04:16.54 | Beirdo | untied maybe |
| 04:17.15 | |Vulture| | Anyone know if the Dell 2850 is 3.3v or 5v PCI? |
| 04:17.19 | implicit | mikegrb: and i'm pretty sure he doesn't care :) |
| 04:17.20 | Jayden | are you making comments about canadians you bastard. |
| 04:17.28 | Beirdo | good luck, I wear sandals :) |
| 04:17.29 | goatmilk | implicit: he need not. |
| 04:17.49 | Jayden | 2850? should definately have some 5's ... |
| 04:17.54 | mikegrb | implicit: I'm sure he doesn't |
| 04:18.06 | nestAr | it's got 3 slots.. not sure on the voltages.. |
| 04:18.07 | |Vulture| | Jayden: its got 3 PCI slots |
| 04:18.10 | |Vulture| | yea |
| 04:18.20 | goatmilk | i'd rather talk about something other than politics though. |
| 04:18.22 | nestAr | guess there's no way to check from linux? |
| 04:18.26 | Jayden | big or little slots |
| 04:18.31 | Beirdo | mikegrb: now you behave, don't let heidi catch you at this |
| 04:18.51 | Jayden | well,, then talk |
| 04:18.57 | MrEntropy | isn't call initiation meant to go something like: INVITE -->, ACK <--, ACK --> ? |
| 04:18.57 | mikegrb | Beirdo: it's okay as long as it isn't in the other channel |
| 04:18.57 | Jayden | ~Gutatamala |
| 04:19.08 | Beirdo | heh. fair enough |
| 04:19.08 | Jayden | damn |
| 04:19.10 | |Vulture| | "1 x 32-bit/33MHz 5v PCI slot" |
| 04:19.11 | mikegrb | but now it is time to take the cats for a walk and smoke a cigarette |
| 04:19.14 | mikegrb | I quit smoking 1 month, 1 week, 3 days, 1 hour, 19 minutes, and 16 seconds ago. During that time, I would have smoked 566 cigarettes. (That's like smoking a 0.03 mile-long cigarette) By quitting, I've saved $99.05! I've avoided inhaling 14 grams of tar, 905 mg of nicotine, and 9 grams of carbon monoxide. |
| 04:19.15 | Jayden | ~Guatamala |
| 04:19.17 | jbot | hmm... guatamala is where the examples end |
| 04:19.22 | mikegrb | I need to update that since I unquit |
| 04:19.28 | mikegrb | I made it a month though! |
| 04:19.31 | Beirdo | heh |
| 04:19.46 | Jayden | :) |
| 04:19.46 | mikegrb | I smoke those too |
| 04:19.54 | mikegrb | I smoke whatever burns |
| 04:19.59 | Beirdo | hehe |
| 04:20.02 | Jayden | you smoke cats? |
| 04:20.08 | mikegrb | anyway cats are anxious to go on thier evening walk |
| 04:20.13 | Beirdo | I smoked grass once as a kid (real grass, not pot) |
| 04:20.40 | Jayden | ewwww |
| 04:20.42 | Beirdo | it was hideous |
| 04:21.10 | tzanger | bjohnson: you around? |
| 04:21.21 | Jayden | smoking sounds good |
| 04:21.26 | Jayden | ~smoking |
| 04:21.27 | jbot | smoking will kill you |
| 04:21.28 | Beirdo | no idea why I did it either |
| 04:21.35 | Jayden | hehe |
| 04:21.40 | Beirdo | ~smoking is good for you |
| 04:21.41 | jbot | ...but smoking is already something else... |
| 04:21.46 | Beirdo | heheh |
| 04:22.07 | Nivex | jbot: no, smoking is hideously gross and will kill you |
| 04:22.08 | jbot | Nivex: okay |
| 04:22.25 | Beirdo | ~smoking |
| 04:22.26 | jbot | smoking is, like, hideously gross and will kill you |
| 04:22.39 | Jayden | ~no, smoking can kill you |
| 04:22.52 | Jayden | ~no, smoking can kill you |
| 04:23.08 | Jayden | jbot.... don't you care? |
| 04:23.10 | Beirdo | jbot: no, smoking can kill you, and it's a disgusting habit. Unless it's cigars. |
| 04:23.24 | Jayden | :( |
| 04:23.36 | Jayden | ok... now everyone |
| 04:23.40 | Jayden | ~rtfw |
| 04:23.41 | jbot | i guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 04:23.48 | Beirdo | bish ain't listening |
| 04:23.53 | Jayden | bish? |
| 04:23.58 | Jayden | fish? |
| 04:24.26 | paulc | babel fish? |
| 04:24.37 | Jayden | goats milk wanted to say somthing.... |
| 04:24.38 | MrEntropy | damn it, is the wiki down again? |
| 04:24.45 | Jayden | ~rtfw |
| 04:24.47 | jbot | i heard rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 04:25.04 | Beirdo | ~foad |
| 04:25.07 | jbot | extra, extra, read all about it, foad is \"fuck off and die\". Considered by many to be impolite. |
| 04:25.07 | Jayden | aparently not bad enough to type |
| 04:25.15 | Beirdo | hehe |
| 04:25.24 | Jayden | Considered by many to be impolite... |
| 04:25.26 | Jayden | nice |
| 04:25.28 | Beirdo | no shit :) |
| 04:25.40 | Jayden | ~AMP |
| 04:25.41 | jbot | rumour has it, amp is an Audio MPEG Player. [non-free] |
| 04:26.08 | Jayden | ~no, AMP is Asterisk Management Portal, a GUI for * |
| 04:26.10 | JerJer | Big floppy donkey dick |
| 04:26.14 | Beirdo | who owns jbot? |
| 04:26.24 | Jayden | JerJer, what an entrance |
| 04:26.44 | doughecka | lol |
| 04:26.56 | tzanger | bfdd is nothing compared ot a nice firm moosepenis |
| 04:27.01 | doughecka | Beirdo: timriker |
| 04:27.13 | MrEntropy | is the wiki down? |
| 04:27.25 | Jayden | is the wiki running |
| 04:27.33 | doughecka | tzanger: nice soft noodle |
| 04:27.40 | Beirdo | umm, I think timriker wrote it, didn't he? He's certainly not in the channel as such. |
| 04:27.40 | JunK-Y | yes it's up on my side. |
| 04:27.49 | Jayden | better go catch it |
| 04:28.01 | tzanger | I *knew* you were waiting for that |
| 04:28.11 | Jayden | y |
| 04:28.12 | doughecka | Beirdo: yes, hes timriker of bzflag fame :P |
| 04:28.18 | Jayden | can't help it sometim |
| 04:28.20 | Jayden | es |
| 04:28.33 | Beirdo | I mean who on this channel :) |
| 04:28.42 | Jayden | everyone |
| 04:28.45 | doughecka | Beirdo: I was the one that sorta asked that it be added |
| 04:28.48 | Jayden | ~Guatamala |
| 04:28.49 | jbot | somebody said guatamala was where the examples end |
| 04:28.53 | Beirdo | cool |
| 04:28.54 | Jayden | ~JerJer |
| 04:28.59 | doughecka | I have no control over it |
| 04:29.06 | Jayden | sure you do |
| 04:29.10 | doughecka | ~JerJer is nufone |
| 04:29.11 | jbot | doughecka: okay |
| 04:29.19 | doughecka | no administrative control |
| 04:29.22 | Jayden | watch this |
| 04:29.53 | Jayden | Jbot: no, JerJer is the guy who just said Big floppy donkey dick |
| 04:29.54 | jbot | Jayden: please, watch your language. |
| 04:30.00 | Jayden | hey, he did |
| 04:30.11 | goatmilk | Jayden: yeah you really need to watch your language. |
| 04:30.12 | Jayden | well.. fine, be that way |
| 04:30.14 | doughecka | ~nufone is <action> kicks voicepulses butt |
| 04:30.15 | jbot | ...but nufone is already something else... |
| 04:30.18 | doughecka | ah |
| 04:30.20 | doughecka | ~nufone |
| 04:30.21 | jbot | methinks nufone is Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 04:30.40 | MrEntropy | a 200 response is not a method is it? |
| 04:30.41 | Beirdo | ~voipjet |
| 04:30.59 | Jayden | ~goatmilk is mad at me cuz I don't like his name... |
| 04:31.00 | jbot | ...but goatmilk is already something else... |
| 04:31.08 | Jayden | ~goatmilk |
| 04:31.09 | jbot | goatmilk is, like, silly silly |
| 04:31.18 | *** join/#asterisk lamtran (~lamtt77@210.245.42.226) |
| 04:31.19 | mikegrb | Jayden: stop playing with the bot |
| 04:31.19 | Jayden | yeah... ummmm |
| 04:31.23 | Jayden | sorry |
| 04:31.32 | Jayden | :( |
| 04:31.33 | Beirdo | jbot: no, goatmilk is better for you than cowmilk |
| 04:31.34 | jbot | okay, Beirdo |
| 04:31.46 | Jayden | hey, I did not set that |
| 04:31.49 | Beirdo | there we go |
| 04:31.49 | doughecka | hopefully me asterisk box will magically fix itself tomorro |
| 04:31.50 | doughecka | w |
| 04:32.12 | Jayden | hey, doug, you dropped a w |
| 04:32.29 | Jayden | so, goatmilk.. you said you wanted to discuss somthing |
| 04:32.34 | Jayden | wassup |
| 04:33.43 | Jayden | and on his way to sleep |
| 04:33.43 | Jayden | nice |
| 04:33.43 | doughecka | Jayden, thou wilt be as valiant as the wrathful dove, or most magnanimous mouse |
| 04:33.54 | Jayden | heehee |
| 04:34.33 | *** join/#asterisk The_Ball (~alex@dialup-166.27.221.203.acc51-wick-bne.comindico.com.au) |
| 04:34.50 | Jayden | Ok... now I am going too... |
| 04:35.11 | brenda | Does anyone have a link to * codes for Qwest? |
| 04:37.00 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 04:37.58 | mrgoby | anyone had success with * on mini-itx ? |
| 04:38.11 | klasstek | voip-info.org |
| 04:38.34 | JerJer | mrgoby: most certainly |
| 04:38.43 | mrgoby | want to run a fanless box with a tdm-400... do you think that will be too hot ? |
| 04:38.45 | klasstek | sup JerJer? |
| 04:38.59 | JerJer | ssdd |
| 04:40.14 | doughecka | JerJer: how many locations do you have? |
| 04:40.49 | JerJer | all of them |
| 04:41.09 | doughecka | hah |
| 04:42.22 | Beirdo | all your locations are belong to JerJer |
| 04:42.35 | doughecka | mine dont |
| 04:43.19 | Beirdo | I wouldn't be so sure :) |
| 04:45.22 | implicit | JerJer: how are you? |
| 04:45.31 | implicit | JerJer: why don't you call me any more? |
| 04:45.48 | MrEntropy | what method are 200 responses sent in? |
| 04:45.55 | mikegrb | implicit: he is too busy calling me |
| 04:46.10 | implicit | mikegrb: he is too busy sucking himself off |
| 04:46.23 | doughecka | in soviet russia, JerJer calls YOU |
| 04:46.24 | *** join/#asterisk clark_ (clark_@ip68-7-102-220.sd.sd.cox.net) |
| 04:46.33 | *** join/#asterisk santiago (~santiago@63.245.86.121) |
| 04:47.31 | Beirdo | ~last implicit |
| 04:47.39 | JunK-Y | ~seen implicit. |
| 04:47.40 | jbot | i haven't seen 'implicit.', JunK-Y |
| 04:47.40 | JunK-Y | ~seen implicit |
| 04:47.41 | jbot | implicit is currently on #asterisk (9h 30m 49s). Has said a total of 9 messages. Is idling for 1m 31s |
| 04:47.49 | catbutt | anyone having probs with voicepulse and CID? |
| 04:47.50 | implicit | hi junk-y |
| 04:47.54 | JunK-Y | mooo |
| 04:47.58 | implicit | ooom |
| 04:48.06 | JunK-Y | sup? |
| 04:48.09 | Beirdo | ~nii |
| 04:48.10 | jbot | i guess nii is a cool guy |
| 04:48.10 | implicit | pus |
| 04:48.14 | Beirdo | heheh |
| 04:48.19 | Beirdo | mistyped |
| 04:48.22 | Beirdo | ~moo |
| 04:48.23 | jbot | mooooooooo! I am cow, hear me moo, I weigh twice as much as you. I am cow, eating grass, methane gas comes out my ass, or http://www.linuks.mine.nu/moo/ |
| 04:48.31 | implicit | ~jbot |
| 04:48.32 | jbot | i guess jbot is the shipboard computer, but you may call me eddie if it helps you relax |
| 04:48.33 | implicit | ~karma |
| 04:48.33 | jbot | implicit has karma of 2 |
| 04:48.38 | implicit | wooooooooohoooooooooo |
| 04:48.42 | implicit | ~karma junk-y |
| 04:48.42 | jbot | junk-y has neutral karma |
| 04:48.44 | Beirdo | ~karma |
| 04:48.44 | jbot | beirdo has neutral karma |
| 04:48.58 | implicit | why do you have neutral |
| 04:49.02 | implicit | ~karma junk-y |
| 04:49.02 | jbot | junk-y has neutral karma |
| 04:49.06 | implicit | ~karma beirdo |
| 04:49.06 | jbot | beirdo has neutral karma |
| 04:49.15 | JunK-Y | ~karma junky |
| 04:49.15 | jbot | junky has neutral karma |
| 04:49.26 | JunK-Y | isnt neutral |
| 04:49.27 | catbutt | so no voicepulse connect users? |
| 04:49.31 | implicit | ~karma beirdo |
| 04:49.31 | jbot | beirdo has neutral karma |
| 04:49.34 | JunK-Y | im at like +23 |
| 04:49.44 | implicit | JunK-Y: this is a different karma |
| 04:49.48 | implicit | this is jbot karma |
| 04:49.49 | implicit | any wya |
| 04:49.57 | implicit | anyway, brb |
| 04:50.03 | Beirdo | ~mikegrb++ |
| 04:50.18 | Beirdo | hmph |
| 04:50.34 | _Vile | ~karma _Vile |
| 04:50.34 | jbot | _vile has neutral karma |
| 04:50.42 | tzanger | oh wow |
| 04:50.47 | implicit | ~karma _Vile |
| 04:50.47 | jbot | _vile has neutral karma |
| 04:50.51 | _Vile | yeah man I'm rollin' |
| 04:50.51 | tzanger | zap transfers are attended by default!! |
| 04:50.54 | implicit | _vile++ |
| 04:50.58 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 04:51.00 | pcm | ~karma |
| 04:51.00 | jbot | pcm has neutral karma |
| 04:51.03 | catbutt | twisted? |
| 04:51.11 | implicit | ~karma _Vile |
| 04:51.11 | jbot | _vile has neutral karma |
| 04:51.19 | martinp | ~karma |
| 04:51.19 | jbot | martinp has neutral karma |
| 04:51.26 | Beirdo | ~trout |
| 04:51.27 | jbot | ACTION fills beirdo's pants with of day-old trout |
| 04:51.36 | Beirdo | nice |
| 04:51.38 | pcm | karma sux |
| 04:52.12 | implicit | ~karma |
| 04:52.12 | jbot | implicit has karma of 2 |
| 04:52.14 | implicit | good |
| 04:52.22 | implicit | ~kram |
| 04:52.23 | jbot | Looking for the elusive BishopChicken. |
| 04:52.24 | shido6 | boink |
| 04:52.30 | implicit | ~karma kram |
| 04:52.30 | jbot | kram has karma of 3 |
| 04:52.50 | Beirdo | ~boink |
| 04:52.51 | catbutt | ~twisted |
| 04:52.52 | jbot | [twisted] twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net |
| 04:52.59 | implicit | ~implicit |
| 04:53.00 | jbot | it has been said that implicit is fun |
| 04:53.06 | paulc | ~paulc |
| 04:53.19 | paulc | no one said I'm a british wanker? |
| 04:53.19 | Beirdo | jbot |
| 04:53.39 | implicit | i dont know |
| 04:53.44 | Beirdo | jbot: paulc says that he is a British wanker. Who are we to argue? |
| 04:53.46 | jbot | Beirdo: okay |
| 04:53.52 | implicit | ~paulc ? |
| 04:53.53 | jbot | well, paulc is a british wanker |
| 04:53.57 | pcm | does anyone have some impossible project to do ? |
| 04:54.08 | catbutt | paulc - you are a british waneeer |
| 04:54.15 | catbutt | wanker |
| 04:54.19 | catbutt | oops |
| 04:54.19 | paulc | ah, but does jbot know that? |
| 04:54.23 | paulc | pcm: bored? |
| 04:54.29 | Beirdo | because I just told him? |
| 04:54.33 | pcm | paulc: looks like it hehe |
| 04:54.37 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 04:54.39 | catbutt | no but catbutt knows it |
| 04:54.44 | Beirdo | or because you did |
| 04:54.50 | implicit | ~paulc |
| 04:54.51 | jbot | extra, extra, read all about it, paulc is a british waneeer (says catbutt) |
| 04:55.08 | Beirdo | I am such a dumbass some days |
| 04:55.10 | catbutt | hehe |
| 04:55.11 | paulc | pcm: Hmm.. how about I give you my 200 LEDs and Basic Stamp 2 and you go can build me my colour washing light wall thing? |
| 04:55.13 | catbutt | oops |
| 04:55.17 | Beirdo | ~Beirdo |
| 04:55.27 | pcm | pualc: that's not impossible |
| 04:55.34 | pcm | paulc: that's simply boring .... |
| 04:55.37 | paulc | jbot, no, paulc is aBbritish WANKER not a British waneeer |
| 04:55.38 | jbot | paulc: okay |
| 04:55.55 | paulc | jbot, no, paulc is a British WANKER not a British waneeer, and often can't type |
| 04:55.56 | jbot | okay, paulc |
| 04:55.59 | paulc | ~paulc |
| 04:56.00 | jbot | you are probably a British WANKER not a British waneeer, and often can't type |
| 04:56.01 | implicit | jbot, no paulc is insane |
| 04:56.02 | jbot | implicit: okay |
| 04:56.14 | Beirdo | jbot: Beirdo is a dumbass some days, and irritable on Mondays |
| 04:56.15 | jbot | Beirdo: okay |
| 04:56.15 | paulc | ButLovelyWithIt(tm) |
| 04:56.16 | implicit | ~paulc |
| 04:56.17 | jbot | i heard paulc is insane |
| 04:56.43 | paulc | Hmm.. American Idol.. Constantine - what were you thinking?! |
| 04:56.46 | catbutt | catbutt is a real catbutt |
| 04:57.01 | paulc | Is it me, or are there a handful of guys in the final 12 who are just fecking awful? |
| 04:57.19 | Beirdo | I refuse to watch that tripe |
| 04:57.23 | catbutt | bo bice |
| 04:57.28 | mrgoby | jbot is the next american idol |
| 04:57.33 | catbutt | will win |
| 04:57.49 | catbutt | bo i a neighbor |
| 04:57.54 | mrgoby | ~dance |
| 04:57.55 | jbot | ACTION does a disco dance. |
| 04:57.55 | catbutt | is |
| 04:58.02 | mrgoby | ~sing |
| 04:58.03 | jbot | "Night fever, night fever. You know how to do it!" |
| 04:58.06 | Beirdo | ~disco |
| 04:58.07 | jbot | burn, baby, burn |
| 04:58.21 | catbutt | man.. i need a beer |
| 04:58.31 | catbutt | but I don't have one |
| 04:58.35 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:59.06 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
| 04:59.50 | JunK-Y | some1 with zap cards wants to make a small test w/ me ? |
| 05:00.15 | pcm | what test ? |
| 05:00.30 | tzanger | ONE POINT TWENTY ONE JIGGAWATTS INTO THE RJ45!! |
| 05:01.53 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 05:02.27 | *** join/#asterisk Newbie___ (some@211.24.146.10) |
| 05:04.23 | Newbie___ | hi all, i am running * for calling cards on an E1, is it possible to make * seperate between office number and acess number on the same E1 ? |
| 05:05.10 | *** join/#asterisk siome (~siome@222.124.54.122) |
| 05:05.37 | siome | hi |
| 05:05.52 | siome | can i ask newbie question here? |
| 05:06.41 | JunK-Y | siome: sure.é |
| 05:07.30 | siome | can asterisk talk to switching equipment with e1 using mfc signaling protocol? |
| 05:07.53 | pcm | what does mfc stand for ? |
| 05:08.00 | JunK-Y | i dont know e1. |
| 05:08.03 | Beirdo | ~mfc |
| 05:08.04 | jbot | Microsoft Foundation Classes, or crap |
| 05:08.15 | Beirdo | heh |
| 05:08.33 | pcm | siome: then propably NO unless some comercial company sells it ... |
| 05:08.34 | Beirdo | I don't think that's the MFC you had in mind? |
| 05:08.49 | siome | its a telephony signaling protocol in E1 |
| 05:08.59 | pcm | siome: is it digital or analog ? |
| 05:09.22 | Beirdo | if it's on an E1, it's digital |
| 05:09.27 | siome | signaling is digital |
| 05:09.54 | Beirdo | E1 being a wholly digitial media :) |
| 05:10.08 | heison | can someone help me test my iaxtel number? |
| 05:10.09 | siome | how can asterisk talk to another equipment using e1? |
| 05:10.19 | siome | E&M ? |
| 05:10.30 | Beirdo | no clue |
| 05:11.22 | pcm | asterisk does E&M and R2 and PRI(dss1) |
| 05:11.48 | pcm | and loopstart |
| 05:11.55 | siome | pcm : how about ss7? |
| 05:12.07 | pcm | siome: that's the taugh one |
| 05:12.11 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 05:12.45 | pcm | pcm: there's only www.openss7.org .... they have 'something' but noone knows what .... unless you buy from them |
| 05:13.01 | pcm | ups, i'm not talking to myself ...hehe |
| 05:13.07 | siome | :) |
| 05:13.08 | Beirdo | how is that "open"? |
| 05:13.19 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 05:13.27 | pcm | 'virtually' it's open hehe |
| 05:14.04 | siome | even if it is, it can answer only part of my needs |
| 05:14.36 | pcm | if you want to use ss7 you shouldn't want to seek asterisk stuff |
| 05:14.38 | siome | i need an open system to convert an MFC-R2 E1 signaling to SMFC-R2 E1 signaling |
| 05:15.03 | siome | i thought asterisk can do that :) |
| 05:15.06 | pcm | well that might be implemented .... you need to check steve underwood's stuff |
| 05:15.15 | pcm | he implemented R2 for many countries ... |
| 05:15.22 | neopher | anyone know how to extract a .msi file, won't install, tring to get 30 vip firmware from cisco call manager install file |
| 05:16.00 | pcm | look for libr2 on cvs.digium.com .... |
| 05:16.07 | pcm | or somewhere on the web |
| 05:16.09 | siome | pcm : where can i found steve underwood's stuff? im sorry :) who is he? |
| 05:16.13 | neopher | wow, i'm getting allot of lag tonight |
| 05:16.26 | pcm | look for libr2 on cvs.digium.com .... |
| 05:17.05 | siome | thanks pcm, Beirdo, JunK-Y |
| 05:17.21 | Beirdo | no problem. not that I helped much :) |
| 05:17.37 | |Vulture| | whats the best codec for communicating via two * servers through IAX2? 729? with ilbc in second? |
| 05:17.43 | pcm | sieme: actually here: http://www.opencall.org/ |
| 05:18.01 | Beirdo | best for what? |
| 05:18.05 | siome | tks |
| 05:18.13 | |Vulture| | for data/voice balance |
| 05:18.20 | |Vulture| | I mean data/quality balance |
| 05:18.49 | Beirdo | I would imagine that 729 and speex would be good for that |
| 05:19.04 | siome | the cvs.digium is authenticated page by the way |
| 05:19.18 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 05:19.20 | pcm | are there any business ppl here ? |
| 05:19.37 | algorithmn | question? |
| 05:19.54 | pcm | question ? |
| 05:19.59 | neopher | anyone know how to extract a .msi file, won't install, tring to get 30 vip firmware from cisco call manager install file |
| 05:20.00 | pcm | siome: do you know what's the cvs ? |
| 05:20.21 | neopher | sorry to ask a second time, mirc keeps disconnecting |
| 05:20.36 | pcm | neopher: it propably might be encrypted .... |
| 05:20.51 | pcm | sieme: go to opencall.org for http stuff |
| 05:21.35 | neopher | anyone have call manager installed |
| 05:25.19 | neopher | never mind, i just found the switch to disable hardware check |
| 05:26.32 | siome | pcm : i found the mfc/r2 support for asterisk in opencall.org thanks |
| 05:26.47 | siome | pcm : im halfway trough |
| 05:26.49 | siome | :) |
| 05:26.54 | siome | what is cvs? |
| 05:27.56 | Beirdo | ~cvs |
| 05:27.57 | jbot | cvs stands for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS can be found at http://asterisk.espia-net.net/horde/chora/cvs.php |
| 05:29.56 | roamer323 | anyone know if music-on-hold actually depends on timer to work properly? I can't install ztdummy and have no zaptel card :-( |
| 05:31.15 | heison | roamer323: i believe so... |
| 05:32.27 | pcm | are there any business ppl here ? |
| 05:32.32 | shido6 | ? |
| 05:32.39 | shido6 | what do you mean? |
| 05:33.02 | roamer323 | thx - heison |
| 05:33.06 | *** join/#asterisk goodnewscd (~goodnewsc@S010600095b316c67.cg.shawcable.net) |
| 05:34.43 | roamer323 | anyone going to be @ the miami tradeshow over the next few days? |
| 05:35.29 | pcm | shido: you know sales ppl ... business makers .... |
| 05:36.36 | shido6 | what are you looking to do? |
| 05:37.30 | a1fa | whats a good free sip server |
| 05:37.51 | roamer323 | alfa - SER |
| 05:38.15 | a1fa | SER.? |
| 05:38.32 | Beirdo | ~ser |
| 05:38.33 | jbot | i heard ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 05:39.03 | techie | roamer323: yeah |
| 05:39.32 | pcm | is Beirdo a bot too ? |
| 05:39.33 | a1fa | no |
| 05:39.36 | roamer323 | jbot - are you actually a bot (pretty intelligent if you are) |
| 05:39.37 | jbot | roamer323: what are you talking about? |
| 05:39.39 | a1fa | i want one available on net |
| 05:39.43 | a1fa | ~fwd |
| 05:39.44 | jbot | it has been said that fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 05:40.09 | pcm | ~tor2 |
| 05:40.21 | pcm | ~wcfxo |
| 05:40.28 | roamer323 | ~sipphone |
| 05:40.41 | Beirdo | ~Beirdo |
| 05:40.42 | jbot | from memory, beirdo is a dumbass some days, and irritable on Mondays |
| 05:40.49 | roamer323 | hmm - not so intelligent after all |
| 05:40.49 | Beirdo | heh |
| 05:40.51 | techie | oh. |
| 05:41.17 | Beirdo | if nobody's taught the bot, he can't reply |
| 05:41.40 | Beirdo | ~beer |
| 05:41.41 | jbot | methinks beer is ummm, ummm good!, or good for you! |
| 05:41.43 | roamer323 | ~asterisk |
| 05:41.44 | jbot | hmm... asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
| 05:42.01 | a1fa | stop |
| 05:42.01 | a1fa | ":) |
| 05:42.10 | a1fa | i need a free sip registrar |
| 05:42.11 | a1fa | a good one |
| 05:42.19 | techie | i need free cash |
| 05:42.26 | a1fa | i got it |
| 05:42.45 | pcm | techie: print it heh |
| 05:45.09 | Beirdo | roll bums for their dough |
| 05:46.38 | a1fa | ok |
| 05:46.38 | a1fa | people |
| 05:46.47 | a1fa | who should be my free sip registrar |
| 05:46.51 | a1fa | i plan to buy |
| 05:46.56 | a1fa | vonage soon |
| 05:47.28 | Beirdo | why? |
| 05:47.40 | Beirdo | vonage isn't asterisk friendly |
| 05:48.19 | a1fa | its cheap tho |
| 05:48.44 | Beirdo | so why are you asking here if you aren't even planning on using asterisk? |
| 05:48.45 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:48.56 | a1fa | i am |
| 05:49.08 | a1fa | i will be using lingo with asterisk |
| 05:50.29 | *** part/#asterisk santiago (~santiago@63.245.86.121) |
| 05:51.16 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 05:51.18 | *** join/#asterisk jjg (~bvc@dsl081-101-201.den1.dsl.speakeasy.net) |
| 05:51.20 | jjg | hi |
| 05:52.09 | a1fa | Beirdo : this is for my home :) |
| 05:52.19 | a1fa | grandstream |
| 05:52.30 | Beirdo | and? |
| 05:52.46 | jjg | is it possible to increase txgain for a inbound sip call then IAX2 outbound to FWD? |
| 05:52.50 | a1fa | nvm.. what free sip registrar do your ecommend |
| 05:53.00 | Beirdo | I don't use SIP |
| 05:53.30 | Beirdo | well I do from my softphone to Asterisk, but that's it |
| 05:53.43 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 05:54.09 | Defraz | What do you use? |
| 05:54.13 | Beirdo | IAX |
| 05:54.20 | Beirdo | what else? :) |
| 05:54.26 | Defraz | haha |
| 05:54.37 | Beirdo | a lot more trouble-free |
| 05:54.38 | Defraz | I am having some trouple with h323 stuff |
| 05:54.43 | Defraz | yea I can imagine |
| 05:54.56 | Nugget | livejournal's power failure earlier this year (over 24 hours of downtime) and wikipedia's power failure yesterday (still down) are poster children for mysql's lack of robustness. Why do people still use mysql? it's such a horrible platform. |
| 05:54.57 | Defraz | I want to tie my two asterisk boxes together. |
| 05:55.10 | Defraz | IAX will be used for the tif I can figure it out. |
| 05:55.11 | Beirdo | WTF? |
| 05:55.26 | Beirdo | power failures don't show squat about mysql. |
| 05:55.43 | Nugget | sure they do. |
| 05:55.57 | *** join/#asterisk santiago (~santiago@63.245.86.121) |
| 05:56.03 | Nugget | when your databases all come back corrupted with no recovery options available but experimenting, it sure as hell is a fault of the database. |
| 05:56.04 | Beirdo | how? |
| 05:56.12 | Beirdo | ah |
| 05:56.22 | Nugget | when recovery, even with backups, takes 40 hours -- the database is at fault |
| 05:56.26 | Beirdo | I'm not sure that that's what happened to either though |
| 05:56.39 | Nugget | the power failed and the databases came back with errors |
| 05:56.49 | Nugget | even using innodb with "transactions" |
| 05:57.00 | Beirdo | I bet I know why |
| 05:57.12 | Beirdo | I bet they had write caching on on the drives |
| 05:57.35 | Beirdo | and where were the UPSes? That's what it indicates to me |
| 05:57.35 | Nugget | livejournal's postmortem mentioned that as a factor in one case, but not in the others. |
| 05:57.37 | GreyFoxx | Power failed, with no UPS or automated safe shutdown ?, no replicated alternate DB backup? |
| 05:57.56 | Beirdo | don't blame mysql for lack of UPS |
| 05:58.03 | techie | pure hell |
| 05:58.05 | a1fa | ok.. who uses sip here? |
| 05:58.20 | Beirdo | why are you stuck on SIP? |
| 05:58.22 | Nugget | UPSs are rarely permitted in datacenter facilities. |
| 05:58.34 | a1fa | Beirdo : i am trying to find a free sip registry |
| 05:58.40 | Beirdo | most datacenters INCLUDE UPS last I heard. |
| 05:58.47 | Nugget | yes, and they don't permit them. |
| 05:58.55 | Beirdo | mine sure as hell does |
| 05:58.55 | GreyFoxx | The few Datacenters I've dealt with included UPSs and onsite generators |
| 05:58.57 | techie | you're in the wrong colo |
| 05:59.27 | Nugget | no serious colo will permit you to run your own UPS. it's a hazard because it prevents them from doing an emergency shutdown in the event of a catastrophe. |
| 05:59.43 | Nugget | they provide redundant power to the machines, centrally. but sometimes that fails. |
| 05:59.48 | Beirdo | no serious colo will have a lack of UPS |
| 05:59.50 | Nugget | (as in these cases) |
| 06:00.27 | Nugget | they're not going to let you load a 2200va tripplite in your rack. |
| 06:00.27 | Beirdo | and redundant power rarely fails (thank GOD) |
| 06:00.40 | Nugget | it's a hazard |
| 06:00.56 | Beirdo | how is is a hazard? |
| 06:01.10 | DJ-Pyro | Beirdo: the EPO in the datacenter has to cut power to everything |
| 06:01.16 | Nugget | as I said earlier, it prevents them from being able to shut down the machines quickly in the event of a fire |
| 06:01.21 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 06:01.23 | Nugget | right |
| 06:01.25 | DJ-Pyro | a ups in a rack is still providing power that they can't disconnect |
| 06:01.36 | techie | you need your own cage |
| 06:01.57 | Beirdo | why would they EPO when they should have fire suppressant systems? |
| 06:02.01 | DJ-Pyro | and since it was the EPO that was hit, and not a UPS failure, it did its job of turning off everything |
| 06:02.04 | GreyFoxx | Exactly |
| 06:02.10 | GreyFoxx | Gas based firesupression |
| 06:02.17 | DJ-Pyro | Beirdo: because the fire department comes in and they use water |
| 06:02.20 | DJ-Pyro | water + electricity = bad |
| 06:02.24 | *** part/#asterisk Defraz (~t0tal@65.103.222.4) |
| 06:02.29 | DJ-Pyro | despite the fact that they may have gas supression |
| 06:02.29 | Beirdo | DJ-Pyro: no |
| 06:02.45 | a1fa | ups this that, |
| 06:02.53 | a1fa | lets get back |
| 06:02.54 | DJ-Pyro | well, and for people who may accidently have grabbed that nice 480V rail while doing some work |
| 06:02.59 | a1fa | on track |
| 06:03.04 | Beirdo | trust me, the fire department doesn't just come in like that, unless you cave a craptacular FD |
| 06:03.20 | Beirdo | have rather |
| 06:03.24 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
| 06:03.30 | DJ-Pyro | Beirdo: well, they're not going to storm the place with water, but they're still going to want the power cut before someone can get hurt |
| 06:03.41 | iceyp | reading sip and iax realtime with extconfig.conf , is this a default module of asterisk? |
| 06:03.44 | Beirdo | if you have gas suppression system, the FD won't come in unless the gas system fails |
| 06:04.25 | Beirdo | having witnessed a halon dump when our alarm went off once, I can personally attest to it |
| 06:04.28 | GreyFoxx | No UPS + writecaching enabled on important DB without at least 1 replicated backup which doesn't use write caching is plain stupid. BUT at this point I have no evidence that it is the situation they found themselves in :) |
| 06:05.02 | DJ-Pyro | halon scares me |
| 06:05.06 | Nugget | GreyFoxx: no argument there -- but it's still a failing of the database that the only way to ensure data integrity is to never have an accident. |
| 06:05.24 | Beirdo | and as I've powered off my mysql databases hard many times.... |
| 06:05.26 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-80-91.sentechsa.net) |
| 06:05.31 | DJ-Pyro | GreyFoxx: they had a ups, that wasn't the problem |
| 06:05.46 | GreyFoxx | Nugget: I've killed an Oracle DB on a system with write caching enabled. Is it my fault or Oracles ? |
| 06:06.14 | Nugget | with write-caching enabled all bets are off. |
| 06:06.25 | Beirdo | yup |
| 06:06.25 | GreyFoxx | DJ-Pyro: I was in a NOC at a previous job (Cable ISP + Telco) when the halon system was set off with me locked in the noc... freaked me out hehe |
| 06:06.40 | Beirdo | so don't blame mysql for DBA stupidity :) |
| 06:06.58 | DJ-Pyro | GreyFoxx: yeah, I would be scared as hell too |
| 06:07.09 | Beirdo | GreyFoxx: one of my coworkers had to go into the machine rooms during the dump to make sure there was no fire |
| 06:07.11 | Nugget | as I said, write caching was not a factor for the majority of failed machines in livejournal's crash. |
| 06:07.16 | Beirdo | it was an accidental dump |
| 06:07.17 | *** join/#asterisk Inv_arp (junya@adsl-8-231-38.mia.bellsouth.net) |
| 06:07.29 | GreyFoxx | DJ-Pyro: Needless to say I hit the ground and breathed through my shirt all bundled up while they scrambled to unlock the door :) |
| 06:07.52 | Nugget | there were two cases of human error they identified (failure to disable write caching and failure to migrate away from myisam tables on some clusters) but several of the corrupt clusters were configured "properly" |
| 06:08.04 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:08.07 | jjg | exit |
| 06:08.09 | jjg | quit |
| 06:08.28 | Nugget | nobody knows yet what the problem is with wikipedia |
| 06:08.43 | Nugget | but mysql's track record for large sites is looking more and more grim. |
| 06:08.53 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 06:08.53 | Nugget | (which is hardly surprising) |
| 06:08.54 | DJ-Pyro | GreyFoxx: wow, impressive war story |
| 06:09.12 | GreyFoxx | Nugget: Which makes me question just how "properly" it was done. Was there a replicated backup to another machine? do they do binary backups of the tables ? Or just sqldumps which would take forever to recreate indexes on something the size of wikipedia I';m sure |
| 06:09.36 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 06:10.15 | iceyp | anyone here using sip/iax/extensions realtime? |
| 06:15.47 | implicit | (: |
| 06:15.51 | implicit | hehehehe |
| 06:16.58 | a1fa | can anybody recommend me a good sip reg? |
| 06:17.00 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 06:17.03 | a1fa | for the 10000^2 time |
| 06:17.08 | Nugget | what is a "sip reg"? |
| 06:17.24 | a1fa | somethign sip comes out of |
| 06:18.09 | Nugget | I think perhaps you are confused. |
| 06:18.13 | a1fa | sip server |
| 06:18.20 | Nugget | fwd is probably what you want. fwd.pulver.com |
| 06:18.38 | Nugget | it's a good way to play with voip before you're ready to do something worthwhile |
| 06:18.41 | a1fa | i dont want to run a sip server for my home connection |
| 06:19.01 | marc32344 | why people still buy quintum/cisco hardware... if you can do the same with asterisk/digium for 1/5 of price? |
| 06:19.14 | Nugget | ok, that's fine. but why did you come into a channel about a voip server to tell us that you don't want to run your own voip server? |
| 06:19.40 | Nugget | I doubt you'll find many people who share your perspective in here. |
| 06:19.51 | Essobi | Mmm. Is there any festival voices that don't sound like my old Speak-N-Spell? |
| 06:19.54 | a1fa | no |
| 06:19.56 | Nugget | Essobi: no |
| 06:19.59 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
| 06:20.01 | Essobi | Bahaa |
| 06:20.02 | a1fa | Nugget : i will be running asterisk |
| 06:20.06 | iceyp | anyone use asterisk realtime ? |
| 06:20.08 | a1fa | for my business |
| 06:20.17 | Essobi | Is there any alternative to Festival currently? that's decent? |
| 06:20.22 | iceyp | err iax realtime or extension realtime or sip realtime |
| 06:20.39 | Nugget | Essobi: not that I've ever seen. but clearly we all have different ideas about what's "decent" |
| 06:20.41 | |Vulture| | iceyp: what are you asking? |
| 06:20.53 | Nugget | a1fa: then play with fwd. |
| 06:20.56 | roamer323 | marc32344 -> because they work right the way, on day one.... save 4x the money but spend 24 hr a day hacking away to get things working is currently not the way business is done |
| 06:21.05 | a1fa | i really didnt like fwd ;) |
| 06:21.07 | Essobi | ice If someone coded app_icemakers, someone would use 999 to put ice in their cup. Someone uses it, somewhere. |
| 06:21.14 | Nugget | it's what you asked for. |
| 06:21.45 | iceyp | mkay |
| 06:21.52 | marc32344 | roamer-- you mean asterisk is buggy? |
| 06:22.32 | iceyp | |Vulture| Extconfig.conf , does this connect to mysql without any additional modules; realtime sip.conf, iax.conf and extensions.conf |
| 06:22.56 | Essobi | roamer323 Unfortunately you spent 40K on a cisco deployment.. then you call them for support, and they ask how many endpoint the bug is affecting and you tell them 3,000, and they say that's a level 1 outage, and they'll get back to you about the bug in 2 months. |
| 06:22.57 | Essobi | :) |
| 06:23.00 | roamer323 | marc32344... it is certainly not "turn key" and certainly a moving target at this time... consultants are available to make sure things work |
| 06:23.51 | Essobi | iceyp Check http://www.voip-info.org and lookup asterisk realtime |
| 06:23.59 | marc32344 | even if it's properly setup from day one? |
| 06:24.02 | Essobi | plenty of info there to keep one busy. |
| 06:24.21 | roamer323 | essobi - true... I am not defending cisco... but just answering marc's question realistically (as of the state of things today) |
| 06:24.26 | iceyp | Essobi i have and it doesnt mention anything additional required, just wanted to double check |
| 06:24.46 | Essobi | Dude. All softswitches are moving targets. I've ran 4000 phone full on cisco deployments. |
| 06:24.59 | Essobi | You upgrade and shit breaks. |
| 06:25.02 | Essobi | or changes |
| 06:25.23 | Essobi | or windows just takes a big ole shit on you and you have to format, reinstall, and restore your backup. |
| 06:25.41 | marc32344 | why would the system require constant debugging?? |
| 06:25.59 | Essobi | We all know windows can do some wierd shit at times. I've seen the unity platforms and everything from CCM3.0 to 4.0. |
| 06:26.02 | marc32344 | install/setup/forget. |
| 06:26.08 | Essobi | MAhaha. |
| 06:26.13 | Essobi | Good luck with that bub. |
| 06:26.41 | Essobi | It's not like a hardware PBX. It is software, it is a machine, it will require maintenance. |
| 06:26.53 | Essobi | and just hope joe hacker doesn't target it. |
| 06:27.19 | marc32344 | can you provide an example? |
| 06:27.20 | Beirdo | and even the hardware ones aren't bullet-proof |
| 06:27.25 | roamer323 | essobi - that's another thing... we should all be thankful that early asterisks adopters are not (yet) vicious |
| 06:27.27 | marc32344 | what can go wrong? |
| 06:27.37 | Essobi | Beirdo True enough. |
| 06:27.46 | Essobi | What what? |
| 06:27.58 | pashah | mornin' |
| 06:28.01 | Nugget | asterisk is great because it's flexible. it isn't great because it's reliable. it gets more and more reliable with each release, but right now it's mostly a flexibility win. |
| 06:28.01 | roamer323 | essobi - re:security |
| 06:28.05 | Essobi | "what" can go wrong? Are you seriously asking that? Have you ever admined a machine? |
| 06:28.13 | marc32344 | poor setup ==> ongoing debugging. |
| 06:28.19 | iceyp | Essobi u used sip realtime? |
| 06:28.31 | pcm | marc: asterisk is 'in-progress' |
| 06:28.37 | pcm | and it propably will never change |
| 06:28.38 | iceyp | Essobi because none of the examples show how to add a user/pass/host for mysql |
| 06:28.45 | iceyp | just says sipfriends => mysql,asterisk,sip_buddies |
| 06:28.54 | Essobi | Well, gee.. He pwns your root, looks at your FXO's and adds a sip peer and starts jerking off to 1900 numbers. |
| 06:29.03 | Essobi | iceyp Nope.. I never used it. |
| 06:29.08 | Essobi | Just the old-skool ones. |
| 06:29.21 | Essobi | where you'd dump a mysql database to sip.conf in cronjob |
| 06:30.18 | Essobi | marc So what are we arguing about here? :) |
| 06:30.44 | iceyp | Essobi yeah i currently using that, just worried what would happen when i have over 100 000 users |
| 06:30.50 | Essobi | How many endpoints are you thinking you're going to install and just "forget"? |
| 06:31.19 | Essobi | iceyp What's to worry? I've got a sql database with over 3.8 million records in it. :) |
| 06:31.26 | iceyp | Essobi how do u get asterisk to reload in crontab? |
| 06:31.45 | iceyp | Essobi so the sip.conf can have 1000 00 users with no slow lookups etc? |
| 06:31.47 | Essobi | Heh. |
| 06:31.51 | iceyp | i thought a database would be better |
| 06:31.55 | Essobi | Nah. |
| 06:32.00 | Essobi | sip.conf = memory |
| 06:32.03 | marc32344 | sounds like it's alpha version. |
| 06:32.07 | iceyp | sweet then |
| 06:32.10 | *** join/#asterisk whui (~whui@202.55.45.34) |
| 06:32.15 | iceyp | so it uses heaps of memory :S |
| 06:32.18 | iceyp | that not good |
| 06:32.27 | Essobi | Umm.. dude. |
| 06:32.28 | Damascene | well, you can harden the asterisk box i'd imagine? and i suppose a big disadvantage is now that it has moving parts (ala, hdd) you have to worry about that. i wonder if you can have a lightweigth asterisk box on flash disk and external logging to a mysql box? |
| 06:32.32 | Essobi | Memory is cheap. |
| 06:32.44 | iceyp | k |
| 06:32.52 | Essobi | Disk bandwidth that is required to run big indexes and databases.. ain't. |
| 06:32.53 | JerJer | don't use technology that has moving parts |
| 06:33.23 | marc32344 | damascene-- hdd a problem? |
| 06:33.23 | Essobi | BINGO! WE GOT A WIENER! |
| 06:33.23 | iceyp | Essobi when u add an additional user via the db and it recreates sip.conf, how do u reload asterisk? |
| 06:33.25 | JerJer | my linux distro is 26 megs, uncompressed |
| 06:33.36 | Essobi | asterisk -x cmd works |
| 06:33.41 | iceyp | sweet |
| 06:33.48 | Damascene | marc32344: sure for 'set it and forget it' setups. so i think i'd use a flash card/ide disk instead if i could. i'm new to asterisk... but i cant' imagine the default setup requiring that much disk space. |
| 06:33.51 | iceyp | so u reload every 5 mins or something? |
| 06:33.54 | Essobi | JerJer You're a pimp. Drop is somewhere and lemme dl the busybox install. |
| 06:33.57 | JerJer | -rx 'comand' |
| 06:34.00 | iceyp | reload doesnt cause any issues to the current users |
| 06:34.06 | Essobi | oh rx.. my bad. :) |
| 06:34.24 | Essobi | so "asterisk -rx sip reload" should be money |
| 06:34.27 | JerJer | iceyp: only reload when there is a change to be made |
| 06:34.42 | JerJer | Essobi: in due time |
| 06:34.46 | JerJer | patience is a virtue |
| 06:34.57 | Essobi | There's no channel dropping in sip on reload is there? |
| 06:35.02 | marc32344 | essobi-- in other words, you are saying dont bother with asterix. |
| 06:35.24 | Essobi | marc I'm saying any install is going to have it's problems being that it IS a softswitch. |
| 06:35.54 | Essobi | Granted a cisco platform might be all hand holding and huggy nice stuff.. but it has it's problems too. I take that from experience. |
| 06:35.55 | Damascene | Essobi: hehe, you are scaring me since i'm considering moving someone from 5 analog lines to asterisk. :) |
| 06:36.24 | |Vulture| | all that hand holding costs BIG $$ |
| 06:36.29 | Essobi | It really depends on the user base and how much they are paying you, if you're going to hold their hands and .. |
| 06:36.32 | Essobi | exactly. |
| 06:36.38 | Essobi | solve all their problems. |
| 06:36.47 | marc32344 | I have a $3 markup/user |
| 06:36.52 | JerJer | Its all about the Benjamin's |
| 06:36.54 | Essobi | hey jer.. if I got two Cisco 5400s.. |
| 06:36.59 | Essobi | marc is that it? Muhaha. |
| 06:37.02 | Damascene | Essobi: well, have you seen serious reliability issues for asterisk though? |
| 06:37.02 | |Vulture| | JerJer: hey you guys use 2850s right? |
| 06:37.03 | JerJer | now, bitch go get me some pie |
| 06:37.11 | |Vulture| | Dell 2850 I mean |
| 06:37.14 | JerJer | i don't even know what a 2850 is |
| 06:37.19 | a1fa | hmn |
| 06:37.21 | a1fa | its not working |
| 06:37.23 | a1fa | fwd |
| 06:37.27 | JerJer | 1750 |
| 06:37.35 | Essobi | hey jer.. if I got two Cisco 5400s.. can I use sip reinvite to bypass the rtp betreen the peers when routing from one to the other via an asterisk dialplan/server? |
| 06:37.45 | JerJer | sure |
| 06:37.48 | Essobi | I'm not up to speed on the current state of chan_sip |
| 06:37.53 | Essobi | Rockass. :) |
| 06:37.57 | JerJer | Essobi: if that is really all you want to do fire up SER |
| 06:37.57 | |Vulture| | JerJer: oh oky, I was trying to find out if the 2850s are 5v or 3.3v PCI |
| 06:38.19 | marc32344 | how many hours/month /T1 does a asterix configuration require... average... |
| 06:38.26 | JerJer | then route to asterisk when u need to for pbx logic, gateway, applications, etc |
| 06:38.36 | JerJer | A S T E R I S K |
| 06:38.45 | Essobi | JerJer Well.. I need to do some fixup stuff. CDR/CID/Dial-by-name/Record-a-call-when-needed.. |
| 06:38.45 | marc32344 | asterisk |
| 06:38.51 | JerJer | everybody lets play "How do we spell Asterisk" |
| 06:38.58 | Essobi | A! |
| 06:39.00 | Essobi | S! |
| 06:39.03 | Essobi | T! |
| 06:39.04 | |Vulture| | marc32344: not much after the initial config... just adding users really... |
| 06:39.05 | Essobi | E! |
| 06:39.07 | Essobi | R! |
| 06:39.10 | Essobi | I! |
| 06:39.14 | Essobi | Uhhh.. |
| 06:39.24 | marc32344 | according to essobi... it's an ongoing battle to keep it up an running. |
| 06:39.26 | |Vulture| | Essobi: Givme an A! |
| 06:39.30 | JerJer | hey a drunk cheerleader |
| 06:39.44 | Essobi | marc I didn't say that. I said depending on the deployment and the number of users |
| 06:39.50 | Essobi | there's always handholding |
| 06:40.03 | Essobi | and softswitch problems/bugs |
| 06:40.11 | |Vulture| | marc32344: if you have 200 users... your gunna have problems |
| 06:40.14 | Essobi | I'm out peeps |
| 06:40.18 | Essobi | Night all |
| 06:40.20 | |Vulture| | night |
| 06:40.23 | marc32344 | i plan to put 200users/ T1 |
| 06:40.36 | Damascene | |Vulture|: i plan on only using it for 4-5 users. i guess i wont' have issues then? ;) |
| 06:40.42 | |Vulture| | yea... end users always find a way to mess things up |
| 06:40.45 | |Vulture| | Damascene: lol |
| 06:40.51 | a1fa | hjm |
| 06:40.54 | a1fa | it just authed |
| 06:40.54 | a1fa | ;) |
| 06:41.00 | roamer323 | damascene - * is solid with 5 users |
| 06:41.06 | |Vulture| | the fewer end users... the more spaced out the problems |
| 06:41.11 | roamer323 | night essobi |
| 06:41.18 | Damascene | roamer323: okay cool, thanks |
| 06:41.45 | a1fa | sip is choppy ;) |
| 06:41.46 | a1fa | wtf |
| 06:42.28 | |Vulture| | Damascene: I have 6 servers with 5-20 users... they require little upkeep using TDM400P 4FXO with overflow on IAX2 |
| 06:42.36 | JerJer | sip isnt choppy |
| 06:42.45 | JerJer | it is the codec you are running using sip signalling |
| 06:42.49 | marc32344 | 5-20 users??? making any money?? |
| 06:42.50 | |Vulture| | JerJer: it is over a 14.4 :P |
| 06:42.58 | a1fa | what should i use |
| 06:43.01 | a1fa | what coded |
| 06:43.18 | JerJer | MPEG-4 |
| 06:43.22 | roamer323 | haha |
| 06:43.28 | a1fa | no |
| 06:43.31 | |Vulture| | marc32344: they are being used as office PBX systems... the $$ was in the installs and upkeep |
| 06:43.38 | a1fa | G723? |
| 06:43.39 | roamer323 | AVC |
| 06:43.46 | JerJer | Start with G.711 |
| 06:43.55 | JerJer | then try gsm, speex and iLBC |
| 06:45.21 | a1fa | this is nice |
| 06:45.28 | a1fa | too bad i cant recieve calls |
| 06:46.11 | marc32344 | Quad T1 on asterisk.... that must create lots of work. |
| 06:47.19 | |Vulture| | just channels... |
| 06:47.19 | a1fa | this is some funny shit |
| 06:47.22 | a1fa | i got a phone call |
| 06:47.25 | a1fa | but nothing happened |
| 06:47.51 | *** join/#asterisk pranav (pranav@202.149.48.205) |
| 06:48.03 | |Vulture| | what do you mean nothing happened? |
| 06:48.14 | a1fa | fwd test call |
| 06:48.27 | a1fa | my phone just rang.. but i guess thats normal |
| 06:48.43 | a1fa | i answered it.. and there wanst a voice saying this is just a test |
| 06:49.09 | *** join/#asterisk IsMe (some@211.24.146.10) |
| 06:49.14 | |Vulture| | if there is suppose to be.. its possible that your firewall isn't setup correctly |
| 06:49.22 | a1fa | yup |
| 06:49.38 | a1fa | there is no passthrough |
| 06:50.01 | a1fa | this is so funny |
| 06:50.03 | a1fa | ha hahaha |
| 06:50.05 | |Vulture| | ? |
| 06:50.09 | a1fa | the phone is ringing |
| 06:50.21 | a1fa | there you go |
| 06:50.24 | a1fa | it just ranggggg |
| 06:50.28 | a1fa | :) |
| 06:50.35 | pcm | you have the signalling working ... |
| 06:50.36 | pcm | hehe |
| 06:50.41 | |Vulture| | a1fa: you are easily ammused |
| 06:50.44 | a1fa | i am |
| 06:50.45 | a1fa | : |
| 06:50.47 | a1fa | :) |
| 06:50.47 | IsMe | has anyone done * with PABX ? |
| 06:50.58 | a1fa | this used to cost thousands of dollars |
| 06:51.11 | IsMe | Mavvie, can i pm u ? |
| 06:51.22 | a1fa | can anybody gimme a call on my sip? |
| 06:51.22 | Mavvie | only if you speak proper english. |
| 06:51.37 | a1fa | haha Mavvie u are such a nazi |
| 06:51.46 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 06:51.51 | Mavvie | a1fa: recht. |
| 06:52.06 | Mavvie | well that was a bad translation. |
| 06:52.23 | a1fa | ehhe deutschen leute |
| 06:52.33 | a1fa | who can gimme a call on my sip |
| 06:53.27 | |Vulture| | fwd? |
| 06:53.36 | a1fa | yup |
| 06:53.49 | marc32344 | ne1 knows what hardware packet8 is using? |
| 06:53.51 | |Vulture| | :( no fwd here |
| 06:54.04 | a1fa | you can still call me |
| 06:54.07 | *** join/#asterisk Capouch (501@12.176.248.4) |
| 06:54.12 | Beirdo | who is ne1? |
| 06:54.17 | a1fa | sip:number@fwd... |
| 06:54.32 | viLeR | somebody recommends some good linux asterisk client |
| 06:54.33 | |Vulture| | k msg me Ill do it |
| 06:56.47 | a1fa | wow |
| 06:56.50 | a1fa | i cant believe this works |
| 06:57.03 | |Vulture| | haha |
| 06:57.08 | a1fa | i am behind a packetshaper |
| 06:57.21 | |Vulture| | wait till you learn about T1 PRI :) |
| 06:57.29 | a1fa | nat, and |
| 06:57.37 | a1fa | 2nd vlan |
| 06:57.40 | a1fa | wow ;) |
| 06:57.50 | |Vulture| | hahaha thats where the latency was coming from |
| 06:57.54 | a1fa | |Vulture| you are my new best friend :) |
| 06:58.28 | *** join/#asterisk terraHome (~cjs@cpe-66-25-94-95.satx.res.rr.com) |
| 06:58.33 | |Vulture| | :) |
| 06:58.35 | a1fa | i am going to QOS these packets ;) |
| 06:59.11 | a1fa | vulture |
| 06:59.15 | a1fa | what is your number |
| 06:59.18 | terraHome | good evening |
| 06:59.19 | a1fa | i need to write this down :) |
| 06:59.31 | |Vulture| | I don't have a FWD # |
| 07:00.10 | |Vulture| | should prolly get one though.... just never bothered since I have a buncha DIDs |
| 07:00.48 | a1fa | fwd doesnt make outside phone calls |
| 07:00.50 | a1fa | that is bullshit |
| 07:00.54 | |Vulture| | lol |
| 07:01.17 | a1fa | i mean to other sip networks |
| 07:01.20 | |Vulture| | broadvoice my friend... $20 for unlimited LD+Most international |
| 07:01.23 | Beirdo | of course it doesn't |
| 07:02.14 | a1fa | nice |
| 07:02.59 | a1fa | iii likeee it |
| 07:04.14 | a1fa | |Vulture| : do you know a sip server (free) that will let users make phonecalls outside of their sip? |
| 07:04.35 | |Vulture| | not sure I follow |
| 07:04.43 | cool_blade | ASTERISK FREAKING ROCKS |
| 07:05.05 | marc32344 | cool_blade-- how many channels you're running? |
| 07:05.05 | a1fa | you know fwd.. i cant dial people 45555@outside.com, for example |
| 07:05.06 | |Vulture| | like you want to call another * server? |
| 07:05.12 | a1fa | yup |
| 07:05.20 | cool_blade | marc32344: 25 |
| 07:05.30 | marc32344 | a full t1 |
| 07:05.30 | |Vulture| | well if you want to do that, they have to setup an account for you, then you have to set it in your sip.conf |
| 07:05.38 | cool_blade | marc32344: well e1 |
| 07:05.38 | |Vulture| | but * to * I would recommend IAX2 not SIP |
| 07:05.53 | a1fa | i see |
| 07:05.56 | a1fa | what service do you use? |
| 07:05.58 | marc32344 | cool_blade-- how many users? |
| 07:06.09 | cool_blade | marc32344: about 40 |
| 07:06.24 | marc32344 | cool_blade-- thats not much |
| 07:06.25 | |Vulture| | a1fa: I use broadvoice and voicepulse connect |
| 07:06.48 | cool_blade | mac32344: yes i know |
| 07:07.04 | a1fa | nice :) |
| 07:07.05 | |Vulture| | a1fa: but I have TDM400Ps to bring in local POTS, and I am about to install a 2 T1 PRIs to run all the offices |
| 07:07.10 | cool_blade | marc32344: i'll start working right away to get the company to get more users |
| 07:08.30 | a1fa | sweet |
| 07:08.41 | a1fa | i am buying two TDM400s |
| 07:08.54 | a1fa | no FXOs atm |
| 07:09.16 | |Vulture| | a1fa: why buy the TDMs without the FXO or FXS cards? |
| 07:09.27 | a1fa | FXS |
| 07:09.31 | |Vulture| | ah |
| 07:09.51 | a1fa | its a *B.. i dont remember on top of my head |
| 07:10.04 | |Vulture| | I use IP500 phones, well I use to use 7960/40s but IP500s are better for the price |
| 07:10.15 | *** part/#asterisk Capouch (501@12.176.248.4) |
| 07:12.10 | a1fa | i use that grandstream handytone 486 |
| 07:12.15 | a1fa | i am very very impressed |
| 07:12.31 | cool_blade | good - because the older grandstream looked like a kids toy |
| 07:12.50 | a1fa | yeah.. it comes with a router |
| 07:12.53 | marc32344 | essobi--- you quit |
| 07:12.54 | marc32344 | ? |
| 07:13.03 | a1fa | and i only have one ethernet jack in my room |
| 07:13.09 | a1fa | so i can still have my computer |
| 07:16.27 | Beirdo | heheheh |
| 07:16.38 | Beirdo | Pierre Berton on how to roll a joint |
| 07:17.16 | Qwell | a1fa: 40b |
| 07:17.27 | a1fa | 40b? |
| 07:17.38 | Qwell | first digit is # of FXS ports, second digit is # of FXO |
| 07:17.42 | Qwell | 4 FXS ports is 40b |
| 07:17.50 | a1fa | yeah |
| 07:18.02 | *** join/#asterisk ckruetze (~ckruetze@i3ED6843F.versanet.de) |
| 07:18.04 | a1fa | <a1fa> its a *B.. i dont remember on top of my head |
| 07:18.08 | a1fa | 400B |
| 07:18.12 | a1fa | thats right |
| 07:20.32 | a1fa | heh |
| 07:20.38 | a1fa | i can call vonage from fwd |
| 07:20.46 | Qwell | yeah, pretty cool feature |
| 07:20.49 | a1fa | and packt8 |
| 07:21.08 | Qwell | Jeff Pulver co-founded Vonage, afaik |
| 07:21.13 | Qwell | so it makes sense that there is a link |
| 07:21.20 | Qwell | My boss couldn't get it working though...dunno |
| 07:21.21 | a1fa | :) |
| 07:21.27 | a1fa | vonage? |
| 07:21.48 | Qwell | yeah, he got an IP phone, connected it to fwd, and tried calling his house...didn't work |
| 07:21.52 | Qwell | (his house is a vonage account) |
| 07:21.59 | marc32344 | is vonage using as5300? |
| 07:22.28 | Qwell | vonage can call fwd too. its some long number...011393..etc |
| 07:22.40 | a1fa | yup |
| 07:22.55 | Qwell | fwd can also go to iaxtel |
| 07:22.58 | a1fa | **2431 Vonage 011 0 393 (w/intl. dialing enabled) |
| 07:23.07 | a1fa | whos got vonage here |
| 07:23.11 | a1fa | or packet8 |
| 07:23.14 | a1fa | i want to test this |
| 07:23.50 | marc32344 | ne1 knows what hardware packet8 uses? |
| 07:23.52 | a1fa | iptel? |
| 07:23.56 | marc32344 | for their network |
| 07:25.35 | *** part/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 07:26.20 | *** join/#asterisk h3x0r (Justino@ip68-108-176-196.lv.lv.cox.net) |
| 07:26.59 | a1fa | night |
| 07:27.08 | a1fa | i got class in 6h |
| 07:29.15 | Firestrm | anyone here ever deal with navigata? im concidering their PRI service.. |
| 07:36.00 | *** join/#asterisk chiko (~chiko@202.162.220.243) |
| 07:41.38 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 07:44.04 | *** join/#asterisk magg (~magg@80.203.139.96) |
| 07:44.50 | *** part/#asterisk BBRodriguez (~alex@pD95631BB.dip.t-dialin.net) |
| 07:50.38 | a1fa | too bad |
| 07:50.45 | a1fa | i cant schedule a wakeup call at 6 am |
| 07:50.49 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 07:53.00 | terraHome | evening shido |
| 07:53.14 | terraHome | or rather... |
| 07:53.16 | terraHome | good morning |
| 07:53.44 | a1fa | somebody wake me up |
| 07:53.46 | a1fa | at 7 CST |
| 07:53.55 | terraHome | eh |
| 07:54.55 | |Vulture| | a1fa: set a wakup call |
| 07:54.57 | |Vulture| | with * |
| 07:55.28 | a1fa | how? |
| 07:55.32 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 07:55.34 | sezuan | hi |
| 07:55.56 | a1fa | :) |
| 07:56.00 | a1fa | i am going to sleep |
| 07:56.13 | a1fa | i am going to have a hard time making to my 8 o'clock class |
| 07:56.15 | |Vulture| | http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20wake-up |
| 07:56.19 | sezuan | Slighty OT, can someone help me with a ENUM record? |
| 07:56.19 | |Vulture| | later |
| 07:56.51 | a1fa | |Vulture| : i dont have * yet.. its gonna be here next monday ;) |
| 07:58.21 | iceyp | whats required to insert a src ph number? In my master.csv there is no src |
| 08:07.46 | iceyp | mm |
| 08:07.53 | iceyp | everyone must be asleep ;( |
| 08:08.27 | *** join/#asterisk schurig (~schurig@p54B28804.dip0.t-ipconnect.de) |
| 08:09.58 | chiko | i try install asterisk with ata186 |
| 08:10.22 | chiko | any problem with that? |
| 08:10.44 | *** join/#asterisk qiu (~andrei@home-073519.b.astral.ro) |
| 08:13.57 | viLeR | chiko: no |
| 08:14.29 | viLeR | chiko: my ata186 with version 2.16 works fine. |
| 08:14.50 | chiko | my ata186 version 3.1 |
| 08:15.30 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
| 08:15.52 | Ron-Na | Anybody up and listening? |
| 08:16.05 | viLeR | chiko: fine |
| 08:16.14 | Ron-Na | I have SuSE 9.2 installed and got problems |
| 08:16.54 | Ron-Na | I swapped the Digium card from another server to this system and it gives me an error at modprobe wcfxs |
| 08:23.41 | djin | lspci sees the board? |
| 08:23.53 | djin | and did you do an 'modprobe zaptel' first? |
| 08:24.13 | Qwell | it would also help to know what the error is |
| 08:24.29 | iceyp | how do i force the 'source' in master.csv , i've tried adding callerid in sip aswell as fromuser, however my source from master.csv is still empty |
| 08:24.38 | djin | no Qwell, it the challenge to guess for the error. |
| 08:24.49 | Qwell | djin: Its too late for a challenge. :p |
| 08:25.10 | djin | :) |
| 08:25.30 | Qwell | and on that note... |
| 08:25.40 | rvhi | hi, is it possible to match multiple extension in one line in extensions.conf? |
| 08:25.52 | rvhi | for example, to match 492, 294, 234 |
| 08:25.54 | Qwell | rvhi: With wildcards and such, yes |
| 08:26.01 | Qwell | those 3, and ONLY those 3? |
| 08:26.11 | rvhi | more, just an example |
| 08:26.16 | djin | _XXX,1,etc. |
| 08:26.22 | Qwell | but, those X and only those X? |
| 08:26.22 | rvhi | (492|294|234)? |
| 08:26.40 | Qwell | ie, you want to match 123, but not 321? |
| 08:26.45 | rvhi | right |
| 08:26.53 | Qwell | thats beyond me then, heh |
| 08:27.19 | djin | rvhi, there must be some simularity that identifies them. |
| 08:27.27 | Qwell | 123,1,Goto(abc,1) |
| 08:27.32 | Qwell | 234,1,Goto(abc,1) |
| 08:27.33 | Qwell | dunno |
| 08:27.41 | rvhi | there is no similarity in this case |
| 08:27.50 | rvhi | i have dozens of numbers ported to my pri |
| 08:27.52 | djin | that you cannot combine them. |
| 08:28.12 | rvhi | they are all unique 7 digit numbers |
| 08:28.32 | Qwell | 123,1,SetVar(calledexten=${EXTEN}) 123,2,Goto(abc,1) |
| 08:28.37 | Qwell | 234,1,SetVar(calledexten=${EXTEN}) 234,2,Goto(abc,1) |
| 08:28.41 | Qwell | something like that should work...in theory |
| 08:29.21 | Qwell | hell, make it a simple macro |
| 08:29.28 | djin | rvhi, I'd guess that they are all unique . . . |
| 08:29.53 | rvhi | macro seems to be the easiest way |
| 08:30.10 | djin | but without overlapping simularities, you're scrwewed. |
| 08:30.10 | rvhi | does realtime extension support macro? |
| 08:30.20 | Qwell | off to bed |
| 08:30.30 | Qwell | /detach |
| 08:30.33 | djin | Qwell, night night. |
| 08:30.38 | rvhi | thx, good night! |
| 08:31.47 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:32.18 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 08:32.58 | libpcp | hi all |
| 08:33.14 | libpcp | is anyone can help me with TE410P? |
| 08:34.11 | djin | state your problem. |
| 08:34.27 | djin | then we can see if we can help you. |
| 08:37.43 | *** join/#asterisk meppl (~mephisto@pD9542F94.dip.t-dialin.net) |
| 08:37.45 | meppl | guten morgen |
| 08:39.24 | libpcp | i would like to ask if its possible to configure the TE410P even without the real connected from ISDN provider? |
| 08:39.58 | djin | sure, you can test bij using crosscable from one port to another. |
| 08:40.14 | djin | and configure zapata.conf properly, off course |
| 08:44.35 | libpcp | djin: ah so i really need to loop back the E1 port so I could test it |
| 08:45.23 | libpcp | i thought i could configure the zapata.conf and zaptel.conf and run the asterisk without a real connection or loopback test |
| 08:46.32 | libpcp | djin: is that the reason why im getting Feb 22 16:48:53 ERROR[3870]: Unknown signalling method 'pri_cpe' because the E1 ports are not connected or not being loopback? |
| 08:57.54 | djin | yes, it is. |
| 08:58.08 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:58.46 | djin | You need a PRI crosscable and to configure one port as 'pri_net'. |
| 09:06.52 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
| 09:07.27 | IronHelix | anyone else having broadvoice issues tonight? |
| 09:09.06 | *** join/#asterisk Newbie___ (some@211.24.146.10) |
| 09:10.08 | ta[i]nted | IronHelix i'm always having broadvoice issues |
| 09:10.24 | *** join/#asterisk Beirdo_ (~gjhurlbu@beirdo.user) |
| 09:10.35 | ta[i]nted | did you try call tech support? |
| 09:10.46 | ta[i]nted | call(ing) |
| 09:10.48 | IronHelix | nah, i just signed up today |
| 09:11.13 | IronHelix | i've been trying various ways of doing sip.conf |
| 09:11.53 | IronHelix | but the only one that seems to work is register -> (info) @proxy.chi.broadvoice.com with host=proxy.chi.broadvoice.com |
| 09:12.01 | IronHelix | the others either dont register, or dont work |
| 09:13.03 | IronHelix | it was working fine earlier, and i figured it *might* be on my end because between then and now i upgraded asterisk to 1.0.5, just checked that it compiled ok and then went off to dinner |
| 09:13.46 | magg | quit |
| 09:13.50 | magg | ops ;D |
| 09:14.13 | IronHelix | (btw, the broadvoice chan_sip patch got merged into asterisk release right? there was a very brief note saying it had on voip-info.org with no details and it was dated November, so i figured it had) |
| 09:17.12 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 09:22.54 | Ron-Na | SuSE 9.2 recognizes digium card TDM22B as "Communication controller: TigerJet Network Inc. Tiger 3XX Modem/ISDN interface" How can I fix that!!!! |
| 09:23.14 | Inv_arp | IronHelix: follow these directions http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 09:23.40 | Inv_arp | i had a hard time with broadvoice until i followed this page |
| 09:24.19 | IronHelix | ron-na- thats fine. just install the zaptel package |
| 09:24.33 | IronHelix | inv-arp- wow thats an ugly hack, but i'll give it a shot :) |
| 09:25.48 | Ron-Na | ok, modprobe zaptel is going in but modprobe wcfxs gives me an error!!! |
| 09:26.53 | IronHelix | if you already did make and make install for zaptel |
| 09:26.55 | IronHelix | also do make config |
| 09:27.00 | Ron-Na | ztcfg -vvv gives me the channel map Channel 01 ~ 04, but than also: "ZT_CHANCONFIG failed on channel 1: No such device or addrss (6) |
| 09:27.01 | IronHelix | that sets it up as a service |
| 09:27.11 | IronHelix | hmmm |
| 09:27.31 | IronHelix | i had a similar problem with a fxo... |
| 09:27.52 | Ron-Na | I have done cvs downloaded and used make clean; make update; make install |
| 09:28.24 | Ron-Na | module wctdm unsupported by SUSE/Novell, tainting kernel. |
| 09:28.50 | Ron-Na | wctdm: disagrees about version of symbol zt_receive and ... |
| 09:28.50 | fishboy1669 | morning peeps |
| 09:29.39 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 09:31.21 | Ron-Na | IronHelix what do you mean with "make config" ? (I did a make cloneconfig in /usr/src/linux before I worked on zaptel) |
| 09:31.28 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 09:32.08 | IronHelix | make config installs a service in rc.d |
| 09:33.19 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:33.52 | CMike | anyone in here heard anything about grandstream and different indication tones / Callerid FSK/DTMF ? |
| 09:34.32 | CMike | I know they were taking about is as uppcoming in new firmware last year.. But I haven't heard anything |
| 09:35.49 | *** join/#asterisk Mike_TK (~Mike_@213.180.245.62) |
| 09:36.31 | IronHelix | amazing |
| 09:36.37 | IronHelix | it seems to now be working on BOS |
| 09:36.43 | IronHelix | :wtf: |
| 09:37.20 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 09:41.16 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 09:41.19 | Ron-Na | IronHelix do you have another hint for me to go a step further ;-) |
| 09:41.39 | IronHelix | im tryin to remember what i did to fix it |
| 09:41.52 | IronHelix | although im not sure it would help as im on fedora |
| 09:42.00 | IronHelix | i think it had something to do with the startup scripts |
| 09:42.05 | IronHelix | ie modprobe vs insmod |
| 09:42.09 | IronHelix | for inserting modules |
| 09:42.59 | IronHelix | try googling your error |
| 09:43.03 | IronHelix | google spiders the digium list |
| 09:43.09 | IronHelix | or google it with site:lists.digium.com |
| 09:44.52 | IronHelix | ooh |
| 09:45.04 | IronHelix | http://www.voip-info.org/wiki-Asterisk+Linux+SuSE try that |
| 09:51.11 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:53.43 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 09:53.51 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 09:58.14 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 09:58.57 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:58.58 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 10:02.30 | Zeeek | où suis-je |
| 10:04.14 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 10:07.11 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:07.25 | Zeeek | no one cares :( |
| 10:08.16 | chiko | is ata186 only sccp? |
| 10:08.28 | Mike_TK | chiko: no |
| 10:08.56 | Mike_TK | chiko: h323,sip too |
| 10:09.25 | chiko | how to make sip? |
| 10:09.32 | RoyK | make sip? |
| 10:09.35 | RoyK | as in make love? |
| 10:09.59 | chiko | :) |
| 10:10.49 | IronHelix | hehe |
| 10:11.09 | IronHelix | if you have to get new firmware- good luck |
| 10:11.31 | chiko | i must upgrade?? |
| 10:11.32 | IronHelix | from what i hear, getting cisco firmware subscriptions is only slightly easier than swimming through lava |
| 10:11.37 | IronHelix | not sure |
| 10:11.40 | IronHelix | pray you dont |
| 10:14.46 | *** join/#asterisk qiu (~qiu@andrei.digicom.ro) |
| 10:18.16 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 10:19.03 | Zeeek | . |
| 10:21.02 | *** join/#asterisk pranav (pranav@202.149.48.216) |
| 10:21.42 | pranav | hi |
| 10:21.53 | *** join/#asterisk idnar (mithy@idnar.user) |
| 10:21.59 | Zeeek | hi |
| 10:22.24 | idnar | is there any way for me to increase the connection timeout on outgoing calls over IAX? Asterisk seems to be giving up after 1000ms |
| 10:24.39 | pranav | i am trying to connect to a server , i registered in it, but it says "wrong password on authentication for REGISTER |
| 10:24.47 | *** join/#asterisk codebreaker (~codebreak@flexserv.de) |
| 10:25.15 | pranav | its my own another server which i am trying to connect it to |
| 10:26.26 | pashah | pranav: looks like you are registering with wrong password =) |
| 10:26.33 | codebreaker | hello how to change exten => _99.,1,Dial(CAPI/9420576:${EXTEN},30,r) so that dialed number 990123456 get outbound dialed 0123456? i know the was something with inserting a 2 but i didnt rember |
| 10:26.40 | pranav | no i selected a random password |
| 10:26.57 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:27.06 | idnar | codebreaker: use ${EXTEN:2} |
| 10:27.17 | codebreaker | idnar: thanks |
| 10:27.45 | pranav | which password do i have to put in, actually i tried few passwords, but in all i was getting this error |
| 10:28.20 | kks | i always get dialstatus=CONGESTION, if no such sip account and the peer is not registered. when i will able to get dialstatus=CHANUNAVAIL? |
| 10:31.04 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 10:36.22 | pranav | tell me what should i do? |
| 10:38.25 | |Vulture| | is it possible to limit the number of voice channels that a single DID can occupy on a T1 PRI? |
| 10:39.43 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 10:39.50 | djin | You want to limit incoming calls for one specific DID? |
| 10:40.20 | |Vulture| | yes so say that the DID can only take 10 of the 23 channels |
| 10:40.27 | |Vulture| | I just wanted to know if it is possible |
| 10:41.02 | RoyK | does asterisk support video conferencing over SIP? |
| 10:41.10 | djin | You can limit things in your extensions. It's not so much tot limit physically, but more what happens to the 11th caller. |
| 10:41.21 | |Vulture| | ah I see |
| 10:41.51 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 10:41.53 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 10:42.18 | pranav | do i need to put the password of the server |
| 10:42.21 | RoyK | deaf customers want video conferecing.... |
| 10:42.22 | |Vulture| | PRI is deff. the thing to do... |
| 10:42.22 | RoyK | hm |
| 10:42.55 | djin | I have a SIP image v7.3, but require v6.3 first to upgrade from 4.x. Can anyone help my on this image? |
| 10:43.05 | djin | FOr the Cisco 7940, that is. . . |
| 10:43.37 | RoyK | djin: app_groupcount |
| 10:44.08 | djin | RoyK, I'm not sure what you mean. |
| 10:44.35 | RoyK | djin: look up app_groupcount on the wiki |
| 10:44.44 | RoyK | that can do all sorts of limitations |
| 10:45.18 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 10:45.21 | zoa | yooo |
| 10:45.29 | djin | Are you sure you're answering my question instead of Vulture's? |
| 10:45.37 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 10:46.29 | |Vulture| | he already answered mine |
| 10:47.26 | pranav | when i am trying to register to a sip server,wrong password on authentication for registration |
| 10:47.42 | pranav | it says,wrong password on authentication for registration |
| 10:47.59 | djin | yes pranav, and what does that tell you? |
| 10:48.23 | pranav | i tried with another password , but again the same thing comes |
| 10:48.50 | djin | yes pranav, and what does that tell you? |
| 10:49.12 | pranav | do i need the password of that sip server |
| 10:49.26 | pranav | to whom i am trying to register |
| 10:49.36 | djin | that would be my guess. |
| 10:50.24 | Zeeek | pranav still trying with FWD? |
| 10:50.31 | [ro]nic3try | is possible to ask for a passwd before making a call ? |
| 10:50.47 | djin | Authenticate(1234) |
| 10:50.59 | [ro]nic3try | doc's ? |
| 10:51.04 | djin | ocs |
| 10:51.08 | djin | ~docs |
| 10:51.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 10:51.26 | [ro]nic3try | :) ok |
| 10:52.09 | djin | Trust me, it's all there ;) |
| 10:53.37 | pranav | i tried that but i am getting the same error |
| 10:55.23 | pranav | i have made the entry in the sip.conf using register => user:password@sipprovider.com |
| 10:55.30 | codebreaker | how can i make all incoming calls from capi ringing/directing to one user/iaxphone. |
| 10:55.58 | djin | are you using chan_capi? |
| 10:56.15 | codebreaker | djin: yes |
| 10:56.44 | codebreaker | djin: but i also have some hosts with hfc cards. so its more a general question. |
| 10:57.13 | codebreaker | i am thinking about something exten => s,1,call user |
| 10:57.18 | pranav | zeeek: for some time i have stopped to work with fwd, i am trying to connect to my own sip server |
| 10:57.31 | djin | In the capi context in extensions put something like exten => _X.,Dial(IAX. . . . . |
| 10:57.51 | djin | You're almost right, only wrong about the 's'. |
| 10:58.06 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 10:58.10 | djin | and I missed the '1' :) |
| 10:58.34 | codebreaker | djin: without the s? |
| 11:02.56 | djin | change it something like _X. that handle all incoming MSN's |
| 11:03.20 | djin | Nobody calls 's' ;) |
| 11:07.45 | qiu | hi guys ... i try to compile asterisk with chan_h323 |
| 11:07.52 | qiu | and i have some errorr |
| 11:08.21 | qiu | with cvs : chan_h323.o(.data.rel+0x40): undefined reference to `h323_show_codec' |
| 11:08.43 | qiu | and with stable 1.0.5 : make[1]: *** No rule to make target `h323/libchanh323.a', needed by `chan_h323.so'. Stop. |
| 11:09.05 | qiu | does anyone compile succesfuly chan_h323 ? |
| 11:10.07 | qiu | (i have : pwlib: v1.8.1 and openh323 v.1.15.1) |
| 11:11.40 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 11:11.41 | pranav | djin: i tried giving the password of that server still i am getting the same error |
| 11:12.06 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 11:13.12 | qiu | anyone with chan_h323 ? |
| 11:13.22 | RoyK | ~h323 |
| 11:13.23 | jbot | well, h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 11:13.43 | RoyK | asterisk + h323 = spaghetti |
| 11:14.13 | qiu | well ... i need h323 |
| 11:14.20 | clive- | qui buy cisco |
| 11:14.42 | zoa | hello there all of you |
| 11:15.07 | clive- | zoa, hows the jitter bufferring going |
| 11:15.13 | qiu | clive : well ... why not huawey |
| 11:15.18 | qiu | ? |
| 11:15.25 | zoa | dunno |
| 11:15.31 | zoa | looks like there is some persistant rumor |
| 11:15.34 | clive- | qui never used huaw, |
| 11:15.44 | qiu | but i need to compile asterisk with chna_h323 |
| 11:16.08 | qiu | clive : why ? do you used it ? |
| 11:16.15 | clive- | qiu use * for ivr and stuff, and cisco for h323 |
| 11:16.47 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.61) |
| 11:16.49 | qiu | well ... i need h323 only for compatibility |
| 11:16.56 | qiu | for small things |
| 11:16.57 | *** join/#asterisk FocusRite (~fbg@81.6.195.82) |
| 11:17.16 | shadebob | Hi, someone known if RHINO channel bank have ISDN fxs option? |
| 11:17.18 | eipi | about digium boards: its true that if i dont have a mboard that doesnt support pci 2.2, the digium cards wont work? |
| 11:17.55 | FocusRite | hi folks. does anyone have experience using asterisk with pika boards (if it works with pika gear that is) ? |
| 11:18.51 | eipi | about digium boards: its true that if i dont have a mboard that support pci 2.2, the digium cards wont work? |
| 11:22.40 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 11:22.54 | *** join/#asterisk brazil (~cleber@200.198.105.37) |
| 11:23.14 | brazil | hi all! |
| 11:24.00 | RoyK | hi |
| 11:28.41 | Zeeek | Hej! |
| 11:29.24 | PoWeRKiLL | hi |
| 11:29.47 | PoWeRKiLL | How can I permit all incoming SIP traffic from a based on a IP ? |
| 11:31.17 | eipi | http://www.voip-info.org/wiki-Asterisk+sip+permit-deny-mask |
| 11:32.25 | eipi | powerkill search the same words on google and you will find that url |
| 11:33.31 | *** join/#asterisk karman (~karman@196.46.71.170) |
| 11:34.59 | karman | hallo all |
| 11:35.10 | karman | anyone got any idea how e&m works? |
| 11:35.25 | karman | or how to get it to work ;-) |
| 11:41.51 | RoyK | e&m? |
| 11:41.59 | RoyK | as in S&M? |
| 11:43.47 | Zeeek | ok, I'm back. Who ordered the caramel chicken? |
| 11:45.57 | RoyK | Zeeek: wot? |
| 11:46.08 | Zeeek | I guess I'll have to eat it myself |
| 11:46.26 | clive- | . |
| 11:46.30 | RoyK | , |
| 11:46.31 | Zeeek | .. |
| 11:46.45 | RoyK | \/ |
| 11:47.09 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 11:47.23 | RoyK | * * |
| 11:47.23 | RoyK | |_| |
| 11:48.02 | Zeeek | <--------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^--------------------> |
| 11:48.38 | karman | e&m wink |
| 11:48.58 | karman | ok, let me put this way.. i need asterisk to send the dialed number to the pbx |
| 11:49.24 | karman | current configL *fxo ---> co-line pbx |
| 11:49.49 | Zeeek | drums stop. no good. |
| 11:51.23 | Zeeek | in the early part of the last century there was a large wave of Norwegian immigration in the US |
| 11:51.33 | Zeeek | they all ened up in Minnesota |
| 11:52.05 | Zeeek | I guess immigration officers figured they wouldn't mind the cold |
| 11:54.45 | karman | mmm.. seems like everyone is fast asleep |
| 11:55.00 | Zeeek | or worse |
| 11:56.04 | Zeeek | I hate anonymous callerid |
| 12:01.44 | djin | <PROTECTED> |
| 12:02.49 | Zeeek | these Cisco SIP images seem to be a real PITA to deal with |
| 12:03.00 | Zeeek | I think I may buy a Poly |
| 12:03.56 | djin | Zeeek, Cisco's are good phones. |
| 12:04.28 | djin | Only firmware upgrade takes soem getting used to. |
| 12:04.29 | Zeeek | ya but if you have to beg and steal every time you need an image... |
| 12:04.33 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.249) |
| 12:04.36 | Shrink | djin, I've been through the flashing process once, a couple of months ago, it was just a case of persevering |
| 12:05.06 | djin | Zeeek, I didn't steal 7.3. Our password is home sick ;) |
| 12:05.28 | djin | Schrink, did you upgrade to 7.3? |
| 12:05.58 | Zeeek | dunno, just what I've observed here and elsewhere |
| 12:06.23 | *** join/#asterisk negativecreep (~yama@202.147.174.98) |
| 12:06.51 | negativecreep | hi all |
| 12:07.39 | negativecreep | can i connect a X100P to an office pbx.. |
| 12:07.54 | negativecreep | will * pick up the call if respective extension is dialed...? |
| 12:08.13 | djin | If * receives the request, yes. |
| 12:08.52 | negativecreep | djin: there is no special configuration required for that right?? |
| 12:08.58 | *** part/#asterisk Pkunk (tsvatr@mbbs.munnabhai.info) |
| 12:14.12 | [ro]nic3try | i'm tring to redirect my calls to a ser server.. don't quite understand how does that works |
| 12:14.16 | karman | anyone still alive in this place? |
| 12:14.32 | Delvar | no im dead |
| 12:14.39 | CMike | me too |
| 12:14.40 | CMike | :) |
| 12:14.57 | Delvar | ~dead |
| 12:14.58 | jbot | yes :( |
| 12:15.00 | FocusRite | does anyone have experience using asterisk with pika boards (if it works with pika gear that is) ? |
| 12:15.04 | karman | thought so. |
| 12:15.05 | Delvar | see everyones dead |
| 12:15.14 | karman | cause not getting any responce ;-( |
| 12:15.15 | Mike_TK | I was never alive. |
| 12:15.30 | Zeeek | no one has ever heard of e&m only S&M |
| 12:15.33 | [ro]nic3try | if i sai exten => _9XXXX,1,dial(sip/username:pass@ser_server) .. doesn't work |
| 12:15.40 | Delvar | i ws alive this morning, then i had some breakfast then i died |
| 12:15.48 | FocusRite | i'm dreaming this and when i wake up the chan will be lively and full of wisdom |
| 12:15.57 | Mike_TK | [ro]nic3try: and where is your dial number? |
| 12:16.09 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 12:16.09 | Mike_TK | [ro]nic3try: and it will never work, you must define peer |
| 12:16.14 | karman | ok... so e&m is unknow |
| 12:16.17 | Samoied | Hello! |
| 12:16.21 | karman | so lets try other technology |
| 12:16.29 | Zeeek | e&j Gallo |
| 12:16.43 | karman | what technologies are able to detect/send number dialled? |
| 12:16.46 | karman | for DID |
| 12:17.01 | Mike_TK | karman: e&m? ear and mouth |
| 12:17.10 | [ro]nic3try | in sip.conf.. if i say register=> user:passwd@ser_server / 1234 .. then ... how should my dialplan look ? |
| 12:17.10 | RoyK | karman: SetCallerID? |
| 12:17.11 | RoyK | :P |
| 12:17.14 | Samoied | I have a tdm04B |
| 12:17.19 | eipi | mike i read the same... acronym finder? |
| 12:17.20 | Samoied | with 4 fxo ports |
| 12:17.31 | [ro]nic3try | to me it looks like i'm calling back myself :( |
| 12:17.39 | Delvar | [ro]nic3try: have you looked on voip-info ? |
| 12:17.44 | Mike_TK | karman: No, it's special interface type |
| 12:17.56 | karman | ok, callerid.. do in need to do, setcallerid = ${exten}? |
| 12:18.02 | Mike_TK | karman: Hmm, what was a question? |
| 12:18.27 | Samoied | So, why this have a RJ-45 jack? |
| 12:18.36 | [ro]nic3try | yes, but i don't understand how do i send the number to server |
| 12:18.52 | Samoied | Is possible to use more than 2 pins? |
| 12:19.11 | Delvar | [ro]nic3try: ah, it snot like iax where you can do IAX2/user@entity/number |
| 12:19.27 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 12:19.40 | Delvar | [ro]nic3try: you have to have an accoun tin sip.conf registered to the corect server, then do a dial to SIP/number@entity |
| 12:20.17 | Delvar | [ro]nic3try: or SIP/entity/number if you prefer, they both do the same thing |
| 12:20.36 | [ro]nic3try | so.. i register in sip.conf.. then i dial number@ser_server in extension.conf ? |
| 12:20.50 | Delvar | [ro]nic3try: exactly |
| 12:21.26 | [ro]nic3try | ohh.. thx |
| 12:22.54 | eipi | about digium boards: its true that if i dont have a mboard that support pci 2.2, the digium cards wont work? |
| 12:23.23 | eipi | about digium boards: its true that if i dont have a mboard that support pci 2.2, the digium cards wont work? |
| 12:25.45 | karman | eipi.: that is true |
| 12:26.12 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 12:26.50 | eipi | then i have to throw my p2 233mhz? :( |
| 12:28.45 | karman | i'm not sure about the x100p |
| 12:28.47 | karman | that might work |
| 12:28.53 | karman | the others i know wont work |
| 12:30.27 | Zeeek | I think the X1200 may work in older boxes |
| 12:30.27 | Zeeek | errr x100 |
| 12:30.27 | Zeeek | it worked find in my old PIII |
| 12:30.27 | Zeeek | the TDM400 did not |
| 12:30.35 | RoyK | anyone that knows a softphone that works with SIP and video? |
| 12:30.44 | djin | xten |
| 12:30.47 | eipi | eyebeam |
| 12:30.50 | eipi | xten |
| 12:30.57 | djin | eyebeam |
| 12:31.29 | *** join/#asterisk dotcoder (~dotcoder@81.88.192.7) |
| 12:31.43 | zoa | tsss |
| 12:33.30 | *** part/#asterisk dotcoder (~dotcoder@81.88.192.7) |
| 12:35.13 | CMike | hehe.. |
| 12:35.34 | CMike | I think I have a cideoconf unit somewhere.. for ISDN .. a Sony I think |
| 12:35.42 | CMike | maybe I should play with that.. |
| 12:35.48 | CMike | videoconf. unit, even |
| 12:36.25 | CMike | BBL |
| 12:38.31 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 12:43.55 | tih | About the PCI 2.2 question just discussed: the point is, if I |
| 12:44.10 | tih | have understood correctly, that you have to choose the right card. |
| 12:44.36 | tih | If you have PCI 2.2 and interrupt chaining, you can use the 3.3v boards, and get better performance. |
| 12:44.53 | tih | In older hardware, you have to use the 5v boards, and accept that the CPU will have to work harder. |
| 12:48.03 | tzanger | tih: huh? |
| 12:48.14 | tzanger | PCI 2.2 doesn't say shit about 3.3v or 5v cards |
| 12:48.34 | tzanger | I have MANY PCI2.2-compliant systems that are 5V PCI only (which pisses me off to no end) |
| 12:49.32 | tih | Really? |
| 12:49.50 | tih | Not currently sold systems, surely? |
| 12:51.28 | PoWeRKiLL | how can i reload cdr_mysql while in a call |
| 12:52.51 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 12:53.03 | pr0m | morning. |
| 12:53.15 | tih | The TE405P should work in any PCI bus, at a performance cost. The TE410P demands a modern, PCI 2.2, 3.3v, interrupt-chaining, PCI bus -- and will be more efficient. Right? |
| 12:53.30 | pr0m | i'm following the various newsgroup postings online about the pap2 being locked to vonage. |
| 12:53.42 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 12:53.46 | *** join/#asterisk benno2 (~benno2@host194-15.pool80182.interbusiness.it) |
| 12:53.48 | pr0m | has anyone found a solution to unlock the linksys pap2? |
| 12:55.08 | benno2 | zaphfc with hfc-s , bristuff 0.2.0RC5 asterisk stable. every few secs I get this message (running * with high verbosity): received TEI check request for TEI = 67 ; Feb 22 13:53:50 WARNING[3836]: chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 |
| 12:55.17 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 12:55.42 | benno2 | any idea ? the card seems working perfectly |
| 12:56.20 | *** join/#asterisk akrall (~akrall@201.129.249.161) |
| 12:57.31 | akrall | Hi Guys.... I have a question.. anybody had problems with line noise on x100p's? I can perfectly call anybody but after 4 or so minutes of talking, line noise comes in and I have to hangup and call again.. |
| 12:58.24 | Essobi | Mmm. |
| 12:58.44 | akrall | yea.. its weird.. this happens wether I placed the call or the call came in |
| 12:58.49 | Essobi | What's the X100P? the single FXO port or the single T1? I never can remember. |
| 12:59.17 | Essobi | That and I havn't had coffee yet. |
| 12:59.44 | Essobi | Is it a know-off FXO? |
| 12:59.52 | Essobi | Knock-off card? |
| 13:00.55 | akrall | yep, x100p clone... a lot of people have been using them and they have worked great so far |
| 13:00.59 | akrall | why you ask? |
| 13:01.01 | Essobi | Sounds like you might have a ground loop problem on your machine and FXO line. You have a two-to-three prong adapter anywhere inline in your machine.. monitor, pc, powerstrip, anywhere near? |
| 13:01.17 | Essobi | akrall cause the clones are all different and all ... well different. |
| 13:01.35 | akrall | prong adapter? |
| 13:01.58 | akrall | prong? |
| 13:02.35 | Essobi | you know how a standard pc cable has 3 prongs |
| 13:02.40 | Essobi | 2 AC and 1 ground |
| 13:02.56 | akrall | ok, I know what you mean.. I just never heard the word prong |
| 13:03.02 | Essobi | tehe |
| 13:03.14 | akrall | so you ask if I have any of those near the computer? |
| 13:03.18 | Essobi | PRONG. |
| 13:03.25 | akrall | well yea... a lot :) about 5 or so around |
| 13:03.43 | Essobi | Yea.. any on the machine or monitor, printer, phone, etc, on the same plug or circuit as that plug? |
| 13:03.50 | Essobi | HAHA. there you go man. |
| 13:04.15 | Essobi | Get rid of those and try again. |
| 13:04.31 | Essobi | You're eliminating the ground from your PCs grounding path. |
| 13:04.38 | akrall | really? can those be causing the line interference? |
| 13:04.46 | Essobi | so it sees the phone line and goes.. CRUNCHY! I GOT GROUND! |
| 13:04.48 | akrall | but why exactly after 4 or so minutes of talk time? |
| 13:05.07 | Essobi | electrical resistance one would assume. |
| 13:05.18 | akrall | damn! |
| 13:05.18 | Essobi | you're lucky nothing went up in smoke. Heh. |
| 13:05.26 | akrall | worth taking a look... what do you recommend? |
| 13:05.35 | Essobi | 3 prongs |
| 13:05.38 | Essobi | Heh. |
| 13:05.46 | Essobi | You need a ground. |
| 13:05.48 | akrall | moving every away from the computer and pluging the computers on their own outlets? |
| 13:06.15 | Essobi | well that helps too sometimes.. but sometimes "away" isn't away. it's on the same circuit in the house. |
| 13:06.21 | akrall | the asterisk computer is conected to a no break (all 3 prong outlets) and that in turn is connected |
| 13:06.27 | akrall | to thewall outlet which has ground |
| 13:06.39 | akrall | but, I have a lot of those outlet bars around.. |
| 13:06.46 | Essobi | Hmm. |
| 13:06.52 | *** part/#asterisk didz_ (didz_@200.218.192.52) |
| 13:07.30 | Essobi | yuo should have ground then if the box and the monitor and all components do |
| 13:07.33 | Essobi | printer and etc |
| 13:07.46 | akrall | the noise effect I get is funny... your can talk for 4 minuites without echo .. no problems.. and then.. a loud |
| 13:08.16 | Essobi | ? |
| 13:08.18 | akrall | interference noise kicks in.. sometimes I can hear the other person between the noise but they can hear me.. sometimes I cant hear anything at all and they also hear the noise |
| 13:08.31 | Essobi | what's in sound like? |
| 13:08.59 | Essobi | make sure your card is seated good too.. while the machine is off obviously. ;) |
| 13:08.59 | akrall | a very loud hiss noise, like static noise of ham radio.. |
| 13:09.08 | akrall | hahahaha |
| 13:09.16 | akrall | thats one of the first things I did |
| 13:09.27 | Essobi | White noise.. Yea, I doubt that's anything software.. atleast one wouldn't think it was. |
| 13:09.33 | Essobi | you might have gotten a bad card. |
| 13:09.36 | Essobi | *SHRUG* |
| 13:09.45 | *** join/#asterisk Abbas (Abbas@203.81.194.242) |
| 13:09.50 | Abbas | Hi |
| 13:09.52 | Essobi | I got to hop in the shower, then get the kids ready for school. LAter on all. |
| 13:10.12 | Abbas | can we configure Linksys PAP2 to dial Ip to IP |
| 13:10.22 | akrall | could be.. I have 2 in place.. Im going to do more testing to see if there is some pattern |
| 13:10.29 | tzanger | tih: untrue |
| 13:10.35 | akrall | l8r essobi |
| 13:10.37 | akrall | thx |
| 13:10.58 | tzanger | the TE405P will ONLY work in a 5V PCI slot and is (from a logical point of view) identical in function to the TE401P |
| 13:11.00 | akrall | guys.. anybody configured their * for FWD using IAX2? |
| 13:11.16 | tzanger | tih: no digium card supports interrupt chaining |
| 13:11.25 | tzanger | it *may* work but it is certainly not recommended |
| 13:12.12 | tzanger | I have a TE405P and had to physically modify it to fit in a 3.3V PCI slot: www.mixdown.ca/~asterisk |
| 13:13.26 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 13:18.55 | akrall | guys.. anybody configured their * for FWD using IAX2? |
| 13:24.19 | [ro]nic3try | Delvar: i still canot make my call work, i registered in sip.conf, and in extension.conf, i have exten => 778989,1,Dial(SIP/778989/ser_server,20) , and doesn't work |
| 13:24.37 | [ro]nic3try | it says that canot find host 778989 |
| 13:24.51 | Delvar | you got it the wrong way round |
| 13:25.02 | Delvar | exten => 778989,1,Dial(SIP/ser_server/778989,20) |
| 13:25.09 | [ro]nic3try | upz :( |
| 13:25.18 | Delvar | or, exten => 778989,1,Dial(SIP/778989@ser_server,20) |
| 13:26.28 | [ro]nic3try | man... your gold |
| 13:26.44 | [ro]nic3try | :) |
| 13:26.53 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:27.12 | [ro]nic3try | now it's working.. thx a lot :-* |
| 13:27.20 | [ro]nic3try | }{}{}{ |
| 13:28.46 | Ron-Na | How to add 100 voicemailboxes at once???? |
| 13:29.04 | epoch | write a script! |
| 13:29.30 | Ron-Na | maybe somebody has already written the script - hehehehe |
| 13:29.41 | epoch | well yeah, look in contrib/scripts/ ;P |
| 13:29.47 | epoch | but that one's only good for one at a time |
| 13:29.53 | epoch | you could modify it very easily |
| 13:30.07 | ManxPwr | ~docs |
| 13:30.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:30.14 | Ron-Na | I saw it once in the list, ... but I have not found it anymore |
| 13:31.35 | slePP | anyone know how to get IE to stop expanding a <div> just because there's a float inside it that is larger than the div? |
| 13:41.58 | *** join/#asterisk whui (~whui@202.55.45.34) |
| 13:42.54 | *** join/#asterisk gabb0 (~gabb0@indo1.indosoft.unb.ca) |
| 13:43.00 | gabb0 | hello all |
| 13:44.29 | gabb0 | I know how to do voicefile conversions with sox but I'm having troubles finding the right settings to convert a file generated from the monitor app. The file is pcm and I want to convert it to either a wav or mp3 |
| 13:44.44 | gabb0 | has anyone here done this |
| 13:44.58 | ManxPwr | gabb0, Yes. And they have documented it on the Wikit |
| 13:45.26 | gabb0 | I've looked on the wiki and didn't see it for pcm |
| 13:47.08 | gabb0 | plus I can't get to the wiki for obvious reasons |
| 13:48.13 | vaewyn | obvious reasons? ... it seems to be working just fine |
| 13:49.04 | gabb0 | oops, obvious to me I guess. It's on my end |
| 13:49.22 | gabb0 | what is the ip addr of the wiki, my dns is down at the moment |
| 13:49.45 | vaewyn | 66.151.54.101 |
| 13:50.18 | gabb0 | thanks |
| 13:54.30 | tzanger | morning |
| 13:56.30 | ManxPwr | I'm tracking a package sent via DHL from Hong Kong to New Orleans. You would think that an international carrier like DHL would store tracking information in a way that handles timezones and the international date line. |
| 13:56.36 | ManxPwr | *grumble* |
| 13:56.46 | tzanger | ManxPwr: why would you think that? |
| 13:56.59 | elric | i am having problems with SIP softphone connecting to an IAX2 softphone. SIP is on the internal network and there is no NAT between * and SIP. IAX is on the internet. Both can access ZAP interfaces. If IAX calls SIP, * says cannot create channel SIP/1009. If SIP calls IAX, IAX can hear SIP but SIP hears nothing. |
| 13:57.01 | ManxPwr | tzanger, Sometimes I'm too much of an optimist. |
| 13:59.02 | *** join/#asterisk Abbas_ (Abbas@203.81.194.242) |
| 13:59.05 | *** join/#asterisk `Sauron (sauron@rrcs-24-153-164-117.sw.biz.rr.com) |
| 14:04.54 | ManxPwr | LOL! It's rained more on Los Angels this season than in Seattle. |
| 14:08.20 | ariel_ | <PROTECTED> |
| 14:08.22 | ariel_ | moring all |
| 14:08.30 | Luhiwu | anyone knows what does 'Got SIP response 481 "Invalid CSeq Number"' means? |
| 14:09.20 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 14:09.30 | gabb0 | ManxPwr, I know you had said it was on the wiki but I haven't seen it. My issue is the file extension is actually pcm and I can't seem to convert that to anything |
| 14:11.10 | ManxPwr | gabb0, that's really a sox question. |
| 14:11.23 | bjohnson | elric: sounds like codec problems |
| 14:11.29 | ManxPwr | What application generates the file? |
| 14:11.38 | gabb0 | monitor |
| 14:11.42 | elric | bjohnson, using ulaw |
| 14:11.44 | *** join/#asterisk ast_freak (~yircme@hades-out.universalsystems.net) |
| 14:12.03 | ManxPwr | gabb0, What is the ACTUAL monitor command? |
| 14:13.03 | gabb0 | exten => s,2,Monitor(pcm,${MYVAR}-${TIMESTAMP}) |
| 14:13.22 | ManxPwr | WHY are you using PCM? |
| 14:13.39 | ariel_ | elric do you have disalow=all then allow=ulaw |
| 14:13.40 | bjohnson | elric: don't know. Both can use zap to call out and calls are fine? maybe try turning off reinvites? |
| 14:13.41 | gabb0 | long story |
| 14:13.50 | ManxPwr | I don't think I've ever herd of anyone using PCM with monitor. |
| 14:13.58 | elric | bjohnson, ok |
| 14:14.01 | ManxPwr | gabb0, Stop using PCM. |
| 14:14.24 | ManxPwr | use GSM (of you want smaller files) or WAV (if you don't care about file size) |
| 14:14.27 | elric | ariel_, let me check |
| 14:14.27 | gabb0 | it plays back fine with Playback so you would think I would be able to convert it to anything |
| 14:15.00 | ManxPwr | gabb0, Oh, I'm sure you can. But since nobody seems to know how to convert that file type.... |
| 14:15.01 | gabb0 | the plan is to use gsm but the thing is I have this pcm file and need it |
| 14:15.26 | ManxPwr | gabb0, According to "man sox" at least 4 different file formats use pcm |
| 14:15.46 | gabb0 | I wonder which * uses |
| 14:15.59 | ManxPwr | looks like it uses raw adpcm |
| 14:16.19 | ManxPwr | gabb0, but if you want to use a file format nobody else uses, you should expect no support. |
| 14:16.53 | ManxPwr | Much like my problem with E&M wink I had earlier this week. Almost nobody uses E&M Wink so I had a LOT of problems getting it fixed. |
| 14:17.06 | elric | bjohnson, yeah both can use zap, i will turn off reinvites now and check |
| 14:17.29 | elric | ariel_, yeah it was disallow=all and then allow=ulaw |
| 14:17.56 | Abbas_ | hi guys |
| 14:17.56 | ManxPwr | elric, Do you have NO OTHER allow= lines? |
| 14:18.44 | gabb0 | I know what you are saying and agree. It was a must at the time to use pcm but now we don't need to. And I know exactly what you mean about e&m wink. I had a tough time getting it set up a while back. |
| 14:19.09 | elric | ManxPwr, i have allow=gsm |
| 14:19.23 | tzanger | wow |
| 14:19.39 | tzanger | I just had the county ambulance service tell me that the proper way to test 911 addressing is to CALL 911 |
| 14:19.57 | tzanger | I asked her if she was sure, and she said yes -- she used to be one of the dispatchers and that's how they did it |
| 14:20.01 | ManxPwr | tzanger, That suprizes you? |
| 14:20.05 | tzanger | I dunno something does NOT sound right |
| 14:20.06 | tzanger | yes |
| 14:20.10 | tzanger | absolutely it sounds wrong |
| 14:20.15 | tzanger | 911 is for emergency calls, not tests |
| 14:20.17 | ManxPwr | tzanger, Call when they are not busy. |
| 14:20.23 | Abbas_ | can we configure LinkSys PaP2 to make IP tp IP calls by dialing te other device IP only? |
| 14:20.28 | vaewyn | tzanger: yeah... that is the norm... state right from the start that this is a "test call" and they are cool with it |
| 14:20.29 | ManxPwr | tzanger, Yes, but the only to test 911 is to call 911 |
| 14:20.39 | tzanger | ManxPwr: hmm okay I will make sure there isn't an emergency before I call |
| 14:20.49 | tzanger | vaewyn: yes that is what she told me |
| 14:20.52 | ariel_ | tzanger, that is what I got from the 911 service here too. They said the perfer to get a test call and stated as that to make sure the system is working. |
| 14:20.56 | ManxPwr | We plan on doing a monthly 911 test soon. |
| 14:20.58 | tzanger | she said state very clearly right at the start that it is NOT an emergency |
| 14:21.06 | vaewyn | My sister in law is 911 dispatch... gets those calls all the time |
| 14:21.12 | tzanger | wow okay |
| 14:21.15 | tzanger | it just does NOT seem right |
| 14:21.37 | vaewyn | hehehe |
| 14:21.56 | vaewyn | this is "emergency preparedeness" so it is ok :P |
| 14:22.13 | vaewyn | can't let those terrorists catch us with our trousers down ;P |
| 14:22.13 | ManxPwr | BTW, I assume 911 will confirm the address that they see for the call? |
| 14:22.21 | vaewyn | yeah |
| 14:22.43 | vaewyn | and cell calls they will confirm the lat/lon |
| 14:23.01 | vaewyn | (if you have a-gps capable phone) |
| 14:23.45 | vaewyn | in fact most can tell you nearest address to that lat/lon |
| 14:23.48 | vaewyn | but not all |
| 14:24.32 | gabb0 | ManxPwr, figured out how to get the pcm converted |
| 14:24.51 | vaewyn | heck... one I called in DC area read my battery life back to me from the @#$#@ phone |
| 14:25.32 | ManxPwr | gabb0, how? |
| 14:25.32 | vaewyn | rename it blah.raw probably :P |
| 14:25.32 | ManxPwr | vaewyn, Scary. |
| 14:25.56 | tzanger | vaewyn: how do they differentiate from a true test and someone with a knife to their neck being told to say that |
| 14:25.57 | vaewyn | ManxPwr: yeah... but hey... more power to them... they still didn't know who called them ;P |
| 14:26.27 | bjohnson | tzanger: likely not to many attackers would force a victim to call 911 |
| 14:26.32 | tzanger | bjohnson: true enough |
| 14:26.39 | vaewyn | tzanger: probably by the faint "tell them and you die" in the background :P |
| 14:26.39 | tzanger | this was the perth county ambulance service in stratford |
| 14:26.46 | tzanger | vaewyn: ;-) |
| 14:27.04 | gabb0 | ManxPwr, used audacity. I imported raw data. then used the settings U-law, little-endian, 8000 Hz |
| 14:27.18 | gabb0 | then I can export it as wav or do whatever |
| 14:27.39 | vaewyn | mv blah.pcm blah.raw; sox blah.raw blah.wav |
| 14:27.59 | gabb0 | ah, where were you ten minutes ago |
| 14:28.01 | gabb0 | haha |
| 14:28.15 | vaewyn | ahh.. little different then mv blah.pcm blah.ulaw; sox blah.ulaw blah.wav |
| 14:28.24 | vaewyn | waiting for you to find it in the wiki :} |
| 14:28.26 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 14:28.57 | vaewyn | really wish sox would at least try to guess on the sound format before borking |
| 14:29.05 | karman | i need to send dialled number to PBX, what signaling technoly should i use on analog line? |
| 14:29.05 | vaewyn | although raw is a little hard to pin down |
| 14:29.54 | Corydon76-home | karman: you need to know how the PBX is signalling to you |
| 14:30.26 | *** part/#asterisk brazil (~cleber@200.198.105.37) |
| 14:30.31 | Corydon76-home | If the PBX is signalling FXO Loopstart, then you need to signal FXS Loopstart |
| 14:30.36 | vaewyn | hey... anyone know of a good tftp client to test servers with? |
| 14:30.41 | vaewyn | (linux of course) |
| 14:30.58 | djin | tftp |
| 14:31.00 | Corydon76-home | vaewyn: should be just 'tftp' |
| 14:31.05 | Shrink | one comes with tftpd, called tftp |
| 14:31.32 | vaewyn | ahh... seperate package called tftp in debian |
| 14:31.36 | vaewyn | that was too obvious |
| 14:31.53 | Corydon76-home | vaewyn: wait until you try to use it. It's less obvious |
| 14:31.53 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:32.10 | djin | Shrink, not sure if you responded to a question earlier. Did you update from 6.3 to 7.3 on cisco 7940? |
| 14:32.28 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:32.46 | *** part/#asterisk telme (~teliax@c-67-166-37-218.client.comcast.net) |
| 14:32.51 | Corydon76-home | "Whaddya mean, I can't LIST the TFTP directory?" |
| 14:32.56 | vaewyn | Corydon76-home: worked fine for me... connect... get... exit |
| 14:33.01 | vaewyn | hahaha |
| 14:33.29 | vaewyn | well.. that I knew already... tftp doesn't support that |
| 14:33.29 | djin | Corydon76-home, yes that's TFTP |
| 14:33.33 | vaewyn | get put... that is about it |
| 14:33.33 | djin | just PUT and GET stuff. |
| 14:33.36 | Corydon76-home | djin: Yah, I know... first impressions last... |
| 14:33.51 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 14:34.24 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 14:34.41 | vaewyn | now to see if I can get this WIP-5000 to upgrade it's firmware |
| 14:34.45 | Abbas_ | Hi |
| 14:34.49 | *** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net) |
| 14:35.00 | Abbas_ | can we configure LinkSys PaP2 or Cisco ATA 186 to make IP tp IP calls by dialing te other device IP only? |
| 14:35.14 | karman | Corydon76-home, the signaling is loop start. but how does this send the dialled number? |
| 14:35.16 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 14:35.19 | mAsH` | hi all |
| 14:35.23 | ManxPwr | Abbas_, That would be up to the SIP device! |
| 14:35.26 | Casper_UA | hi |
| 14:35.28 | karman | Corydon76-home, do i need to use E&M? |
| 14:35.30 | RoyK | ~nickometer mAsH` |
| 14:35.30 | jbot | 'mAsH`' is 66.000% lame, royk |
| 14:35.31 | Corydon76-home | karman: it doesn't |
| 14:35.49 | Corydon76-home | You don't get DNIS on Loopstart lines |
| 14:35.50 | Abbas_ | ManxPwr i have cisco ATA-186 and Linksys PAP2 aswell |
| 14:36.05 | ManxPwr | Abbas_, Then you should be able to check the documentation for those devices. |
| 14:36.18 | karman | Corydon76-home; and e&m lines? |
| 14:36.25 | ManxPwr | Dialing by IP would bypass Asterisk. |
| 14:36.27 | Abbas_ | ManxPwr have u expereienced ever |
| 14:36.42 | Corydon76-home | Ugh, E&M... I hated having to use E&M... use PRI instead |
| 14:37.23 | Corydon76-home | I mean, if you're going to go for digital signal... at least use the best |
| 14:37.38 | Abbas_ | ManxPwr actually we wanna use it at the place with no internet connectivity they have VPN |
| 14:38.02 | karman | Corydon76-home: can't, money issues.. need cheap solution. got tdm400p.. but do not know if this is even able to doe E&M. it works if set to it.... |
| 14:38.16 | ManxPwr | Abbas_, What part of "read the documentation for the device" do you not understand? |
| 14:38.34 | Corydon76-home | karman: well, then, you've already made your decision, then. |
| 14:38.40 | Abbas_ | :$ |
| 14:38.42 | Abbas_ | ok |
| 14:38.48 | ManxPwr | karman, I don't know of anyone using the TDM400P for analog E&M Wink. |
| 14:39.08 | karman | Corydon76-home: umm.. not really... |
| 14:39.24 | ManxPwr | The TDM400P is ANALOG ONLY, of course. |
| 14:39.39 | Corydon76-home | karman: I'm leaving anyway. Talk to the channel. |
| 14:39.47 | karman | Corydon76-home: thanks!! |
| 14:40.20 | bjohnson | having a little trouble understanding CID here. If I answer a call and get CID, do I lose it if I goto a different context? |
| 14:40.41 | ManxPwr | bjohnson, no. |
| 14:40.43 | |Vulture| | bjohnson: not unless you pass new CID |
| 14:40.54 | karman | ManxPwr: what is the options of sending DID info over anolog? the only one found thus far is E&M |
| 14:40.55 | bjohnson | maybe that's my problem |
| 14:41.07 | ManxPwr | karman, There are really no real options for that. |
| 14:41.41 | karman | ManxPwr: mmmm.... stupid then... pbx able to do it.. but asterisk not.. funny. |
| 14:41.49 | ManxPwr | karman, Correct. |
| 14:41.58 | ManxPwr | Asterisk has SIGNIFICANT limitations. |
| 14:42.16 | ManxPwr | karman, You could prolly hack something togather with your LEC. |
| 14:42.20 | karman | ManxPwr: ok, back to drawing board.. need to learn how to detect hangups on fxs channels then.. aarrghhh |
| 14:42.20 | RoyK | ManxPwr: thinking of what? |
| 14:42.53 | ManxPwr | karman, Um, a hangup on an fxs channel is a hangup. |
| 14:43.11 | ManxPwr | it's FXOs that you have to worry about. |
| 14:43.16 | bjohnson | if I set a var and another call comes in that setvar's the same var, does the first get overwritten or does it call get it's own var space? |
| 14:43.35 | ManxPwr | bjohnson, only if you SetGlovalVar |
| 14:43.41 | bjohnson | Iok |
| 14:43.42 | ManxPwr | SetVar is local to the current channel. |
| 14:43.45 | karman | ManxPwr: not this case... this thing only plays sounds. not any singalling changes.. and sounds to long for busy detect |
| 14:44.08 | ManxPwr | karman, Asterisk FXS -> PBX FXO? |
| 14:44.26 | karman | ManxPwr: umm.. i think i'm confusing myself here.. let me check again |
| 14:44.50 | karman | ManxPwr |
| 14:45.17 | karman | ManxPwr: Asterisk fxs---> pbx |
| 14:45.29 | bjohnson | yeah. I have SPA 3ks that keep calling back into my * system |
| 14:45.38 | karman | ManxPwr: that config gives hangup issues |
| 14:45.40 | mAsH` | anyone can help me wrecording a conversation? |
| 14:45.50 | ManxPwr | And when the PBX hangs up asterisk should see it just fine. |
| 14:46.00 | ManxPwr | It's Asterisk FXO -> PBX FXS that can cause problems. |
| 14:46.04 | bjohnson | in my case * sees the hangup and hangs up |
| 14:46.14 | bjohnson | but the fxo reinitiates an incoming call |
| 14:46.17 | karman | ManxPwr: NOPE.. just beeps, beeps. |
| 14:46.38 | bjohnson | ManxPwr: that's what I have |
| 14:46.43 | ManxPwr | karman, There is only so much Asterisk can do to handle terribly broken PBX portsd. |
| 14:46.54 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 14:46.55 | karman | ManxPwr: LOL!! |
| 14:47.05 | Darwin35 | yes the g729s.so for fbsd rocks |
| 14:47.16 | Darwin35 | 729a |
| 14:47.18 | karman | ManxPwr: it works fine if i use the fxo on asterisk.. |
| 14:47.23 | ManxPwr | Wehn an FXO port hangs up it hangs up. Simple as that. Just like hanging up a phone. |
| 14:47.43 | karman | ManxPwr: but then i can't DID. |
| 14:48.09 | ManxPwr | karman, Give up. Asterisk is not a good solution for your requirements. |
| 14:48.26 | ManxPwr | If you change your requirements then you should reconsider Asterisk. |
| 14:48.44 | karman | ManxPwr: well.. its working, the only thing its rining at switchboard. so she has to transfer to correct extension |
| 14:49.08 | ManxPwr | karman, It's not working if it doesn't do what you want it to do. |
| 14:49.31 | karman | ManxPwr: True.. but if you look at cost of other systems.. ITS WORKING!! |
| 14:49.55 | JerJer | define working |
| 14:49.57 | ManxPwr | karman, Other systems do not have broken FXO ports. |
| 14:50.30 | ManxPwr | But your PBX has broken BPX ports. |
| 14:50.44 | ManxPwr | It's suprizing it even works with just your PBX and your telco. |
| 14:50.52 | karman | ManxPwr: nope, pbx working as it should.. |
| 14:51.03 | ManxPwr | karman, then it should work with Asterisk. |
| 14:51.59 | ManxPwr | karman, But you really can't do much until you know EXACTLY how the telco delivers the DID to Asterisk. |
| 14:52.40 | karman | ManxPwr: if you go and think about it: fxo on the pbx side will thing the fxs on asterisk is normal telephone ie, human.. so will play sound to human that other human hang up the line |
| 14:52.53 | ManxPwr | NO NO NO! |
| 14:53.03 | ManxPwr | The FXO port on the PBX will think Asterisk is a normal telephone LINE. |
| 14:53.35 | ManxPwr | FXO port = expects to hear dialtone and expects to receive ring voltage. |
| 14:53.38 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 14:53.47 | *** join/#asterisk st4vs (~root@212.106.106.29) |
| 14:53.49 | ManxPwr | FXS port = expects to PROVIDE dialtone and provide ring voltage. |
| 14:53.53 | karman | ManxPwr: think we talking about device, nog signaling(or what ever) |
| 14:54.05 | *** part/#asterisk st4vs (~root@212.106.106.29) |
| 14:54.42 | ManxPwr | So you plug an asterisk port that provides dialtone into a PBX port that expects to hear dialtone. |
| 14:54.44 | karman | ManxPwr: yup.. talking wrong way round.. i was looking at configs: |
| 14:55.03 | ManxPwr | karman, I can't help you if you don't know the correct terms to use. |
| 14:55.23 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 14:55.34 | karman | i know correct terms.. |
| 14:56.07 | ManxPwr | karman, So you are plugging Asterisk (sends dialtone) port -> PBX (expects dialtone) port? |
| 14:56.21 | bjohnson | karman: are you trying to send the incoming DID to the fxs for the PBX to display somehow? |
| 14:57.57 | bjohnson | if so, here's what I was thinking of doing: preppend a number or char to the CID so that the person answering can see where it is coming from and who is calling in one string |
| 14:57.57 | karman | ManxPwr: yes.. Asterisk dialtone(fxoks=1) into PBX, co line. |
| 14:58.18 | ManxPwr | karman, Then Asterisk should see when the pbx hangs up the line. |
| 14:58.21 | karman | that part is working fine for hangups |
| 14:58.46 | ManxPwr | So what's the hangup problem? |
| 14:58.56 | karman | when configured as: asterisk (fxsks=4) to pbx extension, then problem |
| 14:59.04 | ManxPwr | karman, That won't work well. |
| 14:59.14 | JerJer | Asterisk should be the 'telco' |
| 14:59.16 | karman | the latter i can do "did: dialing, seeing that i'm just dialing internal extension. |
| 14:59.19 | *** join/#asterisk meppl (~mephisto@pD9542F94.dip.t-dialin.net) |
| 14:59.21 | ManxPwr | You will have hangup problems. |
| 14:59.53 | ManxPwr | karman, Obviously your CO ports can do DID or it would not work when you had the telco directly plugged into the PBX. |
| 15:00.12 | bjohnson | hang up problems on a fxo device connected to a PBX ATA (ie fxs) port? |
| 15:00.15 | karman | never had telco into that coline. |
| 15:00.19 | ManxPwr | karman, You already know that "asterisk (fxsks=4) to pbx extension" does NOT work. |
| 15:00.49 | ManxPwr | karman, What brand of PBX? |
| 15:01.01 | karman | lg 162 |
| 15:01.09 | ManxPwr | never heard of it. |
| 15:01.12 | karman | lol |
| 15:01.19 | jsolares | the first thing you should do is.... |
| 15:01.24 | jsolares | throw that into the garbage :X |
| 15:01.46 | ManxPwr | karman, How were your DID lines connected into the PBX? |
| 15:02.37 | karman | ok , lets get back to original question... what is the way of doing DID.. i need to find out what card is needed in PBx. We never used DID |
| 15:02.53 | karman | but, as far as i can see, the pabx can do e&m wink by default |
| 15:03.05 | karman | thus: it should work on normal coline card. |
| 15:04.19 | ManxPwr | karman, You are trying to do two different projects at once. |
| 15:04.21 | a1fa | fwd is ok |
| 15:05.34 | karman | ok.. so if you would do DID on anolog lines.. how would you go about it? |
| 15:06.34 | jsolares | how many lines? |
| 15:06.55 | ManxPwr | karman, I wouldn't do analog DID lines. Our telco charges us the same for analog and digital lines. |
| 15:07.18 | karman | hehe.. this is ZA.. voip is only been legal for 3 weeks. |
| 15:07.25 | jsolares | lol |
| 15:07.38 | jsolares | still, how many lines |
| 15:07.42 | karman | 80 lines. |
| 15:07.48 | jsolares | all analog? |
| 15:07.55 | karman | but this is not telco issue |
| 15:07.56 | bjohnson | ManxPwr: my problem against digital is quantity .. not worth it for 3 lines |
| 15:08.05 | karman | i not even interfacing with telco . |
| 15:08.11 | karman | this us purely internal |
| 15:08.16 | ManxPwr | I would do fake DID. Call -> Asterisk IVR -> PBX CO port going into an IVR where Asterisk can dial the required extension. |
| 15:08.32 | jsolares | so you want to replace the pbx? |
| 15:08.41 | karman | nope.. not replace.. |
| 15:08.48 | karman | just make interbranch calls over voip |
| 15:08.52 | jsolares | then i dont get what you're trying to do |
| 15:08.57 | jsolares | ah |
| 15:09.05 | karman | ManxPwr: mmm. need to see if PBX can do that.. |
| 15:09.08 | jsolares | how many concurrent calls do you want to do? |
| 15:09.13 | karman | 1 |
| 15:09.32 | jsolares | then connect one of the pbx extension into a fxo port on the asterisk |
| 15:09.44 | bjohnson | karman: on my pbx it's under call attendant and DISA |
| 15:09.53 | jsolares | i have that set up right now, altho only for testing, and its an avaya pbx |
| 15:09.59 | karman | jsolares: i did.. hangup problems |
| 15:10.07 | ManxPwr | karman, no you didn't. |
| 15:10.17 | jsolares | i thought you connected the extension to fxs |
| 15:10.27 | ManxPwr | bjohnson, just said plug asterisk into the PBX CO port and you already told us that does not have hangup problems. |
| 15:10.28 | karman | jsolares: i did both |
| 15:10.30 | jsolares | connect it to the red module on the card you did? |
| 15:10.48 | bjohnson | ManxPwr: actually, you said it. I concur that it makes senses |
| 15:11.17 | jsolares | what's a pbx co port? i still am green when it comes to terms in telephony |
| 15:11.18 | karman | bjohnson: dont we just love the confusion that fxo/fxs can create ;-) |
| 15:11.30 | bjohnson | karman: what kind of fxs and fxo units are you using? can you monitor line voltage? |
| 15:11.32 | jsolares | the red module is fxo on your card :p |
| 15:11.48 | jsolares | bjohnson: afaik digium ones |
| 15:12.03 | *** join/#asterisk bkw_ (nobody@bkw.developer.and.friend.of.asterisk) |
| 15:12.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:12.05 | karman | bjohnson: tdm400p one red.. one green.. |
| 15:12.05 | bjohnson | my SPA 3000 has helped me solve a couple of problems because you can monitor line voltage |
| 15:12.28 | bjohnson | karman: maybe the PBX isn't doing a big enough change in voltage to signal a drop |
| 15:12.48 | jsolares | i need to monitor line voltage, either the avaya pbx is screwed up sending me phantom calls every 15mins, or the digium card is too sensitive to voltage changes |
| 15:12.51 | *** join/#asterisk zno (~zeno@ip-160-79-174-98.autorev.intellispace.net) |
| 15:13.11 | karman | bjohnson: i connect to co line on pbx (the one expectin dialtone) it works fine for hangups, but DID not working (YET) |
| 15:13.50 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 15:13.51 | karman | bjohnson: when connecting to extension on pbx (the one giving dialtone) it does not detect dialtones, but "DID" works |
| 15:14.15 | karman | bjohnson: i should rather call DID as : dial the extension that originator dialed |
| 15:14.21 | bjohnson | I used my SPA 3000 for 2 problems. 1. one telco termination was reversed wired (so positive voltage instead of negative) and 2. an fxo hooked into a data/fax/phone switch wasn't getting enough variation in the line voltage (was configurable on the SPA) |
| 15:14.44 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.232) |
| 15:14.46 | sudhir492 | hi all |
| 15:14.50 | bjohnson | karman: I on't know what you mean by DID not working yet. Can you explain that more? |
| 15:15.02 | zno | I'm having a call parking problem: whenever i transfer someone to my parking extension 700, I don't get the park extensions read back to me, it says Playing 'digits/7' (language 'en') WARNING[12204]: file.c:550 ast_readaudio_callback: Failed to write frame |
| 15:15.28 | zno | when I call my parking extension, I get the parked extension read back to me fine |
| 15:15.42 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:15.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:15.50 | karman | bjohnson: well, my first questin was: how to do DID on anolog lines... the only why i can think of is e&m wink.. but i do not know if digium cards is abnle to do this |
| 15:16.03 | bjohnson | yeah .. I saw that |
| 15:16.06 | bjohnson | what do you mean |
| 15:16.17 | bjohnson | "how to do DID on anolog lines" |
| 15:16.26 | bjohnson | a DID is a phone number correct? |
| 15:16.36 | jsolares | you hook up the telco analog line to the fxo and off you go |
| 15:16.45 | karman | let me give you my complete config.: |
| 15:17.02 | Nugget | here comes the paste, brace yourself guys. |
| 15:17.08 | jsolares | pastebin!!! |
| 15:17.09 | karman | pbx ---> asterisk -- IP ---> asterisk ---> pbx |
| 15:17.24 | jsolares | kick the pbx until it works |
| 15:17.28 | jsolares | pbx's* |
| 15:17.29 | karman | so, the one pbx picks up, dial 9 (to get dialtone which is asterisk) |
| 15:17.31 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:17.39 | gabb0 | a ball bat is often effective too |
| 15:17.50 | bjohnson | here is mine: telco analog line <-> SPA fxo <-> * <-> SPA fxs <-> Nortel analog line in (fxo) |
| 15:17.53 | karman | then asterisk wait for dtmf from the caller, then calls the other asterisk box. |
| 15:18.19 | karman | ie dial/iax2 exten 109 |
| 15:18.30 | karman | other asterisk dial zap/g2/109 |
| 15:18.54 | *** join/#asterisk Mike_TK (~Mike_@213.180.245.62) |
| 15:19.13 | jsolares | you dont need fxs on asterisk for that, just a pbx that works with fxo |
| 15:19.18 | gabb0 | jsolares said kick the pbx until it works, I find a bat more effective |
| 15:19.25 | karman | LOL |
| 15:19.46 | karman | the problems comes in that the pbx then dials tha swb, for it does not see the number dialed (109) |
| 15:19.55 | karman | swb = switchboard |
| 15:20.02 | jsolares | did you configure the zaptel to use the tones from your country? |
| 15:20.32 | karman | tones for my country.. bwwahhaahaa.. this is ZA.. |
| 15:20.47 | bjohnson | karman: so you are trying to directly dial a specific internal extensions from the pbx CO in line? Do any pbx allow that? mine doesn't unless you authenticate |
| 15:21.16 | karman | bjohnson: that is exactley it. |
| 15:21.21 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 15:21.23 | karman | but authenticate? |
| 15:21.51 | ariel_ | karman, put in the zapata.conf under the port your connecting to the pbx relaxdtmf=yes |
| 15:22.44 | karman | i think i should go about the easy rout.. ivr, and do dial(zap/g1/exten, D(109)) |
| 15:22.52 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 15:23.05 | karman | thanks all for you insights... |
| 15:23.16 | bjohnson | karman: on mine .. to dial an internal extension from an incoming call you first have to config the pbx to answer that line and wait for the extension to be dialed. Typically that is done with an authentication password. On my system that is called using auto-attendant and getting DISA access |
| 15:24.04 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
| 15:24.05 | bjohnson | ie .. have you confirmed your pbx is answering the call and allowing the caller to dial an extension? |
| 15:24.11 | karman | bjohnson: this one is same for DISA...., but DID should not work in that manner |
| 15:24.31 | karman | bjohnson: pbx not answering then waiting for extension... |
| 15:24.44 | karman | bjohnson: i was hopeing not to go that route |
| 15:25.04 | bjohnson | I don't understand at all what you mean by DID then. |
| 15:25.08 | [TK]D-Fender | Got a problem starting * could use a hand. I just compiled * without PRI/ZAP and on load it whines about not having PGSQL or ODBC (both are technically installed) and I'd rather it just use the CSV CDR's. those 2 warning are the last thing I see before it dumps me back to the shell. Any hints? |
| 15:25.10 | karman | ok |
| 15:25.43 | karman | the normal meaning for DID is: you have say 4 anolog lines.. all in hunting group. you have say 10 numbers that will hunt on the 4 lines |
| 15:26.10 | karman | when someone dials one of these 10 numbers, it will take the first available line (of the 4) |
| 15:26.24 | karman | the telco then provides the "dialed number" |
| 15:26.39 | karman | the pbx then knows that theat dialed number maps to inernal extension |
| 15:26.46 | karman | then auto routes it |
| 15:27.04 | karman | <PROTECTED> |
| 15:27.13 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 15:27.13 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 15:27.23 | tzanger | holy shit |
| 15:27.28 | tzanger | F*1 works wiht norstar |
| 15:27.29 | jsolares | but you want to be able to call extensions from one branch to the other using voip |
| 15:27.44 | stevekstevek | F*1? |
| 15:27.51 | jsolares | i think it all boils down to this, your current lg pbx's suck :p |
| 15:27.54 | dsmouse | terrapen: ping? |
| 15:28.01 | bjohnson | I, for one, did not know you meant direct inward dialing everytime you said DID |
| 15:28.21 | stevekstevek | what else would DID stand for? |
| 15:28.22 | karman | bjohnsonL: sorry ;-( |
| 15:28.22 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 15:28.35 | dsmouse | ~did |
| 15:28.37 | jbot | rumour has it, did is Direct Inward Dialing |
| 15:28.45 | karman | lol |
| 15:28.47 | bjohnson | stevekstevek: buy a DID, people dial it, the call comes into * |
| 15:28.59 | jsolares | that's the same |
| 15:29.06 | stevekstevek | yes, same meaning. |
| 15:29.23 | stevekstevek | but, I guess they mean some kind of origination service providing DID... |
| 15:29.47 | FocusRite | does anyone have experience using asterisk with pika boards (if it works with pika gear that is) ? |
| 15:29.51 | karman | stevekstevek: originatin service in this case is asterisk |
| 15:30.06 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 15:30.51 | bjohnson | wouldn't you just dial into the pbx like a regular phone number then? |
| 15:30.58 | karman | ps: how do one debug zap channels: like to see if sending dtmf etc. |
| 15:31.21 | karman | bjohnson: yes.. i would. on digital systems its there already |
| 15:31.31 | karman | bjohnson: but this is analog.... |
| 15:31.37 | bjohnson | or .. like you would be dialing to a telco through a SIP device |
| 15:31.50 | bjohnson | so the fxs ports do not support that? |
| 15:32.07 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 15:32.09 | ariel_ | karman, wait just lets do this in your context that the inbound zap port put exten => s,1,NoOp(${DNIS}) to see what is being sent. |
| 15:32.34 | karman | ariel_: i'm not receiving call by asterisk,.. i'm making the call... |
| 15:32.45 | karman | bjohnson: i really wont know. |
| 15:33.09 | ariel_ | karman, your going from your lg pbx to asterisk then via voip to other asterisk correct? |
| 15:33.13 | karman | bjohnson: if someone out ther is able to tell me this.. |
| 15:33.27 | karman | ariel_, yes, but then to another lg pbx. |
| 15:33.44 | karman | pbx * --* pbx |
| 15:33.45 | yashax | Is it me or www.race.com is down? |
| 15:34.00 | Nugget | http://slacker.com/things/race.php |
| 15:35.27 | ariel_ | yashax, humm I was just on there site a few minutes ago. let me check. |
| 15:35.34 | yashax | thanks... |
| 15:36.43 | ariel_ | yashax, there web site is down but not there network. |
| 15:37.07 | yashax | thank you... |
| 15:37.20 | yashax | Does anyone have any experience or use one of the following services: broadvoice.com, nufone.com, livevoip.com - I am trying to find an inexpensive and reliable termination for *. I would love any comments. |
| 15:37.28 | yashax | race.com as well |
| 15:38.06 | jsolares | i've had nufone for a week, and so far it's good |
| 15:38.15 | ariel_ | karman, ok lets see your able to go from lg1 to asteisk then at the asterisk it is able to go to the 2nd asterisk. But that one is not able to go to the 2nd lg? |
| 15:38.18 | bjohnson | yashax: try them all |
| 15:38.26 | jsolares | if you only want to make one call at a time, perhaps broadvoice is more for you |
| 15:38.36 | bjohnson | yashax: voipjet is also popular |
| 15:38.58 | jsolares | if you want to make multiple calls then i'd recommend nufone, and i'm currently in the process of acquiring an account at livevoip.com |
| 15:39.05 | bjohnson | IMO the choice to use broadvoice depends on your call volume |
| 15:39.10 | Zeeek | who has ploycomm phones? |
| 15:39.20 | bjohnson | to sell? |
| 15:39.29 | karman | ariel_: able to go to second pbx.. but DID not working |
| 15:39.35 | Zeeek | what is the biggest diff between the ip500 and 600 beides # of lines |
| 15:39.36 | bjohnson | or to give you user feedback |
| 15:39.45 | Zeeek | FEEEEEEDback |
| 15:39.56 | ariel_ | In my view voipjet is good but has problems with callerID and being able to send faxes, VPC is working for me just fine. And so si Race.com. Nufone works for outbound ld as well. |
| 15:40.03 | Zeeek | $100 more for 600 - why is it [not] worth it |
| 15:40.09 | ManxPwr | Sometimes users REALLY piss me off. |
| 15:40.10 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 15:40.10 | *** mode/#asterisk [+o kram] by ChanServ |
| 15:40.22 | Zeeek | there are no users - only non-admins |
| 15:40.25 | bjohnson | ManxPwr: I find that hard to imagine |
| 15:40.29 | ariel_ | karman, so your did is dropping between the 2nd asterisk and the 2nd pbx. |
| 15:40.37 | Zeeek | bjohnson you have any? |
| 15:40.41 | Zeeek | polys? |
| 15:40.46 | bjohnson | Zeeek: no |
| 15:40.46 | ManxPwr | ariel_, I refuse to use VoipJet because of their nasty messages on the -biz mailing list about other VoIP companies. |
| 15:40.56 | *** join/#asterisk moonboi (~chatzilla@64.18.161.212) |
| 15:41.09 | karman | ariel_: wont say dropping.. i just dont know how to do it over anolog lines |
| 15:41.28 | ariel_ | ManxPwr, great to know. I am no longer using them due to there callerID problems. |
| 15:41.35 | ManxPwr | bjohnson, I told the customer that they should give every agent a DID number and have that DID handle both voice and fax for that user. They all hated the idea. Now they have come up with the idea that each agent gets their own fax machine. |
| 15:41.41 | moonboi | is it possible to use meetme if only one pstn line on say a x100p or do i need PRI's to do so ? |
| 15:42.08 | ManxPwr | mooboi, You can use meetme with VoIP only, as long as you have a zaptel device. |
| 15:42.10 | Qwell | mooboi: How are you getting calls? PSTN? |
| 15:42.17 | ariel_ | mooboi, just one pstn line will only give you one call. others are via voip then. |
| 15:42.52 | moonboi | ty for the info |
| 15:42.56 | moonboi | from pstn i recieve |
| 15:42.58 | moonboi | bbl |
| 15:43.10 | bjohnson | ManxPwr: I don't understand the difference |
| 15:43.47 | yashax | so with broadvoice.com you can only make 1 call at a time? |
| 15:43.49 | ManxPwr | bjohnson, There isn't really a difference. |
| 15:43.58 | jsolares | yashax: as far as i can tell, yes |
| 15:44.02 | ManxPwr | bjohnson, hence me being pissed off. |
| 15:44.24 | bjohnson | yashax: it is unclear in most voip providers terms of service. I think with bv they charge per minute for more than one concurrent call |
| 15:45.09 | bjohnson | ManxPwr: so .. they finally saw the brilliance emenating from you and have adopted your solution. |
| 15:45.26 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 15:45.31 | ManxPwr | bjohnson, not really. The came up with the idea after forgetting I suggested it in the first place. |
| 15:46.08 | bjohnson | they likely don't realize it's the same thing. |
| 15:46.43 | bjohnson | you're suggested focused on incoming while their suggestion focused on outgoing |
| 15:46.54 | bjohnson | man .. bad grammar |
| 15:46.58 | bjohnson | and spelling |
| 15:47.18 | yashax | Would it make any difference as far as choosing a provider if I want to make both, local & long distance? |
| 15:47.40 | bjohnson | kind of |
| 15:48.12 | bjohnson | it depends if any providers you are looking at have any kind of package that even refers to local calling |
| 15:48.22 | jsolares | by local and long distance do you mean whitin the us? |
| 15:48.35 | bjohnson | local vs long distance is typically no difference to voip providers |
| 15:48.42 | qiu | hi ... does anyone use asterisk with gnugk in proxy mode ? |
| 15:48.51 | yashax | yes in US only |
| 15:48.54 | qiu | the call to be initiated from asterisk |
| 15:49.29 | jsolares | yashax: yeah most providers only differentiate alaska and hawaii |
| 15:49.37 | jsolares | the others are same rate |
| 15:49.46 | bjohnson | yashax: I think the ones you mentioned do not make a distinction between local and long distance calls |
| 15:49.54 | yashax | yeah... sorry for confusion. I notice that it is very common for * boxes to route the local calls via POTS and long distance via VOIP, but I would like to do both via VOIP |
| 15:50.09 | bjohnson | certainly possible |
| 15:50.17 | jsolares | broadvoice has a 9.95 unlimited in-state plan |
| 15:50.29 | bjohnson | I use voip as a backup outgoing for local calls if my pots are busy |
| 15:50.37 | yashax | but I absolutely have to have more than one call at a time... |
| 15:50.43 | BrianR___ | grr.. |
| 15:51.03 | bjohnson | yashax: 1. inquire at BV |
| 15:51.04 | jsolares | yashax: then livevoip seems the cheapest at 0.012$ per minute compared to nufone's 0.020$ |
| 15:51.06 | BrianR___ | Aparently there's some bug with the festival module which causes asterisk to hang the machine if running with realtime priority :( |
| 15:51.08 | jsolares | that too |
| 15:51.20 | bjohnson | 2. consider volume of calling and decide if pay-as-you go is cheaper |
| 15:51.23 | jsolares | i'm using festival with text2wave :D |
| 15:51.39 | BrianR___ | jsolares: Is it reliable? |
| 15:51.41 | vaewyn | livevoip is cheaper... but nufone is still more stable... |
| 15:51.45 | ^Fenris | can I tell if I have a dial tone on a POTS line that I have connected to *? |
| 15:51.46 | yashax | bjohnson: sorry...BV? |
| 15:51.48 | vaewyn | pick your poison |
| 15:51.50 | jsolares | so far so good |
| 15:51.56 | bjohnson | yashax: broadvoice |
| 15:52.15 | yashax | k... but someone said that it is 1 call max at a time? |
| 15:52.33 | bjohnson | yashax: 1. inquire at BV |
| 15:52.42 | *** part/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 15:52.43 | yashax | about it... got it.. |
| 15:52.45 | jsolares | i said that, but there's conflicting info out there, so do as bjohnson |
| 15:52.46 | BrianR___ | I think the festival server is busted too. It winds up using 100% CPU... |
| 15:53.17 | jsolares | i dont even start the festival server :D, text2wave and the perl script on the wiki rules |
| 15:53.29 | sudhir492 | my asterisk just core dumped :-( |
| 15:53.40 | bjohnson | any voip provider can limit how many concurrent channels you can use. That max number changes from provider to provider .. you have to ask |
| 15:54.02 | bjohnson | yashax: once you have a definite answer .. post it on the wiki |
| 15:54.04 | BrianR___ | I'm wondering why the machine gets hung when asterisk runs as realtime though. It's as if the festival daemon is somehow inheriting realtime priority from the asterisk server, which seems completely impossible. |
| 15:54.22 | yashax | Will do.... |
| 15:56.27 | jsolares | odd my brother called canada using nufone yesterday evening and still hasnt showed up on their cdr |
| 15:56.43 | Zeeek | shhhht |
| 15:56.56 | a1fa | later |
| 15:58.28 | bjohnson | hey .. where would be a good place on the wiki to post some info about the iax.cc (sixtel) service? |
| 15:58.39 | bjohnson | I'm kind of surprised they don't already have stuff there |
| 15:59.09 | *** join/#asterisk Firebird_ (~xxx@130.40.39-62.rev.gaoland.net) |
| 15:59.55 | Firebird_ | Hi, Is there anyone her ewho can help me with a problem of audio quality using the monitor application ? |
| 15:59.59 | ManxPwr | bjohnson, there's a voip serv\ice provider page |
| 16:02.07 | Firebird_ | guess everybody sleeping... ZZzzzzzzz |
| 16:02.24 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 16:02.33 | benno2 | do you think an analog modem connection (laptop) -> sipura spa 2000 -> * -> ISDN card will work ? |
| 16:02.58 | benno2 | basically all I need is to dial a local ISP over the PSTN (not over a voip provider to pstn) |
| 16:03.09 | *** part/#asterisk karman (~karman@196.46.71.170) |
| 16:03.34 | benno2 | the local LAN is not loaded at all so I guess the packet loss rate is near zero and since I'm using uncompressed g711 it should work right ? |
| 16:04.16 | mikegrb | why ask, just try |
| 16:05.15 | ^Fenris | do or do not, there is no try |
| 16:06.14 | Zeeek | ManxPwr what phones do you use/like? |
| 16:07.37 | Zeeek | besides Cisco |
| 16:07.44 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 16:07.44 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 16:08.28 | *** join/#asterisk LittleRobbie (~rob@pcp09255610pcs.olathe01.ks.comcast.net) |
| 16:08.30 | tzanger | bjohnson: what kind of info |
| 16:09.32 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 16:10.28 | yashax | Guys, Just done talking to Broadvoice - signing up. They do NOT provide instructions for configuring * with their service. Will someone, please, be able to walk me through the necessary config to make it work? |
| 16:10.54 | Zeeek | try the ml |
| 16:11.03 | tzanger | yashax: out of curiosity, why are you using them if they're so unhelpful? |
| 16:11.22 | Zeeek | BV specific problems have been discussed on the ML |
| 16:11.28 | Samoied | How to wait for digits in zap channel? |
| 16:11.34 | Samoied | I have used WaitExten |
| 16:11.39 | LittleRobbie | yashax: i saw a good thread on setting them up in the asterisk-users archives... link coming... |
| 16:11.53 | Samoied | But this not give the signal |
| 16:12.46 | yashax | thank you... so some of you saying that BV is not so good? (Zeek) |
| 16:12.48 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:13.03 | Zeeek | I'm not as I never used them |
| 16:13.06 | LittleRobbie | yas hax: http://lists.digium.com/pipermail/asterisk-users/2005-February/087215.html |
| 16:13.21 | Zeeek | BV has SPECIFIC issues with asterisk |
| 16:13.36 | tzanger | I just want to know why you'd use them if they're so difficult to work with |
| 16:13.40 | LittleRobbie | that thread describes a problem with a certain int'l dial plan, but good results overall |
| 16:13.48 | *** join/#asterisk zipp (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 16:14.07 | Firebird_ | Who can tell me why I have a very bad quality in the audio file resulting from a monitoring ? |
| 16:14.20 | yashax | ok thank you.. I will give it a try and cancel if the service is unacceptable |
| 16:14.30 | LittleRobbie | now that I have done *my* good deed for the day, could someone help me out with zaptel on sparc? |
| 16:14.48 | jsolares | anyone have experience with vario hardware? |
| 16:15.29 | zipp | anyone having intermittent outage issues with nufone, i.e. "the person you are trying to reach is not available" msgs when calling inbound 8XX number? |
| 16:15.39 | zipp | it works about 30% of the time |
| 16:15.39 | Darwin35 | yes g729 on fbsd rocks |
| 16:16.20 | |Vulture| | has anyone devised a php/sql interface to recording calls on *? |
| 16:16.29 | jsolares | zipp: do you have callerid set to a "real" phone number? |
| 16:16.46 | yashax | Still have the same question, if possible. Will someone be able to please give me a hand in configuring the BV account with *? |
| 16:16.59 | zoa | Darwin35: you got it from digium > |
| 16:17.00 | zipp | jsolares, this is inbound, I am calling my nufone number from a cell phone |
| 16:17.02 | zoa | ? |
| 16:18.04 | Zeeek | yashax someone just went to the trouble of giving a link to read |
| 16:18.05 | zipp | jsolares, this is on 2 seperate systems at 2 seperate data centers, both work fine with voicepulse connect... |
| 16:18.19 | jsolares | ah, no idea then... |
| 16:18.46 | Zeeek | zipp is callerid blocked by any chance? |
| 16:18.48 | LittleRobbie | yes, yas, read the link i posted... it has snips from sip.conf and extensions.conf. |
| 16:18.50 | heison | i'm experiencing problem with iaxtel... i can make outgoing calls but can't seem to accept calls, people get "user not registered", sounds like my register line in iax.conf isn't working, but iax2 show registry does show Registered. And my registration to Nufone works just fine... any clue? |
| 16:18.54 | yashax | sorry.. but I thought that link describe the problem with the BV and not the instructions. I am sorry if I missed the other correct link? |
| 16:19.09 | Zeeek | GO LOOK AT THE MAILING LIST |
| 16:19.13 | yashax | Did I miss something? sorry.. |
| 16:19.25 | whui | hi heison |
| 16:19.45 | LittleRobbie | so, anyway, anybody know of anybody using zaptel on a sparc? |
| 16:19.52 | LittleRobbie | besides me that is? |
| 16:20.06 | BrianR___ | g729 really sounds like ass... |
| 16:20.12 | yashax | Zeek: for setup instuctions? |
| 16:20.33 | zoa | littlerobbie: does it work for you ? |
| 16:22.14 | bjohnson | ManxPwr: the listings of voip providers on the wiki are confusing. In practice there is no distinction between residential and commercial or division by country .. but I guess you have to start somewhere |
| 16:23.03 | bjohnson | tzanger: info like .. will forward to a pstn number if they are unable to contact you, they agree your DID number from them is tranferrable to other service providers |
| 16:23.18 | tzanger | bjohnson: nice |
| 16:23.23 | tzanger | stevekstevek: depends on the input |
| 16:23.24 | LittleRobbie | ewll, in a word, no. zaptel and wcfxo load, but the x100p does not seem to initialize properly and ztcfg fails. plus the system sort of "freaks out" |
| 16:23.25 | Beirdo | you need a demonstration, stevekstevek? |
| 16:23.36 | Beirdo | eat some chili |
| 16:24.56 | bjohnson | yashax: I think someone even updated a wiki page for bv configuration |
| 16:25.04 | BrianR___ | Anyone know where I can find a table of how many X a given codec runs on a given CPU? |
| 16:25.10 | LittleRobbie | i had posted to the asterisk-dev list and actually got a couple bites... suggestion was to bring it up heer as there was rumor of solairs-sparc successes |
| 16:25.30 | mikegrb | bjohnson: a database with all that info is being created |
| 16:25.43 | mikegrb | bjohnson: see the asterisk-biz archives |
| 16:25.48 | BrianR___ | like how many g729's vs how many gsm's vs. how many speex's? |
| 16:26.03 | *** join/#asterisk ayzee (mario@supermario.org) |
| 16:26.23 | bjohnson | Beirdo had chili for lunch yesterday |
| 16:26.26 | *** join/#asterisk Mike_TK (~Mike_@213.180.245.62) |
| 16:26.37 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 16:26.43 | mikegrb | bjohnson: no, he had spaghetti o's |
| 16:26.48 | mikegrb | bjohnson: at mi casa |
| 16:27.03 | Beirdo | hehe |
| 16:27.13 | Beirdo | that was good chili |
| 16:28.00 | zoa | brianR, wait a little more |
| 16:28.02 | zoa | we are making that |
| 16:28.08 | zoa | some results can be found on astertest.com |
| 16:28.34 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 16:28.46 | Mike_TK | Hello, Small question. What's estimated time of sending g729 license key after payment? |
| 16:28.50 | zoa | 1 day ? |
| 16:29.04 | BrianR___ | zoa: Aah. |
| 16:29.40 | *** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it) |
| 16:30.16 | BrianR___ | zoa: I'm using the experimental G729 codec. But for deployment I want to use GSM or a licensed g729 codec. Unfortunately the digium g729 codec has an annoying activation scheme which may be incompatible with our availability requirements. |
| 16:30.49 | BrianR___ | It seems that only a few sip hardphones support compressed codecs other than g729. |
| 16:31.05 | zoa | haha |
| 16:31.11 | ariel_ | BrianR___, have you tried g726 |
| 16:31.14 | zoa | the digium g729 codec will be compatible |
| 16:31.41 | zoa | and if its not, they will make it compatible for your weird os |
| 16:31.59 | zoa | im sure |
| 16:31.59 | BrianR___ | zoa: Doesn't the codec refuse to work if you need to change ethernet cards and your internet connection is down? |
| 16:32.02 | zoa | they are nice guys |
| 16:32.14 | zoa | yes |
| 16:32.17 | zoa | i think so yes |
| 16:32.23 | BrianR___ | Yeah.. That's a problem.. |
| 16:32.26 | zoa | but it might be that you can recycle them |
| 16:32.32 | BrianR___ | ariel_: Which hardphone shave g726? |
| 16:32.39 | zoa | well the other option is a big lawsuit |
| 16:32.39 | zoa | :p |
| 16:32.58 | BrianR___ | zoa: I haven't actually bought any hardphones yet, so the other option might be gsm :) |
| 16:33.03 | zoa | besides this, the quality of the g729 y digium will be better |
| 16:33.08 | BrianR___ | Or i could fix the asterisk g729 passthru mode... |
| 16:33.10 | zoa | the gsm is only on snoms i think |
| 16:33.14 | ariel_ | BrianR___, supura do |
| 16:33.14 | zoa | which are very good |
| 16:33.22 | zoa | aha so sipura and snom |
| 16:33.29 | BrianR___ | I like the look of the snom phones - they seem solid. |
| 16:33.35 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 16:33.48 | ariel_ | sipura has g726 |
| 16:34.08 | Pinhole | When using SIP with *, how do I get * to www-authenticate? |
| 16:34.09 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 16:34.18 | *** join/#asterisk starbucks (~Alex@m815f36d0.tmodns.net) |
| 16:34.28 | Pinhole | It should do that on register, but it doesn't |
| 16:34.37 | zno | snoms are nice |
| 16:35.05 | zipp | anyone have any comments on the hitachi WIP-5000? |
| 16:35.06 | ariel_ | I wish more phones out there would use gsm |
| 16:35.19 | zipp | ariel_, I wish more phones out there would use iax2 |
| 16:35.24 | sudhir492 | is there a way to limit a dial command to say 5 minutes of talk time? |
| 16:35.35 | chipux | zipp: hmm. Ive been pondering ordering one.. |
| 16:35.44 | ariel_ | zipp, yes but that is not going to happen any time soon |
| 16:36.04 | zipp | ariel_, yea, notice farfon... |
| 16:36.09 | chipux | zipp: they look nice.. but I haven't heard many first hand accounts :-/ |
| 16:36.18 | zoa | sudhir492: there is read the manual |
| 16:36.23 | zipp | chipux, half the price of the cisco |
| 16:36.34 | sudhir492 | zoa: thats what I am doing right now. |
| 16:36.50 | Pinhole | sudhir492: look for absoulte timeout |
| 16:36.52 | sudhir492 | zoa: having difficulty in finding that one. Maybe my asterisk is old |
| 16:37.24 | chipux | http://www.bitstruct.com/hitachi/wirelessip-5000 |
| 16:37.35 | zipp | is sipura > budgetone?? |
| 16:37.41 | zipp | I have a few budgetone |
| 16:38.04 | ariel_ | sudhir492, lookup absolutetimeout |
| 16:39.36 | loud | yashax, kinda late but, http://www.broadvoice.com/support_install_asterisk.html |
| 16:39.36 | ariel_ | zipp, the Sipura-841 is about 85 dollars and mine is working just fine. |
| 16:40.00 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 16:40.01 | zoa | yeah its only in there in the last few months |
| 16:40.05 | zoa | maybe 6 months or so |
| 16:40.14 | yashax | loud: You GENIOUS!!!!!!!!!!! |
| 16:40.26 | loud | patch is included in the latest asterist stable. no need to apply it. :) |
| 16:40.35 | yashax | Thank YOU! |
| 16:40.43 | loud | you are welcome. |
| 16:40.53 | |Vulture| | yashax: they misspelled proxy.mia.broadvoice on there... watch out |
| 16:40.53 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 16:41.01 | yashax | I am using asterisk@home, latest version - do you know if the patch is in there? |
| 16:41.20 | ariel_ | yashax, yes |
| 16:41.59 | Pinhole | Is there a way of making asterisk authenticate sip peers when they register? |
| 16:42.05 | yashax | thank you guys.. will let you know when it is, hopefully, working as to how... |
| 16:42.59 | zoa | Pinhole: they do by default |
| 16:43.44 | Pinhole | zoa, no, they don't. watch the packets. there is no WWW-Authenticate headers. |
| 16:44.04 | Pinhole | thus I am probably missing something. |
| 16:44.12 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
| 16:45.05 | Mike_TK | So, noone knows how much time sending a license key for g729 will take after payment? |
| 16:46.06 | loud | 4, 5 hours. |
| 16:46.36 | *** join/#asterisk jero (~SFLphone@199.243.85.90) |
| 16:46.39 | jero | hi |
| 16:47.40 | jero | hey, can I plug a digium quad-T1 card on a dell server that has PCI-X or PCI-express slots ? |
| 16:47.49 | *** join/#asterisk human39 (~human39@chewie.fyi.net) |
| 16:47.58 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:48.23 | human39 | morning folks, has anybody ordered a sipura IP phone? |
| 16:48.42 | ariel_ | yashax, www.race.com is backup for there webs. |
| 16:48.50 | tzanger | jero: is PCIX/PCIe hardware compatible with PCI 2.2 (5v or 3v)? |
| 16:48.52 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
| 16:48.56 | *** join/#asterisk carsim (~carsim@203.115.184.38) |
| 16:48.57 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 16:48.59 | jero | tzanger, I have no idea |
| 16:49.16 | pimpwell | whats the cheapest message recording company? |
| 16:49.24 | tzanger | jero: find that answer and you'll find your answer. :-) |
| 16:49.25 | jero | tzanger, the only servers I can find now are pcix or pcie |
| 16:49.28 | pimpwell | I don't even like Allison anyway |
| 16:49.32 | pcm | pimpwell: that would be yourself :) |
| 16:49.32 | moonboi | what is the cheapest FXS for asterisk ? i want to connect an analog phone to my existing setup on a shoestring .... |
| 16:49.35 | tzanger | jero: actually just ask Dell |
| 16:49.50 | bjohnson | tzanger: http://www.voip-info.org/tiki-index.php?page=iax.cc |
| 16:49.52 | coppice | tzanger" PCI-X goes in the 66MHz 64 bit slots, so its always 3.3V. PCIe has completely different slots |
| 16:49.55 | pimpwell | I sound like I'm from brooklyn |
| 16:50.11 | jero | arent you ? |
| 16:50.13 | moonboi | straight from tha sewa ! |
| 16:50.13 | jsolares | i have _1NXXNXXXXXX in extension for calling in the US, would a _1800NXXXXXX be matched before the other one? |
| 16:50.18 | pimpwell | gutta |
| 16:50.19 | pimpwell | I am |
| 16:50.31 | moonboi | hehe |
| 16:50.52 | jero | coppice: so you mean a digium 3.3v card will fit in a pci-x slot ? |
| 16:50.56 | jero | and work |
| 16:50.58 | pcm | pimpwell: patch the voice through voice modulation :) |
| 16:50.59 | bjohnson | jsolares: we have no way to tell |
| 16:51.00 | coppice | yes |
| 16:51.05 | jero | great |
| 16:51.15 | pimpwell | I'll just hire a stripper for cheap |
| 16:51.16 | pimpwell | fuck it |
| 16:51.17 | pimpwell | thx guys |
| 16:51.20 | *** part/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 16:51.21 | bjohnson | jsolares: there is a wiki page about pattern matching order .. basically use includes |
| 16:51.33 | jero | PCI-X 2.0: High Performance, Backward Compatible PCI for the Future |
| 16:51.36 | jsolares | hrm meh, livevoip charges for 1800 but nufone doesnt, so i want to make outbounds go with livevoip but 800's with nufone |
| 16:51.41 | jsolares | bjohnson: k thanks... i'll go look |
| 16:51.42 | qiu | hi gays ... i need some help with an phone=SIP=asterisk=gnugk=destination |
| 16:51.43 | jsolares | :D |
| 16:51.51 | coppice | the PCI-X slots throttle back if a 33MHz card is plugged in |
| 16:52.11 | bjohnson | jsolares: btw .. problems with that plan if you are from Canada |
| 16:52.12 | qiu | i heve in asterisk the next error : |
| 16:52.13 | qiu | channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/ |
| 16:52.23 | qiu | app_dial.c:1007 dial_exec: Had to drop call because I couldn't make SIP |
| 16:52.27 | jsolares | i'm not from canada |
| 16:52.31 | *** join/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 16:52.32 | bjohnson | and with that , I bid adeiu |
| 16:52.33 | tzanger | the only thing I don't like about my iax.cc DID is that the first second of audio is cut off |
| 16:52.36 | jsolares | cyas |
| 16:52.41 | tzanger | if I Answer and then Background (silence/1) it works fine |
| 16:52.52 | Connor- | I've been having problems with asterisk becoming unregistered with one of my providers.. what do I need to look at? |
| 16:52.56 | bkw_ | tzanger, that happens with all asterisk installs |
| 16:52.57 | bkw_ | really |
| 16:52.59 | tzanger | Answer and Wait(1) does not, their DID provider is doing some kind of voice delay |
| 16:53.01 | BuckRogers | hey has anyone configured *66 |
| 16:53.03 | tzanger | bkw_: uh, no |
| 16:53.05 | BuckRogers | the busy signal fix |
| 16:53.06 | bkw_ | uh yes |
| 16:53.07 | tzanger | bkw_: my PRI DIDs don't do that |
| 16:53.08 | bkw_ | I see it all the time |
| 16:53.14 | bkw_ | ours does |
| 16:53.22 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 16:53.23 | tzanger | bkw_: you don't have echotraining turned on do you? |
| 16:53.24 | bkw_ | its like aserisk sends audio before the channel is up all the way |
| 16:53.29 | bkw_ | no |
| 16:53.32 | tzanger | bkw_: hmm interesting |
| 16:53.36 | BuckRogers | im pulling my hair out trying to get my sip clients to work with it |
| 16:53.36 | tzanger | smells like a bug |
| 16:53.41 | bkw_ | but then again I have a PRI in LA that doesn't |
| 16:53.52 | bkw_ | so its very odd I tell ya |
| 16:54.33 | _Brian | Morning all.....does anyone know of a application with Asterisk that would allow me to stream the audio from a phone conversation (or meetme room) and allow it to be broadcast so that other parties can listen to the stream? |
| 16:54.52 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 16:55.12 | guugmember | hello guys, who has worked with the Varion V400P-E 4 Port E1 Digital Interface Card |
| 16:55.14 | Darwin35 | icecast |
| 16:55.29 | moonboi | what is the cheapest FXS for asterisk ? i want to connect an analog phone to my existing setup on a shoestring .... |
| 16:55.30 | _Brian | Darwin35: but can icecast be setup to listen to a voice stream? |
| 16:55.47 | BuckRogers | go with the digium |
| 16:55.54 | Darwin35 | I believe it can |
| 16:55.55 | BuckRogers | u get great support |
| 16:55.59 | mtqh | _brain: search the wiki |
| 16:56.00 | CleanerX | well, you can put it out to console, so you should get it out to icecast |
| 16:56.05 | tzanger | moonboi: cheap + VOIP don't often get good results |
| 16:56.15 | mtqh | THere is an asterisk app for icecast |
| 16:56.20 | BuckRogers | tzanger i adgree |
| 16:56.26 | BuckRogers | go with quality |
| 16:56.36 | BuckRogers | less trouble down the road |
| 16:56.46 | moonboi | i alsoo agree tzanger, but im short on cash and need a cheap solution indeed |
| 16:56.52 | Connor- | bkw, you not talking to me anymore? |
| 16:57.00 | moonboi | i tough of a sipura spa3000 maybe ... |
| 16:57.01 | BuckRogers | get your self an ATA |
| 16:57.07 | _Brian | mtqh: thanks....i looked real quick on the wiki.....i will keep looking ...... |
| 16:57.15 | BuckRogers | the 1001 will do the trick |
| 16:57.20 | moonboi | a cheap ATA connector would do |
| 16:57.26 | Hmmhesays | anyone know if call back should be written "call-back" "callback" or "call back" ? |
| 16:57.32 | moonboi | ill check , ty |
| 16:57.34 | BuckRogers | or grand stream 286 |
| 16:57.36 | guugmember | is varion totally compatible with * ? |
| 16:58.09 | BuckRogers | has anyone configured *66 the busy signal fix with * |
| 16:58.14 | xkev | hmmhesays, I'd take option B or C, leaning to B |
| 16:58.30 | Hmmhesays | yeah that's what I'm thinking |
| 16:58.34 | tzanger | very, very cool |
| 16:58.44 | tzanger | hookflash, dial extension, talk, hangup |
| 16:58.49 | tzanger | or hookflash for 3-way call |
| 16:59.00 | BuckRogers | just got those working my self |
| 16:59.04 | *** join/#asterisk m3d (~medberry@ftcrel4.hp.com) |
| 16:59.37 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
| 17:00.06 | *** join/#asterisk Spigoto (~chatzilla@207.59.131.38) |
| 17:00.24 | BuckRogers | So any one ever get Automatic redial on busy or *66 working? |
| 17:00.26 | Firebird_ | Anyone can explain me why I've got such a bad quality in monitor wav files ? |
| 17:00.32 | BuckRogers | i really could use some help |
| 17:01.32 | *** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com) |
| 17:02.17 | guugmember | no varion users in this channel? |
| 17:02.25 | BuckRogers | Ive looked in the Wiki and no luck |
| 17:02.45 | BuckRogers | any administrators have any advice |
| 17:03.25 | ionix | u would have to do an AGI |
| 17:03.42 | ionix | i.e on busy, call an agi and offer the possibility to redial on busy |
| 17:03.44 | BuckRogers | Thats what im having difaculty with |
| 17:03.57 | ionix | oh |
| 17:04.27 | BuckRogers | ive written agi's before but this is being stubborn |
| 17:04.58 | Darwin35 | we neeed 1 gian extensions file that has all the options in it that you just urn on and off as you need them |
| 17:05.10 | BuckRogers | once the busy number is ringing i need to bridge the two calls |
| 17:05.32 | BuckRogers | and have the origantor ring with a diffrent ring tone, the ring tone change i have covered |
| 17:05.49 | tzanger | whoa |
| 17:05.51 | tzanger | IAX Packet 712 from circuit ids 7->3conflicts with earlier call with circuit ids 3->4 |
| 17:05.55 | Pinhole | BuckRogers, does the manager api have bridging functionality? |
| 17:05.56 | tzanger | from tethereal watching a call |
| 17:05.59 | tzanger | wtf does that mean |
| 17:06.16 | BuckRogers | asterisk manager? |
| 17:06.25 | BuckRogers | perl agi? |
| 17:07.23 | Pinhole | BuckRogers: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 17:08.14 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 17:08.15 | BuckRogers | thanks |
| 17:08.17 | BuckRogers | good start |
| 17:08.19 | Pinhole | You should be able to "originate" and then "transfer". |
| 17:08.50 | Pinhole | phpagi has manager functions (cvs). I'm not sure if perl agi does. |
| 17:09.20 | Samoied | I have a fxo port (digium) connected in PABX |
| 17:09.33 | Samoied | I make calls for others phones connected in PABX |
| 17:09.38 | Trionnis | can someone point me in a direction to find out who actually provisioned an 800 number? |
| 17:09.46 | Samoied | but I want to dial 0,222222 |
| 17:09.57 | Trionnis | I need to port one away from Vonage, but I'm going to have to side step them to do it :( |
| 17:10.23 | `Sauron | Grr. |
| 17:10.24 | tafazzi | Anybody using iConnectThere.com sip provider connected to an asterisk? |
| 17:10.28 | |Vulture| | Vonage will release a number? |
| 17:10.30 | `Sauron | I forgot my phone bill at home |
| 17:10.33 | |Vulture| | good luck lol |
| 17:10.33 | Trionnis | they won't |
| 17:10.41 | Trionnis | but their PRI provider will |
| 17:10.45 | Trionnis | ;) |
| 17:10.47 | Samoied | How to i make a stop between numbers? |
| 17:10.52 | |Vulture| | haha how you going to do that? |
| 17:11.21 | Trionnis | vonage doesn't have to abide by the regs, but gblx/level3/et al. do |
| 17:11.28 | *** part/#asterisk m3d (~medberry@ftcrel4.hp.com) |
| 17:11.36 | BuckRogers | Pinhole , thankyou ill get on it right away |
| 17:12.02 | Trionnis | I've already talked to an engineer at my new provider, and they claim to have done it, and able to do it if I can tell them who provisioned the number to vonage |
| 17:12.11 | Trionnis | can't hurt to try |
| 17:12.13 | Trionnis | ;) |
| 17:12.25 | carsim | Hello I have a VoIP gateway with fxo ports and lan ports for IP Phones. I use gnugk to route IP Phone to IP Phone Calls and IP Phone to PSTN. my problem is i dont have an autoattendant...can i use asterisk to process all inbound calls from PSTN? i really need ur comment on this. thanks. |
| 17:13.30 | BuckRogers | yes you can we doit |
| 17:13.53 | BuckRogers | its all in the inbound out bound call settings |
| 17:14.08 | BuckRogers | im not the expert our head tech is |
| 17:14.23 | BuckRogers | and im sure some other places |
| 17:14.29 | BuckRogers | but it is doable |
| 17:14.36 | carsim | do i still need gnugk for this buck? |
| 17:14.38 | BuckRogers | like pamala anderson |
| 17:14.52 | BuckRogers | i would just switch completely to asterisk |
| 17:15.08 | BuckRogers | if posssible |
| 17:15.32 | carsim | so asterisk is also a gatekeeper is that what ur saying buck? |
| 17:15.45 | Trionnis | no, it's the keymaster |
| 17:15.52 | Trionnis | ./rimshot |
| 17:15.56 | carsim | :) |
| 17:16.04 | LittleRobbie | no, that was rick moranis |
| 17:16.25 | BuckRogers | yes |
| 17:16.27 | BuckRogers | could be |
| 17:16.28 | *** part/#asterisk human39 (~human39@chewie.fyi.net) |
| 17:16.30 | BuckRogers | or client |
| 17:16.38 | BuckRogers | very flexible |
| 17:16.41 | BuckRogers | like a gymnist |
| 17:16.50 | carsim | ok thanks.... |
| 17:16.50 | Trionnis | are you wanting an IVR carsim? |
| 17:16.55 | Trionnis | er |
| 17:16.58 | Trionnis | IVR menu, that is |
| 17:17.00 | jero | I really have callerID problems... Is that digiums fault? |
| 17:17.00 | carsim | yes i want an IVR |
| 17:17.05 | Trionnis | so make one :) |
| 17:17.14 | BuckRogers | Proble not Jero |
| 17:17.29 | Trionnis | check pm |
| 17:17.31 | Trionnis | ;) |
| 17:17.34 | jero | I only get 1 of 10 callerID |
| 17:17.34 | ManxPwr | This PA168 phone looks pretty cool, but the firmware.....is not ready for prime time. |
| 17:17.35 | BuckRogers | check your configs |
| 17:17.40 | *** join/#asterisk Can0beans (~root@c-24-3-113-223.client.comcast.net) |
| 17:17.43 | jero | Im pretty sure my config is ok |
| 17:17.57 | *** part/#asterisk Can0beans (~root@c-24-3-113-223.client.comcast.net) |
| 17:18.00 | BuckRogers | have u called digium |
| 17:18.11 | *** join/#asterisk Legend (~legend@office.bgcfreedom.com) |
| 17:18.12 | BuckRogers | they are very good with support jero |
| 17:18.13 | Legend | is nufone donw? |
| 17:18.23 | jero | BuckRogers, not yet |
| 17:18.30 | BuckRogers | Give em a ring |
| 17:18.52 | BuckRogers | they may need to ssh into your machine |
| 17:18.56 | jero | yes |
| 17:18.58 | vaewyn | Legend: working fine for me |
| 17:19.06 | jero | Ill consider this option, thanks BuckRogers |
| 17:19.22 | BuckRogers | no problemo |
| 17:20.27 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 17:21.30 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 17:22.17 | Hmmhesays | waiting for a phone call is the worst thing EVAR |
| 17:23.07 | vaewyn | nonono... getting a phone call is the worst thing ever |
| 17:23.10 | vaewyn | :} |
| 17:23.21 | Hmmhesays | heh |
| 17:24.25 | Hmmhesays | a phone call I missed yesterday |
| 17:24.32 | Hmmhesays | because my drunk friend had my phone |
| 17:25.08 | BuckRogers | Ahh the downfalls of the devils water |
| 17:25.35 | vaewyn | no downfalls for me... :} I just burn it or cook with it :} |
| 17:26.06 | BuckRogers | Nice |
| 17:26.14 | BuckRogers | penny ala vodka |
| 17:26.31 | BuckRogers | lemon chicken with a white wine sauce |
| 17:26.54 | vaewyn | flamed white whine vinagrette |
| 17:26.58 | vaewyn | -h |
| 17:27.12 | BuckRogers | fresh motsarella |
| 17:27.22 | BuckRogers | sliced with tomato |
| 17:27.27 | vaewyn | all good |
| 17:29.18 | *** join/#asterisk akrall (~akrall@201.128.92.118) |
| 17:29.25 | akrall | guys.. anybody configured their * for FWD using IAX2? |
| 17:29.56 | |Vulture| | how much do PRI lines with unlimited LD commonly run? |
| 17:30.27 | jsolares | i have a question regarding the 4 E1/T1 cards, can i connect 2 E1's coming from the telco to the asterisk box, and the other 2 ports outgoing to another pbx? |
| 17:30.31 | BuckRogers | ehh 490-750 |
| 17:30.32 | *** join/#asterisk jdg (~jdg@CA03F909.adsl.mana.pf) |
| 17:30.49 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 17:31.00 | BuckRogers | Josolares, why not just use network cards |
| 17:31.03 | Trionnis | <akrall> guys.. anybody configured their * for FWD using IAX2? <-- yes |
| 17:31.07 | BuckRogers | there much cheaper |
| 17:31.08 | Trionnis | whoops |
| 17:31.17 | BuckRogers | keep the 2 ports open for expantion |
| 17:31.18 | Trionnis | akrall: yes |
| 17:31.27 | FuRR_ | jsolares: yes you can |
| 17:31.34 | jsolares | BuckRogers: the other pbx is not voip, it's expectinv voice e1's |
| 17:31.41 | *** join/#asterisk search_learn2005 (~Miranda@adsl-68-127-105-86.dsl.pltn13.pacbell.net) |
| 17:31.43 | BuckRogers | ahh |
| 17:31.48 | BuckRogers | limited to ss7 |
| 17:31.52 | jsolares | but it can be done right? |
| 17:31.52 | BuckRogers | yeah sure |
| 17:31.53 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 17:31.56 | BuckRogers | you can |
| 17:32.15 | jsolares | ok, dont want to spend the cash only to find out you cant. lovely |
| 17:32.27 | BuckRogers | call digium to make sure |
| 17:32.43 | search_learn2005 | Hi ! Anybody with any idea on Multicom 2000. My school has it and I am trying to find out if it can be used as a channel bank. |
| 17:32.49 | Madkiss | What could cause a lateny from up to two seconds from me to my asterisk box in LAN? |
| 17:32.49 | jsolares | i'm holding you and FuRR_ to their words :P j/k |
| 17:33.19 | BuckRogers | right on |
| 17:34.04 | BuckRogers | bubbafet? |
| 17:38.17 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 17:38.19 | puzzled | hi all |
| 17:38.26 | BuckRogers | howdy |
| 17:39.07 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:39.15 | BuckRogers | algorithmn |
| 17:39.19 | BuckRogers | i hate that guy |
| 17:39.25 | BuckRogers | thinks he knows everything |
| 17:39.28 | *** part/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 17:39.42 | *** join/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 17:40.22 | Madkiss | Oh come on, somebody must have had this problem already. |
| 17:40.28 | Sedorox | ? |
| 17:41.07 | ManxPwr | Madkiss, What is your specific problem? |
| 17:41.25 | Madkiss | My asterisk Server is in the same LAN the client is in, and in the echo test, I get up to two seconds delay. |
| 17:41.42 | ManxPwr | I have never heard of that problem. |
| 17:41.56 | ManxPwr | What are the ping times between the server and the phone? |
| 17:42.25 | Madkiss | 64 bytes from 192.168.0.4: icmp_seq=2 ttl=255 time=0.284 ms |
| 17:42.42 | *** join/#asterisk Xoubir (~fsdfsd@edatissa.net4.nerim.net) |
| 17:42.57 | search_learn2005 | Hi ! Anybody with any idea on Multicom 2000. My school has it and I am trying to find out if it can be used as a channel bank. |
| 17:43.08 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 17:43.24 | Xoubir | hi all |
| 17:43.47 | ManxPwr | ~google "Multicom 2000" |
| 17:44.41 | ManxPwr | Madkiss, hardphone or softphone? |
| 17:44.46 | Madkiss | ManxPwr: Softpjome |
| 17:44.50 | ManxPwr | Madkiss, If it's a softphone then I HAVE heard of that problem. |
| 17:45.00 | ManxPwr | It means "your damn OS or sound drivers really suck" |
| 17:45.04 | ManxPwr | NEXT! |
| 17:45.06 | Madkiss | aharm? |
| 17:45.10 | Madkiss | erm ... |
| 17:45.34 | jsolares | hehehe |
| 17:45.58 | Madkiss | wait, you seem to be right |
| 17:46.20 | *** join/#asterisk criptos (~criptos@dsl-200-78-97-55.prod-infinitum.com.mx) |
| 17:46.21 | Trionnis | lol |
| 17:46.26 | ManxPwr | Why do you think many people say "softphones suck"? |
| 17:46.26 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 17:46.32 | Trionnis | 'cause they do? |
| 17:46.33 | Trionnis | :) |
| 17:46.34 | ManxPwr | Oddly enough, because they do. |
| 17:46.41 | Madkiss | I never heard anybody saying that. |
| 17:47.23 | criptos | any one have used Pleiades Channel Bank 30 FXO Ports and 1-E1 with * ? |
| 17:47.25 | criptos | comments? |
| 17:47.54 | harryvv | Saw the spa 841 was not really impressed. Did pickup my spa 1000 though. |
| 17:48.47 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 17:49.07 | zipp | ManxPwr, mostly because they don't use a good mic/speaker i.e. a headset |
| 17:49.15 | search_learn2005 | ManxPwr: I read all those websites before, but none of them explain if it can be used as channel bank or not. The real issue is that my school has 7 PSTN lines coming in, and already have a 10/100 network and analog phones in each of the 50 rooms. I have been trying to find out what solution is better, buying VOIP phones, or buying a channel bank and using the analog phones. |
| 17:49.26 | zipp | using a cheap mic and labtec speakers would make anyone think a softphone sucks |
| 17:50.17 | Trionnis | ;) |
| 17:50.23 | harryvv | zip I like the xlite but I wonder if there was a way to map hotkeys for it. :) |
| 17:50.39 | ionix | sip phones search_learn2005 |
| 17:50.40 | redder86 | Is there a reason that you cannot retrieve both new and old/saved messages from Comedian mail in one call? I have to call in, listen to new messages. Hang up. Call in again, listen to old messages. How can I do that in just one call? |
| 17:50.48 | *** join/#asterisk jhavard (~jhavard@ryouko.7SP.net) |
| 17:51.04 | harryvv | you dont need to do that |
| 17:51.09 | *** join/#asterisk Othello (Othello@nusnet-216-182.dynip.nus.edu.sg) |
| 17:51.26 | search_learn2005 | ionix: Are SIP phones VOIP phones? |
| 17:51.50 | Trionnis | search_learn2005: I'd think using the voip phones would be a more forward-looking choice (excuse the buzzword) |
| 17:51.59 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:52.22 | Trionnis | now there's a quote for bash.org |
| 17:52.26 | Trionnis | => |
| 17:52.38 | Trionnis | search_learn2005: yes, they are |
| 17:52.46 | search_learn2005 | Trionnis: What about the echo problem? |
| 17:53.00 | Trionnis | on a LAN, you shouldn't have much, if any |
| 17:53.15 | Trionnis | IMHO, of course |
| 17:53.15 | criptos | which channel bank for onlly fxo channel would recommend me? |
| 17:53.34 | ManxPwr | Using VoIP phones lets you leverage your synergy! |
| 17:53.48 | Trionnis | lol |
| 17:53.56 | jhavard | and increase your ROI on your CRM and ERP systems. |
| 17:54.00 | Trionnis | yes!! |
| 17:54.14 | Zeeek | ManxPwr what phones do you use/like? |
| 17:54.19 | Zeeek | besides Cisco? |
| 17:54.43 | Trionnis | *cough* |
| 17:54.58 | *** join/#asterisk cmslaght (~cmslaght@admin.ambt.net) |
| 17:55.01 | search_learn2005 | Trionnis: When I looked at Digicom website they only have 4 fxo and 4 fxs cards. How am I going to serve 7 fxo and 50 fxs? |
| 17:55.14 | Trionnis | 2 cards, and ip phones |
| 17:55.15 | Sedorox | channel banks |
| 17:55.18 | terrapen | i'm still trying to decide between the Cisco 7960 and the IP500 |
| 17:55.19 | Trionnis | ;) |
| 17:55.21 | ManxPwr | Reminds me of the time a customer wanted a "company mission statement" from my consulting company. I went to the RedHat web site, found their mission statement, changed it slightly and gave it to the customers. |
| 17:55.21 | terrapen | its a tough call |
| 17:55.37 | harryvv | What Poe hubs/switches are aviable to power the ip phones. |
| 17:55.40 | terrapen | the cisco is built better but the LCD is not as good and the phone is a bitch to get the latest fw on |
| 17:55.49 | cmslaght | has anyone seen this: Feb 22 12:47:03 WARNING[19750]: chan_zap.c:4503 zt_indicate: Don't know how to set condition 17 on channel Zap/2-1 |
| 17:55.52 | ManxPwr | Zeeek, Polycom Soundpoint 300 & 500, Cisco 7905G/7940G/7960G, SIPura SPA-841 (once they get the gain problem fixed) |
| 17:56.03 | ariel_ | harryvv, for the price I like the Netgrear Swith's |
| 17:56.05 | Trionnis | brb |
| 17:56.10 | terrapen | the polycom does not feel as well built but is a cinch to install and the price is right |
| 17:56.28 | Zeeek | ManxPwr thanks for that - I was wondering if anyone really thinks the ip600 is worth much more than the 500? |
| 17:56.43 | Zeeek | ($100 more) |
| 17:56.44 | terrapen | its probably made from the same materials |
| 17:56.50 | terrapen | and at that price, get a 7960 |
| 17:56.52 | harryvv | Manx, seen the spa 841 looks cheap and whats with the wierd symbols :) Seen a new Astra which is owned by another company looks really nice |
| 17:56.58 | search_learn2005 | Sedorox : Why channel bank but not ip phones with two cards? |
| 17:57.18 | Sedorox | If you can.. do all IP Phones.. would be better |
| 17:57.25 | Sedorox | but if you have to include lots of regular telephones |
| 17:57.27 | Sedorox | use a channel bank |
| 17:57.50 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:57.53 | criptos | Serodox: which channel back would you recommend for only fxo usage? |
| 17:58.24 | ionix | Quintum is cheap |
| 17:58.28 | Sedorox | I haven't used any... |
| 17:58.28 | benno2 | Sedorox: what channel banks do have a good price/performance ratio ? I used a few sipura spa 2000 and they work very well and are cheap. but if you have lots of analog phones to convert its a mess because of all the power plugs zou need |
| 17:58.28 | ManxPwr | Zeeek, The 600 has more call apperances and an XML browser. Neither are features we need. |
| 17:58.30 | Sedorox | but if you look on |
| 17:58.40 | Sedorox | http://www.voipsupply.com |
| 17:58.45 | Sedorox | they have a decent listing |
| 17:58.48 | terrapen | i wonder if i can get a good price on 30-35 7960's |
| 17:58.53 | terrapen | voipsupply is good |
| 17:58.58 | search_learn2005 | Sedorox: what do you think the echo problem on VOIP , will channel banks solve this problem? |
| 17:58.58 | Sedorox | yea |
| 17:58.59 | Zeeek | ManxPwr - me neither - thx |
| 17:59.00 | terrapen | friendly folks |
| 17:59.08 | harryvv | Manx, how much for the 600 |
| 17:59.10 | Sedorox | echo problems? |
| 17:59.12 | ariel_ | you can get some good adtrans 750 and 850 for 400 or 500 dollars on ebay they work. |
| 17:59.24 | ManxPwr | harryvv, I don't know. I just said we didn't use them. |
| 18:00.12 | Sedorox | for a channel bank.. your looking around $99/channel |
| 18:00.25 | Sedorox | which is on par with getting a spa2000 or similiar for each phone |
| 18:00.49 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 18:01.23 | ManxPwr | A channel bank gives you a central place to admin phones. |
| 18:01.33 | ManxPwr | Using ATAs give you 24 places to admin phones. |
| 18:01.34 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 18:01.36 | Sedorox | yea |
| 18:01.46 | search_learn2005 | Sedorox: Instead of paying $99 a channel, do you think I should pay $99 a VOIP phone and make the installation a little bit less cumbersome, and future proof? |
| 18:02.05 | Sedorox | search_learn2005: future proof.. maybe |
| 18:02.07 | Sedorox | but it depends |
| 18:02.12 | terrapen | if you had to deploy between 30-50 phones, would you choose Cisco 7960s or Polycom |
| 18:02.15 | ManxPwr | search_learn2005, Because $99 VoIP phones suck. |
| 18:02.17 | terrapen | err Polycom IP500 |
| 18:02.22 | terrapen | its such a tough call |
| 18:02.23 | Sedorox | do you have cat5 drops at every location you want to have a phone? |
| 18:02.28 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 18:02.31 | benno2 | search_learn2005: with the sipura spa-2000 you pay around $45 per channel |
| 18:02.42 | benno2 | but you need an ata for each 2 phones |
| 18:02.44 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 18:02.50 | __Sparks_ | Wonder if someone can help me here! - I have an X101P card in my asterisk box, I have played with the rx and tx gain, and also the echotraining, but still i am getting an echo (about 1/10th of a second delay) is there any magic formula! |
| 18:02.52 | Sedorox | because say.. if you already have a exsisting setup with all phones coming back to a room.. you could setup a channel bank to move all the phones over to voip |
| 18:02.53 | benno2 | so converting 30 phones uses 15 ethernet ports on the switch |
| 18:02.56 | terrapen | search, you definitely want IP phones.... |
| 18:03.40 | ManxPwr | Once SIPura fixes their firmware you could go with a SPA-841. Gives you 2 lines for $99 and 4 lines for $130 |
| 18:03.51 | ManxPwr | But no PoE and no switch port. |
| 18:04.03 | Sedorox | depends on his enviroment tho... |
| 18:04.05 | *** join/#asterisk fearnor (~alex@66.250.55.66) |
| 18:04.14 | ManxPwr | This PA168 phone I just got today might be a good solution if I can figure out how to configure the damn things. |
| 18:04.15 | Sedorox | if its like a school.. it might be hard to have a fxo for 4 rooms |
| 18:04.26 | fearnor | !summon atacomm |
| 18:04.38 | search_learn2005 | Sedorox: We have Cat-5 at every location, so will the final word be VOIP phones, two cards, no channek bank? |
| 18:04.50 | terrapen | search, that's what i would do |
| 18:04.56 | terrapen | take some time and try different phones |
| 18:05.04 | Sedorox | well VOIP Phones.. and if you have two PSTN lines.. then yes |
| 18:05.16 | Sedorox | yea.. terrapen has a good point |
| 18:05.18 | ManxPwr | I try to buy one of each inexpensive IP phone for testing. |
| 18:05.20 | terrapen | i bought a couple of different phones to try |
| 18:05.30 | terrapen | and will sell the ones i dont want on eBay |
| 18:05.33 | search_learn2005 | Sedorox: We have 7 PSTN lines coming in |
| 18:05.47 | dsmouse | terrapen: you were asking about testing 911 yesterday? |
| 18:05.54 | terrapen | but now i cannot decide between the Cisco 7960 and the Polycom IP500 |
| 18:05.56 | terrapen | yeah mouse |
| 18:06.10 | Sedorox | then get three TDM400P's |
| 18:06.13 | Sedorox | all with FXO modules |
| 18:06.25 | benno2 | ManxPwr: anyway I am sometimes wondering if PoE makes sense. I have servers that are not on an UPS and they have uptimes of 30 days and more and if you have a backup generator, the probability of longer power outages is very small. so ok for keeping the servers and switches under UPS but if IP phones loose sometimes the power (very seldom) all what can happen is a dropped call which happens on the PSTN too. |
| 18:06.27 | Sedorox | that'll give you 8 incoming.. unless of course its a 8 lines on a T1 |
| 18:06.31 | terrapen | why does he need three? |
| 18:06.36 | terrapen | aren't they four liens each? |
| 18:06.38 | search_learn2005 | Sedorox: What kind of a server do you thin we need to host asterisk for 7 fxo, 50 fxs with two cards? |
| 18:06.43 | Sedorox | he's got 7 PSTN lines... |
| 18:06.46 | Sedorox | oh |
| 18:06.48 | terrapen | yes |
| 18:06.48 | Sedorox | duhhh |
| 18:06.49 | Sedorox | yea.. sorry |
| 18:06.49 | Sedorox | 2 |
| 18:06.50 | dsmouse | terrapen: I called the police number and asked, and they said as long as I called their communications office a few minutes in advance it would be ok... ofcouse, that's Raleigh PD... |
| 18:06.50 | Sedorox | :-p |
| 18:06.55 | Sedorox | my math sucks today |
| 18:06.57 | benno2 | and since almost everyone has a cellphone even if there was a power outage you can always call emergency numbers |
| 18:06.58 | Sedorox | forgive |
| 18:07.06 | terrapen | mouse: good deal |
| 18:07.12 | Sedorox | 50 fxo with two cards?!?! |
| 18:07.31 | benno2 | anone that agrees with me ? (PoE still costs alot (PoE switches, midspans, PoE enabled phones etc) |
| 18:07.49 | terrapen | mouse, my fear is that something will happen in one of our stores and an employee will need to call 911 and we'll end up w/ a lawsuit |
| 18:08.09 | dsmouse | terrapen: /me nods. |
| 18:08.10 | terrapen | which is why i wanted to test |
| 18:08.11 | Sedorox | benno2: I haven't looked at PoE stuff much.. so can't say either way |
| 18:08.33 | dsmouse | terrapen: my feer is I'll have just stabbed a robber. |
| 18:08.36 | Sedorox | terrapen: like I said yesterday.. if you call 911 and tell them its a test.. they have no problem with it |
| 18:08.39 | dsmouse | fear |
| 18:09.20 | Sedorox | like I said.. my poppop does it whenever he gets a new phone.. last time the woman at 911 actually appreciated him making sure it worked |
| 18:09.44 | terrapen | I DONT CALL 911, I CALL .357 |
| 18:09.52 | dsmouse | Sedorox: she was prolly just happy it wasn't another drug overdose :) |
| 18:09.56 | Sedorox | ummm ok... |
| 18:09.57 | tzanger | terrapen: :-) |
| 18:10.04 | Sedorox | lol |
| 18:10.12 | tzanger | help, help I can't find the '.' on my phoneset! |
| 18:10.17 | Trionnis | lol |
| 18:10.17 | terrapen | (actually, .45, but that's neither here nor there) |
| 18:10.28 | Trionnis | "where's the any key!!" |
| 18:10.37 | tzanger | .357 is more than enough to stop someone. especially copper-jacket hollowpoints |
| 18:10.38 | terrapen | tz: just dial star |
| 18:10.45 | qiu | hi ... i need, again, some help with oh323 |
| 18:10.51 | terrapen | i don't own a .357 tho |
| 18:10.59 | terrapen | i have an M1911 |
| 18:11.02 | qiu | i get this "warning" |
| 18:11.04 | qiu | WARNING[21557]: chan_oh323.c:2218 oh323_write: OH323/L6245: Unable to write to fd 46 (32, Broken pipe) |
| 18:11.04 | Trionnis | feh |
| 18:11.08 | Trionnis | get a Glock |
| 18:11.14 | terrapen | fuck glock :P |
| 18:11.15 | Trionnis | much better made than gov't crap |
| 18:11.16 | qiu | and is no sound |
| 18:11.19 | terrapen | i have a Sig Sauer, too |
| 18:11.22 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 18:11.23 | Trionnis | ah |
| 18:11.25 | tzanger | my dad owns two. I fired it when I was 6 years old. (we were at the range and he wanted to put the fear of god into me about handguns) -- it worked. i won't touch a handgun. I have no problem whatsoever with rifles though |
| 18:11.27 | Trionnis | now that I'll agree with :) |
| 18:11.39 | harryvv | My ISP dns server went down. |
| 18:11.45 | Trionnis | so use another one? |
| 18:11.46 | Trionnis | :) |
| 18:11.49 | harryvv | hehe |
| 18:11.49 | *** part/#asterisk search_learn2005 (~Miranda@adsl-68-127-105-86.dsl.pltn13.pacbell.net) |
| 18:11.52 | ManxPwr | I'm not a fan of guns, but admit they are useful. |
| 18:11.53 | benno2 | does the cisco 7960 work well with asterisk ? |
| 18:11.54 | WGFreewill | qiu: what version of asterisk / oh323 / pwlib / openh323 are you using |
| 18:12.00 | terrapen | if you want a home defense weapon, you buy a 12-gauge shotgun, end of story |
| 18:12.01 | harryvv | well there was two going to use more. |
| 18:12.08 | |Vulture| | benno2: yes |
| 18:12.08 | terrapen | benno: yes it does |
| 18:12.13 | benno2 | thanks |
| 18:12.16 | Trionnis | 198.6.1.2 and 198.6.1.3 |
| 18:12.18 | ManxPwr | "benno2 does the cisco 7960 work well with asterisk ?" <-- now there's another question we get asked every fucking day. |
| 18:12.21 | qiu | owlib 1.15.2 |
| 18:12.21 | vaewyn | terrapen: amen! |
| 18:12.25 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 18:12.27 | qiu | openh323 1.8.1 |
| 18:12.28 | dsmouse | terrapen: I want a home big enough that I NEED a rifle |
| 18:12.28 | Trionnis | if those are down, there's a lot bigger problems on the internet |
| 18:12.32 | Trionnis | ;) |
| 18:12.34 | terrapen | mouse: heh |
| 18:12.35 | tzanger | I don't remember this very well but apparently he was squatting behind me with his hands over mine and told me to squeeze the trigger... After it went off I ran and hid in the car until it was time to go home :-) |
| 18:12.36 | qiu | and asterisk 1.0.5 |
| 18:12.38 | vaewyn | dsmouse: hahaha |
| 18:12.39 | WGFreewill | (but the error usually mens that you cant access a device or file, some permissions problem) |
| 18:12.44 | terrapen | i'm glad im not your neighbor :P |
| 18:12.52 | tzanger | I have to say it was very effective |
| 18:12.57 | terrapen | "I have a deer rifle for home defense!!!" |
| 18:13.14 | ManxPwr | terrapen, Deer try to attack your home? |
| 18:13.16 | harryvv | anyone know of one root dns server address I can use. |
| 18:13.17 | pcm | terrapen: and how often do you have to use it for defense ? |
| 18:13.18 | dsmouse | outtolunc: bows. |
| 18:13.18 | tzanger | I personally don't see the need for handguns. Rifles/shotguns/etc though, sure |
| 18:13.20 | qiu | yes ... i know ... but i dont know where is the problem file/socket |
| 18:13.22 | terrapen | tzanger: at least your stepfather wasn't a militia guy |
| 18:13.26 | *** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net) |
| 18:13.28 | Trionnis | harryvv: use those I gave you |
| 18:13.29 | tzanger | terrapen: heh |
| 18:13.30 | pcm | terrapen: it's it better to live in a safe neighberhood |
| 18:13.37 | Trionnis | they're uunet cache servers |
| 18:13.39 | vaewyn | outtolunc: ohh no... only criminals will have guns then... so it is... "stop... don't shoot!" |
| 18:13.43 | WGFreewill | i guess the recommendation is to run cvs stable, oh323 0.65, with the janus patch4 releases |
| 18:13.58 | ManxPwr | WGFreewill, I would never recommend that. |
| 18:14.04 | ManxPwr | But others might. |
| 18:14.08 | Inv_arp | if one is behind a modem what codec is best to use? |
| 18:14.09 | terrapen | my stepfather, before my mother divorced him, ended up going to federal prison for possession of plastic explosives and a silencer |
| 18:14.13 | terrapen | err silencers |
| 18:14.17 | WGFreewill | I would be excited to hear of other working combinations |
| 18:14.19 | WGFreewill | for chan_h323 |
| 18:14.22 | WGFreewill | or oh323 |
| 18:14.30 | WGFreewill | I have been swimming in h323 hell |
| 18:14.31 | WGFreewill | myself |
| 18:14.33 | Zeeek | heh check this: http://del.icio.us/mrbill/asterisk |
| 18:14.41 | ManxPwr | WGFreewill, ANYONE that uses H323 is in hell. |
| 18:14.41 | vaewyn | Inv_arp: ilbc, gsm or g.729 |
| 18:14.47 | terrapen | mrbill heh |
| 18:14.53 | Zeeek | and this: http://del.icio.us/nothing2005 |
| 18:14.55 | dsmouse | mr_bill! |
| 18:14.58 | WGFreewill | I have legacy equipment |
| 18:14.59 | vaewyn | ManxPwr: amen! |
| 18:15.01 | WGFreewill | that demands it |
| 18:15.02 | Zeeek | asterisk is almost mainstream |
| 18:15.16 | ManxPwr | WGFreewill, Then expect to be in hell. |
| 18:15.22 | WGFreewill | many thousands of circuits |
| 18:15.25 | Zeeek | who among you is mrBill ? |
| 18:15.29 | ManxPwr | Accept H323 hell. Embrace your H323 hell. |
| 18:15.31 | WGFreewill | I am |
| 18:15.32 | WGFreewill | but |
| 18:15.36 | terrapen | i know mrbill |
| 18:15.40 | WGFreewill | chan_h323, the latest |
| 18:15.40 | Inv_arp | vaewyn: gotten ok results with them? |
| 18:15.43 | vaewyn | waster even |
| 18:15.44 | WGFreewill | low volumes |
| 18:15.45 | qiu | sorry .... pwlib: 1.6.6 and openh323 1.13.5 .... |
| 18:15.45 | WGFreewill | works okay |
| 18:15.53 | WGFreewill | and oh 323 |
| 18:15.53 | qiu | with patch |
| 18:15.53 | terrapen | he's a unixnet guy |
| 18:15.55 | WGFreewill | 0.65 |
| 18:15.56 | Zeeek | SNL mrBill or mrbill asterisk? |
| 18:16.13 | WGFreewill | yep |
| 18:16.16 | WGFreewill | inaccessnetworks |
| 18:16.19 | WGFreewill | has their versions |
| 18:16.22 | WGFreewill | on the website |
| 18:16.24 | WGFreewill | for download |
| 18:16.25 | ManxPwr | WGFreewill, chan_oh323 does not use Asterisk's RTP stack. |
| 18:16.36 | WGFreewill | agreed |
| 18:16.38 | Inv_arp | i cant believe aol's chat tcp voip sounds fine under a modem |
| 18:16.40 | vaewyn | Inv_arp: pray you never are forced to use a modem... :} |
| 18:16.42 | WGFreewill | soundcard dsp interface |
| 18:16.42 | ManxPwr | What I recommend is to use the chan_h323 that's included with Asterisk. |
| 18:16.46 | terrapen | CNN depresses me today |
| 18:16.52 | terrapen | read the headline |
| 18:16.53 | Xoubir | Do any1 know if there is a way of making * 1.0.5 support attended transfers ? (feature is currently in CVS HEAD, and i can't update to cvs cause i'm using bristuff drivers) |
| 18:16.55 | terrapen | very very sad |
| 18:17.05 | Zeeek | this is really interesting, the number of asterisk related stuff on this site |
| 18:17.12 | *** part/#asterisk akrall (~akrall@201.128.92.118) |
| 18:17.20 | WGFreewill | I have crashes |
| 18:17.35 | qiu | ManxPwr: i tried to compile chan_h323 and it didn't work on cvs asterisk and asterisk-1.0.5 |
| 18:17.36 | ManxPwr | Xoubir, Use real words. There will NEVER EVER EVER be a 1.0.x asterisk that supports attended transfers using "t" or "T" on the Dial line. Never. Ever. |
| 18:17.39 | WGFreewill | when I get moderate call flow |
| 18:17.51 | tzanger | ManxPwr: out of curiosity, why? |
| 18:17.52 | Inv_arp | vaewyn: im on dsl but my friend in domncan repubic is on modem ... aol's solution works fine... but want to try iax based ones |
| 18:17.57 | ManxPwr | qiu, then something else is wrong. Perhaps you didn't follow the instructions to the letter. |
| 18:17.59 | tzanger | ManxPwr: too much changed? |
| 18:18.10 | ManxPwr | tzanger, No. 1.0.x is for bug fixes only. |
| 18:18.16 | ManxPwr | No new features. |
| 18:18.21 | tzanger | ManxPwr: ahh |
| 18:18.26 | vaewyn | Inv_arp: give it a try then... ilbc and g.729 should work the best... gsm is a close second |
| 18:18.35 | tzanger | ManxPwr: so the "Bounty: $50 - Backport Latest Dundi to Stable" title on -dev is never gonna be filled? |
| 18:18.44 | Xoubir | okay i see, then, is there a way of compiling bristiff drivers against cvs head of * ? :) |
| 18:18.51 | ManxPwr | tzanger, not for an official Digium 1.0.x no. |
| 18:18.55 | tzanger | :-) |
| 18:19.06 | ManxPwr | If it gets filled it will be a fork. |
| 18:19.08 | tzanger | ManxPwr's the 1.0.x mastah |
| 18:19.25 | ManxPwr | tzanger, Actually drumkilla is the 1.0.x master. |
| 18:19.26 | qiu | ManxPwr: openh323 - 1.15.1; pwlib -1.8.1 and in cvs asterisk i get the error "undefined reference to `h323_show_codec" |
| 18:19.32 | tzanger | ManxPwr: ah |
| 18:19.38 | tzanger | you're just the cheering section and party whip :-) |
| 18:19.47 | redder86 | Anyone know how to access both new and old voicemail messages in Comedian Mail in one call? |
| 18:20.18 | vaewyn | redder86: umm... change folders |
| 18:20.39 | ManxPwr | qiu, What part of "This code runs on Open H.323 v1.12.2 and PWLib v1.5.2. If you use different |
| 18:20.39 | ManxPwr | versions, you are on your own. See the Makefile for more details." do you not understand in /path/to/asterisk/channels/h323/README?? |
| 18:20.53 | vaewyn | hehehe |
| 18:21.03 | ManxPwr | It doesn't get much clearer than that. |
| 18:21.23 | ManxPwr | "I didn't follow the instructions and now it doesn't work! chan_h323 sucks!" |
| 18:21.32 | WGFreewill | head cvs |
| 18:21.43 | tzanger | ManxPwr: well h323 *does* suck mightily |
| 18:21.45 | WGFreewill | uses 1.15.1 and 1.8.1 |
| 18:21.46 | qiu | ManxPwr: openh323 - 1.15.1; pwlib -1.8.1 |
| 18:21.48 | ManxPwr | I am of course referring to the README in 1.0.x |
| 18:22.00 | WGFreewill | stable uses 1.12.2 and 1.5.2 |
| 18:22.08 | ManxPwr | I guess qiu doesn't understand that. |
| 18:22.10 | qiu | MancPw: pwlib: 1.6.6 and openh323 1.13.5 are on another machine |
| 18:22.19 | terrapen | why do so many people want to use H.323? |
| 18:22.22 | terrapen | i don't understand |
| 18:22.30 | WGFreewill | 864 terminal line(s) <<<< need to get there |
| 18:22.33 | terrapen | every couple of hours, there is a question about it |
| 18:22.34 | WGFreewill | h323 is the only way |
| 18:22.35 | ManxPwr | terrapen, They don't. The equipment they connect to require it. |
| 18:22.36 | vaewyn | cause it is a cancer from the past that is hard to vanquish |
| 18:22.42 | qiu | in cvs from README : "You must run Open H.323 v1.15.1 and PWLib v1.8.1." |
| 18:23.02 | ManxPwr | qiu, CVS is doesn't always work. |
| 18:23.10 | ManxPwr | If you want something that works and is stable then use 1.0.x |
| 18:23.12 | WGFreewill | qiu: right I have that working stable on half a dozen machines, with low call volume |
| 18:23.17 | ManxPwr | CVS-HEAD works most of the time. |
| 18:23.43 | redder86 | vaewyn: thanks. The voice prompts after deleting new messages does not say "press 2 to change folders". |
| 18:23.48 | WGFreewill | but more than 10 or so calls and the CVS-HEAD takes a header |
| 18:23.51 | redder86 | vaewyn: although pressing 2 does work |
| 18:24.03 | qiu | ManxPwr: vrerry sory ... i looked now on stable and indeed "Open H.323 v1.12.2 and PWLib v1.5.2" |
| 18:24.13 | qiu | i will try these |
| 18:24.18 | ManxPwr | WGFreewill, I suspect that's a problem with not using Asterisk's RTP stack. |
| 18:24.18 | qiu | thanks |
| 18:24.20 | vaewyn | redder86: if you get the help recap at the end it says it... but yeah other than that it is a bit obtuse |
| 18:24.33 | WGFreewill | thats with chan_h323 |
| 18:24.43 | WGFreewill | I have a stack of chan_h323 boxes |
| 18:24.47 | WGFreewill | and a stack of oh323 boxes |
| 18:24.50 | WGFreewill | debian sarge |
| 18:24.52 | djin | are callgroups limited to contexts. Iaw. can a callgroup '10' exist in multiple contexts? |
| 18:24.54 | *** join/#asterisk mgomes_mpg (1000@Froes.microlink.com.br) |
| 18:25.03 | ManxPwr | WGFreewill, Then report the problem with chan_h323 to bugs.digium.com. |
| 18:25.21 | WGFreewill | yeah I am getting ready to try stable myself here |
| 18:25.31 | WGFreewill | 1.0.x |
| 18:25.48 | mgomes_mpg | hello! is Chris Hozian here in chat, or other digium staff member ? |
| 18:26.38 | WGFreewill | just was wondering if other had successful chan_h323 usage with large call volumes |
| 18:26.49 | WGFreewill | (or oh323 for that matter) |
| 18:27.09 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:27.36 | JerJer | do not use -stable and H.323 |
| 18:27.41 | JerJer | well at least chan_h323 |
| 18:27.54 | JerJer | WGFreewill: define large call volumes |
| 18:28.13 | WGFreewill | right now I am working on moving about a 20 calls max |
| 18:28.20 | WGFreewill | but I have DS3s of voice |
| 18:28.27 | WGFreewill | waiting behind legacy gear |
| 18:28.31 | WGFreewill | for the guture |
| 18:28.34 | WGFreewill | future |
| 18:28.56 | WGFreewill | 864 terminal line(s) |
| 18:29.13 | WGFreewill | a farm of asxxxx that look about like this |
| 18:30.02 | WGFreewill | latest asterisk with Open H.323 v1.15.1 and PWLib v1.8.1 |
| 18:30.30 | WGFreewill | with chan_h323 has been the closest I have gotten |
| 18:30.35 | JerJer | WGFreewill: and you are sadly mistaken |
| 18:30.41 | JerJer | There are better ways than H.323 |
| 18:30.50 | WGFreewill | couple time a day though I have to restart the asterisk box |
| 18:30.55 | WGFreewill | I agree |
| 18:31.02 | WGFreewill | when chan_h323 crashes |
| 18:31.10 | WGFreewill | my SIP and IAX calls are still flowing |
| 18:31.14 | JerJer | then provide some valid debug |
| 18:31.25 | WGFreewill | asterisk doesnt core |
| 18:31.31 | WGFreewill | it just sits there at the prompt |
| 18:31.39 | tzanger | hmm |
| 18:31.40 | WGFreewill | I have to ctrl-c |
| 18:31.47 | JerJer | i simply do not understand why so many people have problems with chan_h323... we have abused the hell out of it for outbound calling and cannot make it fail |
| 18:31.54 | WGFreewill | and killall -9 asterisk ; killall -9 mpg123 |
| 18:31.58 | tzanger | I have txgain=-1 and rxgain=-1 in my /etc/asterisk/zapata.conf (and yes it's been reloaded) |
| 18:32.09 | tzanger | but zttool shows the levels as 0 and 1 |
| 18:32.11 | JerJer | granted we don't setup assloads of calls in a short period of time, but still |
| 18:32.11 | WGFreewill | outbound works better than inbound |
| 18:32.12 | mikegrb | the killall -9 mpg123 is not necessary |
| 18:32.26 | mikegrb | they will die when asterisk dies |
| 18:32.27 | WGFreewill | what OS |
| 18:32.36 | *** join/#asterisk AsteriskNooB (AsteriskNo@207-114-232-10.gen.twtelecom.net) |
| 18:32.36 | JerJer | um Linux |
| 18:32.36 | AsteriskNooB | good morning all! |
| 18:32.36 | WGFreewill | is your base box built with |
| 18:32.45 | WGFreewill | Redhat 8.0, debian sarge, etc |
| 18:32.51 | JerJer | all of the above |
| 18:32.55 | mikegrb | WGFreewill: those are not oses |
| 18:33.32 | WGFreewill | i know, apologize |
| 18:33.33 | AsteriskNooB | hey, if I have DID number definitions and I dont have a definition for a number that is sent in... will it default out to the s? |
| 18:33.43 | WGFreewill | more concerned with voice, but every once in a while something wont compile because of some wierd library location that let say redhat moved |
| 18:33.49 | mgomes_mpg | hello! is Chris Hozian here in chat, or other digium staff member ? |
| 18:33.49 | JerJer | i even lowered myself to install debian in house and fired up a call generator for like 2 weeks without so much as a hiccup |
| 18:33.55 | mikegrb | AsteriskNooB: no, add a catchall |
| 18:34.15 | AsteriskNooB | mikegrb: hmm, i might need to re-structure my dialplan then |
| 18:34.39 | JerJer | again outbound H.323 calls... inbound there is still some strange issues when using certain endpoints |
| 18:34.47 | AsteriskNooB | mikegrb: under default i have my main extensions, and then i have include didnumbers and in didnumbers i've defined my main did's |
| 18:34.51 | WGFreewill | most of my traffic is outbound |
| 18:34.53 | WGFreewill | termination |
| 18:35.06 | WGFreewill | very small amounts inbound to asterisk |
| 18:35.06 | mikegrb | no, after the list of did's just add an include with a goto(default,s,1) or whatever with a pattern match to catch whatever else |
| 18:35.33 | WGFreewill | when the peaks come |
| 18:35.40 | AsteriskNooB | but what happens when they are in the auto-attendant pressing numbers, then it wont say invalid but just keep repeating the menu right? |
| 18:35.43 | WGFreewill | we almost know to the hour bases on usage patterns |
| 18:36.07 | WGFreewill | h323 hangs, asterisk is there, they can even call me over the SIP channel to tell me to restart it |
| 18:36.19 | WGFreewill | but no core gets dropped |
| 18:36.26 | JerJer | when it hangs attach with gdb |
| 18:36.28 | JerJer | then run |
| 18:36.34 | JerJer | thread apply all bt full |
| 18:36.37 | JerJer | and fire up a bug |
| 18:36.39 | JerJer | then tell me |
| 18:36.39 | *** join/#asterisk clive- (~pirch@rrba-146-90-178.telkomadsl.co.za) |
| 18:36.46 | WGFreewill | sweet, will do |
| 18:37.20 | JerJer | make damn sure you are on cvs -head or it will be closed |
| 18:37.34 | Essobi | Hah. |
| 18:38.00 | WGFreewill | grabbed it 2 days ago |
| 18:38.02 | WGFreewill | np |
| 18:38.17 | *** part/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 18:38.30 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 18:38.39 | Essobi | Oh god. |
| 18:38.39 | AsteriskNooB | mikegrb: would it be better to have DID's in default and then have it Goto() another context, like CompanyADefault? |
| 18:38.51 | Essobi | Yet another linux install for ricers. http://freshmeat.net/projects/mygeos/?branch_id=54678&release_id=188593 |
| 18:39.36 | Essobi | AsteriskNooB Completely up to you. It's all preference. do you want one big context for all your SIP/ZAP trunks to land in and route out? |
| 18:39.54 | Essobi | Or would you rather have different ones land in different places? |
| 18:40.24 | AsteriskNooB | Essobi: well the problem I'm seeing is if I do a catch-all on [didnumbers] which is included in default and I send the catch all to default, I'm going to have a loop right? |
| 18:40.41 | Essobi | Hmm. That reminds me. I should write a MySQLGoto app. |
| 18:40.41 | harryvv | whats the difference between the spa 1000 and 1001? |
| 18:41.00 | JerJer | 1 |
| 18:41.03 | AsteriskNooB | Would it help if I pastebin'd my extensions.conf? |
| 18:41.05 | tzanger | harryvv: a 1 |
| 18:41.11 | jsolares | ok, a question, i have an intel 915 board, should i get 5v or 3.3v pci cards? |
| 18:41.20 | tzanger | jsolares: are the PCI slots 5v or 3.3v? |
| 18:41.22 | mikegrb | AsteriskNooB: I would have the DIDs in a different context but it is totally up to you |
| 18:41.29 | jsolares | how do i find out :p |
| 18:41.34 | jsolares | that's a better question i guess |
| 18:41.35 | Essobi | AsteriskNooB I think there's context matching preferences.. it's only going to land on one unless you do something silly like s,1,Dial(Local/s@default) |
| 18:41.41 | tzanger | jsolares: look at the mobo |
| 18:41.43 | |Vulture| | jsolares: digium has a picture on their site |
| 18:41.49 | AsteriskNooB | mikegrb: so what would I do, change my incoming context to the DID context? |
| 18:41.54 | Essobi | It's been awhile since I been down that road. |
| 18:42.24 | jsolares | ohh i see the picture, thanks |
| 18:42.24 | AsteriskNooB | zapata.conf would point to didnumbers instead of default? |
| 18:42.39 | |Vulture| | no.. extensions.conf |
| 18:42.52 | AsteriskNooB | like my incoming sip points to [infromnet] then strips a diget and points to [didnumbers] ? |
| 18:48.26 | bjohnson | harryvv: the 1001 is smaller |
| 18:48.49 | bjohnson | harryvv: might be some other minor differences |
| 18:49.36 | bjohnson | harryvv: they're both 1 port fxs. My guess is that the 1000 will be discontinued .. but I don't know for certain |
| 18:49.41 | harryvv | bj, got the 1000 locally in west vancouver |
| 18:50.03 | AsteriskNooB | http://www.pastebin.com/245042 |
| 18:50.16 | bjohnson | harryvv: cool. where? |
| 18:50.23 | AsteriskNooB | Essobi and mikegrb... http://www.pastebin.com/245042 |
| 18:51.45 | AsteriskNooB | if I just point my zapata.conf to that config above (when I get the PRI) then it will work, correct? |
| 18:52.18 | AsteriskNooB | (the context in zapata i mean) |
| 18:53.15 | __Sparks_ | What's the best thing to use to interface Asterisk with a normal PSTN telephone line? |
| 18:53.28 | vaewyn | TDM400P with FXO interface |
| 18:53.54 | bjohnson | __Sparks_: a fxo |
| 18:54.04 | __Sparks_ | vaewyn, would that cure my echo problem! |
| 18:54.30 | vaewyn | FXO you are gonna have some echo... that's just the way it is |
| 18:55.24 | __Sparks_ | I see, I currently have a x100p and have echo I want rid of! |
| 18:55.30 | Connor- | anyone know how to get into the advanced/admin menu on a WRT5RGP2 ? |
| 18:55.57 | AsteriskNooB | __Sparks_: I have no echo with my two X100's do you have cancellation on? |
| 18:56.11 | harryvv | sparks i have echo on my x100p. IT self learns and the echo will diminish. |
| 18:56.43 | modulus_ | bleh |
| 18:56.50 | modulus_ | tset: standard error: Invalid argument |
| 18:57.05 | __Sparks_ | AsteriskNooB. yes I have tried playing with the gain, and also the echotraining, but it's still there |
| 18:57.34 | mgomes_mpg | hello! is Chris Hozian here in chat, or other digium staff member ? |
| 18:57.50 | __Sparks_ | harryvv, during the call, the echo does improve, but it is still there, and annoying! |
| 18:58.36 | *** part/#asterisk Xoubir (~fsdfsd@edatissa.net4.nerim.net) |
| 18:58.45 | AsteriskNooB | __Sparks_: I have 3 or 4 options turned on to get rid of it, it doesnt work whenplugged into an older PBX port but it works great straight to the telco, echocancelwhenbridged=yes echocancel=128, and echotraining=200 |
| 18:59.29 | AsteriskNooB | thats just what i'm using, soon i'll be out of that boat :D |
| 18:59.55 | __Sparks_ | AsteriskNooB, my setup is direct to the telco - I will try those setting! |
| 19:00.08 | __Sparks_ | AsteriskNooB, what do you have your txgain and rxgain set to? |
| 19:01.08 | AsteriskNooB | __Sparks_: I had to crank them up because my audio was low, I'd suggest using ztmonitor <channum> -v to watch the graph as someone is NOT the phone and plan with the numbers until the line reaches the very top (100%) |
| 19:01.17 | AsteriskNooB | __Sparks_: (both of my channels are different) |
| 19:01.39 | AsteriskNooB | sorry, as someone is NOT on the phone. :) |
| 19:01.54 | *** join/#asterisk shuric (alexander@62.89.245.9) |
| 19:01.56 | AsteriskNooB | **cant type today |
| 19:02.45 | __Sparks_ | okay, I will have another play! - thanks! |
| 19:02.53 | AsteriskNooB | __Sparks_: so anyway, both my channels have a txgain=2.0 and one has rxgain 2.0 and the other rxgain 8.0 |
| 19:03.13 | vaewyn | ouch... that is some harsh gain |
| 19:03.25 | AsteriskNooB | yeah, one of the cards is a POS |
| 19:03.33 | AsteriskNooB | but it works great |
| 19:04.10 | __Sparks_ | POS? - PointOfSale!?! |
| 19:04.20 | AsteriskNooB | the telco had to adjust their gain levels before for our old system... they have a box upstairs that turns our analog lines VOIP sends them over the T1 so we can burst with net when we arent talking, its really nice |
| 19:04.36 | AsteriskNooB | __Sparks_: POS in this case = Piece of Shit |
| 19:04.46 | __Sparks_ | lol, i see! |
| 19:05.37 | yashax | in CLI, how can I find out if my registration with termination service is successful? |
| 19:05.48 | __Sparks_ | Would getting an ISDN line be a better option? |
| 19:05.54 | KalD|Work | yashax, iax2 show registry or sip show registry |
| 19:06.26 | yashax | k.. shows, "request sent" |
| 19:06.32 | AsteriskNooB | __Sparks_ ISDN is ALWAYS the better option in my opinion, but how many lines do you need? |
| 19:06.35 | vaewyn | If you already have a T get a voice/data integrated one and dro pyour lines opn that |
| 19:06.46 | KalD|Work | yashax, so the other side has not replied yet... perhaps iptables is blocking? |
| 19:07.05 | AsteriskNooB | vaewyn: you talkin to me or sparks? |
| 19:07.07 | yashax | damn... of course... what ports on the FW do I need to open... thanks.. |
| 19:07.07 | *** join/#asterisk shuric (alexander@62.89.245.9) |
| 19:07.12 | __Sparks_ | AsteriskNooB, need one. but two would be better! |
| 19:07.21 | AsteriskNooB | Sparks: there ya go :) |
| 19:07.25 | vaewyn | not sure :} |
| 19:07.27 | KalD|Work | yashax, hehe happens all the time to me =) what proto you using? iax2? |
| 19:07.32 | yashax | SIP |
| 19:07.37 | yashax | :) |
| 19:07.39 | *** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it) |
| 19:07.42 | benno2 | vaewyn: your WIP phone still working well or are you discovering flaws ? |
| 19:07.45 | KalD|Work | yashax, hmm... try opening 5060 udp |
| 19:08.08 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 19:08.19 | vaewyn | benno2: the web setup interface still hoses it... but as long as I stick to configuring it via the menu it is running great |
| 19:08.24 | AsteriskNooB | vaewyn: I have a T1, but for the lines they put modules in, I can have PRI, or I can have Analog, we havent moved from our old pbx, so everything is Analog, once I convince people that the Asterisk/Cisco system is better, then I will be paying another 100 a month and getting that card switched to PRI :) |
| 19:08.46 | vaewyn | benno2: did upgrade to the rc1 formware and will try the web config again with it but... |
| 19:09.12 | vaewyn | AsteriskNooB: yeah... PRI is VERY much the way to go |
| 19:09.31 | AsteriskNooB | vaewyn: but the nicest thing is they packetize everything, when nobody's on the phone I have a FULL T1, when people start talking it takes away a little bit of bandwidth for voice... not much, i usually am seeing 1400K on speedtests with all lines lit up |
| 19:09.36 | benno2 | vaewyn: but apart the webinterface problems, does the rest work well ? (I'm considering purchasing one) |
| 19:10.35 | AsteriskNooB | vaewyn: most other T1 services I've seen take away channels completely for voice, all the time, on the phone or not, so you loose Nx64 on your bandwith, that sucks |
| 19:11.01 | vaewyn | benno2: so far... so good |
| 19:11.52 | KalD|Work | yashax, that work for you? |
| 19:12.03 | benno2 | vaewyn: the only thing what worries me is the roaming ... if the roam time could be max 1sec then it would be ideal. |
| 19:12.06 | yashax | KalD: Can you force the registration, instead of waiting for 2min retry? |
| 19:12.16 | KalD|Work | yashax, try reload |
| 19:12.18 | yashax | don't know yet... Just opened the FW... |
| 19:12.53 | yashax | still same.. |
| 19:12.55 | yashax | hmm.. |
| 19:13.10 | *** join/#asterisk _PiGreco_ (~a@adsl-120-46.38-151.net24.it) |
| 19:13.14 | _PiGreco_ | hello |
| 19:13.15 | KalD|Work | yashax, try sip reload |
| 19:13.22 | _PiGreco_ | silly question |
| 19:13.36 | vaewyn | benno2: I havn't had time to play with the switches yet... but it seems to be them... I should try 2 aps on a true hub and see how well the roaming goes... |
| 19:13.36 | _PiGreco_ | i have asterisk on my nat |
| 19:13.42 | *** part/#asterisk qiu (~qiu@andrei.digicom.ro) |
| 19:13.43 | KalD|Work | _PiGreco_, no silly questions only silly answers =) |
| 19:13.54 | bjohnson | like that one ^^? |
| 19:13.55 | benno2 | vaewyn: yes please do and let us know. |
| 19:14.06 | KalD|Work | bjohnson, exactly =) |
| 19:14.06 | _PiGreco_ | 2 iaxy clients, one inside LAN one outside |
| 19:14.23 | _PiGreco_ | asterisk sees one public ip and one private |
| 19:15.02 | _PiGreco_ | if i call in one way its all ok |
| 19:15.15 | bjohnson | err become |
| 19:15.18 | bjohnson | became? |
| 19:15.25 | bjohnson | something like that |
| 19:16.02 | _PiGreco_ | the other doesnt work |
| 19:16.20 | Himeko | the fruit? |
| 19:16.29 | yashax | hmm.. no luck.. |
| 19:16.31 | KalD|Work | _PiGreco_, do you have port forwarding for 4569 udp to asterisk? |
| 19:16.44 | KalD|Work | brb |
| 19:16.58 | Himeko | i've always like cranberry stuff |
| 19:16.58 | _PiGreco_ | yes thats not the problem |
| 19:17.02 | Himeko | er liked |
| 19:17.15 | _PiGreco_ | the problem is communications starts |
| 19:17.27 | _PiGreco_ | the public ip iaxy hears the other end |
| 19:17.29 | harryvv | bj, you have a 1000? |
| 19:17.41 | _PiGreco_ | doesnt hear, pardon |
| 19:17.51 | _PiGreco_ | the private one instead hears all ok |
| 19:18.50 | shido6 | _PiGreco_ ? |
| 19:19.07 | bjohnson | harryvv: 3 3ks and 3 2ks |
| 19:19.15 | bjohnson | and 2 x100p |
| 19:19.33 | _PiGreco_ | shido6: read above :) |
| 19:19.49 | vaewyn | bkw_: you around? |
| 19:20.03 | FuRR_ | is res_mysql missing from asterisk-addons ? |
| 19:20.03 | KalD|Work | k back |
| 19:21.45 | KalD|Work | _PiGreco_, so you get one-way communication? |
| 19:22.02 | _PiGreco_ | yes one way |
| 19:22.27 | KalD|Work | _PiGreco_, ok so the internal client works for both directions and the external only can transmit? |
| 19:22.43 | vaewyn | ok... who wants to play SIP guru? |
| 19:22.57 | _PiGreco_ | no |
| 19:23.08 | harryvv | bjohnson did you follow a quick wiki to get those working with asterisk. I am looking at the 1000 pdf documentation and its fairly leanghty. |
| 19:23.18 | _PiGreco_ | internal one just hears |
| 19:23.23 | _PiGreco_ | ext just speaks :) |
| 19:23.34 | _PiGreco_ | i dont know if its a problem of routing |
| 19:23.56 | __Sparks_ | Are there any SIP providers out there that offer free calls to UK 0800 numbers? (Other then SipGate, as the CLID is broken!) |
| 19:24.14 | Trionnis | doesn't fwd do that? |
| 19:24.29 | yashax | Guys, can anyone give me a hand in trying to figure out why my BV registration is not completing? |
| 19:24.39 | bjohnson | harryvv: there's a link on the wiki (I think on the SPA 2000 page) to a voxilla tutorial for a 2000 .. should be exactly the same for a 1000. That got me going and I've been tweaking it a bit. |
| 19:25.05 | bjohnson | __Sparks_: check the wiki |
| 19:25.10 | __Sparks_ | Trionnis, I thought fwd were only voip, not voip to pstn? |
| 19:25.28 | bjohnson | fwd offers some outbound 800 |
| 19:25.40 | bjohnson | and some transfers to other voip systems (like vonage) |
| 19:25.45 | bjohnson | and even some inbound |
| 19:25.53 | Trionnis | fwd => vonage doesn't work, last I checked |
| 19:25.55 | Trionnis | :( |
| 19:26.03 | harryvv | okay |
| 19:26.08 | harryvv | thanks bjohnson |
| 19:28.05 | Trionnis | can someone point me in a direction to find out who actually provisioned an 800 number? |
| 19:28.13 | Trionnis | i.e. gblx, level3, etc? |
| 19:28.23 | yashax | guys, anyone? |
| 19:28.53 | Inv_arp | yashax: hold up i get u a working config |
| 19:29.43 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 19:29.49 | mutilator | afternoon all |
| 19:29.56 | bkw_ | ok who here has seen false busy detection when someone is yelling over the phone? |
| 19:30.04 | bkw_ | if you have busydetect=yes |
| 19:30.06 | bkw_ | and busycount=8 |
| 19:30.20 | bkw_ | if someone is loud or yelling over the phone it could still detect that as a busy condition? |
| 19:30.23 | moonwick | huh |
| 19:30.26 | bkw_ | and kill over the call? |
| 19:30.35 | modulus_ | alll of my liiiiife |
| 19:30.38 | mutilator | not i |
| 19:30.39 | modulus_ | where have you beeeeen? |
| 19:30.43 | xkev | so if someone answers and says "die, die, die, die ..." 8 times? :) |
| 19:30.45 | Inv_arp | yashax: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup this setup wprked for me for BV |
| 19:30.45 | modulus_ | i wonder if i'll ever see you again... |
| 19:30.56 | bkw_ | well it takes loud noise to cause it to do it |
| 19:30.59 | vaewyn | bkw_: I've had that problem |
| 19:31.11 | vaewyn | busycount=8 fixed it at least for now |
| 19:31.18 | bkw_ | vaewyn, I recall someone talking about this in here.. thats what made me think of it |
| 19:31.21 | bkw_ | well we had 8 |
| 19:31.30 | bkw_ | which could be on the edge of the threshold |
| 19:31.40 | bkw_ | but it was doing it so randomly and I couldn't catch it doing it |
| 19:31.50 | bkw_ | so I thought jacking up the busycount would help |
| 19:31.54 | bkw_ | because thats all I can think that causes it |
| 19:32.14 | Inv_arp | bkw_: quick ques can * compile on gcc 2.95? |
| 19:32.19 | vaewyn | fixed mine... going from default to 8 has kept me drop free for a co9uple months now |
| 19:32.19 | bkw_ | doubt it |
| 19:32.22 | bkw_ | their is a bug open on that |
| 19:32.25 | bkw_ | go check bugs.digium.com |
| 19:32.26 | bkw_ | DUH |
| 19:32.36 | yashax | What ports should be open on FW for SIP to work? |
| 19:32.37 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 19:32.37 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 19:32.46 | bkw_ | http://bugs.digium.com/bug_view_page.php?bug_id=0003639 |
| 19:33.22 | Inv_arp | well it complained about incomplete type for me ... i put an integer in their and it compiled |
| 19:33.51 | vaewyn | Hey bkw_ any ideas how to debug getting "Feb 22 14:19:08 NOTICE[1601]: chan_sip.c:7806 handle_request: Unable to create/find channel" on the first call from a device... but after that it is fine (ie I want to find if it is the phones fault or *s) |
| 19:35.39 | zipp | anyone know if I can setup IAX trunking with nufone? |
| 19:35.44 | zipp | trunk=yes breaksit |
| 19:35.47 | zipp | trunk=yes breaks it |
| 19:35.59 | vaewyn | I think you have to contact them first |
| 19:36.08 | vaewyn | IIRC |
| 19:36.55 | vaewyn | You can ask JerJer if he is still around |
| 19:37.31 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
| 19:37.32 | modulus_ | bleh |
| 19:37.55 | zipp | vaewyn, wouldn't it only help them? |
| 19:37.55 | xkev | since I switched to NI-2 (from DMS100, or maybe it was a cvsup that caused this), I can't do inband progress without indicating Ringing() first |
| 19:38.03 | modulus_ | alll of my liiiiife |
| 19:38.07 | modulus_ | where have you beeeeen? |
| 19:38.11 | tzanger | modulus_: I hate that song |
| 19:38.21 | modulus_ | tzanger, i love you |
| 19:38.34 | yashax | Inv_arp: Thank you.... I missed one line.... NOw, I have the BV registered on *... But when I call the DID, I get, the party busy and can not answer the call...any ideas? |
| 19:38.39 | tzanger | modulus_: ha |
| 19:38.54 | clive- | has anyone set up multiple iax clients that can call each other ? |
| 19:39.40 | KalD|Work | clive-, you mean directly or via *? |
| 19:39.46 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 19:40.11 | Inv_arp | yashax: setup your sip client to answer from-BV? |
| 19:40.23 | clive- | I mean , initially via *, then hopefully native bridge comes into effect |
| 19:40.46 | zipp | clive-, if you don't care about cdr's |
| 19:40.49 | yashax | Inv_arp: where is that? |
| 19:41.03 | clive- | zipp, no cdrs is cool,,, |
| 19:41.06 | KalD|Work | clive-, yeah it is easy =)... make sure that the iax.conf is correct and makesure you put them all in a context that allows them to dial each other |
| 19:41.18 | yashax | I am using Asterisk@home and the config files are slightly different, so I am having problems finding things.. |
| 19:41.25 | Inv_arp | yashax: in sip.conf and extensions.conf ... what sip client ya use? |
| 19:41.50 | KalD|Work | clive-, setup your clients and iax.conf so they all register (verify with iax2 show peers in the CLI) |
| 19:41.54 | clive- | thats bascially my question, ..does each one need a dial match in extensions.conf ? |
| 19:42.08 | KalD|Work | clive-, then simply put exten => 1000,1,Dial(IAX2/client1) etc for each one |
| 19:42.34 | clive- | if it gets big, that can become ugly |
| 19:42.38 | yashax | /msg context = from-sip-external ; Send unknown SIP callers to this context |
| 19:42.43 | yashax | oops... |
| 19:42.49 | Darwin35 | when are they going to make a good iax2 based phone ? |
| 19:42.50 | KalD|Work | clive-, depends on how you have them registered - if you have them as [1000] to [1999] in iax.conf then you can do Dial(iax2/${EXTEN}) |
| 19:42.55 | yashax | /msg Inv_arp context = from-sip-external ; Send unknown SIP callers to this context |
| 19:43.24 | yashax | should this line have context=from-broadvoice ? |
| 19:43.28 | clive- | ahh...great one...thats excellent, thanks Kald |
| 19:43.34 | KalD|Work | clive-, once you get this working for testing you might want to migrate to a mysql based registration |
| 19:43.35 | zipp | Darwin35, the only thing is the iaxy |
| 19:43.55 | Inv_arp | yashax: doesnt matter on name just need a matching one in extension.conf |
| 19:44.00 | KalD|Work | clive-, no prob. =) |
| 19:44.05 | Darwin35 | there was a phone at 1 point being developed down under but it seems to have died |
| 19:44.09 | clive- | Kald, yup, it will have to go the database route eventually |
| 19:44.13 | zipp | Darwin35, farfon? |
| 19:44.17 | Inv_arp | yashax: lemme give u an ex... one moment |
| 19:44.20 | Darwin35 | I think so |
| 19:45.00 | zipp | Darwin35, one could be made w/ an sbc and the iax lib quite easily |
| 19:45.08 | zipp | wouldn't be 80 bucks though |
| 19:45.20 | zipp | s/an/a/ |
| 19:45.26 | Darwin35 | sbc ? |
| 19:45.28 | yashax | Inv_arp: k.. plz msg me... |
| 19:45.28 | *** join/#asterisk akrall (~akrall@201.128.92.118) |
| 19:45.45 | akrall | Guys.. anybody has an URL on how to program a find-me dialplan? |
| 19:45.57 | *** join/#asterisk machinehd (~machinehd@storm.bcgroup.net) |
| 19:46.09 | Darwin35 | find me fallowme is in the wiki |
| 19:46.17 | Darwin35 | go read |
| 19:46.26 | akrall | followme.. good.. thx |
| 19:47.15 | zipp | Darwin35, single board comp -> http://www.embeddedarm.com/epc/ts7200-spec-h.html |
| 19:47.24 | zipp | that would be my choice of dev board |
| 19:47.47 | zipp | had to lookup url |
| 19:48.17 | vaewyn | zipp: seen the $$$ on that? |
| 19:48.28 | *** part/#asterisk akrall (~akrall@201.128.92.118) |
| 19:48.35 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 19:48.36 | vaewyn | ain't bad is it ? :} |
| 19:48.46 | zipp | vaewyn, that is one of the reasons for picking it |
| 19:48.56 | zipp | however, how can you compete w/ a budgetone |
| 19:49.04 | zipp | for $80 |
| 19:49.04 | zipp | ? |
| 19:49.32 | zipp | you still have to have the handset, speakerphone, dialpad, screen... |
| 19:49.35 | vaewyn | quality... make a firmware that is 99.999% stable instead of 99.999% broken |
| 19:49.37 | machinehd | Anyone having problems calling ext to ext with the 5.22 GS firmware? I can call out, get calls in but can't dial EXT to EXT. |
| 19:49.41 | vaewyn | plus get volume |
| 19:49.50 | zipp | you could however make a _quality_ phone |
| 19:49.56 | vaewyn | case and point ^^^ :} |
| 19:50.02 | zipp | 320x240 color screen |
| 19:50.10 | zipp | you could do a lot w/ a complete linux distro... |
| 19:50.43 | zipp | priced around 350-450, I just would like to see a 100 cheap iax phone also |
| 19:50.45 | *** part/#asterisk _PiGreco_ (~a@adsl-120-46.38-151.net24.it) |
| 19:50.52 | vaewyn | give it a decent res screen... color optional... and a development kit for making apps... and you are set... can be anything from a phone to a timeclock to a...etc..etc.. |
| 19:50.57 | zipp | you need to know cpld for that |
| 19:51.08 | zipp | for making a cheap one... |
| 19:51.13 | vaewyn | *nods* |
| 19:51.45 | zipp | I called a hardware development company, upwards of 200K to develop a cheap IAX phone |
| 19:51.51 | zipp | I just don't have that lying around :) |
| 19:51.56 | Darwin35 | hmmm |
| 19:52.07 | `Sauron | zipp: I bet it could be done cheaper. |
| 19:52.12 | Darwin35 | I will look into it . after I finish my wifi unit |
| 19:52.13 | zipp | `Sauron, me too |
| 19:52.25 | zipp | if you found someone with the knowledge and interest |
| 19:52.26 | Darwin35 | ibut hmm a iax phone would be better off |
| 19:52.26 | vaewyn | yeah... 200k will get you a couple protos... and all the designs for full on production |
| 19:52.33 | `Sauron | zipp: gumstix connex, the etherstix and the audiostix - and you've got a phone. |
| 19:52.36 | *** join/#asterisk Rick_Hunter (~rhunter@05-046.008.popsite.net) |
| 19:52.47 | zipp | `Sauron, looking... |
| 19:53.06 | vaewyn | `Sauron: yeah... but do you have enough CPU for the good codecs? |
| 19:53.24 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
| 19:53.27 | `Sauron | vaewyn: I dunno. How many MIPS you need? |
| 19:53.29 | vaewyn | No one has been able to tell me if that CPU can handle it |
| 19:53.37 | `Sauron | I could always try after I get one. |
| 19:53.44 | clive- | zipp, I hear the pa168 phones can do iax, havent tested it myslef |
| 19:53.46 | vaewyn | not sure... depends on FP/INT etc... |
| 19:53.51 | `Sauron | Although I'm looking at the 200MHz one, not the 400MHz one |
| 19:53.59 | `Sauron | no FP unit in embedded computers |
| 19:54.01 | *** join/#asterisk asterisknewibe (asteriskne@adsl-068-213-121-038.sip.chs.bellsouth.net) |
| 19:54.07 | `Sauron | so you'd need codecs that are non-fp |
| 19:54.26 | zipp | clive-, can you easily purchase them, and is it iax2, not iax? |
| 19:54.51 | clive- | zipp, I have the phones, just havent had the time to try the iax2 firmware |
| 19:55.09 | asterisknewibe | One question for you pro's out there...? using rh9 how do you make the wcte11xp autoload at boot ? I can modprobe and load it..but can't get it to auto load at boot..any help would be great... |
| 19:55.44 | `Sauron | The 400MHz PXA255 performs roughly equivalent to a 233MHz K6, or about 4-6 times faster than a P90 |
| 19:55.52 | zipp | in my opinion, grandstreams greatest move would be to open up the firmware, and put together a free sdk |
| 19:55.57 | `Sauron | vaewyn: Think that's enough to do enc/decoding? |
| 19:56.09 | zipp | `Sauron, yes |
| 19:56.13 | vaewyn | `Sauron: at least for a single line... not sure on multiple |
| 19:56.21 | vaewyn | but easily 1 liner |
| 19:56.38 | `Sauron | Well, you'd only encode/decode one line at a time |
| 19:56.39 | vaewyn | zipp: agreed... |
| 19:56.42 | `Sauron | so it wouldn't matter |
| 19:56.47 | zipp | grandstream's firmware sucks, and opening it would make SIP better, and people would create IAX2 firmware |
| 19:56.55 | `Sauron | any conferencing would happen in */somewhere else |
| 19:57.19 | `Sauron | when you put a line on hold, it doesn't continue to decode that line |
| 19:57.19 | Darwin35 | call GS and offer to develop for them |
| 19:57.27 | `Sauron | it goes to decode the line you pick up |
| 19:57.29 | `Sauron | etc |
| 19:57.36 | vaewyn | Darwin35: been there... done that... no reply yet |
| 19:57.40 | `Sauron | So you only ever have to do the encoding/decoding for a single line |
| 19:58.05 | zipp | clive-, which pa168 phone do you have? |
| 19:58.06 | Darwin35 | Vae youhave to call and ask for Brian or Richard |
| 19:58.08 | `Sauron | Shrug, maybe I'll pick up the audiostix extra when I place my order, and play around with it. |
| 19:58.16 | Darwin35 | to get anywhere |
| 19:58.20 | `Sauron | And then just use the 802.11 cf card for talking on the wire |
| 19:59.00 | zipp | Darwin35, would be a good idea, but them opening it would be better |
| 19:59.16 | `Sauron | Yum Yum. |
| 19:59.28 | Darwin35 | openign src would be nice but these comanies are all about making money |
| 19:59.29 | zipp | `Sauron, gumstix.org? |
| 19:59.35 | `Sauron | .com |
| 19:59.36 | Darwin35 | thats life |
| 19:59.40 | `Sauron | .org is their Wiki |
| 19:59.40 | zipp | Darwin35, they would sell more hardware |
| 19:59.45 | Darwin35 | I know |
| 19:59.54 | Darwin35 | you know bt they dont grasp |
| 20:00.00 | clive- | zipp, its the one with the tilt up screen and the huge buttons |
| 20:00.16 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 20:00.43 | vaewyn | :} |
| 20:00.46 | vaewyn | tempted even |
| 20:00.48 | zipp | clive-, from china :) |
| 20:01.07 | `Sauron | Hum. |
| 20:01.07 | vaewyn | need to figure out how to do the menuing stuff on the polycoms though |
| 20:01.16 | Darwin35 | I wish I knew what os they used and would reverse enginier thier .bin pkgs |
| 20:01.19 | `Sauron | is iax2 just the session protocol, similarily to sip |
| 20:01.21 | zipp | `Sauron, gumstix+audio/eth is > $200 |
| 20:01.23 | Darwin35 | and get src |
| 20:01.24 | vaewyn | anyone got examples of the polycom menu stuff? |
| 20:01.33 | `Sauron | and the audio transport is done with whatever codec, using RTP |
| 20:01.33 | vaewyn | `Sauron: yep |
| 20:01.59 | `Sauron | zipp: If you can proof-of-concept it, you could build units for-sale at less cost |
| 20:02.03 | clive- | zipp, they are all from china |
| 20:02.05 | zipp | `Sauron, I don't think so, iax2 only uses 1 port |
| 20:02.27 | vaewyn | zipp: is still rtp though... just better done :} (IIRC) |
| 20:03.03 | Darwin35 | ok who has a extensions file with almost every option mapped |
| 20:03.21 | shido6 | HAH |
| 20:03.22 | shido6 | funny |
| 20:03.35 | vaewyn | the sampel is about as big a chunk of "everything" as you get |
| 20:04.19 | Darwin35 | i need overhead paging via the sound/dsp |
| 20:04.28 | mutilator | not really voip related at all, but anyone know anythin about portmaster 4's, i wanna get some stats on disconnects for modmes, show modems doesn't give much help |
| 20:04.34 | Darwin35 | I need dial by name |
| 20:04.48 | *** join/#asterisk jdg (~jdg@CA03F9F4.adsl.mana.pf) |
| 20:04.54 | zipp | `Sauron, you also need DIO for the buttons, which gumstix doesn't have |
| 20:04.54 | hardwire | anybody here interfacing to an NEC pbx? |
| 20:04.55 | Darwin35 | I need IDL |
| 20:04.58 | hardwire | via t1 |
| 20:05.00 | zipp | Darwin35, show applications |
| 20:05.15 | shido6 | Damascene "/usr/src/asterisk/configs/*" |
| 20:05.23 | Darwin35 | these need to be mapped in the asterisk extensions.conf |
| 20:05.59 | Darwin35 | I just am burned out wiith how much I have already put in |
| 20:06.10 | Darwin35 | looking for soome cut and paste now |
| 20:07.05 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 20:07.14 | shido6 | Darwin35, rather |
| 20:07.25 | shido6 | Darwin35 look in "/usr/src/asterisk/configs/*" |
| 20:08.10 | thieumS | hi, I'd need to know which processor config I need to complete 4E1 (Sip to Zap) transcoding |
| 20:08.35 | thieumS | has anyone some experience with that ? |
| 20:09.16 | thieumS | the real question is, do I need multi-proc ? |
| 20:09.22 | thieumS | or a single powerfull CPU |
| 20:09.47 | vaewyn | what codecs involved thieumS ? |
| 20:09.52 | thieumS | g729 |
| 20:10.07 | vaewyn | duals... pretty high level ones even |
| 20:10.39 | vaewyn | or have the 4E machine just forward to a couple transcode machines |
| 20:10.40 | thieumS | amd or intel |
| 20:10.41 | Trionnis | quad HT Xeon64's should be enough |
| 20:10.45 | Trionnis | ;) |
| 20:10.51 | vaewyn | ohh geeze |
| 20:10.53 | Trionnis | lol |
| 20:10.53 | thieumS | huhu :p |
| 20:10.56 | Trionnis | :) |
| 20:11.00 | Trionnis | well, it would be |
| 20:11.02 | Trionnis | :P |
| 20:11.10 | thieumS | i would like to eat, as well |
| 20:11.14 | Trionnis | laf |
| 20:11.19 | vaewyn | dual xeon or dual opteron should be fine |
| 20:11.20 | Trionnis | well yeah, that's kinda important too |
| 20:11.22 | *** join/#asterisk rvhi (~root@66.175.65.89) |
| 20:11.41 | vaewyn | or break it up into multiple transcode machines... depends if rack is cheaper or CPUS |
| 20:13.48 | thieumS | ok thanx for your help |
| 20:14.13 | *** join/#asterisk zapa (~zapa@201.128.63.117) |
| 20:14.28 | tzanger | a 'restart when convenient' is sufficient to get new zapata.conf echo/gain settings to take effect, right? |
| 20:14.50 | _Brian | anyone here sucessful in getting Icecast to work with Asterisk? |
| 20:15.14 | zapa | hi all where i can find a Rj48 cable diagram to connect to an e1 telrad to e100p |
| 20:15.51 | *** join/#asterisk pryk (~tmalkut@fire2.orasoft.net.pl) |
| 20:16.07 | denon | zapa: I believe all you need is a standard e1 crossover cable |
| 20:16.20 | machinehd | I have an install of the latest AMP. When dialing EXT to EXT to phones ring, but once picked up there's no sound either way. No NAT. Watching the debug log doesn't seem to show errors. Any ideas? |
| 20:16.31 | denon | afaik, its the same as a T1 crossover cable |
| 20:17.10 | denon | zapa: a quick google found this: http://www.nmscommunications.com/NMS/nms_technotes.nsf/0/91d49c8785b2aab0852566fa0050740a?OpenDocument |
| 20:17.42 | ayzee | I keep getting the request failure: 484 Address Incomplete when I try to make a call locally with sip. what's that about? |
| 20:17.54 | zapa | denon : Thanks !! :) |
| 20:18.27 | jaiger | tzanger, I found I needed to restart * to gain settings to take |
| 20:18.37 | zapa | denon : how do you find it! :) i spent alot of time looking for them |
| 20:18.47 | tzanger | jaiger: I just said "restart when convenient" :-) |
| 20:18.57 | tzanger | jaiger: or did you mean stop and start |
| 20:19.03 | denon | zapa: it was like the first restart of "t1 crossover pinout" |
| 20:19.07 | denon | or something like that |
| 20:19.37 | Trionnis | _Brian: yes, but it's a PITA |
| 20:20.09 | _Brian | Trionnis: i got everything working, but for some reason when I try to make the connection to the stream, winamp just starts and stops again...nothing is heard.... |
| 20:20.36 | ayzee | I can't find any specific information on 484 "Address Incomplete" |
| 20:20.40 | Trionnis | does it show the source connected in the admin page for icecast? |
| 20:21.02 | _Brian | Trionnis: yes |
| 20:21.04 | Trionnis | and you're using ices2 with ogg streaming, right? |
| 20:21.09 | jaiger | tzanger, I needed to stop and start, "reload" didn't work |
| 20:21.10 | _Brian | Trionnis: yes |
| 20:21.21 | Trionnis | pm me the address please |
| 20:21.22 | tzanger | jaiger: I didn't say reload, I said restart -- both commands exist and do different things :-) |
| 20:21.28 | Trionnis | the stream address, that is |
| 20:21.54 | jaiger | tzanger, ok, I've never tried restart |
| 20:21.59 | tzanger | jaiger: ok |
| 20:22.54 | shido6 | boink |
| 20:23.57 | `Sauron | Anyone know how long it usually takes for broadvoice to process a number port request? |
| 20:24.49 | zipp | anyone know where I can get the iaxcomm sources? |
| 20:25.05 | *** join/#asterisk e3eli3h (~e3eli3h@83.168.2.150) |
| 20:25.10 | stevekstevek | zipp: www.sf.net/project/iaxclient/ then look for CVS.. |
| 20:25.11 | shido6 | are you gonna fix it, zipp!?!? |
| 20:25.59 | *** join/#asterisk trym (~trym@linux.debian.us) |
| 20:27.45 | zipp | stevekstevek, I got it, thx |
| 20:29.02 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 20:30.20 | *** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net) |
| 20:33.54 | stevekstevek | vaewynAFK: zipp: `Sauron: This is 20 mins late, but IAX2 does _not_ use RTP. |
| 20:34.09 | `Sauron | steve: :) |
| 20:34.13 | `Sauron | Then what does it use? |
| 20:34.14 | terrapen | is iaxclient no good? |
| 20:34.22 | stevekstevek | It uses IAX2 :) |
| 20:34.32 | `Sauron | Hum |
| 20:34.37 | stevekstevek | terrapen: huh? |
| 20:34.40 | bjohnson | that is why it is nat friendly |
| 20:34.41 | `Sauron | find out if there's an integer version of that codec ;) |
| 20:34.50 | stevekstevek | what codec? |
| 20:34.53 | bjohnson | one port |
| 20:35.03 | `Sauron | the iax2 stuff |
| 20:35.07 | bjohnson | protocol |
| 20:35.09 | clive- | stevek, hi , did you figure out that iax2 transfer wierdness with the 81,82ms timestamping |
| 20:35.10 | stevekstevek | IAX2 isn't a codec. |
| 20:35.11 | bjohnson | not codec |
| 20:35.15 | `Sauron | Sigh. |
| 20:35.33 | `Sauron | iax has to transfer audio somehow |
| 20:35.35 | `Sauron | duh |
| 20:35.39 | bjohnson | codec = ulaw, alaw, gsm, etc |
| 20:35.42 | `Sauron | and it's not streaming pcm audio |
| 20:35.45 | stevekstevek | clive-: yeah, delete the lines in, I think, fake_timestamp that do the +1 business, that are marked "SJD thinks this shouldn't be here". |
| 20:35.55 | `Sauron | bjohnson: That's what I was asking about 20 minutes ago |
| 20:36.02 | bjohnson | what? |
| 20:36.05 | bjohnson | I just got back |
| 20:36.11 | stevekstevek | `Sauron: it uses whatever codec you want it to use (well, any of the 10 or so that are defined). |
| 20:36.11 | `Sauron | So it uses the same codecs as everything else, just encapsulates it into something else |
| 20:36.13 | clive- | stevek and then it works great ? |
| 20:36.17 | bjohnson | yes |
| 20:36.21 | bjohnson | `Sauron: yes |
| 20:36.34 | stevekstevek | `Sauron: right. But it does not use RTP. RTP is a different protocol. |
| 20:37.04 | dca[laptop] | speaking of RTP |
| 20:37.04 | `Sauron | iax uses udp across known ports |
| 20:37.07 | `Sauron | dum did um |
| 20:37.12 | stevekstevek | clive-: There still can be a discontinuity when you start/end the native bridge (or a regular bridge, actually), but that fixes the issue during the native bridge. |
| 20:37.27 | dca[laptop] | anyone know if it is possible to prevent asterisk from releasing the RTP stream? |
| 20:39.32 | tzanger | I just disable bridge optimization |
| 20:39.53 | bjohnson | `Sauron: something like that .. one port I think (I'm not an expert on that) |
| 20:39.58 | stevekstevek | tzanger: but that doesn't fix the possible discontinuity when the non-native bridge starts/stops. |
| 20:40.20 | tzanger | hmm |
| 20:40.30 | tzanger | I don't know enough about it |
| 20:40.38 | tzanger | I don't understand what bridge otpimization does then |
| 20:40.40 | tzanger | I tried reading thorugh it |
| 20:40.45 | tzanger | I have a lot of troubel concentrating today |
| 20:40.54 | tzanger | I'm not 100% at all... I think near 50% or 40% even |
| 20:40.56 | stevekstevek | tzanger: you'll still get a discontinuity, if, eg, your dialplan plays some audio, then does Dial, then plays more audio then hangs up. |
| 20:41.21 | tzanger | hmm |
| 20:41.24 | stevekstevek | there will be some "jumps" in the timestamps between the locally generated audio and the bridged audio. |
| 20:41.35 | tzanger | stevekstevek: ahh |
| 20:42.08 | tzanger | but will the new jitterbuffer not see the discontinuity and eventually lop it off since the timestamps jump and then resume (a new, but consistent) pattern? |
| 20:42.37 | stevekstevek | 'cause for the locally generated audio, chan_iax2 generated timestamps for each frame, but for the bridged audio, it is "passing through" the timestamps; chan_iax2 tries to "adjust" those timestamps, but the adjustment doesn't always match what's being sent exactly.. |
| 20:42.48 | stevekstevek | tzanger: yes, it will. But it may take 20 seconds to do that.. |
| 20:43.03 | tzanger | 20 seconds to stabilize on the new pattern? |
| 20:43.11 | clive- | stevek is the jitter buffer / PLC stuff patch in its final form? |
| 20:43.13 | tzanger | what about that iax2 native transfer graphI hsouwed you |
| 20:44.05 | stevekstevek | when a transfer happens, there's a discontinuity, but chan_iax2 and libiax2 both reset everything when the transfer happens. |
| 20:44.12 | *** join/#asterisk pryk (~tmalkut@fire2.orasoft.net.pl) |
| 20:44.18 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 20:44.40 | stevekstevek | but, if the call is just bridged, the receiver doesn't get any notification to reset or anything.. |
| 20:44.41 | zipp | stevekstevek, any idea how to make a call with testcall? |
| 20:44.45 | tzanger | stevekstevek: hmm |
| 20:44.49 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 20:44.55 | tzanger | would htat not be a good idea? (notifying on bridge) ? |
| 20:44.57 | stevekstevek | zipp: type ./testcall guest@misery.digium.com |
| 20:45.08 | stevekstevek | or guest@iax2.fwdnet.net/613 |
| 20:45.14 | buddah | anyone know what is causing this error message? |
| 20:45.14 | buddah | Feb 22 12:45:21 WARNING[17500]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) |
| 20:45.56 | stevekstevek | buddah: someone talked about this earlier.. I don't remember the answer; I think they said it might be VAD data that codec_g729 doesn't understand |
| 20:46.16 | buddah | stevekstevek: ok, any clue how to resolve it? |
| 20:46.57 | stevekstevek | tzanger: it isn't a bad idea, but the better idea is to just eliminate the discontinuity. It's backwards-compatible (no protocol extension needed), and probably not a lot harder. |
| 20:47.06 | stevekstevek | buddah: don't use g729 :) |
| 20:47.35 | buddah | stevekstevek: well we purchased liscences for it, and the message just popped up all of a sudden as of yesterday |
| 20:47.37 | zipp | stevekstevek, thx |
| 20:47.44 | |Vulture| | whats wrong with 729? |
| 20:47.48 | tzanger | stevekstevek: how do you eliminiate the discontinuity, is this the timestamp skewing you were talking about last week I think it was? where you gradually ramp the existing timestamp to the newone? |
| 20:47.54 | stevekstevek | it's parented :) |
| 20:48.30 | stevekstevek | tzanger: so, in chan_iax2, frames that it's about to send out either come with timestamps, or without. |
| 20:48.43 | tzanger | stevekstevek: yes I saw that |
| 20:49.04 | stevekstevek | when they come without timestamps (let's call this implicit timestamps), it generates them (it's called "prediction" in the code). |
| 20:49.22 | tzanger | right |
| 20:50.03 | BrianR___ | hey folks |
| 20:50.04 | stevekstevek | when the come with timestamps (let's call this explicit), the timestamps are theoretically based on the server's timeframe, so it just subtracts the time of the beginning of the call (to get the right timeframe) and sends that. |
| 20:50.26 | stevekstevek | but, this can sometimes still lead to a gap for whatever reason.. |
| 20:50.55 | stevekstevek | so, here's an algorithm to solve the problem: |
| 20:50.58 | __Sparks_ | Other then Sipgate and FWD (Who break CLID!), are there any other SIP providers that offer free calls to UK 0800 numbers? |
| 20:51.03 | BrianR___ | I'm having a bit of trouble with disconnect supervision on a fx0 (fxs signalled) device. It's plugged into an analog port on a PBX which provides dialtone but not open switch interval or polarity reversal... |
| 20:51.12 | stevekstevek | 1) add a new variable to iax2_pvt called "explicit_ts_offset". |
| 20:51.32 | BrianR___ | I turned on callprogress=yes, but it doesn't seem to be disconnecting that line when it plays a dialtone. |
| 20:51.34 | stevekstevek | 2) It gets initialized to zero or something. |
| 20:52.26 | stevekstevek | 3) If we get a frame with an implicit timestamp (and explicit_ts_offset) is zero, we do the same thing as now. |
| 20:53.17 | dca[laptop] | does anyone know how to prevent the RTP stream from being released? |
| 20:53.30 | stevekstevek | 4) If we get a frame with an explicit ts, we figure out what the next implicit ts would be, and calculate offset such that the explicit_ts - explicit_ts_offset = the next implicit ts, and send that out. |
| 20:53.42 | stevekstevek | 5) if we get a frame with an explicit ts, and explicit_ts_offset is set, we subtract that, and send it. |
| 20:53.59 | tzanger | stevekstevek: hmm |
| 20:54.14 | stevekstevek | 6) if we get a frame with an implicit ts, and explicit_ts_offset is set, we clear explicit_ts_offset. |
| 20:54.14 | stevekstevek | (I kinda made that up as I typed, but it's what I had in mind). |
| 20:54.44 | stevekstevek | tzanger: want to put that into a mantis bug, and write a patch for it :) |
| 20:54.51 | dca[laptop] | is it even possible to prevent the RTP stream from being released ... |
| 20:55.16 | stevekstevek | dca[laptop]: I don't do SIP, but I think you want "canreinvite=no" or something. |
| 20:55.57 | dca[laptop] | stevekstevek: that will prevent the call from release (i.e. the SIP signalling) but not the RTP stream |
| 20:56.26 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 20:56.31 | stevekstevek | dca[laptop]: OK, then I guess I don't know the answer. Sorry :) (like I said, I don't use sip...). |
| 20:56.36 | dca[laptop] | np |
| 20:56.47 | dca[laptop] | fyi, iax uses RTP as well ...i think |
| 20:57.07 | stevekstevek | no, it doesn't. |
| 20:57.07 | yashax | In Asterisk@HOME, which config file and where does it store the DialPlan based on the Web GUI? |
| 20:57.11 | stevekstevek | I _do_ know iax. |
| 20:57.22 | dca[laptop] | ah, k |
| 20:57.41 | greg_work | are there any downsides to having nat=yes when there is no NAT? |
| 20:58.25 | greg_work | yashax: *@home uses AMP doesn't it? /etc/asterisk/extensions_additional.conf |
| 20:59.19 | tzanger | stevekstevek: hmm ok this looks rather easy actually |
| 20:59.21 | yashax | got it, thanks... |
| 20:59.54 | yashax | hmm... what I am looking for to find the DialPlan??? |
| 21:00.19 | *** join/#asterisk darby_t (~tom@doa150.neoplus.adsl.tpnet.pl) |
| 21:00.43 | stevekstevek | tzanger: yeah, not too hard. |
| 21:00.55 | stevekstevek | That totally takes care of the transition from implicit -> explicit. |
| 21:00.58 | zipp | stevekstevek, can I build iaxcli w/o tk/tcl? |
| 21:01.08 | stevekstevek | zipp: sure. |
| 21:01.27 | zipp | gdk errors on make iaxcli |
| 21:01.49 | stevekstevek | tzanger: I'm not 100% sure that the transition the other way will be taken care of automatically, but it might require updating one variable (last_voice or something). to do that as well. |
| 21:02.18 | stevekstevek | zipp: if you don't have gtk-devel and whatnot, turn off the HOTKEY stuff. (actually, tkphone doesn't use it either). |
| 21:02.33 | stevekstevek | zipp: I added that for a "push to talk" functionality I use in another client.. |
| 21:02.41 | zipp | thx |
| 21:02.59 | zipp | USE_HOTKEY=0 |
| 21:03.11 | zipp | works, thx |
| 21:03.31 | bjohnson | greg_work: I don't think so |
| 21:04.32 | greg_work | bjohnson: i couldn't think of anything either. it says that "basically it tells * to ignore the address in the SIP request, and use the addres the packet came from instead" |
| 21:04.44 | greg_work | on a LAN, the address is going to be the device that sent it |
| 21:04.59 | bjohnson | yes I think so |
| 21:06.24 | BrianR___ | Any thoughts on my dialtone disconnect supervision problem? |
| 21:06.32 | tzanger | BrianR___: this is for an ATA? |
| 21:06.48 | BrianR___ | tzanger: It's for a VMI, which thankfully provides a dialtone instead of dead air on hangup... |
| 21:06.56 | *** join/#asterisk syslod (~yurplsl@65.114.15.26) |
| 21:06.57 | tzanger | BrianR___: hmm |
| 21:07.03 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 21:07.07 | BrianR___ | tzanger: I'm wondering if the norstar's dialtone is not the standard frequency? |
| 21:07.32 | tzanger | BrianR___: sounds pretty normal to me :-) |
| 21:07.51 | BrianR___ | Turning on callprogress should result in disconnect on dialtone, right? |
| 21:07.57 | tzanger | not sure |
| 21:07.59 | tzanger | I avoid callprogress |
| 21:08.13 | `Sauron | Hum |
| 21:08.24 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
| 21:08.25 | BrianR___ | tzanger: Yes. It's a nasty hack. But there's no other solution if your line doesn't have real diconnect supervision... |
| 21:08.31 | rvhi | hi, how big is the difference between 1.0.5 and cvs-head? |
| 21:09.16 | zipp | rvhi, months |
| 21:09.38 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 21:09.45 | rvhi | if i want realtime database for vm/sip/exten, can i use 1.0.5? |
| 21:09.46 | iceyp | anyone know where and how i can install Asterisk/AGI.pm |
| 21:09.54 | iceyp | I cant find it on CPAN |
| 21:10.00 | zipp | rvhi, I have problems w/ nufone using <= 1.0.5, not with cvs head |
| 21:10.25 | rvhi | read something about a caller id bug in 1.0.5 |
| 21:10.35 | rvhi | is the fix backported? |
| 21:11.05 | rvhi | i really don't want to mess up with cvs head |
| 21:12.04 | greg_work | rvhi: where are you looking to use *? |
| 21:12.07 | kpfleming | rvhi: yes, the current CVS stable has corrected caller ID, and will be released as 1.0.6 soon |
| 21:12.17 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 21:12.25 | FuRR_ | anyone have any exp. with asterisk behind a nat |
| 21:12.31 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 21:12.38 | FuRR_ | i keep getting un authorized messages when the xten and pin are correct |
| 21:12.43 | greg_work | FuRR_: read the wiki |
| 21:12.46 | buddah | anyone familiar with polycom 500s know how to get the caller ID working? |
| 21:12.58 | FuRR_ | greg_work: and which one of the over 1k pages should i start on |
| 21:13.00 | buddah | still showing the default thats set on the t1 |
| 21:13.04 | kpfleming | when does it not work? it's always worked for me |
| 21:13.29 | buddah | the number is showing up as 5622832400 |
| 21:13.38 | buddah | and it should be 3233450128 |
| 21:13.43 | rvhi | i am using * a pstn gateway |
| 21:13.46 | kpfleming | if you are setting callier in zapata.conf, then that's what you will get on your phones, don't set it there |
| 21:13.49 | rvhi | with other sip proxy |
| 21:13.56 | buddah | ahh |
| 21:14.28 | buddah | callier? |
| 21:14.33 | buddah | caller you mean? |
| 21:15.08 | kpfleming | callerid, yes |
| 21:15.10 | buddah | k |
| 21:15.12 | *** join/#asterisk itnomad (~itnomad@net-216-37-66-26.in-addr.worldspice.net) |
| 21:15.23 | greg_work | FuRR_: http://voip-info.org/wiki-Asterisk+SIP+NAT |
| 21:15.24 | kpfleming | that does not control outbound caller id, if you though it did |
| 21:15.44 | *** join/#asterisk emitrax (~emitrax@host55-74.pool80183.interbusiness.it) |
| 21:16.07 | Connor- | anyone have problems with asterisk becoming unregister with another asterisk box? and not re-registering? (only way to get it to re-register is to issue a sip reload command) ?? |
| 21:16.08 | *** join/#asterisk kuonSama (kuon@alragore.goyman.com) |
| 21:16.13 | kuonSama | hi everybody |
| 21:16.52 | kuonSama | I currently have a cisco CME installed, the CME use the router IOS isdn interface to get connected to public phone network. |
| 21:17.14 | kuonSama | I want to use asterisk in place of the CME to add more function (like voice mail, auto respond...) |
| 21:17.35 | kuonSama | is that a good choice? Or should I use Cisco CM? |
| 21:17.40 | kuonSama | I never user asterisk |
| 21:17.42 | shido6 | kick CM to the curb |
| 21:17.57 | shido6 | check your message kuonSama |
| 21:17.59 | kuonSama | I don't like windows |
| 21:18.24 | loud | nor does asterisk, you've got a friend |
| 21:19.43 | buddah | kpfleming: so comment out the usecallerid=yes in zapata.conf? |
| 21:19.58 | kpfleming | no, not that, that's ok |
| 21:20.01 | buddah | ok |
| 21:20.11 | kpfleming | if you are specifying "callerid=" in zapata.conf then you don't want to do that |
| 21:20.16 | buddah | the stuff further down about the channel IDs |
| 21:20.16 | buddah | ahh |
| 21:20.17 | buddah | ok |
| 21:20.20 | buddah | yeah thats all commented out |
| 21:20.40 | kpfleming | are you sre you are receiving CLID from your telco? |
| 21:20.55 | BrianR___ | Is anyone else here familiar with callprogress? |
| 21:22.09 | syslod | callprogress? |
| 21:22.21 | |Vulture| | is there anyway to get a polycom IP500 to read its CID from sip.conf instead of it's personal config file? |
| 21:22.21 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
| 21:22.25 | BrianR___ | syslod: The fake call progress detection for fxs_ls signalled lines. |
| 21:22.52 | syslod | I only have ks here. |
| 21:22.54 | Trionnis | anyone know of a windows program to create the .gsm files for an IVR menu? |
| 21:22.57 | syslod | is that CPC? |
| 21:22.58 | BrianR___ | syslod: From what I can tell, it's also supposed to detect a dialtone and provide fake disconnect supervision on ls lines not equipped with real disconnect supervision.. |
| 21:23.00 | bjohnson | FuRR_: start with the ones that say NAT |
| 21:23.00 | |Vulture| | like Cisco 7960/40s read from sip.conf |
| 21:23.19 | eKo1 | Digium's having some voice and data outtages <--- Ironic, don't you think. |
| 21:23.23 | BrianR___ | syslod: callprogress=yes in zapata.conf |
| 21:23.28 | *** join/#asterisk thoor (~jhlk@cus04-118.cbnstl.net) |
| 21:23.52 | syslod | ?? I think LS is CPC |
| 21:23.54 | *** join/#asterisk zapa (~zapa@dsl-200-95-86-170.prod-infinitum.com.mx) |
| 21:23.56 | thoor | I need some help with my Iaxy unit |
| 21:24.05 | BrianR___ | syslod: What does CPC stand for? |
| 21:24.17 | thoor | it is registered with the asterisk box, but it doesn't recieve a dial tone |
| 21:24.18 | buddah | kpfleming: looks like its an issue with our cisco router |
| 21:24.25 | kpfleming | ok |
| 21:24.25 | eKo1 | Does anybody know of a web service or something to that effect that I can use to determine the location of a number? |
| 21:24.26 | bjohnson | Trionnis: sorry no .. did you check the wiki? you could also use the * record() command |
| 21:24.29 | zapa | denon thanks agains |
| 21:24.33 | Trionnis | yeah |
| 21:24.46 | Trionnis | but that's a pain, since I'm going to end up making a whole bunch |
| 21:24.57 | Trionnis | I looked there, didn't see much of anything really |
| 21:25.02 | Trionnis | :-/ |
| 21:25.07 | syslod | calling party control |
| 21:25.11 | thoor | i dont understand how i would be able to see the device with the show IAX2 peers yet it wont get the dial tone |
| 21:25.15 | thoor | anyone have any ideas? |
| 21:25.17 | bjohnson | thoor: I'm not familiar with IAXy .. but aren't they fxs = create (not receive) a dialtone |
| 21:25.56 | zapa | hi it´s posible to conect a etsi card pri 30 to TE110P |
| 21:25.58 | thoor | they are fxs...but i try dialing into it with a dial and it wont ring |
| 21:26.21 | bjohnson | Trionnis: maybe search the mailing list archives? Likely it's been discussed somewhere |
| 21:26.43 | bjohnson | thoor: do you get a dialtone when you plug a phone into it? |
| 21:26.58 | bjohnson | thoor: maybe try dialing out first |
| 21:27.15 | BrianR___ | syslod: Oh yes.. The device providing the fxs port doesn't have CPC. |
| 21:27.22 | BrianR___ | syslod: It doesn't even provide an open switch interval. |
| 21:27.28 | BrianR___ | syslod: It just plays a dialtone when the caller hangs up. |
| 21:27.36 | thoor | nope...gives a busy/no line connected signal |
| 21:28.08 | Trionnis | bjohnson: looking there now |
| 21:28.13 | Trionnis | thanks for the tip :) |
| 21:28.16 | tzanger | BrianR___: do you know if the NEC Electra systems have a similar kind of 3rd party VM interface (some kind of electra elite 48 ATA?) |
| 21:28.21 | syslod | THat would be tough to do. We have some expensive eq around here and it can't get around CPC on LS to may callprogress work. |
| 21:28.52 | BrianR___ | tzanger: No idea... |
| 21:29.12 | bjohnson | thoor: do you have it defined as a friend or a peer? |
| 21:29.14 | BrianR___ | tzanger: I will be doing integration in a few of our offices with different phone systems though. I think we have an avaya partner somewhere... and a panasonic system... |
| 21:29.23 | thoor | checking |
| 21:29.36 | thoor | friend |
| 21:29.37 | *** join/#asterisk PTG123 (~PTG123@001-735-812.area1.spcsdns.net) |
| 21:29.40 | PTG123 | ? |
| 21:29.47 | bjohnson | ?? |
| 21:30.19 | *** join/#asterisk yasha (~yasha_x@69.15.218.218) |
| 21:31.01 | *** join/#asterisk marcel_ (~marcel@cpc1-shep4-3-0-cust235.leic.cable.ntl.com) |
| 21:32.57 | *** join/#asterisk SirPrize (~blah@host-84-9-105-17.bulldogdsl.com) |
| 21:33.05 | *** join/#asterisk Sconk (~klaus@sconk.dk) |
| 21:33.19 | ast_freak | Anyone know when we're going to see more features from HEAD make it into a release form? perhaps asterisk 1.1? |
| 21:33.56 | thoor | bjohnson: when i just called the device directly, it instantly timed out and went to s in default |
| 21:35.55 | yasha | Ok, I got the Incoming to work with BV, now for outgoing... Any luck where do I look for that? Right now, anytime that I try to dial, it says "circuits are busy now"... |
| 21:36.14 | *** part/#asterisk PTG123 (~PTG123@001-735-812.area1.spcsdns.net) |
| 21:37.07 | Sconk | humm i have problems playing gsm files i gets this error http://sconk.dk/temp/error.txt budt the gsm file is there.. |
| 21:37.20 | rvhi | i tried to get cvs stable, anyone knows how? |
| 21:37.25 | |Vulture| | Anyone know what causes an event where the party on the * server calls # and they just hear ringing, although the # picks up, or they pick up and they can hear but not be heard? |
| 21:37.27 | rvhi | i looked at this wiki page |
| 21:37.30 | rvhi | http://www.voip-info.org/tiki-index.php?page=Asterisk%20Download |
| 21:37.32 | bjohnson | ast_freak: no |
| 21:37.53 | *** part/#asterisk emitrax (~emitrax@host55-74.pool80183.interbusiness.it) |
| 21:38.15 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 21:38.31 | bjohnson | ~cvs |
| 21:38.32 | jbot | cvs stands for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS can be found at http://asterisk.espia-net.net/horde/chora/cvs.php |
| 21:39.46 | bjohnson | rvhi: the cvs co -r v1-0 asterisk on that wiki page should do it |
| 21:39.48 | rvhi | i was told to get cvs stable |
| 21:40.01 | rvhi | b/c it will be the next stable release |
| 21:40.16 | rvhi | i don't want cvs head |
| 21:41.13 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:42.13 | rvhi | in wiki, it says " If you want the latest stable and proven code, use the CVS stable branch. " |
| 21:42.16 | rvhi | what does it mean? |
| 21:43.52 | ariel_ | rvhi, cvs co -r v1-0 asterisk is the current stable. It's the best way to get it. |
| 21:44.08 | zapa | hi all, me again i have a PRI 30 telrad card it's possible to connect to e100p ? |
| 21:44.08 | rvhi | that's stable, right? not cvs stable? |
| 21:44.24 | ariel_ | -r v1-0 is stable |
| 21:44.43 | kpfleming | which is the same thing as "cvs stable" |
| 21:45.05 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 21:45.16 | ariel_ | zapa, if it works like any other PRI yes it should work. just get the right settings on it. |
| 21:45.17 | jsolares | anyone know if the alcatel ip touch 4068 works with h323 in asterisk? |
| 21:45.17 | xkev | ariel_, not 'release' but bugfixes since last release |
| 21:45.18 | rvhi | oh... though -r v1-0 is the latest release, ie. 1.0.5 |
| 21:45.22 | xkev | not ariel, ie: |
| 21:45.59 | xkev | but I love cvs head, but then I'm changing code all the time too :) |
| 21:46.01 | rvhi | ok, anyway, -r v1-0 gets the latest release + bug fix? |
| 21:46.46 | ariel_ | rvhi, yes |
| 21:47.42 | rvhi | cool, thx. i will stick with cvs stable |
| 21:50.14 | dca[laptop] | is VAD actually a switch for iax.conf? |
| 21:50.42 | dca[laptop] | let me rephrase that, will VAD=no actually do anything? |
| 21:53.44 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
| 21:53.58 | pdracevich | hi all!! |
| 21:54.29 | pdracevich | tzanger: I am going great guns working very very well thanks..........BUT |
| 21:54.37 | __Sparks_ | I have another question! - I have signed up with a provider voip.org - they use an outbound proxy at port 5065 - do I need to set this on sip.conf? |
| 21:55.00 | CleanerX | anyone knows this : |
| 21:55.12 | CleanerX | WARNING[34835]: codec_speex.c:166 speextolin_framein: Out of buffer space |
| 21:56.32 | pdracevich | When I dial a number say 4327763 which the dial rule points it to the server "A" and dial into the local PSTN it work, but when i dial an internatiol number i get Rejected connect attempt from 210.54.249.x, request '0044134484717@incoming' does not exist <--- Can anyone help please. |
| 21:59.17 | pdracevich | help? anyone? |
| 21:59.26 | jsolares | what is better, h323 or oh323? |
| 22:03.17 | ionix | anyone has PSAP for montreal ? |
| 22:03.26 | ionix | I keep calling them and they say it doesn't exist |
| 22:03.30 | ionix | which is bullsh*t |
| 22:03.31 | ionix | ;) |
| 22:04.47 | stevekstevek | jsolares: looking for a fight? |
| 22:05.32 | jsolares | stevekstevek: trying to make netmeeting work, but it doesnt seem to |
| 22:05.52 | jsolares | i have no other h323 devices to try, so wanted a headastart for when i "HAVE" to set up h323 |
| 22:06.11 | shido6 | doesnt sjphone |
| 22:06.12 | shido6 | do h323 |
| 22:06.29 | *** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it) |
| 22:06.44 | jsolares | hey greg, i'll try that then |
| 22:06.51 | thoor | is a register statement needed in iax.conf when using an iaxy? |
| 22:07.00 | shido6 | no |
| 22:07.07 | shido6 | just a user and a peer |
| 22:07.13 | shido6 | users have contexts, peers dont |
| 22:07.19 | shido6 | peers have hosts and users dont |
| 22:07.53 | thoor | hmmm |
| 22:08.49 | thoor | i just dont understand why this is not working then |
| 22:09.09 | shido6 | check your pmsg |
| 22:09.52 | thoor | i did debug and it has a progression of subclasses: NEW->Accept->Ack->Ringing->ack |
| 22:10.12 | thoor | pmsg? |
| 22:10.18 | shido6 | private message |
| 22:11.03 | Sconk | is there a way form the cli to see what codec a user is using? |
| 22:12.07 | shido6 | yesssssss |
| 22:12.15 | shido6 | what kinda channel is it? |
| 22:12.17 | shido6 | sip? |
| 22:12.24 | pdracevich | When I dial a number say 4327763 which the dial rule points it to the server "A" and dial into the local PSTN it work, but when i dial an internatiol number i get Rejected connect attempt from 210.54.249.x, request '0044134484717@incoming' does not exist <--- Can anyone help please. |
| 22:12.42 | shido6 | yes |
| 22:12.47 | shido6 | it doesnt exist in your dialplan |
| 22:12.48 | shido6 | read it |
| 22:12.50 | shido6 | look at it |
| 22:12.59 | shido6 | understand that that number doesnt match anything inthe "incoming" context |
| 22:13.18 | Pinhole | Is there an automated tool that can evaulate call quality? Like one * server calls another, plays a message that is echoed, and records the quality. |
| 22:13.49 | bkw_ | haha |
| 22:13.52 | bkw_ | ya its easy to see |
| 22:13.59 | anthm | yacto |
| 22:15.39 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:15.56 | zapa | thanks ariel |
| 22:16.11 | dca[laptop] | okay, i'm at odds with the LEC who is telling me that when i put my call's on hold that the RTP stream goes dead and causes their system to send messages that they don't want to send. My setup is LEC --> * proxy --> SIP device. Any ideas? |
| 22:17.12 | zno | when I park a call, I don't get the parked extension spoken back to me, but when I call the parking extension, I do. |
| 22:18.07 | *** join/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 22:19.01 | BuckRogers | hello |
| 22:19.10 | mmlj4 | who was in Mobile last week? I don't know any of you by nicks |
| 22:20.06 | mmlj4 | s/week/weekend/ |
| 22:21.07 | dca[laptop] | better way to describe it is LEC --> (via SIP) --> * gateway --> (via IAX) --> * proxy --> (via SIP) --> sip device |
| 22:22.00 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 22:22.39 | yasha | GUYS, I am trying to get the OUTGOING to work with BV and placing a call and receiving the following in CLI: Any ideas: |
| 22:22.41 | yasha | <PROTECTED> |
| 22:22.41 | yasha | <PROTECTED> |
| 22:22.41 | yasha | <PROTECTED> |
| 22:22.41 | yasha | <PROTECTED> |
| 22:22.41 | yasha | <PROTECTED> |
| 22:22.43 | yasha | <PROTECTED> |
| 22:22.45 | yasha | <PROTECTED> |
| 22:22.47 | yasha | <PROTECTED> |
| 22:22.48 | Trionnis | ... |
| 22:22.49 | yasha | <PROTECTED> |
| 22:22.51 | yasha | <PROTECTED> |
| 22:23.01 | Trionnis | ever heard of pastebin? |
| 22:23.03 | Trionnis | :| |
| 22:23.05 | yasha | sorry |
| 22:23.15 | Beirdo | ~pastebin |
| 22:23.16 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 22:23.16 | Trionnis | haha |
| 22:23.17 | yasha | did not think about it right away |
| 22:23.21 | BuckRogers | haha just did the same thing |
| 22:23.24 | Trionnis | ;) |
| 22:23.39 | BuckRogers | hey |
| 22:23.53 | BuckRogers | jose consakeco is spilling the beans on 770am |
| 22:23.56 | BuckRogers | WABC |
| 22:24.00 | iceyp | anyone here using astcc ? |
| 22:24.03 | BuckRogers | its pretty sick |
| 22:24.03 | Trionnis | does BV need the 1 on the outbound? |
| 22:24.11 | yasha | Anyone? |
| 22:25.37 | bkw_ | yasha, who the hell told you to dial SIP/peer/exten? |
| 22:25.39 | bkw_ | thats EVIL |
| 22:25.53 | Trionnis | hahaha |
| 22:25.53 | bkw_ | SIP/EXTEN@peer |
| 22:26.20 | yasha | SIP/peer/exten: sorry, what is that? |
| 22:26.25 | dca[laptop] | bkw_ you might now what ails me |
| 22:26.33 | yasha | what am I doing wrong? |
| 22:26.33 | dsmouse | (tho sip/peer/exten would be more consistant with zap) |
| 22:26.37 | bkw_ | so |
| 22:26.41 | bkw_ | thats not the point |
| 22:26.51 | dsmouse | no, it's not :) |
| 22:26.57 | dca[laptop] | okay, i'm at odds with the LEC who is telling me that when i put my call's on hold that the RTP stream goes dead and causes their system to send messages that they don't want to send. My setup is LEC --> * proxy --> SIP device. Any ideas? |
| 22:27.02 | bkw_ | the point is the auth info might not go thru correctl if you dial like that |
| 22:27.11 | *** join/#asterisk thoor (~jhlk@cus04-118.cbnstl.net) |
| 22:27.41 | zapa | thanks for all |
| 22:28.42 | Qwell | dca[laptop]: They're getting mad at you, because their system is sending bad info? |
| 22:29.38 | dca[laptop] | Qwell: no, their system generates and log's error messages when the RTP stream get's dropped, it's something they would like to avoid if possible, so i'm wondering how i can keep the RTP stream going even when my SIP phone is on hold. |
| 22:31.56 | *** join/#asterisk jesse_132 (~chatzilla@207.246.72.150) |
| 22:32.52 | jesse_132 | is there a phone service called a "hot-wire" .... a potential client says they have one that allows all calls in the state to be free... The guy I was talking with didn't really know exactly though... |
| 22:32.53 | |Vulture| | Anyone here use IP500s? |
| 22:33.33 | DJ-Pyro | |Vulture|: yes |
| 22:34.24 | |Vulture| | DJ-Pyro: how do you deal with internal CID? mine always passes the reg.1.displayName and not the CID field in sip.conf |
| 22:34.40 | DJ-Pyro | hmm, ours passes CID from sip.conf |
| 22:35.04 | |Vulture| | strange... maybe it is an option |
| 22:35.19 | *** join/#asterisk JoaoCorreia (~JoaoCorre@81.193.116.63) |
| 22:35.29 | JoaoCorreia | hello |
| 22:35.58 | |Vulture| | DJ-Pyro: any way I could get you to post your IP500 conf file? |
| 22:36.06 | DJ-Pyro | |Vulture|: get rid of reg.1.displayName from the file |
| 22:36.39 | |Vulture| | DJ-Pyro: aaaah... I tried to set it to "" but that didn't work... didn't think of trashing it all together |
| 22:36.53 | iceyp | anyone using a calling card application with asterisk? specifcally astcc ? |
| 22:37.36 | ta[i]nted | i did |
| 22:37.39 | JoaoCorreia | I have an ISDN line in Europe |
| 22:37.40 | ta[i]nted | for a very short while |
| 22:37.54 | JoaoCorreia | are there any cards combatible ? wich one should I use ? |
| 22:42.03 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
| 22:42.09 | *** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
| 22:47.08 | *** join/#asterisk w0w0 (~w0w0@80-28-171-26.adsl.nuria.telefonica-data.net) |
| 22:50.44 | *** join/#asterisk Rick_Hunter (~rhunter@03-040.008.popsite.net) |
| 22:52.13 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:52.37 | CleanerX | JoaoCorreia, what exactly do you want? |
| 22:52.42 | yashax | http://pastebin.ca/6306 - Please give me a hand in trying to figure out what is the problem with this. I am trying to get Outgoing working with BV! |
| 22:56.38 | yashax | guys, anyone? |
| 22:56.59 | hardwire | Ima girl! |
| 22:57.04 | hardwire | in my dreams. |
| 22:58.47 | xkev | if I'm using [queue] strategy = leastrecent, it's only calling the first member (who has least recently taken a call), but if that member doesn't answer, it won't try anyone else |
| 22:59.33 | xkev | ..I could swear this worked in the past, maybe something changed in cvs |
| 23:01.21 | *** join/#asterisk eaperezh (~chatzilla@200.124.6.186) |
| 23:01.50 | *** part/#asterisk yashax (~yasha_x@69.15.218.218) |
| 23:01.59 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
| 23:02.48 | jsolares | yashax: i think your missing a 1 in front of your number |
| 23:04.13 | yashax | I am using a 9 for outgoing..... |
| 23:04.15 | *** join/#asterisk Tili (~Tili@202-133-65-128-dialup.sat.net.pk) |
| 23:04.32 | yashax | I have tried with or without 9 with the same result |
| 23:05.48 | jsolares | it should be an 11 digit number if i'm correct |
| 23:06.04 | jsolares | 1 XXX areacode XXX XXXX phone # |
| 23:06.15 | jsolares | well 1NXXNXXXXXX |
| 23:06.53 | yashax | what do I have in my pastebin? |
| 23:07.10 | jsolares | a 10 digit number |
| 23:07.17 | jsolares | and if you say you're using the 9, an 8 digit number |
| 23:08.02 | ionix | Anyone has the PSAP for montreal ? |
| 23:08.09 | jsolares | you could use _91NXXNXXXXXX in your dialplan yashax |
| 23:08.13 | yashax | no, I am using 9 as a prefix for outgoing line... The reason the pastebin has only 10 digits, it is because I have tried to dial it without 9, but the result is the same... |
| 23:08.38 | iceyp | anyone here using astcc calling card app ? |
| 23:09.35 | jsolares | but what about the 1? |
| 23:09.36 | outtolunc | xkev: there is a new var 'numbusies' in ring_one... check the configs |
| 23:09.52 | xkev | outtolunc, thanks |
| 23:10.17 | *** join/#asterisk wangster (~wangster@S0106000c41aae2bf.wp.shawcable.net) |
| 23:10.38 | yashax | anyone has any ideas? |
| 23:10.47 | *** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net) |
| 23:11.12 | *** join/#asterisk thoor (~jhlk@cus04-118.cbnstl.net) |
| 23:11.12 | jsolares | have you tried with the 1 before your 10 digit number? |
| 23:11.19 | wangster | Is there a way to get asterisk to display DTMF tones on the console as they are recieved? Specifically with regard to SIP. |
| 23:11.20 | jsolares | like dialing a 1-800 |
| 23:11.22 | __Sparks_ | has anyone her setup Asteris with VOIPTalk, suing SIP rather than IAX? |
| 23:11.28 | yashax | without 9? |
| 23:11.46 | jsolares | well 9 doesnt matter, that's asterisk prefix you want to use |
| 23:11.49 | Pinhole | Anybody know of any automated call quality monitoring tools for asterisk? |
| 23:12.02 | jsolares | so it should be 9 1 555 555 5555 |
| 23:12.24 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 23:12.24 | jsolares | so you tell broadvoice to dial 1 555 555 5555 |
| 23:12.32 | shido6 | suing a new voip provider already __Sparks_ |
| 23:12.32 | shido6 | ? |
| 23:12.36 | yashax | yep, just did .... no go... |
| 23:12.37 | xkev | outtolunc, then is this the expected behavior? looks like we were trying to fix penalty |
| 23:13.11 | jsolares | yashax: i'm out of ideas |
| 23:13.21 | jsolares | what does sip show registry tell you? |
| 23:13.34 | jsolares | yashax: ask shido6. he's good |
| 23:13.36 | |Vulture| | registered sip clients |
| 23:13.37 | yashax | registered |
| 23:14.10 | thoor | shido: I made those changes to my iax.conf, now it gives me the error unable to create channel of type IAX2 |
| 23:14.20 | outtolunc | xkev: i've not reviewed this issue, but the quick scan i did.. it seems like it |
| 23:14.32 | jsolares | yashax: have you seen this : http://www.broadvoice.com/support_install_asterisk.html |
| 23:14.32 | outtolunc | i'll see what i can do |
| 23:14.44 | yashax | Yeah, I followed it... |
| 23:14.50 | Grooby | yuke |
| 23:14.52 | yashax | incoming works and outgoing does not.... |
| 23:14.52 | Grooby | don't follow that |
| 23:14.54 | Grooby | it's so out of date |
| 23:15.03 | |Vulture| | yea that page is crap |
| 23:15.15 | |Vulture| | the wiki has a working |
| 23:15.42 | thoor | the broadvoice support page is actually out of date |
| 23:15.50 | thoor | who was having trouble with it? |
| 23:15.52 | yashax | I think I know why.. All of the instructions are based on straight Asterisk config files, but I am running Asterisk@home with ACD so their configs are slightly different.... Hhhhhrrrrrrrrrrrrr |
| 23:15.53 | |Vulture| | and they mispell their name on it :P |
| 23:15.59 | zno | anyone provision their snoms here? |
| 23:16.01 | Grooby | yashax |
| 23:16.04 | Grooby | i can help you |
| 23:16.05 | yashax | About to KILLLLLLLLLLLLLLLLLL MYSELF |
| 23:16.07 | |Vulture| | proxy.mia.brodavoice.com |
| 23:16.11 | Grooby | I run *@home w/ BV |
| 23:16.16 | yashax | NO WAY |
| 23:16.19 | yashax | WOW |
| 23:16.20 | yashax | please |
| 23:16.23 | yashax | :) |
| 23:16.25 | jsolares | hehe |
| 23:16.26 | thoor | nope...those are no longer good the proxies that is |
| 23:16.49 | |Vulture| | proxy.mia.boradvoice.com... yea that one deff is not ;) |
| 23:17.07 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 23:17.10 | |Vulture| | if you use that config for outbound you will get congestion |
| 23:17.11 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 23:17.12 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:17.23 | pdracevich | <PROTECTED> |
| 23:17.23 | thoor | try host=sip.broadvoice.com |
| 23:17.39 | thoor | same for fromdomain |
| 23:17.44 | xkev | outtolunc, I probably ought to file a report. if a member doesn't answer, it should try the next member. that's just dumb if it doesn't. :) |
| 23:18.02 | |Vulture| | http://www.voip-info.org/tiki-index.php?page=Broadvoice#comments |
| 23:18.06 | |Vulture| | that works |
| 23:18.10 | *** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 |
| 23:18.35 | wangster | is there any way to get asterisk to display DTMF keypresses on the console? |
| 23:18.46 | |Vulture| | and make sure to run 1.0.5+ or you will get blacklisted from BV |
| 23:19.05 | |Vulture| | wangster: you mean like viewing in verbose mode? |
| 23:20.38 | thoor | could somone help me with my Iaxy? |
| 23:21.14 | thoor | i am getting a cannot create IAX2 channel error |
| 23:21.19 | wangster | vulture: yes I suppose. I have asterisk in verbose mode but it dosn't display anything when keys are pressed. |
| 23:21.31 | shido6 | thoor |
| 23:21.34 | shido6 | read your pmsg |
| 23:21.35 | thoor | ahh |
| 23:21.51 | IronHelix | for BV- atm ( at least as of last night ) CHI was the only proxy that works correctly |
| 23:21.56 | wangster | The main problem I'm trying to solve is remote retreval of voicemail. |
| 23:22.02 | |Vulture| | wangster: do asterisk -vvvvvvvr |
| 23:22.13 | |Vulture| | wangster: if you don't see them... then they aren't being passe |
| 23:22.14 | |Vulture| | d |
| 23:22.21 | pdracevich | <PROTECTED> |
| 23:22.21 | IronHelix | i have NO IDEA why, but for some reason doing this http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup makes it work with the other proxies |
| 23:24.04 | wangster | vulture: I have verbose set to 9. I assure you they are being passed but nothing displays in the console. |
| 23:24.21 | |Vulture| | strange... mine display.. |
| 23:24.29 | |Vulture| | sip debug? |
| 23:24.38 | fafnir | with asterisk, if i go with say at&t's voip, can i assign all the numbers to the asterisk server so it makes outgoing calls on them? |
| 23:25.07 | wangster | vulture: I have sip debug on as well... Let me try restarting.... |
| 23:25.28 | IronHelix | fafnir- i think ATT voip is locked to an ATA |
| 23:25.31 | *** join/#asterisk JoaoCorreia (~JoaoCorre@81.193.116.63) |
| 23:26.13 | wangster | Vulture: no luck :( THis is Asterisk 1.0.5 |
| 23:26.18 | fafnir | ata? |
| 23:26.33 | fafnir | we're trying to set up a big business with all voip |
| 23:26.38 | fafnir | what would be a good provider? |
| 23:27.01 | Beirdo | hmmmm |
| 23:27.06 | IronHelix | ATA = analog telephony adapter. its a box that basically has ethernet on one end and a RJ11 phone on the other |
| 23:27.11 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 23:27.21 | Beirdo | my music on hold stuff doesn't work if I'm using GSM codec |
| 23:27.26 | Beirdo | but it does for ulaw |
| 23:28.03 | Beirdo | works in speex too |
| 23:28.11 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
| 23:28.23 | IronHelix | ATA = NOT very useful at all if you want to be doing |
| 23:28.25 | IronHelix | VoIP |
| 23:28.35 | IronHelix | IE soft pbx voip |
| 23:28.39 | wangster | vulture: what type of DTMF signalling are you using? RFC, INFO, or INBAND? |
| 23:28.40 | Beirdo | working in ilbc |
| 23:28.58 | IronHelix | you'd want a provider that will let you use your own equipment, which is sometimes referred to a BYOD (bring your own device) |
| 23:29.00 | *** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net) |
| 23:29.35 | IronHelix | so fafnir, to get calling from a * box you might consider looking into a wholesale provider |
| 23:29.41 | IronHelix | it really depends on how much calling you do |
| 23:29.43 | Beirdo | and now it's working with gsm |
| 23:30.14 | RoyK | > |
| 23:30.26 | RoyK | ~jerjer |
| 23:30.27 | jbot | well, jerjer is nufone |
| 23:31.00 | |Vulture| | wangster: what service are you using? |
| 23:31.08 | wangster | Vulture: service? |
| 23:31.26 | |Vulture| | like Zap/IAX2 (VoicePulse)/SIP (Broadvoice) |
| 23:32.08 | wangster | vulture: no service. Its an asterisk box connected to SIP phones. A PRI runs to a Cisco switch which also talks SIP directly to asterisk. |
| 23:32.39 | RoyK | jbot: no, jerjer is that stupid pro american guy that runs nufone |
| 23:32.40 | jbot | okay, RoyK |
| 23:32.42 | |Vulture| | wangster: ah, I've never done those :( |
| 23:33.11 | wangster | vulture: I can't get the DTMF to work when calling in from the PRI so I'm trying to debug it. Its almost certainly a config problem on the Cisco but before I can debug it i need to see if the Asterisk box is registering the keypresses. |
| 23:33.49 | wangster | vulture: DTMF works fine from the SIP phones but I can' t get Asterisk to display the keypresses on the console. |
| 23:34.31 | __Sparks_ | I am having trouble when I have more than one Sip gateway in my sip.conf - i am getting the error "Failed to authenticate on REGISTER" followed by all but one of my accounts - do I need different port numbers for each? |
| 23:34.34 | Beirdo | ~Beirdo |
| 23:34.35 | jbot | somebody said beirdo was a dumbass some days, and irritable on Mondays |
| 23:34.38 | Beirdo | hehe |
| 23:34.51 | Beirdo | good to see nobody's corrected it :) |
| 23:35.24 | Beirdo | ~royk |
| 23:35.25 | jbot | royk is probably mean and shoots people with quantum singularity weapons |
| 23:35.31 | Beirdo | heh |
| 23:35.39 | stevekstevek | ~stevekstevek |
| 23:35.56 | tuxinator_linux | Do someone know of a good reference to learn what options are availble to get dial tone on * (or is willing to talk to me)? I am interested in using the Digium t1 card. |
| 23:36.08 | stevekstevek | ~stevekstevek |
| 23:36.09 | jbot | i heard stevekstevek is someone who told me to say this |
| 23:36.21 | RoyK | ~shoot Beirdo |
| 23:36.23 | jbot | ACTION shoots Beirdo in the eye with a quantum singularity weapon! |
| 23:36.35 | Beirdo | ~trout RoyK |
| 23:37.02 | Beirdo | hmph |
| 23:41.37 | CleanerX | does anybody know that current status of encryption with SIP? |
| 23:41.49 | CleanerX | either SIPS or SRTP ? |
| 23:42.51 | IronHelix | dont think its getting much of anywhere, but i remember hearing that there was experimental iax crypto in the works |
| 23:43.32 | shido6 | tuxinator_linux read your private messages |
| 23:44.07 | CleanerX | yeah - what i wonder about is the aes.c file in * distribution |
| 23:46.04 | stevekstevek | the aes.c is for encryption, but it doesn't mean that SRTP is implemented.. AES is used in other places in *. |
| 23:46.07 | *** join/#asterisk salmandr (~salmandr@h216-170-207-50.216-170.unk.tds.net) |
| 23:46.17 | pointer-gaim | do the sipura spa 3000s have an FXO(<->SIP) port on them? |
| 23:46.21 | |Vulture| | like authenticating IAX2 |
| 23:46.40 | salmandr | does anyone know of a IAX demo server i can call |
| 23:46.50 | salmandr | maybe one that i can listen to MOH for 5 mins or so |
| 23:46.56 | stevekstevek | guest@iax2.fwdnet.net/613 |
| 23:46.58 | Nugget | yes. the one in the sample config files you installed with asterisk. |
| 23:47.03 | Nugget | the one that the asterisk docs describe. |
| 23:47.03 | stevekstevek | guest@misery.digium.com |
| 23:47.54 | salmandr | i called digium but their menu seems like I should be talking to tech support or something |
| 23:48.10 | salmandr | i just want a somewhat lengthy session so i can measure bandwidth usage |
| 23:48.14 | stevekstevek | so don't go to any of the choices. Play in the directory if you want. |
| 23:48.26 | stevekstevek | or, just use the fwd echo test I sent you. |
| 23:48.38 | salmandr | i'll try it out, thanks |
| 23:48.39 | *** join/#asterisk file2 (~file@mctn1-1987.nb.aliant.net) |
| 23:48.56 | IronHelix | yeah use the echo test and if you want to simulate usage just put the phone in front of your cd player |
| 23:49.20 | stevekstevek | that's not likely to make any difference. |
| 23:49.23 | Nugget | I did that when I was diagnosing my voicpulse jitter problems. |
| 23:49.35 | stevekstevek | unless you're using a VBR codec, and only speex supports that at the moment. |
| 23:49.54 | salmandr | does it matter if there is noise or not? looks like misery.digium.com uses gsm |
| 23:50.04 | salmandr | ahh ok |
| 23:50.06 | stevekstevek | not that you're sending.. |
| 23:50.26 | IronHelix | only reason for noise is to keep the codec going in case silence suppression is used somewhere |
| 23:50.28 | stevekstevek | but, if you go to misery.digium.com, it will stop sending you packets after the menu plays. |
| 23:50.59 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:51.05 | stevekstevek | IronHelix: but * doesn't do silence suppression... And "noise" won't fool good VAD anyway :) |
| 23:51.20 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 23:52.29 | BoRiS | ~seen paulc |
| 23:52.35 | jbot | paulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 18h 55m 34s ago, saying: 'Is it me, or are there a handful of guys in the final 12 who are just fecking awful?'. |
| 23:54.42 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 23:55.48 | stevekstevek | why is it that 10% of mails to asterisk-users are in the wrong thread. |
| 23:55.57 | stevekstevek | are people that lazy that they can't start a new mail? |
| 23:56.55 | CoaxD | stevek: because people are stupid and reply to messages to start new ones |
| 23:57.04 | CoaxD | stevek: (of course, leaving the msgid trail intact) |
| 23:57.30 | stevekstevek | is it stupid (well, uninformed), or just lazy.. |
| 23:57.35 | CoaxD | both |
| 23:58.25 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
