| 00:00.08 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 00:01.24 | quidpro | Manx: Basically incoming calls are redirected to DeadAGI, which dials out (no answering specified) another SIP line... nothing on the DIAL line oterh than 60 (timeout) |
| 00:03.07 | quidpro | Manx: Unable to find a codec translation path from g729 to ulaw |
| 00:03.54 | quidpro | Manx: WARNING[32053] channel.c: No path to translate from SIP/out-085bcd18(4) to IAX2/1.2.3.4:4569-101(256) |
| 00:05.17 | ManxPower | quidpro: "show codecs" will tell you what 4 and 256 is |
| 00:05.32 | JunK-Y | isnt ulaw and gsm? |
| 00:05.59 | JunK-Y | ha, 729 |
| 00:06.00 | ManxPower | 4 is ulaw 256 is G729 |
| 00:06.10 | ManxPower | quidpro: so your legs are NOT all g729 |
| 00:06.46 | ManxPower | quidpro: you realize the order of the codecs in sip.conf means nothing, right? |
| 00:06.54 | quidpro | Manx: Let me take a look... |
| 00:06.54 | testdriver12500 | ignore my post...I found a typo in zapata.conf. I feel like a fool. |
| 00:06.56 | ManxPower | see also README.variables, pay special attention to SIP_CODEC (only works for OUTGOING calls) |
| 00:08.48 | quidpro | Manx: In SIP.CONF and iax.conf I have, disallow=all,allow=ulaw,allow=g729,allow=gsm,allow=ilbc |
| 00:09.29 | ManxPower | quidpro: Asterisk will almost always pick ulaw if it has a choice |
| 00:10.03 | quidpro | Manx: When I tested with the Intel G729 binary is about, both calls legs show G729 and the message disappears. But I would rather not use that code for a multitude of reasons. |
| 00:10.40 | ping2921 | Is there a softphone that can do 6-10 concurrent calls? |
| 00:10.55 | JT | xpro and presumable eyebeam can do 6 |
| 00:10.59 | ManxPower | hence my comment about SIP_CODEC |
| 00:11.02 | JT | so can idefisk |
| 00:11.16 | ping2921 | upto 10? |
| 00:11.20 | JT | haven't seen any that do more than 6 |
| 00:11.24 | JT | seems unnecessary |
| 00:11.53 | quidpro | Manx: Hmm, looking at README.variables now... |
| 00:13.16 | quidpro | Manx: So if I am reading this correct, somehow get * to set the SIP_CODEC variable before initiating the second leg? |
| 00:13.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
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| 00:20.24 | *** join/#asterisk Vec (n=Vec@dsl-242-252-39.telkomadsl.co.za) |
| 00:29.08 | flenders | rhombus: did you sort out the echoes on TDM400P? |
| 00:31.39 | JacksLivr | i have a question that may show how little i actually know. Can a person who has no service and is not logged into or in anyway connected to a voip server, use a softphone to dial a sip url that will ring into my * box? I set up a srv record today. |
| 00:31.50 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:31.53 | generalhan | hey all ! |
| 00:32.40 | flenders | JacksLivr: softphone would need to be registered with some service, no? |
| 00:32.58 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 00:33.06 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
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| 00:33.09 | flenders | JacksLivr: I guess you can't dial anywhere without a voip service |
| 00:33.20 | generalhan | i need some ideas on setting up a remote phone other than forwarding SIP at the endpoints. I have tried everything (well apparently everything but the one right way) to get this cisco phone working at a remote location and i just cant get it working ... so i need some other ideas ! |
| 00:33.34 | Skaag | my iax tries to send the password as md5 and my provider does not support that |
| 00:33.47 | Skaag | how do i tell my asterisk to send the passwords as-is, plain? |
| 00:33.54 | JacksLivr | flenders: i thought that the srv record would direct the call to a termination point on my * server |
| 00:34.28 | JacksLivr | so they would not need to know anything but my sip:asdf@domain.com addr |
| 00:34.37 | JacksLivr | but i do not know this for sure |
| 00:35.44 | flenders | JacksLivr: dunno, all the softphones I've seen register to voip service |
| 00:36.24 | JacksLivr | yeah. and that may be the only way to do it. |
| 00:36.34 | flenders | JacksLivr: it wouldn't make sense... people would have free calls all over the world. |
| 00:36.48 | JacksLivr | not to pstn |
| 00:36.49 | JT | registration is only required so that a server knows where you are if it needs to call you if you have a dynamic ip address |
| 00:36.56 | JT | or if you have nat |
| 00:37.03 | JT | it is only for receiving calls |
| 00:37.22 | flenders | JT: you would need at least a gateway, wouldn't you? |
| 00:37.31 | JT | however a few sip providers supposedly won't let you make calls unless you're currently registered |
| 00:37.41 | JT | never seen that behaviour myself |
| 00:37.45 | JacksLivr | JT: can you have a person just dial a sip url without a service? |
| 00:37.56 | JT | what do you mean without a service? |
| 00:38.06 | flenders | not signing up with any ITSP |
| 00:38.26 | JT | oh, he could call someone else with an asterisk box set to receive calls |
| 00:38.36 | JacksLivr | well.... i have this asterisk server and i wanna have someone who knows nothing of voip and has no service to download a soft phone and call me kn my * server |
| 00:38.43 | JT | a lot of people have SIP and IAX urls that can be called by anyone |
| 00:38.56 | JacksLivr | that is what i am trying to do |
| 00:39.03 | JacksLivr | and i think i have it set up |
| 00:39.18 | JacksLivr | i cannot find a softphone that i do not have to register |
| 00:39.20 | JT | you need an entry that allows calls from unauthenticated people |
| 00:39.23 | JT | oh |
| 00:39.27 | JT | idefisk doesn't |
| 00:39.36 | JT | i didn't think xlite needed to |
| 00:39.37 | JT | but not sure |
| 00:40.22 | JacksLivr | yeah i tried with xlite |
| 00:40.29 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:40.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:40.30 | flenders | JT: don't you need a gateway to make the call? |
| 00:40.40 | orlock | JT: know of any 1RU redundant PSU's? |
| 00:40.51 | JacksLivr | im thinking that the client would terminate at my * box |
| 00:40.54 | JT | orlock: psus only or cases? |
| 00:41.02 | JT | flenders: what do you mean gateway? |
| 00:41.10 | flenders | so you could make the call |
| 00:41.11 | orlock | well, the PSU is probably designed for a specific rack case |
| 00:41.20 | JT | orlock: supermicro make a few |
| 00:41.27 | JT | built into cases |
| 00:41.37 | orlock | cool will check it out |
| 00:41.55 | JT | they have dozens and dozens of 1RU cases/barebones systems, it's crazy |
| 00:41.56 | orlock | i was looking at tyan transport servers, and thats the only thing they dont have |
| 00:42.02 | JT | hrm |
| 00:42.12 | orlock | supermicro are mainly intel, right? |
| 00:42.13 | *** join/#asterisk demigod2k (n=joey@cpe-65-29-113-212.twmi.res.rr.com) |
| 00:42.15 | demigod2k | hi |
| 00:42.40 | JT | orlock: supermicro also now put out a dual system dual xeon (dual/quad core capable) 1RU server |
| 00:42.54 | JT | it has 2 computers in 1 RU |
| 00:43.09 | orlock | yeah, just noticed that |
| 00:43.12 | JT | the only bad thing is non-redundant power supply |
| 00:43.15 | orlock | they have some pretty nice systems |
| 00:43.20 | JT | they do mostly intel, but some amd |
| 00:43.21 | JT | yeah |
| 00:44.17 | JT | flenders: an asterisk machine can be that "gateway" |
| 00:44.41 | JacksLivr | JT: do you know of a sip one that doesnt have to be registered? |
| 00:45.13 | JT | not sure, do they all refuse to make calls unless they're successful in registering? |
| 00:45.57 | JacksLivr | the few i have tried |
| 00:46.06 | flenders | JT: so, his asterisk would be the "gateway", so the softphone would need to be setup to use his ip, no? |
| 00:46.21 | JT | ip or hostname, sure |
| 00:46.29 | ManxPower | JacksLivr: many sip phones have a config option to allow calls without registrations |
| 00:46.29 | JacksLivr | flenders: i was thinking that the url would tell the client what the termination point was |
| 00:46.59 | ManxPower | remember registration ONLY tells the remote server what the IP of this client is, it does nothing else |
| 00:47.20 | flenders | JacksLivr: so the callee wouldn't need to configure anything on the softphone? |
| 00:47.27 | flenders | I mean, the caller |
| 00:47.28 | JT | JacksLivr: i'm pretty sure some sip softphones can dial by URI |
| 00:47.50 | Moobius | Is there any way to prevent "empty" voice mail messages? |
| 00:48.16 | Moobius | If a caller hangs up during the VM greeting a ~0 length VM is recorded. |
| 00:48.32 | ltdwk | set a minimum length |
| 00:48.39 | Moobius | ? how? |
| 00:49.05 | JacksLivr | flenders: that is what im hoping |
| 00:49.09 | ltdwk | minmessage=? in voicemail.conf |
| 00:49.22 | Moobius | where "?" is seconds? |
| 00:49.31 | ltdwk | yerp |
| 00:49.33 | Vec | I am tring to setup FOP to simply transfer incoming calls from the ZAP channels to other internal SIP phones, and have all kinds of trouble, do I have to drag a phone from a ZAP channel button to the destination, or can I drag a phone from the 1 internal SIP phone I am talking on to the other SIP phone I want to transfer to ? |
| 00:49.36 | Moobius | thx |
| 00:49.51 | JT | Vec: did you fix your E1? |
| 00:50.30 | Vec | JT : yeh thanks, it turns out there is a 0 coming infront of all numbers going out, and as far as I can tell it is the telco who is putting it there because pri debug shows nothing of the sort |
| 00:50.33 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:51.15 | Vec | JT: To give u an idea of my situation, the time is 2:52 AM and am trying to get this asterisk box ready for 9am :) |
| 00:52.10 | Vec | JT : all I did was removed the first digit which is a 0 90% of the time to sort it out, I am going to ivestigate further with my telco in the morning. |
| 00:54.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:55.07 | Skaag | is there some problem with voip-info.org lately? |
| 00:55.24 | Moobius | Skaag: wiki is down... |
| 00:55.26 | Vec | Skaag : it is down |
| 00:55.33 | Skaag | damn |
| 00:55.40 | ltdwk | if you set nationalprefix=0 that should deal with the 0 i would think |
| 00:55.49 | Moobius | google cache. it is your friend. |
| 00:55.50 | Skaag | I must find out how to disable my asterisk from converting my passwords to md5 hashes |
| 00:56.09 | Vec | ltdwk : thanks will try it out |
| 00:56.21 | Skaag | I need my IAX to use plain auth |
| 00:56.28 | brian | can anyone here get me a free DID for development purposes? |
| 00:56.34 | Vec | Anyone got some FOP experiance ? |
| 00:57.48 | Vec | maybe thats a dumb question. |
| 01:00.54 | `p4r14h | hey just added a TDM100P and when i run ztcfg i get ZT_CHANCONFIG Failed on channel 2: No such device or address (6) |
| 01:01.53 | ltdwk | sounds like you havn't got the driver loaded |
| 01:02.11 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 01:04.40 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 01:08.19 | `p4r14h | ltdwk: you say i dont have the driver loaded, would it show up in lsmod if it wasn't? |
| 01:09.10 | JT | Vec: ah ok |
| 01:11.29 | Vec | JT : now just trying to get FOP working properly |
| 01:12.02 | JT | never used fop |
| 01:12.07 | ltdwk | p414h: i suggest checking lsmod first to see if it's loaded, then if it is, check dmesg to see if the card was detected |
| 01:15.19 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:15.38 | `p4r14h | ltdwk: all of the zaptel drivers are in lsmod and dmesg returns found a wildcard fxo: wildcard x100p |
| 01:16.23 | DrukenLPY | has anyone done a dating service or partyline thing with asterisk ? |
| 01:16.44 | demigod2k | nope but its a clever idea |
| 01:16.52 | JT | i imagine it'd be possible |
| 01:16.55 | Qwell | DrukenLPY: I found some random website which listed some plines, and one was a dating service with Allison as the IVR, heh |
| 01:17.14 | JT | sure, to compete with the 487470324432 other similar services |
| 01:17.33 | demigod2k | just get some good late-nite cable TV commercials and you're set |
| 01:17.34 | JT | you need the marketing dollars |
| 01:17.49 | JT | how cheap are the commercials? can't be that chepa |
| 01:18.19 | `p4r14h | i can get advertising in my town for cheap. we do work for the cable company |
| 01:18.38 | `p4r14h | i might have to steal your idea :P |
| 01:18.45 | demigod2k | whats the pricing like? I'd have guessed like 50-100 bucks a pop for a standard commercial late night |
| 01:19.07 | JT | usually a consumer must hear your company's marketing message at least 6 times before they are likely to respond |
| 01:22.26 | `p4r14h | anyone know why "ZT_CHANCONFIG failed on channel 2: no such device or address (6)" but everything is loaded when checked via lsmod? |
| 01:23.16 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-03012132bcb3bae5) |
| 01:28.50 | [TK]D-Fender | mihinomenest: Yes? |
| 01:29.07 | ManxPower | `p4r14h: the kernel module is loaded, but your config is not valid |
| 01:29.09 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:31.00 | Vec | Anyone know if Voip-info is planning on coming back up, maybe there hdd got corupted and they lost all the data :) ? |
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| 01:34.23 | sumasuma | Vec: i'm sure they must have backup in that case |
| 01:34.39 | sumasuma | vec: you can try with google, and they must have in cache |
| 01:38.52 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 01:39.03 | ttuttle | Will archived callfiles tell me whether a call was busy or answered? |
| 01:40.25 | Moobius | does anyone make SIP phones like this: http://www.bia-bg.com/pictures/red-phone.jpg |
| 01:40.45 | Moobius | buttonless, hotphone only |
| 01:41.38 | Qwell | haven't seen any |
| 01:42.21 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
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| 01:46.02 | gambolputty | Ahhh yes, the red "you're fired" phone. |
| 01:47.47 | ttuttle | Hey, is there any way to detect a "soft busy" signal, i.e. one that is played in audio, but not sent over SIP? (Like the ones American Idol sends?) |
| 01:48.11 | Vec | Whatever I do to try and set the callerid for incoming calls it does not work http://pastebin.ca/394103 ? |
| 01:48.33 | Vec | I want to set it so calls coming in have a specefied callerid ? |
| 01:49.38 | Corydon76-home | 1) you shouldn't be using SetCallerID, as it is deprecated |
| 01:49.46 | ltdwk | you want Set(CALLERID(name)="John Public") and Set(CALLERID(number)=22222) |
| 01:49.55 | ttuttle | Anyone? |
| 01:50.09 | Corydon76-home | 2) This is a PRI circuit, and the name comes in slightly after the initial setup, so you need to Wait(0.5) before you do anything else |
| 01:50.34 | Corydon76-home | otherwise it'll get overwritten |
| 01:51.04 | ltdwk | i didn't know pri's had a Callerid name field? |
| 01:51.49 | ltdwk | how is it populated? |
| 01:51.55 | Vec | ltdwk : never knew either |
| 01:52.08 | Corydon76-home | PRIs don't send calleridname in the initial SETUP message |
| 01:52.45 | ping2921 | is there a timeout for accessing mysql select results? |
| 01:52.47 | Corydon76-home | CallerIDName comes in a separate packet, after the initial SETUP |
| 01:53.04 | ltdwk | varying from telco to telco no doubt |
| 01:53.13 | Corydon76-home | No, it's part of Q.931 |
| 01:53.44 | Corydon76-home | ping2921: using what? |
| 01:54.09 | ping2921 | whenever I update rows in mysql, some of my calls are dropped. Im using asterisk realtime with mysql-addons. |
| 01:54.38 | Corydon76-home | No, there is no timeout |
| 01:55.37 | ping2921 | there must be otherwise all calls will hang due to query delays. |
| 01:56.34 | ltdwk | if a query times out, that is a bad time(tm) |
| 01:56.51 | ltdwk | you dont want whatever lookup method you are using to ever be that latent or unreliable |
| 01:57.07 | ltdwk | s/bad time/bad thing |
| 01:57.20 | Corydon76-home | What query delays? |
| 01:57.55 | Vec | Should ,NoOp(${CALLERIDNUM}), output the callerid number on the cli |
| 01:57.56 | Vec | ? |
| 01:58.12 | ltdwk | ${CALLERID(number)} |
| 01:58.47 | Corydon76-home | If you want output on the CLI, I'd recommend using Verbose(), not NoOp |
| 01:58.47 | ping2921 | each time I do major queries to mysql, asterisk drops calls. |
| 01:58.54 | Vec | ltdwk : thanks |
| 01:59.16 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:59.27 | ping2921 | with 'sent into invalid extension' errors |
| 01:59.29 | ltdwk | sounds like you have bigger issues.... perhaps the query is locking the table? you don't want that |
| 02:00.20 | Vec | When I look at the variable ${CALLERID(number)} it is set correctly before the DIAL, but then it is not being passed to the phone ? |
| 02:00.33 | ltdwk | even if it's the biggest query in the world, you need transactional level locking, not table level locking so you can still read from the table |
| 02:01.18 | ManxPower | Vec: CALLERIDNUM is 1.0 and 1.2, CALLERID(number) is 1.2 and 1.4 |
| 02:01.55 | ManxPower | Vec: where are you dialing? |
| 02:02.05 | ltdwk | Vec: does the handset have CLIP prohibited? |
| 02:02.26 | ManxPower | it is the CallerID NAME that can come as a separate packer |
| 02:03.28 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 02:03.48 | k-man | is voipbuster ok as a sip provider? they seem very cheap for some calls |
| 02:04.17 | Vec | The calling is coming in on the PRI with noCallerID, I then try to set it, but the phone still says unknown ? |
| 02:05.05 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:05.37 | Vec | When I look at the variable using NoOP it is set correctly ? |
| 02:05.53 | ManxPower | Vec: paste the line where you set it |
| 02:07.36 | Vec | exten => _21XX,n,Set(CALLERID(number)=6135551234) |
| 02:08.12 | ManxPower | Vec: now paste the CLI output of that line |
| 02:08.38 | Vec | - Executing [2116@zapIncoming:3] Set("Zap/0-2", "CALLERID(number)=6135551234") |
| 02:08.51 | ManxPower | Vec: what verison of asterisk are you using? |
| 02:08.58 | Vec | 1.4.0 |
| 02:09.03 | Vec | Executing [2116@zapIncoming:6] Dial("Zap/0-2", "SIP/2116@2116|500") |
| 02:09.04 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 02:09.24 | ManxPower | Vec: what SIP device are you using? |
| 02:09.38 | Vec | Grandstrem GXP 2000 |
| 02:09.38 | Corydon76-home | Vec: call SetCallerPres(allowed_passed_screen) |
| 02:10.15 | ManxPower | Corydon76-home: that works for PRI -> SIP calls? |
| 02:10.16 | Vec | Corydon76-home : where must I paste that type that ? |
| 02:10.29 | ltdwk | that is for PRIs only i was lead to believe |
| 02:10.30 | Corydon76-home | Vec: before you set the callerid |
| 02:10.32 | [TK]D-Fender | Vec: get rid of the @(whatever)" you're tacking on the end of your dial. |
| 02:10.46 | Vec | ltdwk : it is a PRI -> SIP call |
| 02:10.53 | ltdwk | i mean outgoing PRI calls |
| 02:11.10 | Corydon76-home | ManxPower: if his caller presentation is set restricted, resetting the callerid doesn't reset the restriction bits, and so the SIP phone won't get the callerid |
| 02:11.10 | Vec | oh ok |
| 02:11.28 | ManxPower | Corydon76-home: any idea why the presentation might be set to restricted? |
| 02:11.40 | Corydon76-home | ManxPower: nope |
| 02:11.43 | quidpro | Hmm.. is there a key sequence to force a certain codec per call on Polycoms? |
| 02:12.14 | [TK]D-Fender | quidpro: Is none, nor can it be set in the dialplan |
| 02:12.38 | quidpro | Damn... chalk up a measly one for Sipura. |
| 02:12.53 | Vec | Checkout http://www.pastebin.ca/394122 its my exten.conf and the cli of the call ? |
| 02:13.38 | [TK]D-Fender | Vec: exten => _21XX,n,Dial(SIP/${EXTEN}@${EXTEN},500) <-ummm |
| 02:13.41 | Vec | Corydon76-home : I think my callerid may be set to restricted on the PRI as I am not getting any CLI from the PRI ? |
| 02:13.45 | [TK]D-Fender | Vec: exten => _21XX,n,Dial(SIP/${EXTEN},500) <-better |
| 02:13.57 | ManxPower | quidpro: *nod*, many consumer ip phones allow you to do that |
| 02:14.34 | [TK]D-Fender | Vec: And why 500s? Thats psycho... |
| 02:15.12 | Vec | [TK]D-Fender : fixed that up thanks, 500 hehehe, just temprory what do u use ? |
| 02:15.14 | ltdwk | according to the docs, the ISDN can set the callerpres bits per call |
| 02:15.24 | flenders | I think sipuras/linksys are only good for their price |
| 02:15.32 | Vec | ltdwk : how can I unset callerpres ? |
| 02:15.39 | Corydon76-home | Vec: you don't. |
| 02:15.49 | flenders | and I have about 20 of those |
| 02:16.06 | ltdwk | So, from that I gather that if a call comes in on the ISDN, and callerpres is prohibited, the SIP handset won't see the caller ID no matter what you set it to |
| 02:16.09 | Corydon76-home | Vec: or you call the telco and ask them to strip the bits before they send calls to you |
| 02:16.18 | ltdwk | you need to SetCallerPres(allowed) |
| 02:16.47 | Vec | do I set that in exten.conf ? |
| 02:16.57 | ltdwk | before you dial the sip handset |
| 02:17.00 | ltdwk | try it and see if it works |
| 02:17.39 | InHisName | looking for stanaphone users with aterisk. |
| 02:17.44 | JunK-Y | Corydon76-home: hhehhahaha |
| 02:17.50 | Vec | Corydon76-home : have to give u the credit, spot on. |
| 02:17.56 | Vec | ltdwk : thanks |
| 02:17.59 | Vec | it worked |
| 02:18.11 | Vec | So I should also shout at my telco to give me callerid. |
| 02:18.22 | Vec | bastards :) |
| 02:18.36 | ltdwk | not hard... but costs a few bucks |
| 02:18.47 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-eef597ae9586345a) |
| 02:19.38 | ManxPower | What sort of evil stupid telco doesn't give callerid number on a PRI? |
| 02:20.04 | ltdwk | Telstra for one, by default. It's always an option |
| 02:20.08 | Vec | ManxPower : funny you should say that, go to www.telkom.co.za and www.hellkom.co.za |
| 02:20.11 | Corydon76-home | ManxPower: actually, they're providing it, just shielding the callerid by restricting the callerpres bits |
| 02:20.32 | ManxPower | ltdwk: it is more work for them to block it |
| 02:21.00 | Vec | Corydon76-home : so r u saying the callerID arrives on the asterisk box but asterisk ignores i because of the callerpres bits ? |
| 02:21.02 | ltdwk | manxpower: never the less, they charge a monthly fee for it |
| 02:21.06 | *** join/#asterisk `Tingles` (n=tingles@S01060050dac95c51.cg.shawcable.net) |
| 02:21.16 | Corydon76-home | Vec: possibly |
| 02:21.55 | Vec | I am not going to be a happy camper if they are going to want to charge me extra for callerid, considering what it already cost for a PRI. |
| 02:22.32 | `Tingles` | What can i go about checking all calls made through my digium card using the analog line, then going out through a sip provider the receipent can hear me perfectly however i do not hear them perfectly... i have eliminated teh sip provider as the other sip hw phones work fine using this setup... it only seems to have an issue when its coming through the card... any ideas? |
| 02:22.35 | `Tingles` | thigns to check ? |
| 02:23.40 | flenders | sip on analog lines? |
| 02:23.50 | *** join/#asterisk hohum (n=dcorbe@71.62.76.68) |
| 02:24.15 | `Tingles` | calling from a land line analog phone using the FXS port on the digium card which then dials out to the Real world using a SIP Provider... |
| 02:25.18 | flenders | are you registered with the provider? |
| 02:25.49 | `Tingles` | yes... and all the other phones about 12 in total are connected and work fine through that sip provider... |
| 02:25.58 | `Tingles` | i am thinking it has something to do with the digium card configuration |
| 02:28.19 | `Tingles` | any ideas? i know there are a bunch of flags etc.. in teh zaptel.conf file... thoe i am unsure which one would specifically be related to this kind of issue |
| 02:29.42 | Vec | If I want to Dial Zap, if error (congestion etc) Dial IAX, can I do that by just going Dial(ZAP) Hangup, Congestion,Dial(IAX). I would try figure this out on my own but I have not slept and have only an hour or so until the sun rises :) |
| 02:30.44 | ltdwk | you'll need a dialstatus macro-y thing |
| 02:31.07 | ltdwk | use the ${DIALSTATUS}, luke |
| 02:31.10 | Vec | ltdwk : ok saw one on voip-info, thanks, is that the way to do it |
| 02:31.17 | sumasuma | how can i detect simultaneous calls in an agi ? |
| 02:31.29 | sumasuma | how can i detect simultaneous calls in an agi, coming for from the same callerid ? |
| 02:32.34 | k-man | jt? |
| 02:32.38 | k-man | are you around? |
| 02:34.55 | `Tingles` | anyone have any ideas? |
| 02:36.14 | ltdwk | have you tried an echo test on the fxs line ? |
| 02:37.15 | ltdwk | that would be my first move to see where the problem lies |
| 02:37.47 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 02:40.15 | *** join/#asterisk kavit (n=kavit@ppp245-70.static.internode.on.net) |
| 02:40.24 | `Tingles` | how would i go about that... sorry first time playing with the card... |
| 02:41.49 | kavit | is anyone having issues connecting to voip-info.org ? |
| 02:43.23 | quidpro | kavit: it's down |
| 02:43.35 | quidpro | been so since atleast this morning EDT |
| 02:44.14 | kavit | ah, I feel crippled of sorts |
| 02:44.16 | kavit | :( |
| 02:44.16 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 02:47.08 | ManxPower | learn to use /path/to/src/asterisk/doc and /path/to/src/asterisk/configs and "show applications" and "show functions" |
| 02:47.36 | k-man | anyone here used oztell? |
| 02:47.45 | k-man | they seem quite good looking at their website |
| 02:47.56 | ManxPower | Also learn to add "site:lists.digium.com" to your google.com searches |
| 02:49.37 | kavit | ManxPower: I am looking for region specific feedback from users.. something /path/to/docs/will/not/have |
| 02:49.54 | kavit | but lists.digium.com helps |
| 02:51.41 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 02:52.54 | *** join/#asterisk Strom_M (n=strom@12-189-87-2.att-inc.com) |
| 02:53.31 | quidpro | WHat is it with these voip sites being down.. voip-info is down aussievoip is down (for FreePBX users). |
| 02:53.40 | *** join/#asterisk monkie (n=lockdown@69-163-133-132.atlsfl.adelphia.net) |
| 02:54.05 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:54.13 | orlock | quidpro: galaxy-solutions is down |
| 02:54.20 | orlock | due to teir upstream being down |
| 02:58.28 | *** join/#asterisk shodan- (n=shodan@ip059.96-113-216.pppoe1.joliette.intermonde.net) |
| 03:00.12 | k-man | i can't work out how to choose a voip provider |
| 03:00.15 | k-man | its too hard |
| 03:00.19 | k-man | it makes my head spin |
| 03:01.46 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE000f66bd2bf3-CM000039529869.cpe.net.cable.rogers.com) |
| 03:01.55 | ttuttle | JunK-Y: Vitelity. |
| 03:01.59 | ttuttle | JunK-Y: whoops |
| 03:02.02 | ttuttle | k-man: Vitelity. |
| 03:02.08 | flenders | k-man: I'm with engin and iinet |
| 03:02.19 | flenders | k-man: and they're both pretty good |
| 03:02.19 | k-man | vitelity? |
| 03:02.30 | k-man | flenders, ok, thats good to know |
| 03:02.43 | k-man | flenders, do either of them provide a voicemail function (if your not usint asterisk) |
| 03:02.57 | flenders | k-man: been with engin for about 2 years, and got another account with iinet 2 months ago |
| 03:03.09 | flenders | k-man: yeah, both do |
| 03:03.31 | flenders | k-man: never used iinet's though. I know it's there |
| 03:03.36 | flenders | but disabled it |
| 03:03.47 | flenders | I've used engin's before |
| 03:04.04 | k-man | ok |
| 03:05.27 | S2AnGeL | any one recommend any good software for seeing who is on what line and who is parked |
| 03:05.31 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 03:05.51 | S2AnGeL | my flash pannel is not working.. |
| 03:15.04 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 03:15.31 | ManxPower | S2AnGeL: FOP is about all there is |
| 03:15.50 | ManxPower | k-man: All VoIP providers SUCK. Teliax seems to suck less than most. |
| 03:15.59 | k-man | hehe |
| 03:16.33 | ManxPower | VoIP providers suckage seems to vary from month to month. |
| 03:16.34 | k-man | ManxPower, i need an australian one |
| 03:16.41 | k-man | but thanks |
| 03:16.48 | k-man | i'm going to go with my net phone |
| 03:18.55 | ManxPower | Whatever you do don't publish your VoIP number until you've had it for a month or three |
| 03:19.26 | flenders | k-man: http://www.mynetfone.com.au/money/ |
| 03:19.33 | orlock | k-man: Nexvoice |
| 03:19.40 | orlock | k-man: or Engin |
| 03:19.44 | orlock | or.. there are others |
| 03:19.51 | orlock | Nexvoice dont sell directly i think |
| 03:20.10 | k-man | flenders, cool |
| 03:21.43 | S2AnGeL | ManxPower:no there are lots.. jus tthe one I been using has turned into a pay one.. and no longer offer it no more.. and the voip wiki seems to be down from where I am and unable to see what else there is aavilable for asterisk |
| 03:21.52 | kavit | I run a voip provider and I dont suck :( |
| 03:21.57 | bkruse_home | lol |
| 03:22.04 | bkruse_home | I R THE NEXT VONAGE! |
| 03:22.10 | S2AnGeL | kavitwhere are you located? |
| 03:22.19 | kavit | S2AnGeL: Sydney Australia |
| 03:22.29 | kavit | we only provision businesses though |
| 03:22.30 | S2AnGeL | oh.. need something in canada |
| 03:22.49 | bkruse_home | S2AnGeL: move to the US? |
| 03:23.01 | S2AnGeL | I am quite happy where I am |
| 03:23.22 | kavit | canada has free medicine for one and all right? |
| 03:23.31 | S2AnGeL | yah |
| 03:23.45 | flenders | kavit: can I have a look at your call rates? |
| 03:23.47 | bkruse_home | US has, a dollar. |
| 03:23.51 | kavit | much like australia...although it is highly discounted here |
| 03:24.04 | S2AnGeL | lotsa stuff we have.. very simular to the USA .. close ly follow them.. though less restrictive.. |
| 03:24.04 | kavit | flenders: we are not the cheapest. |
| 03:24.19 | kavit | flenders: i will priv message you the link |
| 03:25.10 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 03:25.23 | S2AnGeL | so know one knows any good voip providers in canada |
| 03:25.53 | kavit | flenders: we are integrators before providers.... provision started because I was sick of deploying VoIP solutions and dealing with issues with provisioning... so I thought why not start provisioning myself... painless experience for my customers |
| 03:26.01 | kavit | one person to blame |
| 03:26.20 | flenders | kavit: thanks |
| 03:26.40 | orlock | kavit: WHich provider? |
| 03:26.42 | flenders | kavit: we have 4 voip accounts here in the office |
| 03:26.45 | S2AnGeL | is it true linux use is big over in austraila |
| 03:26.57 | kavit | so we prefer provisioning to people who buy pbxes off us or get us to make them a solution |
| 03:27.03 | orlock | S2AnGeL: yeah |
| 03:27.11 | orlock | well, as big as it is anywhere i guess |
| 03:27.17 | kavit | orlock: Resonance Networks ----------> me |
| 03:27.51 | orlock | eurgh |
| 03:27.54 | kavit | orlock: the website isnt upto date with the stuff we do... too busy to update webpages |
| 03:27.55 | orlock | need to patch some rpm's |
| 03:28.01 | bkruse_home | dang, voip-info is still down... |
| 03:28.11 | orlock | yup |
| 03:28.15 | flenders | kavit: mate, you're cheaper than iinet |
| 03:28.17 | S2AnGeL | thats great.. there is one of the places I want to vist.. I hear you got great history and weather.. .. if you ignor all the serious dangerous animals and stuff it sounds like a awesome place |
| 03:28.17 | orlock | cos of a fucked host/colo site |
| 03:28.20 | orlock | sixty days eta |
| 03:29.05 | kavit | flenders: yeah.... we have our own soft switches and infrastructure, we sell rack space and VERY CHEAP PIPE bandwidth |
| 03:29.06 | eald_home | wiki is down? |
| 03:29.07 | S2AnGeL | voip-info yah is it in austraila ? cause freepbx has a wiki thats hosted thats down too its like a bad month for voip because of that mess |
| 03:29.26 | orlock | S2AnGeL: nope, in the states |
| 03:29.30 | eald_home | yeah voip-info |
| 03:29.33 | orlock | check the topic in #freepbx |
| 03:29.48 | S2AnGeL | yah I read it all.. 60 days frig |
| 03:30.09 | S2AnGeL | I hear there is a 30 day old backup so there is hope |
| 03:30.56 | S2AnGeL | even googles cache is not seem to be working.. |
| 03:31.29 | kavit | flenders: how do you find iinet, quality wise? |
| 03:31.33 | flenders | kavit: the web management interface you sell with your servers, is it developed in-house? or you're just reselling someone else's? |
| 03:31.40 | kavit | flenders: I dont like engin.... |
| 03:31.57 | flenders | kavit: iinet is alright... |
| 03:32.08 | orlock | kavit: heard of nexvoice? |
| 03:32.10 | flenders | both sound better than our analog lines, anyway |
| 03:33.01 | kavit | flenders: we provide freepbx with the normal stuff but usually write custom modules for people.... also do stuff with vanilla asterisk and ajax but thats all proprietory and owned by customers, I keep meaning to make open source versions |
| 03:33.24 | kavit | orlock: didnt you tell me about the ADSL pci modem a few weeks ago? |
| 03:34.12 | flenders | kavit: so, for that price, your customers own the server? or you're just leasing? |
| 03:34.25 | kavit | flenders: own it... |
| 03:34.47 | flenders | flenders: good price then |
| 03:35.20 | kavit | yeah... we consult to Corporate Express so they give us mates rates on heavy hardware |
| 03:35.36 | orlock | kavit: yeah, probably |
| 03:35.58 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1167878982.dsl.bell.ca) |
| 03:36.17 | kavit | flenders: I dont like most people who sell pbx hardware... it is shit... just some vanilla box with no reliability.... |
| 03:36.21 | orlock | kavit: actually ust patching some rpm's for centos so it supports it nicely now |
| 03:36.32 | kavit | orlock: dont you work for an ISP as well |
| 03:36.46 | flenders | kavit: I know! |
| 03:36.50 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 03:37.14 | flenders | kavit: all the ones we saw, a few were pretty on the outside, but inside was the same "chinatown" box. |
| 03:37.19 | kavit | orlock: nice.... i have brought up the subject with my peers.... i would love to bundle |
| 03:37.30 | flenders | kavit: that's why we ended up doing it ourselves |
| 03:37.30 | orlock | yeah |
| 03:37.31 | kavit | flenders: we sell HP or SUN server hardware only.... |
| 03:37.44 | flenders | sun running linux, I'd guess |
| 03:37.52 | kavit | flenders: thats the best way to do it.... but not all people are tech savvy |
| 03:37.55 | flenders | actually, does asterisk run on solaris? |
| 03:37.57 | orlock | kavit: the new adsl2 cards tey are making are just going to be an ADSL2 modem on an ethernet card basically |
| 03:37.59 | kavit | flenders: yeh... opterons |
| 03:38.20 | kavit | orlock: sounds nifty....i wouldnt mind trialling them out |
| 03:38.34 | kavit | orlock: i can't buy bulk... too botique for my own good |
| 03:38.40 | kavit | :( |
| 03:38.43 | flenders | kavit: are you planning on selling asterisk support too? |
| 03:38.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:39.02 | kavit | flenders: yeah, we already do... just not up on the site |
| 03:39.37 | kavit | flenders: i need to get one of the guys here to get rid of joomla and use the template i wrote and update stuff |
| 03:39.45 | kavit | joomla is good but google doesnt like it... |
| 03:40.15 | kavit | flenders: start from $100 per hour on site.... complexity of the job decides the rates |
| 03:40.34 | kavit | flenders: syd and melb..... |
| 03:41.17 | orlock | kavit: hey, any suggestions on getting rid of echo with pure sip calls? |
| 03:41.29 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 03:41.29 | orlock | grr |
| 03:41.54 | orlock | this is the second pair of jeans that the metal edging has ripped the ass out of |
| 03:42.15 | kavit | orlock: haven't really come across that issue |
| 03:42.18 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:43.41 | flenders | kavit: I might keep your details, as I'll be going overseas in a few weeks, just in case my manager needs support |
| 03:43.59 | kavit | flenders: no worries, always glad to help |
| 03:44.11 | flenders | kavit: can you msg me your e-mail? |
| 03:44.19 | kavit | flenders: sure |
| 03:45.21 | flenders | kavit: cheers |
| 03:46.17 | kavit | no wuckers matey |
| 03:46.28 | kavit | haha people crack up every time I say that |
| 03:46.34 | kavit | with my very indian accent |
| 03:47.23 | flenders | :D |
| 03:47.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:48.13 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
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| 03:51.53 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 03:52.08 | eald_home | well, then I'll ask here, I need to create a windows fronted for Asterisk Manger API (just some commands) and I need to be very careful with parsing stuff, anybody know a open source program that reliably interacts with Asterisk Manager API? |
| 03:52.56 | *** join/#asterisk bmg505 (n=leon@196.209.248.114) |
| 03:54.54 | JT | k-man: am now |
| 03:54.57 | *** join/#asterisk topping (n=topping@h460783aa.area7.spcsdns.net) |
| 03:55.05 | flenders | JT: too late! |
| 03:55.07 | flenders | :D |
| 03:55.23 | k-man | jt, oh, i was going to ask you about voip providers.. but i signed up with my net fone already |
| 03:55.34 | *** join/#asterisk stridernzl (n=neville@222-155-160-194.jetstream.xtra.co.nz) |
| 03:55.52 | flenders | JT: you don't use voip, do you? |
| 03:56.28 | kavit | oh yeah is shared extensions a goer with asterisk 1.4? |
| 03:56.37 | kavit | I havent played around with 1.4 at all |
| 03:57.14 | hohum | FAIRFIELD CO (Reuters) - Police raided the Stamford Connecticut home of 48 year old Chris Hansen this afternoon. Police have not yet released any information regarding the circumstances of his arrest other than to say that he is being held without bond in a Fairfield County jail. Chris Hansen is best known for his work with Dateline NBC's "How to catch a predator" whose investigations revolve around catching internet sex predators using under cover sting op |
| 03:57.14 | hohum | erations. |
| 03:58.11 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 03:58.55 | JT | flenders: i do |
| 03:59.00 | JT | i use engin |
| 03:59.29 | kavit | JT: how have you found the *.byo.engin servers? |
| 03:59.58 | kavit | JT: a client had a LOT of issues over the last 2 weeks.... |
| 04:00.21 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 04:00.22 | bkruse_home | does a directory have to be empty to have a checkout? |
| 04:00.23 | Mahmoud | hello |
| 04:00.26 | bkruse_home | svn, checkout that is |
| 04:01.09 | Mahmoud | any idea why the call hangs up without asking to recored a voice mail when the person is not available? I'm using macros |
| 04:01.10 | Mahmoud | exten => s,n,voicemail(u${EXTEN}@family) |
| 04:01.48 | Mahmoud | this line is in the macro, if i don't use the actual mail box number it works fine, but doesn't work with ${EXTEN} |
| 04:02.55 | Mahmoud | for example, exten => s,n,voicemail(u000@family) works, where 000 is the mail box number |
| 04:03.52 | ltdwk | show us the output from the CLI |
| 04:04.54 | JT | kavit: past 2 weeks have been fine, there was issues about 3 weeks ago i think |
| 04:05.12 | Mahmoud | hmmm |
| 04:05.12 | JT | kavit: anyone who relies only on VoIP over Internet for business purposes is asking for trouble |
| 04:05.19 | JT | I have ISDN lines too |
| 04:05.39 | bkruse_home | JT: always have backups ;] |
| 04:05.49 | Mahmoud | ltdwk: -- Executing [s@macro-internal:2] VoiceMail("SIP/dawoud-086eb000", "us@family") in new stack |
| 04:05.50 | Mahmoud | [Mar 14 08:02:16] WARNING[36100]: app_voicemail.c:2854 leave_voicemail: No entry in voicemail config file for '' |
| 04:05.51 | JT | yeah |
| 04:05.57 | kavit | JT: correct |
| 04:06.01 | ltdwk | Mah: see a problem there? |
| 04:06.06 | ManxPower | Mahmoud: EXTEN is "s" at that point |
| 04:06.16 | DocHolliday | who do people use here for Canadian call termination? |
| 04:06.37 | ltdwk | Mah: your macro is obviously not being passed the extention in the way you are intending |
| 04:06.49 | ManxPower | What I do is in many of my macros I do exten => s,1,Goto(${MACRO_EXTEN},1) then have exten => X.,1,start of real macro |
| 04:07.09 | Mahmoud | hmmm ltdwk and ManxPower, how to pass the extention? |
| 04:07.20 | ltdwk | Manx just gave a good solution |
| 04:07.22 | Mahmoud | I'm using this: VoiceMail(u${EXTEN}@family) |
| 04:07.25 | ManxPower | Mahmoud: my way is the easiest. There are many other ways |
| 04:07.28 | ltdwk | or you could use MACRO_EXTEN |
| 04:07.55 | ManxPower | ltdwk: In theory my way makes the CDRs look correct. |
| 04:08.10 | `Tingles` | how do i turn up the incoming volume on the FXS port on teh digium card... |
| 04:08.11 | ltdwk | Manx: Aye, that can't be a bad thing |
| 04:08.52 | ManxPower | `Tingles`: see rxgain |
| 04:08.52 | `Tingles` | what conf file? |
| 04:08.52 | `Tingles` | zAPTEL? |
| 04:08.52 | `Tingles` | <PROTECTED> |
| 04:08.57 | ManxPower | `Tingles`: all Digium card stuff is in /etc/zaptel.conf or /etc/asterisk/zapata.conf |
| 04:08.58 | Mahmoud | ManxPower, oh, ${EXTEN} is convertd to letter "s".. i see what you mean |
| 04:09.07 | ManxPower | Mahmoud: No, EXTEN IS "s" |
| 04:09.23 | ManxPower | If exten was not "s" then exten => s,1,whatever would not be run |
| 04:09.28 | Mahmoud | ManxPower, hmmm the macro resuls in us@family, where "s" used to be ${EXTEN} |
| 04:09.55 | ltdwk | Mah: Inside the macro, "s" is the extension looked up, hence why MACRO_EXTEN exists |
| 04:10.08 | Mahmoud | ManxPower, i mean that ${EXTEN} often holds the dialed extention, but within the macro, it holds the letter "s".. right? |
| 04:10.33 | ManxPower | Mahmoud: That is incorrect. EXTEN does not hold the dialed extension. EXTEN holds the currently executing extension |
| 04:11.00 | Mahmoud | ManxPower, so what should I place in voicemail(u????@family)? |
| 04:11.14 | ManxPower | Much of the time the EXTEN and the dialed extension are the same, many times they are not. Goto, Gosub, Macros, all change the value of EXTEN |
| 04:11.16 | ltdwk | Mahmoud: Voicemail(u${MACRO_EXTEN}@family) |
| 04:11.32 | Mahmoud | ltdwk, yeah, but he said that his way is easier, wanna hear it |
| 04:11.44 | ltdwk | not easier, just more friendly to the CDR |
| 04:11.50 | ManxPower | Mahmoud: perhaps you missed this: (23:06:40) ManxPower: What I do is in many of my macros I do exten => s,1,Goto(${MACRO_EXTEN},1) then have exten => X.,1,start of real macro |
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| 04:12.29 | ltdwk | What Manx does, is uses Goto() to CHANGE the _currently_ executuing extension in the current context |
| 04:12.45 | ManxPower | and the current context is the macro |
| 04:12.53 | ManxPower | macros are really just special case contexts |
| 04:13.08 | ltdwk | then you match the extension using _X. and ${EXTEN} will be set correctly |
| 04:14.16 | ManxPower | ltdwk: Actually I usually match on _XXXX,1,Whatever, but that is not relevant to this discussion |
| 04:14.19 | DocHolliday | if i am using the background command is it possible to allow the incoming caller to enter an extension before the sound file finishes? |
| 04:14.38 | ManxPower | DocHolliday: Yes. That is the entire reason to use Background. That is the reason background exists. |
| 04:14.49 | ltdwk | ManxPower: I usually use _X. as it gets everything |
| 04:14.57 | DocHolliday | ManxPower, for some reason when i attempt to dial during the IVR nothing happens |
| 04:15.28 | ltdwk | have you set DIGIT_TIMEOUT ? |
| 04:15.31 | ManxPower | DocHolliday: then you have either a DTMF problem (this is usually the case) or you have no matching exten => line in that context to match the DTMF |
| 04:15.33 | Mahmoud | ManxPower & ltdwk: what about, just passing one extra argument to the macro: voicemail(u${ARG2}@family) |
| 04:15.46 | ltdwk | Mahmoud: the extension is already passed as an argument |
| 04:15.52 | ManxPower | Mahmoud: because it is more confusing that using MACRO_EXTEN |
| 04:16.04 | DocHolliday | ManxPower, if i am able to dial the exten after the IVR does that indicate that it is not a DTMF problem? |
| 04:16.07 | Mahmoud | ltdwk, oh, right, then i'll use ${ARG1} |
| 04:16.54 | ManxPower | DocHolliday: maybe, but not always. For example G729 might make DTMF work %80 of the time when the channel is silent, but %10 of the time when there is audio on the channel |
| 04:17.07 | ManxPower | I'm talking about inband DTMF with G729, of course. |
| 04:17.11 | ltdwk | DocHolliday: make sure you set these... TIMEOUT(digit) etc |
| 04:17.29 | DocHolliday | ltdwk, ahh i hadn't decided whether that was necessary or not :) |
| 04:17.32 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:17.47 | ManxPower | Mahmoud: You need to exit from extensions.conf and read README.variables |
| 04:17.59 | ManxPower | DocHolliday: it isn't if you have a well designed dialplan |
| 04:18.17 | DocHolliday | ManxPower, want to see my dialplan and tell me how crappy it is? |
| 04:18.24 | ping2921 | can anyone tell me where I can find a softphone that has multiple hold lines (other than xlite) ? |
| 04:18.28 | ManxPower | DocHolliday: not in a million years |
| 04:18.40 | DocHolliday | personally i dont think its that ugly :P |
| 04:19.06 | ManxPower | I don't use softphones |
| 04:19.15 | Mahmoud | ManxPower, perhaps you mean, doc/channelvariables.txt? |
| 04:19.32 | ManxPower | Mahmoud: not unless they changed the name of the file in 1.4 |
| 04:20.03 | ManxPower | if you want to see ugly: http://www.fnords.org/~eric/macros.inc |
| 04:20.05 | Mahmoud | they changed doc files names |
| 04:21.16 | DocHolliday | ManxPower, i added those two lines same problem, what dtmf mode should i be setting for my inbound line? |
| 04:21.28 | ManxPower | DocHolliday: what codec are you using? |
| 04:21.41 | ManxPower | What SIP device are you using? |
| 04:21.58 | ManxPower | Is the call coming in on a VoIP carrier? If so, what carrier? |
| 04:22.21 | DocHolliday | ManxPower, DIDWW (carrier), dialing in to the IVR using my cell phone, phones at our office are Cisco 7941Gs |
| 04:23.00 | ManxPower | DocHolliday: USUALLY you would want dtmfmode=rfc2833 in sip.conf |
| 04:23.12 | DocHolliday | if it isnt there will it default to that? |
| 04:23.55 | ltdwk | default is inband i think? |
| 04:23.57 | ManxPower | Some carriers (especially Level 3) [don't count on ANYTHING defaulting] never have reliable DTMF with Asterisk unless you are using ulaw/alaw as your codec and inband as your DTMF mode |
| 04:24.09 | ManxPower | And MANY providers use Level 3 as their carrier |
| 04:24.18 | DocHolliday | ManxPower, same problem |
| 04:24.31 | DocHolliday | ManxPower, mind if i show you my IVR? |
| 04:25.00 | ManxPower | DocHolliday: You are welcome to show it to me, but I won't look at it tonight |
| 04:25.18 | Mahmoud | ManxPower, the reason why you used goto is because exten => X. will save the actual dialed extention in ${EXTEN}, while macros "s" don't? |
| 04:25.20 | ManxPower | DocHolliday: use "sip debug" to see if you are getting the DTMF from the carrier to start out with |
| 04:25.21 | DocHolliday | yeah, im just wondering what the possibilities are, the other issue is that im losing the first 1/2 second of my IVR |
| 04:25.43 | DocHolliday | ManxPower, its coming in over IAX |
| 04:25.44 | *** join/#asterisk Brime (n=b@207-118-116-39.dyn.centurytel.net) |
| 04:26.03 | ManxPower | DocHolliday: do an Answer then a Wait(.5). You sometimes have to do that to let the audio path settle down. |
| 04:26.14 | ManxPower | DocHolliday: iax should not have DTMF problems from a carrier. |
| 04:26.15 | DocHolliday | im doing a wait(1) same issue |
| 04:26.27 | ManxPower | DocHolliday: you have to do an answer before the wait |
| 04:26.44 | ManxPower | no point in waiting for the audio path to settle down if you don't have the audio path yet |
| 04:28.37 | DocHolliday | ManxPower, tried that.. it cuts off literally the first 2 letters of the first word |
| 04:28.54 | ManxPower | DocHolliday: so you do an Answer then a Wait? |
| 04:29.16 | kavit | DocHolliday: without Answer there is no where to go :( |
| 04:29.17 | DocHolliday | already doing that :) |
| 04:29.35 | DocHolliday | kavit, it picks up :P the IVR is just skewed in the first few letters |
| 04:29.52 | Mahmoud | can we use macros in voicemail.conf? |
| 04:29.54 | ManxPower | DocHolliday: You must do the answer before the WAIT or it won't work. |
| 04:29.57 | ManxPower | Mahmoud: no |
| 04:29.57 | Qwell | Mahmoud: no |
| 04:30.17 | Mahmoud | so, i have to mention all the fancy options for every one :/ |
| 04:30.17 | ManxPower | Mahmoud: you are not listening. EXTEN does not contain the dialed number. |
| 04:30.37 | DocHolliday | same problem still ManxPower |
| 04:30.37 | Mahmoud | ManxPower, yeah, it just contains the actual extention being executed in dialplan |
| 04:30.45 | ManxPower | EXTEN contains the currently executing extension. This is sometimes the dialed number, but frequently is NOT |
| 04:30.53 | Mahmoud | yeah, i got that |
| 04:30.56 | ManxPower | DocHolliday: I have no more ideas |
| 04:31.17 | DocHolliday | yeah |
| 04:31.24 | ManxPower | so you are guarnteed that EXTEN will be whatever is after the "exten => " part and before the "," |
| 04:31.31 | Mahmoud | ManxPower, what about when using .X, ${EXTEN} will be the number instead of ".X", right? |
| 04:31.41 | DocHolliday | ManxPower, any other ideas regarding not being able to dial during the IVR? |
| 04:31.57 | ManxPower | Mahmoud: in the case of extensions prefixed with _ it will be whatever value matched _ |
| 04:32.18 | ManxPower | DocHolliday: not when you are using IAX. IAX only has 1 DTMF mode. |
| 04:32.32 | Qwell | 9ugget: You lose. |
| 04:32.34 | ManxPower | do an "iax2 debug" to make sure you are actually GETTING the DTMF from the carrier |
| 04:32.35 | Mahmoud | ManxPower, that's why you wanted to use another wildcard extention as the real macro, so have the dialed extention |
| 04:32.37 | Nugget | d'oh |
| 04:32.57 | ManxPower | Mahmoud: no the reason I did that is so that the CDR records will look correct. There is no other reason |
| 04:33.23 | Mahmoud | ManxPower, I can't use "hint" in macros, should I be using wild card extentions? |
| 04:33.45 | ManxPower | Nugget: shouldn't you be _Nugget. That you can be all things that start with Nugget |
| 04:34.05 | Qwell | _[a-z]ugget. |
| 04:34.13 | ManxPower | Mahmoud: I can think of no valid reason to use the hint priority in a macro |
| 04:34.34 | DocHolliday | ManxPower, i dont see a reference to DTMF |
| 04:34.35 | ManxPower | Qwell: not _[Nn]ugget. |
| 04:35.16 | Mahmoud | ManxPower, i use hint just to provide status information to users.. |
| 04:35.16 | ltdwk | kinda wondering how we went from voicemail to hints |
| 04:35.17 | ManxPower | DocHolliday: then your carrier is fucked up |
| 04:35.17 | DocHolliday | the carrier is DIDWW |
| 04:35.17 | ManxPower | ltdwk: Mahmoud is one of our Special Asterisk users |
| 04:35.44 | ManxPower | DocHolliday: I AM out of practice debugging DTMF issues since I never have any DTMF issues. |
| 04:35.48 | ltdwk | Manx: Special indeed |
| 04:35.51 | DocHolliday | heh |
| 04:35.55 | ManxPower | On the other hand, I never use VoIP carriers either. |
| 04:36.16 | DocHolliday | exten => s,n,Playback(/var/lib/asterisk/sounds/asterisk-recording11) |
| 04:36.16 | DocHolliday | exten => 0,1,Dial(SIP/240&SIP/250&SIP/260|20) |
| 04:36.16 | DocHolliday | <PROTECTED> |
| 04:37.20 | Qwell | You can't dtmf in playback |
| 04:37.24 | kavit | DocHolliday: playback |
| 04:37.28 | kavit | DocHolliday: :( |
| 04:37.36 | DocHolliday | ohh shit i thought i changed that |
| 04:37.42 | DocHolliday | ROFL |
| 04:37.49 | Mahmoud | heh |
| 04:38.30 | DocHolliday | *siren* human error |
| 04:38.47 | kavit | DocHolliday: its called PEBKAC |
| 04:38.54 | ManxPower | There is 20 mins of my life I will never get back. |
| 04:39.02 | ltdwk | rofl |
| 04:39.07 | DocHolliday | ManxPower, i accidentally changed the wrong context from playback to background LOL |
| 04:39.40 | ManxPower | DocHolliday: Things frequently "don't work" at 11pm that do at 8am the next morning. |
| 04:40.00 | DocHolliday | yeah you're telling me.. the problem is i dont like sleeping till things work nicely |
| 04:40.39 | ManxPower | DocHolliday: I have some good tranquilizers that help with that |
| 04:40.40 | DocHolliday | heh i have a dual processor box dedicated to 4 devices and 2 soft phones |
| 04:41.01 | kavit | DocHolliday: overkill dude |
| 04:41.16 | DocHolliday | kavit, sure.. but i already had the machine. |
| 04:41.39 | ManxPower | Looks like I may need them tonight. one of our PBXs is not coming back from a DST update. |
| 04:41.44 | kavit | DocHolliday: use it for something else.... why waste power |
| 04:41.47 | ManxPower | Of course the PBX is 400 miles away |
| 04:41.53 | DocHolliday | kavit, nothing to use it for :) |
| 04:42.00 | ManxPower | YAY! It came back. |
| 04:42.09 | kavit | ManxPower: what type of high availability do you have? |
| 04:42.11 | DocHolliday | ManxPower, work for a VoIP provider? |
| 04:42.39 | ManxPower | DocHolliday: heck no. Consultant for corporations |
| 04:42.59 | ManxPower | kavit: no customer has wanted to pay for "high availability" |
| 04:43.33 | kavit | ManxPower: whats your best approach to get asterisk into corporations.... they are always adamant on something else |
| 04:43.44 | ManxPower | kavit: most of the time it is politics. |
| 04:43.45 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 04:43.53 | kavit | ManxPower: I know... i hate cisco sales people |
| 04:44.13 | DocHolliday | ManxPower, politics where i work consisted of throw everything we own out and move to a new office and let Doc spend a few times the IT budget |
| 04:44.21 | ManxPower | kavit: The shortest time any of my customers has been a customer is 5 years. |
| 04:44.44 | kavit | ManxPower: yeah.... I am looking to go the application development route |
| 04:44.45 | ManxPower | So when I tell my customers something they usually listen. |
| 04:45.07 | orlock | no new customers in 5 years? |
| 04:45.07 | ManxPower | If they do not listen too often then they are soon a Former Customer |
| 04:45.15 | ManxPower | orlock: no need for new customers. |
| 04:45.16 | clyrrad | Hey guys... strange issue... I have a script that detects if asterisk is running, if not it runs /etc/init.d/asterisk start - when this script runs (automatically from cron) the console looses its colors - yet if you do that same command /etc/init.d/asterisk start - the console does NOT loose its colors, any idea why this happens? I been battling this for days now.... |
| 04:45.27 | kavit | ManxPower: make useful stuff around asterisk and sell it for a fraction of what people like cisco sell it for |
| 04:46.05 | Qwell | clyrrad: probably because cron has uses a term without support for color |
| 04:46.19 | clyrrad | Qwell: ah.... |
| 04:46.25 | clyrrad | any way around this? |
| 04:46.38 | Qwell | could try setting one |
| 04:46.52 | ManxPower | I MAY have a new customer soon. I know the person, I know the company, their requirements are a good match for my skills. |
| 04:47.00 | clyrrad | Qwell: how so? |
| 04:47.09 | Qwell | export TERM=linux |
| 04:47.11 | Qwell | or something |
| 04:47.16 | Qwell | ~ymmv |
| 04:47.18 | jbot | ymmv is, like, Your Mileage May Vary |
| 04:47.34 | clyrrad | Qwell: are you suggesting to put that in the crontab? |
| 04:47.42 | kavit | ManxPower: so if you had to give your customer high availability... what would you recommend? dont say sipxpbx.... i dont mind it but I had issues with it |
| 04:47.45 | Qwell | sure, try it |
| 04:47.46 | DocHolliday | ManxPower, skillz are always important. |
| 04:47.59 | ManxPower | kavit: I don't know. I would have to spend a month researching it |
| 04:48.12 | clyrrad | Qwell: thanks - I been wondering why it behaves like that |
| 04:48.20 | clyrrad | gonna try in the crontab |
| 04:48.21 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 04:48.26 | kavit | ManxPower: ah... I dont like relying too much on DNS SRV records.... |
| 04:48.39 | nDuff | hrm. |
| 04:48.52 | ManxPower | If they wanted to failover and not drop calls, I would suggest they find different consultant for the project. |
| 04:49.11 | ltdwk | that kind of failover is tough |
| 04:49.11 | kavit | DocHolliday: what would you use to cluster asterisk |
| 04:49.14 | DocHolliday | ManxPower, yeah i am yet to see a 'no drop' solution |
| 04:49.21 | DocHolliday | kavit, the cisco 28xx :) |
| 04:49.33 | litage | why would asterisk be writing to /var/log/asterisk/messages despite there not being an entry for "messages" in logger.conf?: http://rafb.net/p/gTrixe11.html |
| 04:49.35 | kavit | DocHolliday: is it a load balancing swtch? |
| 04:49.43 | JT | no drop solutions exist. |
| 04:49.53 | ltdwk | show us |
| 04:49.56 | DocHolliday | kavit, its not a 'load balance switch' but it can be used for load balancing :) |
| 04:49.56 | nDuff | curiously, zttool shows TxD as being all high, while "wanpipemon -i w2g1 -c dpr" shows it as low. |
| 04:49.58 | JT | you must spend money close to in the millions |
| 04:50.05 | JT | for telco grade parallel computing gear |
| 04:50.08 | ManxPower | Other than that you have to handle getting the PRI handled by the second PBX, you would have to deal with getting the phones to talk to the 2nd PBX, voicemail mirroring, and I'm sure 6 things I can't think of right now |
| 04:50.18 | kavit | I am thinking of writing a front end for it.... |
| 04:50.23 | JT | none exist for cheap |
| 04:50.31 | JT | they all involve losing active calls |
| 04:50.43 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:50.51 | DocHolliday | ManxPower, thats why you terminate the PRI on a cisco router and then feed the calls to your asterisk machines |
| 04:50.53 | JT | the best that can be done for cheap involves quick takeover of affected lines to accept new calls |
| 04:51.00 | kavit | JT: yeah even cisco who charge through the nose lose active calls\ |
| 04:51.04 | DocHolliday | if one of your machines dies its not a big deal :) |
| 04:51.11 | AJaymn | where are the sound files stored? |
| 04:51.12 | ltdwk | then the cisco is the point of failure |
| 04:51.14 | JT | err yeah the point of faliure is the cisco then. |
| 04:51.16 | ltdwk | how is that redundant? |
| 04:51.19 | ManxPower | DocHolliday: that is still a single point of failure |
| 04:51.25 | JT | that's not an L1 failover solution |
| 04:51.41 | DocHolliday | ManxPower, true but a cisco router fails a maybe 1% as much as an Asterisk box (i'd wager) |
| 04:51.48 | ManxPower | Honestly, I am not the consultant to handle real, carrier grade failover. |
| 04:52.08 | kavit | DocHolliday: there are Asterisk boxes and then there are asterisk boxes |
| 04:52.17 | ManxPower | fake cheap failover would be a lot of work, but something I could do. |
| 04:52.21 | DocHolliday | true |
| 04:52.26 | kavit | DocHolliday: CallManager has had its fair share of bugs and so has Cisco IPCC |
| 04:52.29 | JT | you can get reasonable failover with asterisk |
| 04:52.38 | DocHolliday | kavit, we aren't taking about CCM |
| 04:52.42 | JT | 2 servers, load balancer, L1 failover switch for physical circuits |
| 04:53.00 | kavit | DocHolliday: fair enough |
| 04:53.19 | JT | cisco gear is nowhere near telco grade, especially their voice stuff |
| 04:53.31 | DocHolliday | every platform has its bugs.. my idea is just to minimize the amount of times the failures cause problems |
| 04:53.41 | ManxPower | JT: I *heart* 2621s and Cat 5500s |
| 04:53.49 | kavit | ManxPower: i think a better solution might be hot stand by where all active calls are lost but the down time is miniscle.... |
| 04:53.56 | JT | ManxPower: sure, just don't tell me they're telco grade :P |
| 04:54.10 | ManxPower | JT: never even crossed my mind. |
| 04:54.26 | DocHolliday | JT, well i'd rather hear a call on a Cisco 28xx then asterisk, DSPs sound wayy better |
| 04:54.31 | AJaymn | HELP ---------- where are the sound files stored? |
| 04:54.48 | ManxPower | We have done some work with our telco to have calls to a down PRI be sent to a different office's PRI, the we route them back to the correct asterisk server |
| 04:54.56 | k-man | can someone send me a test email to jason@organictrader.com.au please? |
| 04:54.57 | DocHolliday | <PROTECTED> |
| 04:55.01 | kavit | DocHolliday: does that use CCM express? |
| 04:55.02 | JT | ManxPower: neat |
| 04:55.34 | DocHolliday | kavit, yes it does if you are willing to pay for the licensing :) |
| 04:55.44 | ManxPower | JT: Yeah it is. The same telco routed calls to Teliax for us during Katrina |
| 04:55.55 | JT | hmm |
| 04:56.00 | kavit | DocHolliday: dont they run on pentium 3 chipsets? unless I am mistaken.... |
| 04:56.06 | ManxPower | well, routed them to some Teliax DIDs we set up |
| 04:56.13 | JT | i love the topic of failover/HA, even if it is so hard to acheive :) |
| 04:56.43 | kavit | JT: I am looking to find a decent solution so I can make it easy to administer |
| 04:56.54 | DocHolliday | kavit, the hardware on a router/switch is never a dual core.. the fact is they are engineered for a specific task/ |
| 04:56.57 | ManxPower | I guess I should get to bed soon. Need to do laundry in the morning, then drive to New Orleans in the afternoon |
| 04:57.34 | kavit | DocHolliday: ofcourse... I wasn't bagging it... just making sure we were both talking about the same machine here |
| 04:57.36 | DocHolliday | i'd be surprised if it had a few hundred Mhz of horsepower |
| 04:57.46 | |ryan| | does IAX2 always send DTMF out of band>? |
| 04:58.21 | DocHolliday | 2801 Voice Bundle,PVDM2-8,SP Serv,64F/256D |
| 04:58.33 | ManxPower | |ryan|: always, 100% |
| 04:58.33 | DocHolliday | thats the standard config.. then you add on your DSPs and cards |
| 04:58.35 | kavit | DocHolliday: I had a torrid time getting CCME SIP 3XX requests to work with asterisk 1.2.6 or something... i almost cried |
| 04:59.01 | DocHolliday | haha, well i love asterisk goodness, the cisco phones work extremely well |
| 04:59.18 | DocHolliday | SIP firmware with the new 79x1 platform is excellent |
| 04:59.21 | kavit | DocHolliday: dude not a year or two ago\ |
| 04:59.38 | JT | DocHolliday: what do you mean the hardware is never a dual core? |
| 04:59.38 | ManxPower | |ryan|: I have heard of some carriers that take a SIP connection from THEIR carrier, convert it to IAX2 and send it on to you and their SIP DTMF is screwed up. |
| 04:59.39 | kavit | DocHolliday: SEPXXXXXXXXX>cnf.xml config files right? |
| 04:59.42 | JT | what do you mean by dual core? |
| 04:59.47 | DocHolliday | correct |
| 04:59.48 | kavit | I had to hand edit it |
| 04:59.53 | DocHolliday | JT you aren't going to find a dual core processor in a cisco router |
| 05:00.18 | DocHolliday | chances are the processor will be a few hundred Mhz with ~64-256MB of ram |
| 05:00.19 | JT | what does it matter how many cores the cpu(s) have? :) |
| 05:00.22 | ManxPower | heck, as far as I can tell you are not going to find an Intel CPU in a Cisco router. |
| 05:00.23 | JT | sure |
| 05:00.37 | Qwell | what do cisco routers use? |
| 05:00.40 | DocHolliday | JT just referencing the fact that they aren't state of the art |
| 05:00.48 | JT | DocHolliday: ah yeah |
| 05:00.53 | kavit | 28XX has a katmai DocHolliday if i am not mistaken... p3 450-500 |
| 05:00.59 | DocHolliday | lemme check |
| 05:00.59 | orlock | ManxPower: pretty sure there have been a few in them |
| 05:01.57 | DocHolliday | http://www.linux-mips.org/wiki/Cisco (doesnt list 2801 though) |
| 05:02.11 | DocHolliday | the 2821 is 466Mhz (gives you a rough idea) |
| 05:02.18 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 05:02.29 | ManxPower | "R4700 CPU at 100MHz, Implementation 33, Rev 1.0" and "M860 processor: part number 5, mask 2" |
| 05:02.33 | ManxPower | Whatever those are |
| 05:02.38 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 05:02.43 | Qwell | ManxPower: sounds ARMish |
| 05:03.04 | ManxPower | Qwell: I thought MIPS or Motorola, but I'm not a CPU geek |
| 05:03.10 | Qwell | me neither :P |
| 05:03.18 | DocHolliday | JT how much do you think the average asterisk user spends on the hardware pbx? |
| 05:03.25 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:03.39 | Qwell | DocHolliday: average? $500 for an off the shelf PC |
| 05:03.46 | JT | DocHolliday: on the server or an equivalent traditional pabx? |
| 05:03.53 | DocHolliday | server |
| 05:04.13 | JT | no idea, i've used a few different boxes |
| 05:04.20 | JT | usually under AUD$1000 |
| 05:04.23 | JT | that's USD$800 |
| 05:04.24 | ManxPower | I have no idea what my clients spend currently, but as of a year or two ago it was around $1,000 for the "$customer Special" |
| 05:04.28 | DocHolliday | Qwell, if everyone used server grade hardware do you think the reliability of asterisk would be seen as higher? |
| 05:04.37 | kavit | I get HP Proliants for my customers |
| 05:04.43 | JT | second hand server grade boxes with redundant PSU and RAI1 do fine |
| 05:04.44 | nDuff | If you're going VoIP-only, I suppose. |
| 05:04.46 | JT | RAID1 |
| 05:04.48 | kavit | Proliant ML XXX and DL XXX series |
| 05:04.56 | nDuff | Add a channel bank and a few T1 interfaces and that goes up quite a bit. |
| 05:05.05 | DocHolliday | sure |
| 05:05.11 | ManxPower | and don't forget an external echo canceler |
| 05:05.13 | JT | oh yeah |
| 05:05.15 | kavit | they are more reliable than something someone just throws together |
| 05:05.21 | JT | my interface card cost more than the server |
| 05:05.22 | JT | much more |
| 05:05.29 | JT | on average |
| 05:05.39 | DocHolliday | my standard config is a Dell SC440 1G ram 2*80GB HDs RAID1 + line cards |
| 05:06.00 | kavit | My standard config is Proliant ML 110 G4 |
| 05:06.08 | ManxPower | Heck, half the time I never even SEE the actual server. |
| 05:06.26 | kavit | Server hardware is really cheap now |
| 05:06.33 | DocHolliday | ManxPower, your instruction to the customer is simply to open the SSH port on their firewall? |
| 05:06.36 | JT | yeah |
| 05:06.40 | Qwell | kavit: "cheap"...yeah |
| 05:06.45 | JT | i'm thinkinf of going a more embedded route |
| 05:06.45 | ManxPower | Client tells me the standardized install has been run, the server is on IP address X has the specified interface card in it and I do the rest |
| 05:06.47 | Qwell | good explanation |
| 05:06.58 | Qwell | JT: aadk |
| 05:07.04 | JT | lower power consumption, heat and moving parts |
| 05:07.04 | kavit | Qwell: HP havent lost too much on quality imho |
| 05:07.07 | JT | aadk? |
| 05:07.19 | Qwell | JT: http://www.digium.com/en/products/hardware/aadk.php |
| 05:07.27 | JT | oh ok |
| 05:07.35 | JT | i was thinking more rolling my own :) |
| 05:07.44 | DocHolliday | qwell, that wouldn't work i usually use PoE |
| 05:08.05 | *** join/#asterisk shodan- (n=shodan@ip211.96-113-216.pppoe1.joliette.intermonde.net) |
| 05:08.13 | orlock | my god, either i am loosing it, or awstats is a buggy pos |
| 05:08.23 | kavit | or\ |
| 05:08.30 | Qwell | DocHolliday: there is always injectors |
| 05:08.40 | ManxPower | AAK seems like exactly what my customers do NOT need. |
| 05:08.47 | kavit | orlock: a bit from column A a bit from column b |
| 05:08.48 | orlock | "look at you hacker... panting and sweating as you run through my corridors" |
| 05:08.52 | ping2921 | the load on my asterisk box goes up and down from 0.3 to 1.2, while running the same call sequence. anyone knows why this is happening? |
| 05:08.57 | ManxPower | i.e. they do not need a router, they do not need 5 LAN ports, they do not need analog ports. |
| 05:09.07 | *** join/#asterisk rrrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 05:09.11 | Qwell | ManxPower: 4 lan ports :p |
| 05:09.12 | orlock | kavit: trying to get it to work with virtualdomains.. its fscked! |
| 05:09.23 | Qwell | + wan + craft |
| 05:09.27 | JT | it needs a modular approach |
| 05:09.28 | DocHolliday | qwell, since i use cisco 7941G's, I usually buy an 8 port Linksys PoE switch 10/100 with dual GE uplinks |
| 05:09.36 | JT | so you can put isdn in instead of FXO :) |
| 05:09.38 | kavit | orlock: heh good luck.... |
| 05:09.42 | Qwell | JT: the analog is modules |
| 05:09.51 | JT | can you buy it with isdn? |
| 05:09.55 | Qwell | no |
| 05:09.57 | kavit | orlock: are you guys linked to PIPE? |
| 05:10.07 | Qwell | JT: but, call sales and express an interest |
| 05:10.07 | orlock | kavit: just nextep |
| 05:10.19 | ping2921 | what could cause load spikes in asterisk? |
| 05:10.23 | kavit | orlock: ah ok |
| 05:10.25 | JT | i don't do POTS to a business |
| 05:10.28 | Qwell | ping2921: a number of things |
| 05:10.37 | JT | if you can afford any type of phone system, you can afford isdn |
| 05:10.39 | JT | in australia |
| 05:10.39 | Qwell | JT: isdn isn't too common here |
| 05:10.46 | Qwell | but, yeah, I understand |
| 05:10.48 | JT | isdn means BRI or PRI |
| 05:10.57 | Qwell | well, okay, isdn bri |
| 05:10.58 | ltdwk | isdn BRI costs the same as a normal line in aus |
| 05:11.05 | JT | yeah |
| 05:11.06 | orlock | you just need to submit to a rogering by telstra |
| 05:11.08 | kavit | yeah |
| 05:11.08 | ltdwk | approximately |
| 05:11.09 | orlock | then the isdn is yours |
| 05:11.10 | Qwell | bri is usually pretty close to pri here, heh |
| 05:11.11 | DocHolliday | we have a 10Mbps/10 Metro Ethernet connection (pretty sweet) |
| 05:11.20 | ManxPower | ltdwk: it does in most of the world except USA |
| 05:11.31 | JT | orlock: other carriers rebill bri |
| 05:12.16 | Mahmoud | "comedian mail.. mail box".. Is this what the voicemail says once you attempt to open your vmail box? |
| 05:12.19 | kavit | I get racks and bandwidth real cheap.... especially PIPE and equinix... most ISPS peer now |
| 05:12.30 | |ryan| | is there any way that I can verify that asterisk is getting DTMF from my phone? |
| 05:12.51 | DocHolliday | kavit, considering cogent is now in the 8-10 range.. bandwidth is barely a consideration |
| 05:12.53 | JT | it's a pitty that equinix's facility is so crappy |
| 05:12.57 | JT | but i guess it's cheap |
| 05:13.09 | kavit | DocHolliday: yah |
| 05:13.16 | ltdwk | JT: is it? they talk it up |
| 05:13.22 | ltdwk | JT: the mascot one? |
| 05:13.38 | kavit | JT: I prefer global switch..... we have our own floor space through the people we share offices with |
| 05:13.39 | DocHolliday | kavit, almost any decent provider can be had under $30, can get lots of SIP/IAX sessions on 1Mbps :P |
| 05:13.43 | JT | ltdwk: globalswitch in ultimo is way higher grade in terms of facilities |
| 05:13.58 | JT | kavit: i was just in globalswitch today |
| 05:14.14 | ltdwk | what is the price like? |
| 05:14.19 | JT | expensive |
| 05:14.20 | orlock | JT: ever heard bad things about global crossing or whatever its called now? |
| 05:14.22 | JT | for floorspace |
| 05:14.26 | JT | orlock: no idea |
| 05:14.28 | |ryan| | fsck |
| 05:14.32 | kavit | orlock: asian global crossings? |
| 05:14.42 | kavit | orlock: i hate them |
| 05:14.49 | orlock | damn, they changed their name a few times |
| 05:14.49 | kavit | JT: do you consult? |
| 05:14.50 | ltdwk | equinix is not exactly cheap. they quoted me around $750 per month per half rack |
| 05:14.52 | JT | ltdwk: globalswitch is carrier neutral, bandwidth is up to you, or you go through someone already in there |
| 05:14.55 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:14.55 | orlock | frontier global? |
| 05:14.56 | JT | kavit: a bit |
| 05:15.06 | DocHolliday | ltdwk, most voip providers dont require a lot of space? |
| 05:15.12 | kavit | JT: yeah.... they arent cheap though |
| 05:15.16 | DocHolliday | unless you are installing TDM equipment |
| 05:15.18 | JT | ltdwk: globalswitch is cheaper for half rack, weirdly enough |
| 05:15.33 | ltdwk | globalswitch doesn't have any peering, though? |
| 05:15.47 | JT | ltdwk: globalswitch is an IT real estate provider, that's it |
| 05:16.00 | JT | ltdwk: there's 13 carriers terminated into the meetme rooms in globalswitch |
| 05:16.00 | kavit | ltdwk: We have 2 racks for our use and they are 90% empty.... we have about 5 softswitches |
| 05:16.11 | DocHolliday | kavit, what kind of gear? |
| 05:16.22 | kavit | DocHolliday: Sun and HP mostly |
| 05:16.30 | ltdwk | i have some equipment in AAPT, ross st |
| 05:16.32 | kavit | DocHolliday: DL380s |
| 05:16.37 | JT | is that glebe? |
| 05:16.40 | ltdwk | yeah |
| 05:16.48 | JT | that's such a small datacentre |
| 05:16.50 | JT | been in there |
| 05:16.50 | Qwell | kavit: x86? |
| 05:16.55 | Qwell | or sparc too? |
| 05:16.56 | ltdwk | It is quite small |
| 05:17.05 | DocHolliday | kavit, so would i be correct in saying that most providers really only need a half rack? |
| 05:17.27 | kavit | Qwell: mostly.... I want a sparc... niagra would be nice |
| 05:17.33 | JT | haha |
| 05:17.34 | Qwell | I have one :p |
| 05:17.36 | JT | they're awesome |
| 05:17.49 | kavit | DocHolliday: yeah.... but if you fill up half a rack globalswitch power quota runs out |
| 05:17.52 | Qwell | Sun never took it back |
| 05:18.06 | kavit | DocHolliday: THEY CHARGE ME EXTRA FOR POWER per kw |
| 05:18.11 | DocHolliday | hah |
| 05:18.18 | JT | nice |
| 05:18.25 | JT | then don't waste it :P |
| 05:18.33 | ltdwk | they do that at equinix too... you get a limit |
| 05:18.40 | Qwell | kavit: plug your UPS into your UPS instead of the wall |
| 05:18.40 | kavit | Qwell: feel a bit philanthropical today?? |
| 05:18.41 | DocHolliday | so the cost of operating a voip provider from a hardware standpoint is fairly cheap (a couple servers, bandwidth, rackspace, power) |
| 05:18.43 | Qwell | avoid power costs completely |
| 05:18.57 | JT | Qwell: ? |
| 05:19.09 | JT | DocHolliday: not if you have terminations |
| 05:19.09 | Qwell | that, or power over wireless ethernet from another facility |
| 05:19.19 | kavit | Qwell: or get a cage full of monkeys on cycles |
| 05:19.27 | Qwell | then you'd have to feed the monkeys |
| 05:19.29 | JT | or a fuel cell |
| 05:19.30 | kavit | Qwell: Tesla style |
| 05:19.32 | JT | and a fuel tank |
| 05:20.11 | DocHolliday | JT, ah meaning you interface with the PSTN? |
| 05:20.18 | JT | DocHolliday: yeah |
| 05:20.45 | kavit | I got done by the power clause though... |
| 05:20.53 | DocHolliday | JT, a TLS circuit is like 1250 per month eh? |
| 05:20.54 | kavit | I had to change my service providers agreement |
| 05:21.01 | JT | DocHolliday: tls? |
| 05:21.23 | JT | kavit: you in globalswitch outright, or through someone else? |
| 05:21.49 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 05:21.55 | DocHolliday | basically interfacing directly with a CLECs switch |
| 05:21.56 | kavit | JT: through anchor systems but we share their office and they give shit to us at cost |
| 05:22.03 | JT | ah cool |
| 05:22.10 | kavit | JT: they are great guys /shameless plug |
| 05:22.14 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:22.20 | JT | apparently most people in there sell rackspace at cost |
| 05:22.25 | tengulre | Hi,all |
| 05:22.26 | JT | because they can't compete otherwise |
| 05:22.32 | kavit | JT: but dont tell you about the power |
| 05:22.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:23.01 | JT | heh |
| 05:23.06 | kavit | JT: I get PIPE bandwidth for under 90 bucks a Mb |
| 05:23.09 | DocHolliday | JT heh i pay $75 a month for a business class 10Mbps connection |
| 05:23.11 | kavit | unlimited use |
| 05:23.26 | JT | kavit: in equinix or globalswitch? |
| 05:23.40 | JT | DocHolliday: i'd debate the degree of "business class", but sure |
| 05:23.48 | DocHolliday | JT its a metro ethernet circuit |
| 05:23.50 | kavit | in global switch |
| 05:23.53 | tengulre | I want to developt a very simple VoIP client phone on windows platform, anyone tell me which VoIP protocol suit me? |
| 05:24.05 | ltdwk | there's pipe in globalswitch? |
| 05:24.07 | tengulre | IAX2, SIP, H323? |
| 05:24.15 | JT | kavit: hrm, that's pretty good, you on bandwidth restricted or burstable pipe? |
| 05:24.30 | JT | ltdwk: umm, there's THIRTEEN CARRIERS, pipe is a major player there |
| 05:24.35 | ltdwk | DocHol: That's just for peering bandwidth, not internet |
| 05:24.42 | kavit | JT: its meant to be restricted but the guys love me so its burstable |
| 05:24.46 | Qwell | tengulre: thanks |
| 05:24.46 | JT | ltdwk: globalswitch is probably the biggest carrier neutral datacentre in .au |
| 05:24.50 | JT | kavit: cool |
| 05:25.03 | tengulre | Qwell: thanks for what? |
| 05:25.04 | kavit | JT: yeah I am thinking of getting into the co-location business |
| 05:25.10 | Qwell | For giving me a new QOTD |
| 05:25.10 | JT | nice |
| 05:25.17 | kavit | DocHolliday: we are in australia |
| 05:25.21 | JT | you know there's basically no floor space not spoken for in gs? |
| 05:25.34 | DocHolliday | kavit, wouldnt it be cheaper to host your machines out of the country? |
| 05:25.44 | JT | DocHolliday: bad idea for voip. |
| 05:25.49 | kavit | DocHolliday: latency.... |
| 05:25.49 | ltdwk | Doc: You can't very well do that for voice |
| 05:25.53 | Qwell | host it in NZ :p |
| 05:26.00 | kavit | Qwell: HELL NO |
| 05:26.00 | ltdwk | NZ! Blasphemy |
| 05:26.03 | kavit | :P |
| 05:26.03 | Qwell | heh |
| 05:26.03 | JT | Qwell: it's incredibly more expensive there :P |
| 05:26.22 | kavit | brb... need a durry bro... |
| 05:26.25 | DocHolliday | fair enough |
| 05:26.28 | Qwell | durry? |
| 05:26.28 | DocHolliday | how much bandwidth are you using? |
| 05:26.33 | JT | kavit: do you mind me asking what you currently do with the space there? |
| 05:26.36 | kavit | durry = fag |
| 05:26.44 | JT | haha, tricking the yanks, gw, kavit |
| 05:26.44 | kavit | JT: nothing at the moment |
| 05:26.44 | tengulre | Qwell: can you tell me which voip protocol suit begginner? |
| 05:26.51 | kavit | brb |
| 05:26.57 | JT | kavit: nothing? :o |
| 05:30.06 | clyrrad | Qwell: you were correct... you still here? |
| 05:30.11 | Qwell | yeah |
| 05:30.22 | Qwell | that worked? |
| 05:30.23 | clyrrad | TERM=dumb by default from CRON |
| 05:30.28 | Qwell | nice |
| 05:30.33 | clyrrad | so if you go into the crontab and add TERM=xterm-color |
| 05:30.42 | clyrrad | the colors are there when Asterisk is started from Cron :D |
| 05:30.49 | Qwell | cool, good to know |
| 05:31.09 | clyrrad | thanks for pointing me in the right direction |
| 05:31.11 | JT | starting asterisk from cron? interesting |
| 05:31.21 | tengulre | anybody using iaxclient in here? |
| 05:32.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:35.37 | kavit | JT: by nothing I mean 2 racks with 6 boxes and 4 switches |
| 05:35.48 | JT | for voip? |
| 05:35.50 | kavit | JT: I can get 2 more racks in a hurry |
| 05:35.58 | JT | in a cage? |
| 05:36.35 | kavit | JT: VoIP mostly.... some custom voice app stuff for a cliet |
| 05:36.38 | kavit | client |
| 05:37.24 | kavit | JT: yeah in anchors area...they buy in lots of 15 racks |
| 05:37.31 | JT | cool |
| 05:37.36 | Mahmoud | can I tell asterisk to send emails using SMTP rather than sending its self without using external mail application? |
| 05:37.39 | JT | i hear there's little floor space left |
| 05:37.50 | JT | kavit: itsp or jjust client apps? |
| 05:38.05 | kavit | JT: its almost all gone.... |
| 05:38.12 | kavit | JT: itsp? |
| 05:38.23 | kavit | JT: we host their softswitch |
| 05:38.30 | kavit | if thats what you mean by itsp |
| 05:38.41 | JT | internet telephony service provider |
| 05:38.46 | JT | do you sell voip to the public? |
| 05:38.50 | kavit | JT: ah yeah we do |
| 05:39.05 | JT | business only, managed pabx? |
| 05:39.09 | kavit | JT: to businesses only... and to businesses who are more than 1 seat... |
| 05:39.16 | JT | right |
| 05:39.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 05:39.32 | kavit | JT: getting into managed.... hard to get it into people where it will really be worthwhile |
| 05:39.54 | JT | do you have any terminations? |
| 05:40.25 | kavit | not to PSTN directly.... we go through iVox for that.... |
| 05:40.35 | JT | ah ok |
| 05:40.41 | kavit | straight fibre link into their network though |
| 05:40.44 | JT | terminations cost a bomb in there |
| 05:40.48 | tengulre | JT.. |
| 05:40.55 | JT | due to per metre rental charges |
| 05:41.02 | kavit | yeah :| |
| 05:41.25 | JT | people need to team up to make those charges worth it :) |
| 05:41.29 | kavit | there is not much money i itsp business if you arent consulting and selling hardware and apps |
| 05:41.41 | JT | definately |
| 05:41.55 | JT | i wonder if most itsps just resell someone else's voip |
| 05:42.25 | kavit | JT: people are so short sighted... we are starting an online voip shop... so i approached a few ITSPs to resell their stuff but 80% of them were unreceptive |
| 05:42.33 | kavit | mainly because I have an ITSP |
| 05:43.00 | JT | why would you offer to resell theirs? ;) |
| 05:43.27 | kavit | JT: because if a customer isn't to our suiting...... we can still make money off them |
| 05:43.42 | JT | kavit: i meant more using a major voip provider to get their terminations |
| 05:43.44 | JT | ah ok |
| 05:43.49 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:43.51 | kavit | JT: like someone who wants buisness voip using ATAs |
| 05:44.03 | JT | yeah |
| 05:44.25 | kavit | i hate pointless support calls |
| 05:44.34 | orlock | yeah |
| 05:44.38 | orlock | bane of my life |
| 05:45.01 | kavit | but i have a linksys ata.... :| |
| 05:47.21 | kavit | JT: most ITSPs resell other peoples VoIP |
| 05:47.27 | JT | i have a T1 channel bank at home for my own use, does that count? :P |
| 05:47.59 | JT | kavit: i wonder if there are even half a dozen ITSPs in .au that terminate |
| 05:47.59 | kavit | JT: haha do you run a 1900 number.... |
| 05:47.59 | kavit | single handedly |
| 05:47.59 | kavit | ?? |
| 05:48.01 | JT | nah |
| 05:48.15 | JT | i'd use a sip phone for that :P |
| 05:48.32 | kavit | JT: you will be hard pressed to find a dozen...but there are over 200 ITSPs in australia |
| 05:48.44 | JT | lol, 200, who'd have thunk it |
| 05:48.52 | JT | i find that hard to believe |
| 05:49.04 | kavit | JT: dude it is shocking.... i feel ashamed calling myself a ITSP |
| 05:49.06 | flenders | wow! 200? |
| 05:49.07 | kavit | hang on JT |
| 05:49.13 | flenders | I only heard of 6 or 7 |
| 05:49.13 | JT | where'd you get that data? |
| 05:49.48 | kavit | http://www.marketclarity.com.au/voip/ |
| 05:50.12 | JT | kavit: do you know if engin terminates? |
| 05:50.14 | kavit | 259 |
| 05:50.41 | kavit | JT: i am not sure.... but I'd say they do now if they didn't earlier |
| 05:50.53 | kavit | JT: channel 7 gave them a cash injection |
| 05:51.13 | flenders | kavit: any idea about customer base? |
| 05:51.19 | flenders | I mean, number of customers? |
| 05:51.21 | kavit | i am just pissed off people are selling VoIP as cheap calls and nothing else... no one talks about what else it can do for SMEs |
| 05:51.33 | kavit | flenders: 3% of australian businesses |
| 05:51.42 | orlock | kavit: yeah, IVR's, the flexibilty of *, etc |
| 05:51.46 | kavit | flenders: engin gets about 1500 new customers a week |
| 05:51.54 | flenders | kavit: you're kidding |
| 05:52.02 | JT | kavit: and a marketing plug |
| 05:52.12 | kavit | orlock: in a month we are launching off the shelf asterisk apps... all opening sourcr |
| 05:52.16 | JT | flenders: dude, engin is everywhere, ever been to a dick smith electronics? |
| 05:52.22 | kavit | JT: i dont market :P |
| 05:52.36 | flenders | I know! |
| 05:52.38 | kavit | flenders: their network sucks.... |
| 05:52.40 | JT | kavit: ch 7 gave engin plugs on today tonight |
| 05:52.41 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:52.42 | flenders | impressive |
| 05:52.43 | zeeesh | hi |
| 05:52.47 | kavit | JT: ah yeah |
| 05:53.02 | kavit | JT: and engin boxes dont work with anything else |
| 05:53.08 | kavit | why get rid of an avaya |
| 05:53.21 | JT | yeah i figure who-ever architects the engin network is a complete idiot |
| 05:53.22 | kavit | if i am to be locked down by a 3435355% inferior product |
| 05:53.27 | flenders | kavit: mate, I'll tell you that very few times I had issues with them... and all times, I asked to change something on my account and the fuck up happened |
| 05:53.31 | JT | with their cash, they can easily make it more reliable |
| 05:53.39 | flenders | other than that, they're pretty good to me |
| 05:53.44 | kavit | flenders: do you connect to *.byo.engin? |
| 05:53.50 | JT | kavit: i only use byo |
| 05:54.03 | kavit | JT: i dont like their support.... |
| 05:54.04 | flenders | I only use byo |
| 05:54.09 | flenders | too |
| 05:54.11 | JT | kavit: heh |
| 05:54.30 | flenders | their support is a joke |
| 05:54.35 | ltdwk | i didn't think engin was making any money |
| 05:54.37 | kavit | JT: i had issues with a customer with SIP From headers.. the guy had 50 dids and half of them hand mangled From headers |
| 05:54.48 | ltdwk | that's what the media was saying |
| 05:54.49 | kavit | and then they told me WE DONT SUPPORT ASTERISK |
| 05:54.53 | JT | ltdwk: how could they not? they have so many customers |
| 05:55.08 | kavit | so I asked them it they support SIP |
| 05:55.16 | kavit | and no answer.... |
| 05:55.35 | kavit | JT: they are making a loss on many calls....they dont get flat rates |
| 05:55.48 | JT | kavit: :o |
| 05:55.49 | ltdwk | their pricing is extremely competetive |
| 05:55.59 | JT | on what, mobiles or landlines, they lose money? |
| 05:56.00 | flenders | mate, 10c a call |
| 05:56.15 | kavit | JT: they pay by the minutes and try and work out a statistical model where after X number of users they will make money |
| 05:56.22 | ltdwk | they take a dive on the hope that the average call is not long |
| 05:56.24 | JT | yikes |
| 05:56.30 | kavit | ltdwk: bingo |
| 05:56.43 | kavit | ltdwk: thats why I didnt get my own termination..... |
| 05:56.45 | ltdwk | they use the same providers we do at the end of the day |
| 05:56.49 | kavit | too risky |
| 05:56.54 | JT | kavit: do you have any other data on how they do termination? |
| 05:57.20 | kavit | JT: not really.... they are very secretive |
| 05:57.25 | kavit | one more thing I dont like... |
| 05:57.32 | JT | you know more than I |
| 05:57.41 | kavit | JT: as a provider... i cant get enough of talking about my infrastructure |
| 05:57.48 | kavit | I tell everyone |
| 05:58.01 | kavit | you know what I mean.... |
| 05:58.09 | JT | sure |
| 05:58.19 | JT | they're big boys, they don't need to boast about anything ;) |
| 05:58.24 | ltdwk | so you use engin JT ? |
| 05:58.32 | JT | yes |
| 05:58.36 | ltdwk | what about you kav? |
| 05:58.44 | kavit | ltdwk: I use myself |
| 05:58.52 | flenders | I use it too |
| 05:58.55 | kavit | ltdwk: I run an itsp of sorts |
| 05:59.10 | ltdwk | kavit: what's itsp? |
| 05:59.20 | kavit | ltdwk: ask JT :p |
| 05:59.22 | JT | i'm starting to think there's few ITSPs in .au that actually can send correct customer callerid |
| 05:59.29 | JT | engin is one of the few that can |
| 05:59.32 | kavit | JT: I can do it.... :) |
| 05:59.34 | JT | because the rest resell |
| 05:59.44 | JT | kavit: ss7 or just q.931? |
| 05:59.52 | ltdwk | kavit: okay let me rephrase...where do you send your calls, and by what method |
| 05:59.53 | kavit | JT: not only that I do 1300 numnbers and telstra/optus number porting to voip |
| 06:00.00 | JT | cool |
| 06:00.06 | kavit | JT: ss7 |
| 06:00.07 | orlock | JT: nexvoice can... well, what do you mean by "correct"? |
| 06:00.08 | JT | are you actually registered as a carrier? |
| 06:00.18 | orlock | you can only send the CID of the account you are making the call with |
| 06:00.21 | orlock | you cant spoof it, etc |
| 06:00.22 | ltdwk | kavit: ss7 to telstra wholesale? |
| 06:00.27 | JT | orlock: as in the customer's phone number, you need to be a carrier usually |
| 06:00.42 | kavit | orlock: people buy links of multiple links and then route calls where it is the cheapest |
| 06:01.04 | kavit | ltdwk: i dont terminate myself.... i get termination from iVox who are great |
| 06:01.22 | JT | kavit: so ivox run ss7? |
| 06:01.23 | kavit | ltdwk: so I dont know who their supplier is... but i can guess safely it is telstra |
| 06:01.30 | ltdwk | kavit: so you only use ss7 for incoming DID? |
| 06:01.53 | JT | kavit: or do they just have PRIs that allow sending any callerid? |
| 06:01.58 | kavit | JT: don't quote me on that..... |
| 06:02.17 | JT | heh |
| 06:02.20 | kavit | JT: i think they have hybrid stuff |
| 06:02.26 | kavit | i have seen pris |
| 06:02.26 | JT | kavit: hmm ok |
| 06:02.36 | SwK | *YAWN* |
| 06:02.36 | JT | ss7 is only signalling |
| 06:02.39 | kavit | but I know they deal with telstra as well |
| 06:02.53 | ltdwk | they probably use ss7 over PRIs |
| 06:03.03 | kavit | JT: I can dial 000 from my voip service |
| 06:03.08 | SwK | SS7 doesnt run over PRI |
| 06:03.12 | SwK | it works w/ IMTs |
| 06:03.16 | SwK | which are just trunks |
| 06:03.26 | JT | ss7 isn't technically needed, if their telco allows setting callerid, not sure how ANI fares though |
| 06:03.36 | SwK | you can do SS7 to PRI conversions tho |
| 06:03.39 | ltdwk | well, the ss7 implementations for asterisk claim to support use of Zap PRI interfaces |
| 06:03.43 | kavit | JT: i could find out |
| 06:03.55 | JT | kavit: i'd be interested |
| 06:04.07 | kavit | JT & ltdwk it is an interesting question.. i had never thought of it |
| 06:04.09 | flenders | well gents... gotta go. have a nice evening |
| 06:04.09 | JT | as if you'd use an ss7 implementation for asterisk |
| 06:04.16 | SwK | ltdwk: thats cause it converts from SS7 to something that will work with the PRI stack (in the form of q931) |
| 06:04.19 | kavit | laters flenders |
| 06:04.21 | SwK | vi chan_woomera |
| 06:04.51 | ltdwk | kavit: I use SPT for termination |
| 06:05.03 | kavit | ltdwk: got to got to got to have soul |
| 06:05.06 | SwK | JT actually its supposed to work quite well and how it actually works is sound engineering |
| 06:05.21 | kavit | ltdwk: i talked to them.... too expensive for wholesale and too restrictive... |
| 06:05.22 | JT | SwK: ? |
| 06:05.23 | SwK | its not unlike any other SS7 to PRI converter thats out there |
| 06:05.32 | SwK | the SS7 stuff |
| 06:05.38 | kavit | ltdwk: they have a good network though |
| 06:05.40 | JT | but can asterisk really do anything with ss7 messages? |
| 06:05.46 | JT | or is that another application? |
| 06:06.00 | ltdwk | kavit: I've found the service quite good, rates reasonable but not exceptional |
| 06:06.25 | kavit | ltdwk: i am talking about their bandwidth and hosting stuff as well..... |
| 06:06.28 | SwK | JT: its another application |
| 06:06.38 | ltdwk | kavit: I use them for transit too |
| 06:06.42 | SwK | the one I know about is called SS7 box |
| 06:06.54 | SwK | SS7 is just a signalling protocol like SIP |
| 06:06.55 | kavit | ltdwk: ah....i didn't like them.... |
| 06:07.08 | kavit | ltdwk: plus I hate their CDRs |
| 06:07.23 | SwK | it has extra stuff that can be done like SMS, CNAM, LIDB etc dips |
| 06:07.36 | JT | SwK: i realise |
| 06:07.41 | SwK | SMS there not to be confused with short messaging svs |
| 06:08.09 | `p4r14h | could some one explain to me what would be a simple example of zapata.conf file for just a singe X100P hooked up to a POTS line would be? |
| 06:08.11 | SwK | woomera which is uses is a generic call control bus designed by the guys that did openH323 |
| 06:08.18 | orlock | hmm.. |
| 06:08.26 | orlock | cream bulbs on ebay |
| 06:13.07 | JT | orlock: you can buy them from any catering supply store |
| 06:13.38 | orlock | yeah, or campbells cash+carry if you have a card |
| 06:14.44 | orlock | :) |
| 06:15.55 | Mahmoud | when asterisk is sending voicemail email notifications, the FROM field shows "Asterisk PBX" <email@foo.zone>. How to change the name from Asterisk PBX? |
| 06:18.00 | Mahmoud | ah, got it "fromstring=my new name" |
| 06:18.07 | orkid | that was quick |
| 06:19.12 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:20.02 | tengulre | anyone can tell me where have SIP phone source code and can running under windows platform? |
| 06:20.47 | orlock | hmm |
| 06:21.03 | orlock | open source sip phone under windows.... |
| 06:21.09 | orlock | havent heard of one |
| 06:22.32 | orkid | hah! wow |
| 06:22.55 | *** part/#asterisk bartpbx (n=bartpbx@pD9E88736.dip0.t-ipconnect.de) |
| 06:23.16 | orkid | http://snapshots.voxgratia.org/win32.php ? |
| 06:23.19 | orkid | ekiga is SIP iirc |
| 06:23.31 | orkid | and others, but it should support SIP |
| 06:24.52 | SwK | if you want to spend money of it you can always license the xlite/eyebeam stuff but i hear it aint cheap |
| 06:27.21 | yonahw-work | good morning people |
| 06:28.59 | Mahmoud | any way to access mail boxes without being prompted for mail box name or password? |
| 06:29.17 | kavit | SwK: 60 usd for eyebeam |
| 06:29.48 | tengulre | orlock: thanks. |
| 06:29.49 | kavit | is sjphone open source? |
| 06:30.47 | kavit | ok... to go home or not to go home |
| 06:31.40 | yonahw-work | mahmoud; voicemailmain(s${CALLERIDNUM}) |
| 06:31.54 | yonahw-work | the 's' lets you in without the prompt |
| 06:32.26 | Mahmoud | yonahw-work, and its security is from SIP username/pword authentication? |
| 06:33.07 | yonahw-work | it is letting you into the box for the extension you are calling from |
| 06:33.41 | Mahmoud | so for sip phones, no one cal login unless he knows sip uname/pword. right |
| 06:33.47 | Mahmoud | s/cal/can/ |
| 06:34.02 | yonahw-work | as far as i understand but i have been wrong before |
| 06:35.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:43.03 | Mahmoud | hmmm |
| 06:43.13 | DocHolliday | Mahmoud, when do you sleep? |
| 06:43.29 | Mahmoud | dunno.. until i finish this up |
| 06:45.09 | *** join/#asterisk rrrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 06:46.46 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 06:48.08 | *** join/#asterisk Inez (i=faceoff@devel4.net) |
| 06:48.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:48.43 | Inez | tdi |
| 07:02.27 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 07:06.18 | Mahmoud | i want my phone show how many voice messages i have |
| 07:06.39 | Mahmoud | i used vmexten in sip.conf, but didn't work, probably messed something |
| 07:07.07 | Mahmoud | i said: vmexten=500, where 500 in dialplan is an extension executing voicemailmain() |
| 07:07.38 | Mahmoud | what's wrong? |
| 07:07.58 | Mahmoud | i want WMI notification |
| 07:08.47 | Mahmoud | MWI* |
| 07:09.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:11.08 | *** join/#asterisk TheMahmoud (n=fake@unaffiliated/mahmoud) |
| 07:12.58 | gfraysse | <PROTECTED> |
| 07:15.00 | Vec | I am having a big strange problem, it seems that I can only have 1 simultanious incoming call (ringing) on my pri, any ideas ? |
| 07:20.07 | zeeesh | how to confugure voicemail between 2 peers can anybody guide .. and where receiver will listen ??? |
| 07:20.18 | Vec | It looks like I have ZAP0-2 i.e only using 2 lines |
| 07:25.24 | Vec | Please can someone help me it seems that I have defined my Pri trunk groups incorrectly so I can only have 2 simulatanious calls, I am being attacked here |
| 07:25.37 | JT | err |
| 07:25.42 | JT | how many pris? |
| 07:28.10 | *** join/#asterisk drray (n=drray@c-67-170-9-176.hsd1.wa.comcast.net) |
| 07:29.20 | Vec | JT : just one |
| 07:29.29 | drray | I assume the wiki-go splat? |
| 07:29.47 | Vec | it is odd, it seems when there are 2 calls concurrently, and someone tries to call in they get network busy |
| 07:29.53 | Vec | I am freaking here |
| 07:30.46 | Chris-NB | hi |
| 07:30.53 | Chris-NB | anybody knows what that means? |
| 07:30.57 | Chris-NB | codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 07:30.59 | JT | Vec: it's set up wrong |
| 07:31.09 | JT | Vec: you should not be using trunk groups AT ALL |
| 07:31.51 | Vec | I have group=1 in my zapata.conf ? |
| 07:32.00 | Chris-NB | the weird thing is, this is a sip call and I'm not allowing ilbc ?! |
| 07:32.08 | Vec | and then I am dialing with ,Dial,Zap/g1/${EXTEN:1} |
| 07:32.10 | Vec | ? |
| 07:32.15 | Chris-NB | only gsm and g729 |
| 07:33.06 | JT | Vec: that's a zap group not a trunk group |
| 07:33.09 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 07:33.12 | Vec | Does the PRI automatically search for an open channel on incoming calls, and search for one on outgoing automatically ? |
| 07:33.17 | Vec | oh |
| 07:33.24 | Vec | well are my zap groups stuffed / |
| 07:33.25 | Vec | ? |
| 07:33.39 | JT | Vec: pastebin.ca zapata.conf and zaptel.conf |
| 07:33.46 | JT | Vec: is it a fully populated E1? |
| 07:33.51 | drray | it's first available I think, I use G for outbound and g for inbound |
| 07:34.09 | drray | but that may a holdover from my analog days |
| 07:34.38 | JT | using G/g is first available, they go in different orders though |
| 07:34.47 | JT | r/R is round-robin |
| 07:34.49 | tengulre | anybody developt SIP client applications here? |
| 07:35.05 | drray | I've never tried round robin, is that better than G/g? |
| 07:35.17 | JT | not in my opinion |
| 07:35.18 | Vec | JT : it is, 1-15 B channel 16 - D Channel 17-31 B channels |
| 07:35.41 | JT | it's only good if for some reason you like to keep call volume approximately equal over all your lines |
| 07:35.52 | JT | Vec: ok, what about pastebin? |
| 07:36.10 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:36.36 | drray | JT - ohh.. only one PRI here :) |
| 07:36.49 | JT | drray: lines/channels |
| 07:36.53 | Vec | ok : http://www.pastebin.ca/394351 |
| 07:37.26 | Vec | are the loadzone's only for sounds for the user, or do they have a technical requirement ? |
| 07:37.26 | jojo^ | I'm getting "Unknown RTP codec 90 reaceived from ..".. What codec is that? |
| 07:37.59 | Vec | jojo^ : asterisk does'nt know what it is so its probably not defined, what phone are you calling from or to ? |
| 07:39.08 | jojo^ | tengulre, I'm sure there are a few people who's been building SIP clients here. But it's probably kind of hard to catch them on the right time. |
| 07:39.14 | JT | Vec: CCS and HDB3 should be lowercase, but otherwise it looks ok |
| 07:39.43 | JT | loadzone is for zaptel generated indications |
| 07:39.47 | JT | such as dialtone and busy |
| 07:39.50 | jojo^ | Vec, It's outgoing/incoming calls from my SIP provider. I think they are using Cisco's media gateways |
| 07:40.28 | Vec | jojo^ : it maybe (Since its cisco) that they using G729, and asterisk will not transcode that ? |
| 07:40.48 | Vec | JT : any other thoughts on why people would get network busy signals when trying to call ? |
| 07:40.51 | JT | Vec: show me the dial command you're using? |
| 07:41.17 | jojo^ | Vec, Could be. I'm not sure it's a showstopper though, but everytime I see that message it's about 30 identical NOTICE-rows |
| 07:42.11 | Vec | JT : http://www.pastebin.ca/394357 is my extentions.conf, it is very plain and dead at the moment but just trying to get it working, thanks |
| 07:42.30 | Vec | jojo^ : does everything work fine normally ? |
| 07:42.31 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 07:42.37 | jojo^ | Vec, Yeah |
| 07:42.56 | Vec | jojo^ : I would ignore it |
| 07:42.59 | jojo^ | It's probably time I enable more debugging on this asterisk. |
| 07:43.04 | *** join/#asterisk dseeb_ (n=dcb@58.169.152.56) |
| 07:43.25 | jojo^ | Is it easy to see which calls the NOTICEs refer too? |
| 07:43.46 | JT | what the hell exten => _XX.,n,Dial,Zap/g1/${EXTEN:1} |
| 07:44.03 | JT | afaik that's indefined behaviour |
| 07:44.19 | JT | Dial(Zap/g1/${EXTEN:1}) |
| 07:44.23 | JT | not Dial,Zap |
| 07:44.31 | JT | s/indefined/undefined/ |
| 07:44.59 | Vec | JT : for some reason when I took at the brackets it stopped working before, I'll give it a try again though |
| 07:45.03 | JT | also SetCallerID is deprecated, don't use it |
| 07:45.48 | JT | also using ${EXTEN:1} is bad practice with a _XX. pattern |
| 07:45.53 | JT | that might be the issue |
| 07:46.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 07:47.11 | Vec | JT : it seems like there is only a problem on incoming calls |
| 07:47.21 | JT | Vec: are you using 0 to get an outside line? |
| 07:47.33 | JT | Vec: sorry, is the problem inbound or outbound? |
| 07:47.47 | yonahw-work | JT: what can be used instead of setcallerid? |
| 07:48.20 | JT | Set(CALLERID(num)=) |
| 07:48.32 | JT | CALLERID is a function |
| 07:48.46 | yonahw-work | cool thanks |
| 07:49.32 | JT | Vec: back in half an hour |
| 07:50.06 | JT | actually scratch that |
| 07:50.09 | JT | i'm still here :P |
| 07:50.56 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 07:52.12 | Vec | JT : the problem is inbound |
| 07:52.26 | JT | Vec: ok, that pattern should be fixed too |
| 07:52.38 | JT | what is SIP/2116 |
| 07:53.24 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:54.07 | Vec | JT : I don't dial 0 to get an outside line, but somewhere along the line not sure if its asterisk or the telco a 0 is being added so I have to use the :1 to remove the 0 |
| 07:54.28 | Vec | JT : SIP/2116 is the reception phone |
| 07:54.35 | JT | what sort of phone? |
| 07:54.50 | Vec | Grandstream GXP 2000 |
| 07:54.55 | Vec | 4 lines |
| 07:55.15 | JT | Vec: well the 0 thing, fiddle with national/etc number type |
| 07:55.18 | Vec | I would understand that if those 4 linux are busy then there would be congesion |
| 07:55.21 | JT | maybe the phone is set wrong |
| 07:55.46 | Vec | JT : what is the stripmsd thing for ? |
| 07:56.04 | JT | it's a variable used in freepbx i think |
| 07:56.12 | JT | for its dial macros |
| 07:56.16 | Vec | oh ok |
| 07:57.07 | JT | it'd use it like this: ${EXTEN:${STRIPMSD}} |
| 07:58.00 | Vec | JT : I have just noticed even when no lines are in use I get a congestion signal (network busy) but its irratic ? |
| 07:58.09 | JT | err |
| 07:58.18 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:58.20 | JT | do you have an errors about your D channel? |
| 07:59.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:00.52 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:00.57 | Chris-NB | hi |
| 08:01.24 | *** join/#asterisk lorinc (n=ang@pool-2896.adsl.interware.hu) |
| 08:01.25 | Chris-NB | can someone please look at that and probably give me a hint, what that could be? http://nopaste.info/d15efbe07d_nl.html |
| 08:01.46 | Chris-NB | this is a call between two sip phones, allowed codecs are g729 and gsm |
| 08:02.13 | Chris-NB | but why is there ilbctolin ?! this is for converting ilbc to slin. but I'm not allowing that codecs! |
| 08:05.09 | jojo^ | Vec, This RTP codec-things seems to happen every 10mins or so.. Callwise, that is 1/7 calls I think |
| 08:07.05 | Chris-NB | if I prevent asterisk from loading codec_ilbc.so/format_ilbc.so then I get this, if I try to hit the # key on a phone: |
| 08:07.20 | Chris-NB | WARNING[30436]: channel.c:2380 set_format: Unable to find a codec translation path from ilbc to slin |
| 08:10.25 | Vec | JT : what does ZAP/0-2 mean in the cli log, I am tring to find it again / |
| 08:11.55 | Vec | Executing [2000@zapIncoming:1] Answer("Zap/0-2", "") in new stack << in there not sure what the Zap/0-2 means ? |
| 08:13.55 | JT | it's the channel |
| 08:13.55 | JT | Vec: so is there and D channel errors? |
| 08:18.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 08:19.16 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 08:24.29 | tzafrir_laptop | problem |
| 08:24.49 | JT | s/and/any/ |
| 08:26.04 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 08:37.50 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 08:38.10 | tutt9876 | hi got a problem in my sip.conf |
| 08:38.39 | *** join/#asterisk wwq222 (n=chatzill@c-71-231-5-6.hsd1.or.comcast.net) |
| 08:39.12 | tutt9876 | When I do netstat -anpe |grep -i asterisk on my asterisk server i don't see port 5060 open |
| 08:39.15 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 08:39.28 | tutt9876 | listening I mean |
| 08:40.07 | giasai68 | hello, i'm tring to install mysql (cdr) on my asterisk machine. when i start asterisk i got this error:[Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:541 parse_config: MySQL RealTime: No database user found, using 'asterisk' as default. |
| 08:40.07 | giasai68 | [Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:548 parse_config: MySQL RealTime: No database password found, using 'asterisk' as default. |
| 08:40.07 | giasai68 | [Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:555 parse_config: MySQL RealTime: No database host found, using localhost via socket. |
| 08:40.07 | giasai68 | [Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:562 parse_config: MySQL RealTime: No database name found, using 'asterisk' as default. |
| 08:40.10 | giasai68 | [Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:569 parse_config: MySQL RealTime: No database port found, using 3306 as default. |
| 08:40.13 | giasai68 | [Mar 14 09:35:52] WARNING[2567]: res_config_mysql.c:576 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. |
| 08:40.15 | wwq222 | Hey I installed Asterick with Make Install/Make Samples, added a user to sip.conf, added an extension to extensions.conf, but when I do a sip reload in Asterisk, it says cannot bind to 0.0.0.0:5060 - address in use - i searced the web and it seems lots of people get this error, but no good solutions - anyone got any ideas? Thanks |
| 08:40.17 | JT | giasai68: WTF STOP |
| 08:40.20 | JT | giasai68: no more |
| 08:40.44 | JT | giasai68: it's bad form to flood over 300 people |
| 08:40.49 | JT | use pastebin.ca in future |
| 08:40.58 | giasai68 | ok thank you sorry |
| 08:41.13 | wwq222 | sorry i'm new too what's pastebin.ca? |
| 08:41.29 | JT | a web site to share information that's too big to paste |
| 08:41.46 | JT | anything more than a couple of lines is too big to paste here |
| 08:41.57 | wwq222 | very cool thanks |
| 08:42.04 | tutt9876 | my command was "netstat -anpe |grep -i asterisk" in case you missed it up |
| 08:42.51 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:43.17 | tutt9876 | in case you have been missed out on |
| 08:44.05 | tutt9876 | and no 5060 listening port is my case |
| 08:44.50 | FuriousGeorge | tutt9876: are you ahving nat and sip issues? |
| 08:45.27 | tutt9876 | wwq222: did you make a pgrep asterik ? and killall asterisk and reload? |
| 08:45.43 | wwq222 | hmm no lemem try that |
| 08:45.58 | tutt9876 | I guess your server has already been started |
| 08:47.59 | wwq222 | ok i just tried that - i get Asterisk Ready, but when i do sip reload, i get the same error, and when I try to connect to it using like XLite, it can't seem to hit the asterisk server |
| 08:48.24 | wwq222 | is it normal to get a bunch of warnings off the make samples conf files when you first start up in console mode? |
| 08:51.32 | tutt9876 | wwq222: try with a simple sip.conf/extensions.conf picked on the web? |
| 08:51.32 | wwq222 | does anyone know if there is there a way to test your asterisk independently, w/o 3rd party applications? like a self test or something? |
| 08:52.18 | tutt9876 | save your current extensions.conf/sip.conf before? |
| 08:54.47 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 08:54.49 | wwq222 | hehe htey're so simple that i have nothing to lose |
| 08:55.17 | wwq222 | thing is my softphone is telling me login timed out - which is making me think that it's not even finding the asterisk server |
| 08:55.25 | wwq222 | becuase when I do sip show users all the users in teh sip.conf show up just fine |
| 08:56.23 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 08:56.45 | Simplix | bonjour, what's the meaning of "chan_iax2.c:7910 iax2_poke_peer: Still have a callno..." |
| 08:58.18 | gr0mit | hi - anyone here using voip to the UK from a South African adsl connection? |
| 08:58.26 | telmich | does anyone know about good documentation about zaptel.conf? I am looking at digium for some time, but did not find a good documentation, that explains what one should/must/mustn't set values and which values exclude each other |
| 08:59.12 | *** join/#asterisk orkid_ (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 09:04.15 | giasai68 | i have some problem to install mysql(cdr)on my asterisk i paste the information on http://pastebin.ca/394399 |
| 09:05.28 | giasai68 | jt: is it ok now? |
| 09:07.12 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:07.13 | E-bola | Quick question about queues... What does it mean when u put Local/201 as a member |
| 09:07.32 | E-bola | i normally use SIP/201 syntax tro add my extentions |
| 09:07.55 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 09:08.14 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 09:09.06 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:11.14 | E-bola | nobody awake yet? <:) |
| 09:13.40 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:14.00 | Inez | I'm:) |
| 09:14.03 | Inez | it's 10 am |
| 09:15.20 | E-bola | any idea about my queues question? |
| 09:23.32 | walld | Hi, I've got a quick question regarding trunk usage/selection. When the first selected trunk is busy should the next available trunk be tried before returning busy back to the endpoint? |
| 09:24.37 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 09:24.56 | nDuff | E-bola: Local/* processes the value through your dialplan, so it can eventually end up as SIP/ or Zap/ or whatever else. |
| 09:25.11 | nDuff | E-bola: That's not a queue thing, per se -- you can use Local/* other places as well. |
| 09:25.50 | nDuff | walld: eh? I'm accustomed to using a group (like Zap/g1) for your trunk; in that case, any of the lines in that group will be considered, but Zap/g2 [for instance] won't. |
| 09:27.20 | masked | hi |
| 09:27.27 | walld | nduff: The trunks in question are all SIP point to a multi BRI PSTN gateway. Is there a way to group them? |
| 09:27.50 | E-bola | Nduff: ahh ok, got ya. thanks for clearing that up |
| 09:27.53 | masked | i tried sox for wav to g711 but i could only convert to gsm, the site says it supports g711, is there any tricks to encoding to g711? |
| 09:28.05 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:29.17 | nDuff | walld: No clue. |
| 09:29.46 | walld | nduf: Thanks any way |
| 09:32.28 | nDuff | walld: In a similar kind of situation, I've just built a macro which tries Dial()'ing a string of channels in turn; you might end up doing something similar. |
| 09:33.43 | FuriousGeorge | walld: you want to fail your providers over? |
| 09:33.56 | FuriousGeorge | so if one is not available it tries another? |
| 09:34.57 | E-bola | where can i find a syntax listing and info about dialplan, its been a while since i made mine, and i forgot what some of the characters mean |
| 09:35.10 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 09:36.53 | tzafrir | Any idea which SIP / IAX clients support displaying images sent with SendImage? |
| 09:36.53 | tzafrir | Is it related to video? |
| 09:36.53 | *** join/#asterisk Vec (n=Vec@dsl-242-252-39.telkomadsl.co.za) |
| 09:36.53 | tzafrir | or to sms? |
| 09:36.53 | FuriousGeorge | E-bola: search dialplan or pattern matching in voip-info.org |
| 09:36.53 | E-bola | liek what does this mean? 1|. |
| 09:36.53 | E-bola | FuriousGeorge: i wanted to, but it seems down or terribly slow :/ |
| 09:36.58 | FuriousGeorge | err, that | is new to me too |
| 09:37.02 | Vec | Is it possible 1,Dial(blah,timeout) then do anoth 2,Dial(blah2,timeout), Hangup |
| 09:37.10 | FuriousGeorge | unless its an 'or' of course |
| 09:37.15 | Vec | I don't need any special parameters to check |
| 09:37.27 | E-bola | how would u evauluate an or? |
| 09:37.42 | zoa | tzafir, if its important to you (and the iax protocol actually supports it) i could make sure it gets into idefisk |
| 09:37.42 | E-bola | im looking at a current config on a asterisk at home setup, that i need to copy |
| 09:37.48 | FuriousGeorge | Vec: if you want to proceed regardless of if the original call was answered, use 'g' option with dial |
| 09:37.52 | FuriousGeorge | (i think its g) |
| 09:38.02 | E-bola | ohh its a prefix seperator |
| 09:38.15 | walld | nduff: ITs not necessarily providers but I just want to use all the avilable trunks. At the moment I have a total of, 4 trunks with 2 channels for each and each has the same dial pattern. |
| 09:38.31 | E-bola | so this is how the seperate the trunks :) |
| 09:38.48 | gr0mit | Vec - yes, that should work. did you get your DDI problem solved? |
| 09:39.12 | FuriousGeorge | E-bola: gotoif($[${CALLERID(num)}= 101]|$[${CALLERID(num)}=102]?s,1|s,2) |
| 09:39.46 | sumasuma | how can i connect call with threeway calling ? |
| 09:39.58 | FuriousGeorge | E-bola: there is an 'or' used in expression and as a separator between the to possible results |
| 09:40.04 | sumasuma | Since A flash is required after the dial |
| 09:40.59 | FuriousGeorge | walld: when you say 'trunk' you arent referring to an analog channel? |
| 09:41.30 | FuriousGeorge | walld: is this what you are trying to say: you have 4 providers/routes that allow two channels each? |
| 09:41.59 | E-bola | FuriousGeorge: sure, im sory i wasnt too clear, this was a dialplan containing nothing else than what i pasted |
| 09:42.07 | E-bola | i know you can use or in asterisk in more complex expressions |
| 09:42.51 | walld | FuriousGeorge: Apologies for the poor explanation. The device I'm using as a PSTN gateway has 4 x BRI ports |
| 09:43.04 | giasai68 | is possible to check in a dbtable check if is present a called number and destroy the call: e.g. if i received a call with a number present in a db i want destroy this call |
| 09:43.19 | FuriousGeorge | walld: never used that, what does your dial statement look like |
| 09:43.47 | FuriousGeorge | E-bola: a dialplan containing only what you posted wouldnt do anything, afaik |
| 09:44.17 | E-bola | FuriousGeorge: it would match anything that starts with a 1 and then strip off that 1 and go on |
| 09:44.38 | E-bola | like say if you want a feature like "press 1 to dial a us number" press 2 to dial a french number |
| 09:44.38 | E-bola | etc |
| 09:44.58 | walld | FuriousGeorge: The dial statement is the same for each trunk. _XXXX. |
| 09:45.05 | FuriousGeorge | E-bola: im still not seeing it.. for that feature you would use app_background, i think |
| 09:45.17 | FuriousGeorge | and to strip a digit its ${EXTEN:1) |
| 09:45.26 | E-bola | this is a 1.0 asterisk, maybe the syntax changed |
| 09:45.48 | FuriousGeorge | walld: whates the tech? can you just paste a line |
| 09:46.10 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 09:46.58 | FuriousGeorge | walld: i dont use bri, so clue me in please, maybe i can help you |
| 09:48.33 | E-bola | ohh how anoying, i see what this is FuriousGeorge |
| 09:48.43 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 09:48.52 | FuriousGeorge | E-bola: oh yeah? |
| 09:48.58 | E-bola | apparently asterisk at home gui crap, translates it to the following asterisk conf: exten => _1.,1,Macro(dialout-trunk,5,${EXTEN:1},) |
| 09:49.24 | E-bola | so it does in fat use the ${EXTEN:1) as you said |
| 09:49.29 | E-bola | fact even |
| 09:49.43 | FuriousGeorge | i guess i can see more or less how you could get there from that |
| 09:49.54 | walld | FuriousGeorge: exten => _XXXX.,1,Macro(dialout-trunk,3,${EXTEN},,) |
| 09:49.54 | walld | exten => _XXXX.,n,Macro(dialout-trunk,1,${EXTEN},,) |
| 09:49.54 | walld | exten => _XXXX.,n,Macro(dialout-trunk,4,${EXTEN},,) |
| 09:49.55 | walld | exten => _XXXX.,n,Macro(dialout-trunk,5,${EXTEN},,) |
| 09:49.55 | walld | exten => _XXXX.,n,Macro(dialout-trunk,2,${EXTEN},,) |
| 09:50.10 | FuriousGeorge | so i take it you are using a@h too |
| 09:50.22 | FuriousGeorge | either that or you just like the sound of dialout-trunk |
| 09:50.26 | E-bola | i am migrating an a@h to a normal asterisk 1.2 install |
| 09:50.46 | FuriousGeorge | E-bola: yeah, i was talking to walid before |
| 09:50.50 | E-bola | i have never used trixbox or a@h so im just trying ot make some sence of the huge amount of stuff it puts into conf files |
| 09:50.58 | E-bola | ohh sory :) |
| 09:52.07 | walld | FuriousGeorge:Yes. To try and get my head around the inner workings/copnfigs prior to going live with a standalone *. |
| 09:52.50 | E-bola | i would recomend not starting with any of the gui builds |
| 09:52.50 | walld | FuriousGeorge: The BRI isn't an issue as the gateway talks SIP to *. |
| 09:52.53 | FuriousGeorge | walld: i cant comment on your strategy because i started by learning to do it manually, and never moved on to a GUI way of doing it |
| 09:52.57 | E-bola | in my oppinion they are just confusing |
| 09:53.15 | FuriousGeorge | and, i would think, inherently less flexible |
| 09:53.25 | FuriousGeorge | and i can tell you from experience, a bitch to debug |
| 09:53.45 | FuriousGeorge | walld: take for example now that i ask you what tech you use to dial, and you got no idea |
| 09:53.54 | FuriousGeorge | walld: perhaps its capi? |
| 09:54.57 | walld | FuriousGeorge: I'm begining to realise the error of my ways :) |
| 09:57.51 | FuriousGeorge | those can be set into groups, but otherwise you could hack an array with the dialplan similar to this: http://pastebin.ca/394431 |
| 09:58.21 | FuriousGeorge | so if you have some basic knowledge of simple programming you might could see what is going on there |
| 09:59.03 | FuriousGeorge | that attempts to use 3 different VOIP providers but if it doesnt work it makes a POTS call |
| 10:00.44 | FuriousGeorge | you see, when you use DIAL, you generate a variable DIALSTATUS that is set based on how it goes with the dial |
| 10:01.13 | walld | FuriousGeorge: Thanks for that. It has a better logic than my current setup. The problem with my current config is that when BUSY is returned it just returns BUSY to the endpoint rather than trying an alternate trunk. |
| 10:02.42 | FuriousGeorge | it can be ANSWER,NOANSWER,BUSY,CONGESTION,CHANUNAVAIL. since i specified what to do for busy, and noanswer, and i know that an answer will terminate the call after hangup (really, the call will go to an h extension that i didnt specify), then i know that _s-. will catch all the other conditions (unavail, or congestion) that might be fixed by using another provider |
| 10:03.25 | FuriousGeorge | and i know after 3 im out of providers so try a different tech all together |
| 10:04.49 | walld | The current macro I'm using trys to handle it this way - |
| 10:04.53 | walld | exten => s,22,Goto(s-${DIALSTATUS},1) |
| 10:04.53 | walld | exten => s,108,Noop(max channels used up) |
| 10:04.53 | walld | exten => s-BUSY,1,NoOp(Trunk is reporting BUSY) |
| 10:04.53 | walld | exten => s-BUSY,2,Busy() |
| 10:04.53 | walld | exten => s-BUSY,3,Wait(60) |
| 10:04.54 | walld | exten => s-BUSY,4,NoOp() |
| 10:05.19 | walld | without trying any other trunk/provider |
| 10:05.35 | *** join/#asterisk Alex (i=hauntedu@gentoo/user/Alex) |
| 10:05.38 | *** part/#asterisk Alex (i=hauntedu@gentoo/user/Alex) |
| 10:05.58 | FuriousGeorge | walld: this is one of those situations where i would think a@h does what you want to do, but it might not do EXACTLY what you want EXACTLY how you want it |
| 10:06.11 | FuriousGeorge | that is not to say that your/my method is necessarily better |
| 10:06.27 | FuriousGeorge | but until you learn yourself some dialplan, you wont be in a position to comment either |
| 10:07.06 | FuriousGeorge | if your not interested in learning yourself some dialplan you may want to ask in the a@h irc channel what your options are |
| 10:07.43 | FuriousGeorge | you could always just learn some dialplan and chnage the one a@h generates, but i would think you must eventually break a@h to some extent doing that |
| 10:07.54 | FuriousGeorge | good luck with whatever you decide to do |
| 10:08.11 | walld | I'd rather learn the dialplan. Many thanks for your help :) |
| 10:08.30 | FuriousGeorge | np |
| 10:09.34 | sumasuma | how do i make a threewaycall with asterisk ? |
| 10:09.39 | FuriousGeorge | if you have even a rudimentary understanding of simple logic it should be no sweat for you |
| 10:09.45 | FuriousGeorge | sumasuma: what phone |
| 10:09.54 | sumasuma | FuriousGeorge: i use landline |
| 10:10.07 | FuriousGeorge | analog phone? |
| 10:10.26 | sumasuma | FuriousGeorge: yes |
| 10:10.32 | sumasuma | FuriousGeorge: analog line |
| 10:10.40 | FuriousGeorge | a call comes in/you make a call |
| 10:10.49 | FuriousGeorge | hit flash |
| 10:10.49 | sumasuma | nope |
| 10:10.56 | sumasuma | a call comes in |
| 10:11.03 | FuriousGeorge | i meant in either case |
| 10:11.06 | FuriousGeorge | do this: |
| 10:11.08 | sumasuma | ok |
| 10:11.09 | FuriousGeorge | hit flash |
| 10:11.29 | FuriousGeorge | you then place the original call on hold, and you hear a dialtone again on the ohone |
| 10:11.30 | sumasuma | when i hit flash then if i make a new call it give channel congested |
| 10:11.54 | sumasuma | FuriousGeorge: i want to use asterisk to automate this one |
| 10:11.55 | FuriousGeorge | sumasuma: ok so you have one landline, does it support three way calling itself? |
| 10:12.10 | sumasuma | FuriousGeorge, yes it is a vonage line and it supports |
| 10:12.35 | *** join/#asterisk ai-a[awol] (n=jake@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 10:12.40 | sumasuma | i connected itthrough one of the digium TDM400P card |
| 10:12.40 | FuriousGeorge | how do you make it work with vonage |
| 10:12.43 | rrrobert | i have explicitely enabled the snmp in asterisk1.4. by editing /etc/asterisk/res_snmp.conf. by the asterisk mib doesnot fetches me any value |
| 10:13.06 | FuriousGeorge | sumasuma: if i were you i would get a real voip provider to use with asterisk |
| 10:13.37 | sumasuma | FuriousGeorge: vonage supports sip or any virtual interface with asterisk ? |
| 10:14.26 | E-bola | exten => s,2,playback(custom/1000) |
| 10:14.28 | rrrobert | how can i access the asterisk subagent value |
| 10:14.34 | E-bola | what does the custom/1000 syntax mean? |
| 10:14.53 | ai-a[awol] | Hello, I am looking for a SIP <-> Phone service provider in Hong Kong. |
| 10:15.27 | FuriousGeorge | sumasuma: vonage purposely locks/breaks thier sh!t so it wont work with other devices |
| 10:15.59 | FuriousGeorge | sumasuma: how does three way calling work on a vonage line? |
| 10:16.02 | FuriousGeorge | is it just like a normal analog line where you flash the line and start another call |
| 10:16.07 | sumasuma | yes |
| 10:16.22 | sumasuma | it is just like other normal lines |
| 10:17.04 | FuriousGeorge | so then by flashing on your handset you are pulling up another "line" from asterisk, but that dialtone you hear is generated by asterisk, not by the vonage device when its indicating you have a new "line" from which to start a three way call |
| 10:17.48 | sumasuma | no problem |
| 10:18.03 | sumasuma | FuriousGeorge: i will give you the sequence |
| 10:18.17 | sumasuma | http://pastebin.ca/394450 |
| 10:19.00 | FuriousGeorge | the problem you have, i think, is that while there is an app_flash, it must be called from the agi (perhaps from a script listening for a dtmf of some sort), because after all, if you are in a "dial" in your dialplan, you cannot call other apps while keeping that channel bridged |
| 10:20.26 | sumasuma | FuriousGeorge: any link, that how can i properly use it ? |
| 10:20.30 | FuriousGeorge | sumasuma: "show application flash" in the dialplan |
| 10:20.35 | FuriousGeorge | i mean, "in the CLI" |
| 10:20.56 | FuriousGeorge | as to how to use it in an AGI, that is a bit outside the scope of this conversation |
| 10:21.06 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 10:21.16 | FuriousGeorge | sumasuma: my advice to you remains to get a "real provider" and your problem is solved |
| 10:22.15 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 10:22.36 | FuriousGeorge | if nothing esle, accept the fact that three way calling is not feasible (or an ugly hack at best), and just get a 1 cent per minute provider knowing and enjoy your unlimited channels |
| 10:23.28 | FuriousGeorge | and use that other provider when vonage is all used up |
| 10:24.08 | FuriousGeorge | that's how those unlimited plans get you (btw). sure you get call waiting so it is like two channels for one person, but if you want two channels for two people, suddenly its 50 bucks a month, and so on |
| 10:24.38 | FuriousGeorge | doesnt scale very well |
| 10:24.41 | FuriousGeorge | later all |
| 10:24.42 | rrrobert | is there any difference b/w runnging asterisk with command "astrisk" and "service asterisk start" |
| 10:24.45 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 10:25.16 | FuriousGeorge | one runs a script, the other runs asterisk in that shell (and so asterisk thread dies with the shell) |
| 10:25.49 | FuriousGeorge | you should compare the script and "safe_asterisk" command, and to know the answer to that look at your distro's script (which is not like my distro's, so i cant help you there) |
| 10:25.51 | rrrobert | i want the asterisk subagent to run |
| 10:26.04 | FuriousGeorge | script and "safe_asterisk" probably do that |
| 10:26.34 | sumasuma | FuriousGeorge: thanks for that |
| 10:27.05 | FuriousGeorge | personally i use the scripts, but inevitably end up using safe_asterisk as soon as i stop * and (for some reason) forget to use the scripts |
| 10:27.12 | FuriousGeorge | i guess cuz its less key strokes |
| 10:27.17 | FuriousGeorge | never given me a problem |
| 10:30.11 | rrrobert | FuriousGeorge: how could i configure the asterisk snmp subagent |
| 10:30.19 | gr0mit | any users of voip based in South Africa here? |
| 10:30.47 | rrrobert | i have tried to configure it by editing res_snmp.conf. |
| 10:30.57 | FuriousGeorge | rrrobert: dunno, never tried |
| 10:31.27 | rrrobert | any one tried to run the snmp subagent of asterisk? |
| 10:31.40 | rrrobert | FuriousGeorge: thx anyways |
| 10:32.17 | FuriousGeorge | np.. ask again later. someone will know |
| 10:33.56 | rrrobert | basicall i want to monitor the sip interface and call running over it. |
| 10:34.10 | rrrobert | for your knowlege : http://www.voip-magazine.com/content/view/2877/0/1/2/ |
| 10:37.17 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:37.28 | JT | FuriousGeorge: vonage ATAs do the "3 way calling" |
| 10:40.41 | sumasuma | JT: how can i do that with asterisk ? |
| 10:40.56 | JT | what sort of interface? |
| 10:41.21 | sumasuma | Vonage device is connected to one port of the TDM400P card |
| 10:41.29 | sumasuma | It is an FXO Port |
| 10:41.33 | sumasuma | single call works fine |
| 10:41.42 | sumasuma | I don't know how to get the second call working |
| 10:41.55 | sumasuma | when i dial the second call after a flash, it says Congestion |
| 10:42.11 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:43.23 | JT | maybe vonage does not allow 2 concurrent calls |
| 10:43.39 | sumasuma | i can make it with hand |
| 10:43.41 | sumasuma | works fine |
| 10:43.51 | JT | make it with hand? what? |
| 10:44.03 | sumasuma | with a normal analog phone |
| 10:44.07 | sumasuma | that works fine |
| 10:44.23 | JT | on the vonage device? |
| 10:50.12 | *** join/#asterisk mvictoras (n=soemone@85.75.198.73) |
| 10:50.21 | mvictoras | hi |
| 10:50.33 | mvictoras | have 2 questions |
| 10:51.23 | mvictoras | 1) I have buyed the g.729 license and installed it on my asterisk box. The problem is that the input gain is too low. Can I change it in some way ? |
| 10:51.48 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 10:51.50 | mvictoras | 2) Is there any difference in the audio quality between the licensed and open-source implementation of the g.729 ? |
| 10:52.13 | mvictoras | if anyone could help me on this, I would appreciate it very much |
| 10:52.18 | JT | i don't think the othe one is open source |
| 10:52.22 | JT | i think it is stolen source |
| 10:52.49 | mvictoras | whatever |
| 10:53.17 | Ahrimanes | if you need g729, you probably should go for the licensed one |
| 10:54.26 | mvictoras | i have already installed the licensed one |
| 10:54.27 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 10:54.36 | mvictoras | already got it from digium |
| 10:54.43 | mvictoras | the problem is that the gain is too low |
| 10:54.47 | JT | mvictoras: there's no need to be snappy |
| 10:54.54 | mvictoras | i can't hear the other end |
| 10:55.09 | Ahrimanes | hm doesnt sound like a g729 issue |
| 10:55.22 | mvictoras | ok sorry JT |
| 10:55.40 | mvictoras | let me explain |
| 10:55.46 | mvictoras | I have 2 voip-providers |
| 10:56.52 | mvictoras | the first uses a variety of codecs, and i have configured my *box to use g.726 |
| 10:57.01 | mvictoras | the second one uses ONLY g.729 |
| 10:57.56 | mvictoras | the problem is that when using the second, the input gain on the voip-phones is a little lower than the first one |
| 10:58.25 | kanelbullar | Hi guys, we are experiencing an odd behavior with an NFAS access. We are using a dual span Digium card and getting green lights on both spans. But no B channels come up and therefore it is not possible to make any calls. Has anyone experienced something similar or have any clue regarding what may be happening here? |
| 10:58.26 | mvictoras | if i raise the voip input gain, then it will raise for the both providers, i don't want to. |
| 10:59.07 | mvictoras | so i am asking if there is any option is sip.conf, to raise only the input gain, like in zapata.conf or like in misdn.conf (rxgain) |
| 10:59.24 | mvictoras | i hope i explained it neatly |
| 10:59.50 | Ahrimanes | kanelbullar: hm, maybe signalling issues? |
| 11:00.05 | Ahrimanes | kanelbullar: are you doing pri_cpe or pri_net ? |
| 11:00.15 | kanelbullar | Pri_cpe |
| 11:00.18 | Ahrimanes | ok |
| 11:00.23 | *** join/#asterisk shinux__ (n=shinux@41.222.64.203) |
| 11:00.30 | Ahrimanes | any log output about signalling? |
| 11:00.38 | kanelbullar | Using the cards in T1 mode and signalling=national |
| 11:02.13 | kanelbullar | We are only getting alarms like this: Mar 13 16:14:33 NOTICE[18842] chan_zap.c: Alarm cleared on channel 1 |
| 11:02.40 | kanelbullar | for all the channels in both spans. But we don't see any messages stating the B channels came up. |
| 11:02.52 | JT | do you mean D channels? |
| 11:03.43 | kanelbullar | Not quite, we are also getting similar messages for the remaining channels: Mar 13 16:08:49 NOTICE[18587] chan_zap.c: Alarm cleared on channel 2 |
| 11:03.46 | shinux__ | hey is there an asteriskNOW channel? |
| 11:04.15 | kanelbullar | We are using an NFAS configuration with a primary D channel and a backup D channel (channels 24 and 48) |
| 11:05.37 | kanelbullar | But there seems to be a problem with these as well, since they are only active for a few seconds and toggle |
| 11:05.40 | kanelbullar | Mar 13 16:08:49 NOTICE[18586] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 |
| 11:06.23 | kanelbullar | Mar 13 15:37:09 NOTICE[17173] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 2 |
| 11:12.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:17.00 | tzafrir | "no more alarm": now it is ok, right? |
| 11:19.42 | mvictoras | none knows any solution for the g.729 gain ? |
| 11:21.52 | *** join/#asterisk grndslm (n=grndslm@host-64-234-4-113.nctv.com) |
| 11:23.22 | *** join/#asterisk Gled|Working (n=sguser@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
| 11:24.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:25.45 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 11:30.29 | kanelbullar | We thought it should be ok, but it doesn't appear to be. We are getting that message periodically for D channel 24 and for D channel 48. |
| 11:30.55 | kanelbullar | And when we attempt to make a call, we get the following: |
| 11:30.57 | kanelbullar | Mar 13 16:09:18 WARNING[18586] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! |
| 11:30.58 | kanelbullar | Mar 13 16:09:33 NOTICE[18616] app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 11:31.54 | kanelbullar | Even though the D channel appears to be active, the error says there are no such channels available and attempts to use the D channel that was supposed to be active... |
| 11:32.28 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 11:35.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:39.07 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 11:44.53 | grndslm | you guys know of any wireless routers with a rj11 jack...that's compatible with asterisk?? |
| 11:48.12 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:53.53 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:53.57 | *** join/#asterisk nfi|ermes (n=ermes@217.220.121.62) |
| 11:54.01 | nfi|ermes | hi all |
| 11:54.18 | nfi|ermes | voip-info is falling down, ... |
| 11:54.35 | DrukenLPY | still? |
| 11:55.18 | nfi|ermes | y |
| 11:55.33 | *** join/#asterisk nachoguy (n=boster@208.47.211.203) |
| 11:56.15 | nfi|ermes | if to call through zaptel channel i use Zap/g1, with misdn what should i use ? |
| 11:56.24 | JT | misdn |
| 11:56.36 | nachoguy | I'm calling the math function like this MATH(VAR2,${VAR1}+2000,i) and still getting a float. what am I doing wrong? |
| 11:57.48 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:01.17 | *** join/#asterisk gripner (n=leif@195.178.169.154) |
| 12:01.36 | gripner | hey, anyone up to speed on asterisks compny directory? |
| 12:03.01 | *** join/#asterisk andyd (n=andy@grumps.nosignal.org) |
| 12:03.51 | kanelbullar | Can NFAS be used with e&m signaling? |
| 12:04.22 | JT | i don't think so |
| 12:04.37 | andyd | if you want to advertise myname@sip.myclub.com as an available way to reach you, would you put a [myname] block for all people in your club (only 10), withf type=peer, host=dynamic, context=myname, and then a [myname] context for each in extensions ..... or is there a better way of doing it someone can point me to ? |
| 12:07.20 | grndslm | you guys know of any wireless routers with a rj11 jack...that's compatible with asterisk?? |
| 12:07.37 | creativx | grndslm: keep asking and someone might start manufacturing one |
| 12:08.10 | grndslm | creativx: i wish |
| 12:08.17 | grndslm | creativx: i take it you don't know of any then? |
| 12:08.40 | creativx | good call ;) |
| 12:08.48 | grndslm | i was going to setup an asterisk box...but then realized that the power requirements really would be too much for my apt since i've already got a cell phone |
| 12:08.56 | grndslm | damn |
| 12:08.58 | creativx | i only use sip->asterisk->iax2.. |
| 12:09.12 | creativx | all-ip telefoony |
| 12:09.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:09.36 | JT | just combine an AP and an ATA |
| 12:09.40 | JT | with ethernet |
| 12:09.49 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 12:10.11 | masked | netgear |
| 12:10.16 | masked | have a wifi router |
| 12:10.19 | JT | netshit |
| 12:10.21 | grndslm | hah |
| 12:10.26 | grndslm | i hate netgear too |
| 12:10.34 | grndslm | and dlink |
| 12:10.38 | *** join/#asterisk shinux_ (n=shinux@41.222.64.203) |
| 12:10.39 | grndslm | both can suckz it |
| 12:10.41 | masked | dlink is trash |
| 12:10.44 | JT | yeah |
| 12:10.53 | JT | netcomm is awful |
| 12:11.01 | masked | some netgear is good |
| 12:11.04 | masked | flash it wint openwrt |
| 12:11.12 | masked | good atheros stuff in there |
| 12:11.20 | masked | yeah |
| 12:11.25 | grndslm | masked: is there a netgear with rj11 jack? |
| 12:11.44 | grndslm | i really don't want a whole desktop to run asterisk |
| 12:11.46 | masked | it's got a built in ata |
| 12:11.50 | grndslm | hmm... |
| 12:11.51 | masked | which is that, fxs? |
| 12:11.53 | grndslm | gotta model no? |
| 12:11.57 | masked | i forget which is which |
| 12:12.01 | JT | got google? |
| 12:12.03 | grndslm | heh.. |
| 12:12.11 | masked | 614 |
| 12:12.15 | DrukenLPY | why do you need a wireless router with phoneport? |
| 12:12.17 | JT | fxs or fxo anyway? |
| 12:12.28 | masked | wgt-613u i think! |
| 12:12.42 | masked | JT: 21:11 < masked> i forget which is which |
| 12:12.50 | masked | for hooking in a phone. |
| 12:13.21 | andyd | any ideas for the question i asked ? |
| 12:13.59 | grndslm | DrukenLPY: space & power |
| 12:14.15 | grndslm | just seems easier to have all my networking stuff into one tiny device |
| 12:14.41 | grndslm | maybe it would be better to keep 'em separate... |
| 12:14.43 | grndslm | i dunno |
| 12:14.47 | JT | masked: grndslm has been asking for an RJ11 port, didn't specify FXS or FXO |
| 12:14.49 | DrukenLPY | why all the concern about power and space? you live in a carbord box with a single extension cord? |
| 12:14.52 | grndslm | i haven't used asterisk, so i dunno how reliable it is |
| 12:15.06 | grndslm | JT: i'm not sure what the difference is |
| 12:15.09 | JT | DrukenLPY: he's got a good point, power = heat |
| 12:15.12 | JT | ~fxsfxo |
| 12:15.25 | jbot | [fxsfxo] an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 12:15.25 | masked | fxo |
| 12:15.25 | *** join/#asterisk Merlin83b (n=Merlin83@office.34sp.com) |
| 12:15.39 | masked | lol |
| 12:15.40 | masked | ok |
| 12:15.46 | masked | so it's a fxs port. |
| 12:16.00 | grndslm | fxs fo sho |
| 12:16.20 | masked | yep it's got sip account info in the web interface |
| 12:16.30 | grndslm | awesome... |
| 12:16.32 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:16.38 | grndslm | will have to research some more then |
| 12:16.39 | masked | just put your asterisk server details in there |
| 12:16.50 | JT | better see if anyone else has used its sip function before |
| 12:16.58 | JT | it's netgear remember |
| 12:17.09 | grndslm | well, i'd put openwrt on it |
| 12:17.09 | DrukenLPY | linksys has a router and ata in one, but i don't think it's wireless... unless they have put the ata in the wireless and made a wrt31p2 |
| 12:17.14 | masked | yeah |
| 12:17.16 | masked | i have one |
| 12:17.19 | masked | but have never used it. |
| 12:17.26 | grndslm | hmm.... |
| 12:17.27 | JT | i know |
| 12:17.28 | grndslm | well, thanks |
| 12:17.48 | DrukenLPY | masked, you got an rt31p2? |
| 12:18.04 | masked | no |
| 12:19.12 | masked | grndslm: your probably better getting a soekris or alike board and a ip phone |
| 12:19.27 | masked | i use a wifi voip phone |
| 12:19.46 | grndslm | i actually just screwed up my wrt54g because i thought it was my newer asus router with 2 usb ports..and changed the variable for the memory location |
| 12:19.50 | masked | don't even have a phone line connected |
| 12:20.12 | grndslm | and now i am down one router and still can't get the asus to work 100% of the time, it's flaky for some reason |
| 12:20.21 | JT | i think you can unbrick the wrt54g |
| 12:20.25 | *** part/#asterisk andyd (n=andy@grumps.nosignal.org) |
| 12:20.38 | grndslm | JT: with a jtag cable...which i have, but really don't know how to use...i'm in my first digital devices class |
| 12:20.43 | masked | if boot_wait is on it's easy to unbrick |
| 12:21.09 | masked | shouldn't need a jtag to unbrick it |
| 12:21.11 | grndslm | masked: i've tried tftp and most anything i could think of |
| 12:21.24 | grndslm | i really don't know if i set boot_wait to on or not |
| 12:21.25 | masked | hmm it's been a while |
| 12:21.32 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 12:21.32 | grndslm | was a long time since i first installed openwrt on it |
| 12:21.33 | masked | did you tftp to 192.168.1.1 always? |
| 12:21.38 | grndslm | yup |
| 12:21.39 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 12:22.10 | masked | maybe just the fs is stuffed and failsafe + firstboot will fix it... |
| 12:22.12 | *** join/#asterisk awk (n=awk@blackhole.security.web.za) |
| 12:22.16 | *** join/#asterisk lilwookie (n=lilwooki@30-82-252-216-static.enter-net.com) |
| 12:22.17 | awk | hmm, anyone have come across this problem, I have a sip trunk on my 1 asterisk box, poiting to my 1 of my own carriers, right. I have a dial prefix of say 1, and stripped the 0 once passing to this trunk, so I try dialing |
| 12:22.22 | awk | <PROTECTED> |
| 12:22.25 | awk | <PROTECTED> |
| 12:22.26 | grndslm | problem is that the cfe bootloader or whatever doesn't even come up because it doesn't know where the memory location begins |
| 12:22.29 | awk | <PROTECTED> |
| 12:22.41 | JT | nice, a copy and paste problem |
| 12:22.42 | awk | err, no idea how that came out like that |
| 12:22.53 | grndslm | i know |
| 12:22.58 | masked | grndslm: ohh i did something similar to one of my netgears |
| 12:23.11 | JT | probably the whole not typing it in the irc program thing :P |
| 12:23.15 | masked | grndslm: i put stuff on the wrong mtd's |
| 12:23.22 | tdi | Inez: |
| 12:24.32 | grndslm | masked: yea, i'm on spring break now so i'm gonna try to see if i can get this asus wl-500gP working after i cook my woman an omelet |
| 12:25.02 | grndslm | i'll be back, i'm definitely going to need some guidance with the whole one device or separate ata idea |
| 12:26.07 | lilwookie | is there a way to test an extension status.. from the dialplan? I need to check if the ext is busy or not |
| 12:27.14 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:27.57 | awk | see it doesn't allways do the attempting native bridge between the boxes |
| 12:28.04 | awk | any ideas why it doesn't attempt to do the bridging? |
| 12:28.15 | awk | sometimes it does, sometimes it doesn';t ? |
| 12:29.22 | *** join/#asterisk viperdude (n=jon@195.74.96.120) |
| 12:29.30 | viperdude | hi |
| 12:30.07 | viperdude | has anyone seen this behaviour? After pressing the "Submit All Changes" on a sipuare 1001 it loses all of the "Line 1" info |
| 12:30.36 | awk | upgrade the fireware :) |
| 12:30.49 | viperdude | awk: is it a known problem? |
| 12:31.00 | awk | no, i'm not sure |
| 12:31.04 | awk | just playing with you |
| 12:31.44 | sudhir492 | Anyone knows when the wiki will be up? |
| 12:34.38 | awk | isn';t there a mirror? |
| 12:34.50 | sudhir492 | where is the mirror? |
| 12:36.49 | Inez | ? |
| 12:36.53 | Inez | tdi: |
| 12:37.03 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1167878982.dsl.bell.ca) |
| 12:38.08 | awk | thats what i'm asking |
| 12:38.17 | awk | i'm sure there is more than enough people here willing to mirror |
| 12:39.52 | E-bola | sure is anoying its down |
| 12:40.50 | florz | and their copyright policy is annoying, too, yeah |
| 12:42.10 | sudhir492 | This is the first time I experienced it down. Considering how important it is to me and to the community, I am willing to contribute my mite towards a mirror |
| 12:43.30 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:48.24 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 12:49.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:52.25 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:52.26 | *** join/#asterisk seele_ (n=seele@208.35.116.20) |
| 12:52.27 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 12:52.54 | *** join/#asterisk TaiSHi (n=juanma@zion.dattaweb.com) |
| 12:53.00 | TaiSHi | Hello everyone |
| 12:53.06 | TaiSHi | I'm testing asterisk on a dev-server |
| 12:53.16 | TaiSHi | Planning on switching to a higher-perfomance server in near future |
| 12:53.27 | TaiSHi | Am I able to re-register g729 once I move server ? |
| 12:53.35 | TaiSHi | (bought 3 licences yesterday) |
| 12:54.25 | seele_ | how can I make a 3 minutes call time limit for especific extension??? |
| 12:56.10 | viperdude | TaiSHI: if you bought the g729 codec from Digium read the licence |
| 12:56.22 | TaiSHi | viperdude: I'm reading the README |
| 12:56.25 | TaiSHi | Already solved it |
| 12:56.33 | viperdude | it talks about switching servers |
| 12:56.41 | TaiSHi | I can re-register it once w/o authorization from Digium |
| 12:56.48 | viperdude | seele: set the calll length to 180 seconds |
| 12:56.49 | TaiSHi | From there on, I have to contact them. |
| 12:57.40 | TaiSHi | viperdude: <3 thanks |
| 12:57.47 | viperdude | YW |
| 12:58.10 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 12:59.08 | seele_ | viperdude, where?? |
| 13:00.19 | Ch0Hag | What does the 'notify' application do? |
| 13:00.30 | viperdude | seele: its part of the DIAL cmd |
| 13:00.42 | TaiSHi | seele_: |
| 13:00.44 | TaiSHi | oops |
| 13:00.50 | TaiSHi | seele_: core show application dial |
| 13:00.56 | Ch0Hag | Or, is there a resource other than voip-info.org, which appears to be down, which details this sort of information. |
| 13:01.04 | TaiSHi | www.asteriskguru.com |
| 13:01.17 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 13:01.44 | seele_ | viperdude, ok ... this works for specific extension ?? |
| 13:01.59 | JT | seele_: L |
| 13:02.13 | viperdude | seele_: are you wanting to limit the incoming call to the extension or the outgoing call from the extension? |
| 13:02.27 | JT | florz: what's the go with the copyright policy? |
| 13:02.29 | seele_ | viperdude, the outgoing call |
| 13:02.45 | TaiSHi | Ch0Hag: the above message was for you |
| 13:02.48 | JT | seele_: the answer is L |
| 13:03.17 | viperdude | ok so on exten => 9._,1,Dial(SIP/xxxxxxx@provider,,120,S180) |
| 13:03.28 | viperdude | the S180 limits the call to 180 seconds |
| 13:03.34 | Nobbie | hi, when running IAX2 connections to multple peers from one system, can you run both IAX2 trunks on the same UDP Port ? |
| 13:03.37 | TaiSHi | Mmm, nice to know =D |
| 13:03.46 | Ch0Hag | Yes I saw, sorry. |
| 13:03.49 | Ch0Hag | Cheers. |
| 13:03.51 | JT | viperdude: are you sure? |
| 13:04.04 | viperdude | works for me |
| 13:04.16 | JT | the dial parameter for call length limit is L |
| 13:04.17 | Ch0Hag | I'm trying to read a heavily top-posted email and it's taking all my powers of concentration. |
| 13:04.23 | *** join/#asterisk welby (i=welby@gateway/tor/x-b322ee87452f7bb3) |
| 13:04.45 | seele_ | viperdude, ok thanks |
| 13:04.56 | JT | seele_: HELLO |
| 13:05.03 | viperdude | S(x) - With this option you can specify a period of time after which the line |
| 13:05.03 | viperdude | will be hung up. The timer starts after the called person answered the call |
| 13:05.03 | seele_ | JT, hello |
| 13:05.14 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 13:05.14 | JT | seele_: how many times do i need to repeat |
| 13:05.22 | JT | seele_: the parameter is dial option L |
| 13:05.36 | viperdude | http://www.asteriskguru.com/tutorials/dial.html |
| 13:05.44 | viperdude | See option for "S" |
| 13:05.52 | seele_ | JT, sorry, L where?? |
| 13:06.03 | JT | seele_: as a dial option |
| 13:06.13 | JT | it might be in the book |
| 13:06.24 | viperdude | <JT>: S definately limits the call |
| 13:06.55 | TaiSHi | Ch0Hag: cheers :) |
| 13:07.04 | JT | L has much more options |
| 13:07.07 | TaiSHi | Ch0Hag: If you find out the NOTIFY thing, poke me |
| 13:07.30 | viperdude | <JT>: maybe but the question was how do you limit the call |
| 13:07.42 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 13:07.52 | JT | with L for Limit |
| 13:07.53 | JT | imagine that |
| 13:08.23 | drray | so it's L for limit |
| 13:08.23 | drray | er |
| 13:08.31 | viperdude | <JT> so is this site wrong? http://www.asteriskguru.com/tutorials/dial.html |
| 13:08.43 | JT | it's probably outdated |
| 13:08.50 | JT | take what asteriskguru says with a grain of salt |
| 13:09.01 | viperdude | well it works for me |
| 13:09.40 | [TK]D-Fender | viperdude: "show application dial" |
| 13:09.54 | drray | my asterisk squeaks at me that extensions.conf is obsolete :) |
| 13:09.57 | viperdude | yeah i know |
| 13:11.09 | *** join/#asterisk bmg505 (n=leon@196.209.248.114) |
| 13:11.15 | Ch0Hag | Asteriskguru gives the impression that the the honorific isn't really deserved. |
| 13:11.34 | Chris-NB | anyone tried autoprovisioning with a Thomson ST2030S phone? |
| 13:11.41 | JT | honorific? |
| 13:12.28 | Ch0Hag | Its description of that oh-so-esoteric command, Dial, is incomplete. |
| 13:12.43 | JT | yeah definately |
| 13:13.06 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:15.02 | michael-i | has anyone noticed the "x" file in /usr/local/share/asterisk? It's 6mb tar archive and I want to remove it unless it actually serves some purpose |
| 13:15.04 | ManxPower | That is why "show applications" and "show application X" in the CLI is the best documentation |
| 13:15.21 | michael-i | whoops! /usr/local/share/asterisk/sounds |
| 13:15.49 | ManxPower | michael-i: You must be running a custom install of asterisk |
| 13:16.22 | michael-i | ManxPower, just 1.4.1 on FreeBSD...nothing too custom |
| 13:17.56 | JT | make tar tell you its contents |
| 13:18.07 | JT | asterisk on freebsd is more custom than most ;) |
| 13:18.20 | JunK-Y | michael-i: yes, i already talk about that to Steve Murphy, at digium. |
| 13:18.32 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:18.40 | JT | oooh |
| 13:18.53 | JT | so THAT'S why asterisk has jumped 6MB in size |
| 13:18.57 | JT | to 1.4.1 |
| 13:19.42 | michael-i | JunK-Y, good deal :) I wondered too why my builds jumped up 6mb... |
| 13:19.43 | JunK-Y | * 1.4.1 uses sounds 1.4.6 |
| 13:20.06 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:21.43 | *** join/#asterisk testdriver12500 (n=pete@adsl-072-151-080-066.sip.rmo.bellsouth.net) |
| 13:21.45 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 13:22.28 | testdriver12500 | What's up with voip-info.org? I haven't been able to get in since yesterday. |
| 13:22.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:22.56 | JunK-Y | its down. |
| 13:23.10 | JunK-Y | call m. bush and let him know! |
| 13:23.16 | JT | testdriver12500: either someone doesn't know how to run a web site, or there's a real problem |
| 13:23.25 | JT | either way, it's a bad advertisment for the hosts |
| 13:24.15 | vooduhal | Quick question all. We have a customer with a PRI and when they try to make a call to a number that has been disconnected, instead of getting Bell's message, they get an internal "all circuits are busy" recording. What setting in zapata.conf would be causing this? |
| 13:24.59 | testdriver12500 | vooduhal: are you sure it isn't Bell? |
| 13:25.17 | JT | it wouldn't be bell |
| 13:25.24 | JT | PRIs can give congestion indications |
| 13:25.41 | JT | if it sounds like Alison, it's not Bell :) |
| 13:25.48 | testdriver12500 | good point.. |
| 13:25.50 | vooduhal | JT: It's definitely Alison. |
| 13:25.59 | vooduhal | Is there anyway to turn this off. |
| 13:26.08 | *** join/#asterisk skyhawker (n=info@a62-216-22-13.adsl.cistron.nl) |
| 13:26.16 | JT | mind you, you shouldn't hear alison unless you've set it to play her back somewhere in the dialplan |
| 13:26.27 | vooduhal | We thought they were having a PRI issue because we were getting channel UNAVAILABLE and after upgrading BUSYs. |
| 13:27.02 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 13:28.04 | vooduhal | Well, they are using FreePBX as a management interface. Could the Dial be returning prematurely? |
| 13:28.16 | JT | i already knew that :P |
| 13:28.26 | JT | it's more than a management interface, it changes everything |
| 13:28.41 | JT | nothing would play back alison unless you set it to |
| 13:29.53 | ManxPower | vooduhal: in /etc/asterisk/zapata.conf there is a setting regarding progress. set it to inband. This only works if the telco supports it |
| 13:30.20 | ManxPower | vooduhal: but that will not solve the issue with ALL OTHER DIGITAL CALL TYPES. |
| 13:30.23 | skyhawker | i am having problems with coneference rooms .. i have conference room 800 .. but it just dont work |
| 13:30.25 | vooduhal | K. Will try. |
| 13:30.48 | skyhawker | tells me not avaialble .. any ideas ? i checked google and only advice is to check if the zaptel is runinng |
| 13:30.49 | ManxPower | i.e. if you send a call out a provider (SIP or IAX) you won't hear the disconnect messsage from the telco either. |
| 13:31.09 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 13:31.12 | ManxPower | skyhawker: what is your problem? |
| 13:31.54 | skyhawker | ManxPower: trying to get conference rooms working on asterisk .. got the config done made the room password but when I call it i get address incomplete |
| 13:32.23 | JT | skyhawker: you should probably pastebin.ca your extensions.conf |
| 13:32.24 | skyhawker | the meetme.conf conference i mean |
| 13:32.53 | ManxPower | skyhawker: where do you get the address incomplete? |
| 13:33.03 | ManxPower | that message hasNOTHING to do with meetme. That is an extensions.conf issue. |
| 13:33.09 | skyhawker | JT: http://www.pastebin.ca/394619 is my extensions |
| 13:33.43 | skyhawker | ManxPower I think on asterisk .. but debug doenst show anything really (console) .. everything else works |
| 13:34.00 | skyhawker | ManxPower .. we have no IAX / ZAP lines .. we use a SIP gateway only |
| 13:34.20 | ManxPower | skyhawker: Address Incomplete means "I can't match the digits you dialed to an extension, but if you gave me more digits I might be able to match something" |
| 13:34.23 | JT | skyhawker: where is the MeetMe line, is it from an included file? |
| 13:34.58 | ManxPower | Also I can't help you because your extensions.conf does not contain all the required information. All those #includes make debugging basically impossible |
| 13:35.19 | ManxPower | skyhawker: are you using polycom phones? |
| 13:35.22 | skyhawker | JT: call me stupid .. i dont think i have it |
| 13:35.36 | skyhawker | JT: was i looking the whole time in the wrong place ? |
| 13:35.43 | JT | skyhawker: how's there meant to be a conference then? :P |
| 13:35.51 | JT | what extension are you meant to dial to get one? |
| 13:37.03 | skyhawker | JT : extension 800 is our conference room / meetme |
| 13:37.04 | JT | skyhawker: there's a lot of included files in there |
| 13:37.04 | skyhawker | JT: do you havea example maybe for conference / |
| 13:37.04 | JT | maybe it's in one of them |
| 13:37.04 | JT | did you write this stuff or is it freepbx or similar? |
| 13:37.23 | JT | i don't, but the book does |
| 13:37.24 | JT | ~thebook |
| 13:37.29 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:37.29 | skyhawker | JT: it is from thirdlane |
| 13:37.39 | JT | arrg |
| 13:37.44 | JT | damn proprietary dialplans |
| 13:38.24 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.31 | skyhawker | JT: gonna check out the book oreilly |
| 13:38.43 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 13:39.19 | ManxPower | Asterisk management GUIs are like being handcuffed to a zebra. It might sound cool at first, but it's not really a good idea., |
| 13:39.39 | HarryR | ManxPower, their a nightmare :) |
| 13:39.54 | HarryR | take a look at Telappliants VoIPoffice, it does a pretty good job of it though |
| 13:39.57 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 13:39.59 | skyhawker | ManxPower: i was starting to think the same. Problem is with the limitations ..then debugging the confs cause it just wont work the way you want |
| 13:41.39 | TaiSHi | Hey hey |
| 13:41.43 | TaiSHi | It's alive!! |
| 13:42.44 | TaiSHi | T_T |
| 13:42.47 | TaiSHi | After a long long time |
| 13:42.49 | TaiSHi | It works =) |
| 13:43.39 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:43.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:44.10 | JoNate | Hey guys, I'm still a complete n00b when it comes to asterisk and VoIP. Can someone let me know how many calls a T1 can handle at one time with G711 and G729? |
| 13:45.10 | ManxPower | JoNate: A data T-1 or a voice T-1 |
| 13:45.34 | Ahrimanes | around 24 711 and 120 729? |
| 13:45.39 | ManxPower | A voice T-1 (aka a Channlized T-1) can handle 24 calls. Because this is a PSTN service all calls are ulaw/G711 |
| 13:45.45 | JoNate | ManxPower: Just a data, I know voice can only handle 23 calls at once |
| 13:46.06 | ManxPower | JoNate: G711 takes about 80Kbps per call. |
| 13:46.09 | [TK]D-Fender | JoNate: Even less @ ulaw |
| 13:46.12 | *** join/#asterisk Strom_C (i=strom@nat/digium/x-fc6693315be14282) |
| 13:46.27 | JoNate | 80 KiloBITS right? |
| 13:46.27 | ManxPower | ~codecs |
| 13:46.31 | jbot | codecs is, like, http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 13:46.31 | JoNate | or Bytes? |
| 13:46.44 | JoNate | cool |
| 13:46.45 | [TK]D-Fender | JoNate: 80kbits |
| 13:46.49 | ManxPower | hmm, that what not what I wanted |
| 13:47.12 | Ahrimanes | ~bandwidth |
| 13:47.13 | jbot | it has been said that bandwidth is This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 13:47.16 | Ahrimanes | oh well |
| 13:47.29 | JoNate | ok...I'm like an idiot when it comes to this shit... |
| 13:47.40 | ManxPower | no, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 13:47.48 | JT | 85kbit/s for g.711 SIP imho. |
| 13:47.53 | ManxPower | what moron changed it to say 'allow=all'? |
| 13:48.23 | ManxPower | jbot: no, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 13:48.25 | jbot | ManxPower: okay |
| 13:48.31 | ManxPower | ~codecs |
| 13:48.32 | jbot | somebody said codecs was http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 13:49.35 | JT | you know you can correct it in private :) |
| 13:49.41 | ManxPower | JT I just did |
| 13:49.52 | JT | i saw it! :P |
| 13:51.22 | ManxPower | Hopefully the new wording is strong enough. |
| 13:51.24 | lilwookie | is there a way to test if an extension is BUSY from the dialplan before sending a call to it ? |
| 13:52.07 | ManxPower | lilwookie: "show application chanisavail" |
| 13:52.26 | ManxPower | But if you have call waiting enabled on the phones they won't be busy. |
| 13:52.42 | gr0mit | hey chaps. anyone got any exprience of passing voip traffic from a south african adsl line to UK on a 'shaped' connection from Telkom? |
| 13:54.37 | creature1 | if i want to register a client with asterisk from another network, then what ports would that client need to have opened? port 5060,5061 and the 10.000-20.0000 udp-ports? |
| 13:54.57 | creature1 | hope you understand my question :) |
| 13:55.16 | E-bola | I have a weird problem, my SIP provider routes all incoming calls to our static wan ip on our asterisk server, but for a new pair of numbers, we cant see which number is being called, the to field in the cli interface when u enable sip debugging just says s@ip-address |
| 13:55.27 | E-bola | can i trust the cli interface, or should i try a packet sniffer? |
| 13:55.46 | JT | no 5061 |
| 13:56.02 | Gido-E | E-bola try sip debugging |
| 13:56.09 | creature1 | JT: ok, the udp ports then? |
| 13:56.14 | JT | 5060 |
| 13:56.21 | E-bola | Gido-E: i did, as i said above |
| 13:56.23 | JT | 10000-20000, depending on rtp.conf |
| 13:56.24 | creature1 | 5060 and udp-ports? |
| 13:56.36 | creature1 | great, yeah they are configured like that |
| 13:56.40 | TaiSHi | What's the best free codec ? |
| 13:56.47 | JT | best |
| 13:56.47 | TaiSHi | Quality speaking, I dont really mind the B/W |
| 13:56.47 | JT | ? |
| 13:56.49 | JT | g.711 |
| 13:56.49 | TaiSHi | Mmm |
| 13:56.52 | TaiSHi | Better quality |
| 13:57.03 | JT | g.711 is what the global pstn uses |
| 13:57.14 | TaiSHi | ILBc ? |
| 13:57.15 | creature1 | JT: want to try registering with my asterisk server? |
| 13:57.23 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 13:57.24 | TaiSHi | I dont really mind the b/w |
| 13:57.25 | codazoda | I am using stream_file from the AGI. When I do this, execution seems to continue even while the file plays (i.e. it doesn't wait for the file to stop playing before it continues to the next instruction). GET DATA behaves differently in that it waits for the full file to play before continuing. Is this the normal behavior of STREAM FILE? |
| 13:57.33 | JT | TaiSHi: no, g.711 is the best quality for connecting to standard pstn |
| 13:57.40 | TaiSHi | No PSTN :) |
| 13:57.40 | JT | wtf, ilbc is a low bandwidth codec |
| 13:57.41 | E-bola | looks like: To: <sip:s@213.150.48.5> |
| 13:57.42 | TaiSHi | LAN =D |
| 13:57.44 | JT | it is lower quality |
| 13:57.52 | TaiSHi | Ok |
| 13:57.52 | JT | and uses tonnes of cpu |
| 13:58.01 | TaiSHi | I'm on a LAN, so, still 711 ? |
| 13:58.16 | JT | yeah unless you have everything supporting wideband audio |
| 13:58.46 | TaiSHi | Well, well... wtf? |
| 13:58.57 | JT | g.722 wideband audi |
| 13:58.57 | *** join/#asterisk tklettke (n=tklettke@ip67-155-33-163.z33-155-67.customer.algx.net) |
| 13:59.02 | JT | asterisk hardly supports it |
| 13:59.06 | TaiSHi | Mmm |
| 13:59.08 | JT | and all your phones must support it |
| 13:59.16 | TaiSHi | I'm not looking for something unsupported |
| 13:59.55 | JT | then why are you looking at anything other than g.711? |
| 13:59.57 | TaiSHi | It's just my boss hears a small noise when using a/ulaw |
| 13:59.58 | JT | it's uncompressed |
| 14:00.10 | TaiSHi | And he bugs me with it |
| 14:00.12 | creature1 | I would need help from someone to try to register with my asterisk server, anyones interested? |
| 14:00.22 | TaiSHi | creature1: query me |
| 14:00.28 | creature1 | TaiSHi: ta |
| 14:00.33 | TaiSHi | yw |
| 14:00.37 | JT | either the quality is so good that he can hear a small noise that is in the environment on the other end, or the phone is crap, or your boss is on crack |
| 14:00.39 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:00.50 | TaiSHi | I bet the second! |
| 14:00.54 | TaiSHi | I will keep a/ulaw |
| 14:00.59 | JT | what phone? |
| 14:01.12 | TaiSHi | xten-lite -> grandstream |
| 14:01.12 | b11d | just use slin end to end |
| 14:01.15 | b11d | :| |
| 14:01.16 | JT | ROFL |
| 14:01.19 | JT | grandstream |
| 14:01.25 | JT | lucky there's only a small noise |
| 14:01.27 | *** join/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com) |
| 14:01.32 | JT | never give your boss a grandstream |
| 14:01.42 | TaiSHi | It's a HT486 ATA |
| 14:01.56 | ManxPower | TaiSHi: G711 is what the telco uses for normal PSTN calls. |
| 14:02.05 | JT | a bit less bad than their phones maybe? |
| 14:02.06 | TaiSHi | ManxPower: Ahá |
| 14:02.08 | ManxPower | If your audio is not as good as the PSTN then you have some OTHER issue like timing |
| 14:02.18 | JT | ManxPower: or gs! |
| 14:02.26 | TaiSHi | I think it's more likely one of the computer's sound card |
| 14:03.01 | ManxPower | TaiSHi: Good lord! In the 7th level of Dante's Inferno they give Grandstream phones to telecom geeks! |
| 14:03.20 | ManxPower | TaiSHi: computer sound card???? |
| 14:03.35 | TaiSHi | ManxPower: I'm on one of the workstations |
| 14:03.45 | TaiSHi | And boss was upstairs on a HT486 ATA --> Hardphone |
| 14:03.47 | ManxPower | TaiSHi: Softphones and Grandstream phones. You are trying to get fired, right? |
| 14:04.30 | TaiSHi | u_U |
| 14:04.39 | TaiSHi | He wants softphones-only upstairs |
| 14:04.49 | TaiSHi | That's why he assigned me * project |
| 14:04.55 | ManxPower | TaiSHi: I would recommend that you start looking for another job. |
| 14:05.06 | TaiSHi | ManxPower: already on that u_U |
| 14:05.08 | TaiSHi | Still |
| 14:05.25 | ManxPower | The problem with softphones is that they rely on 1) Windows and 2) random computer hardware -- usually from the lowest bidder |
| 14:05.27 | JT | ~gs |
| 14:05.30 | jbot | gs is, like, South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 14:05.30 | TaiSHi | ITSP --> * --> Queue --> Cute girl operators |
| 14:05.56 | JT | ...why thanks |
| 14:06.00 | TaiSHi | ManxPower: He wont such to windows |
| 14:06.01 | designdream | I trunked with SPA3000 and wasn't pleased with the quality...is there a difference in quality if using a digium wildcard? |
| 14:06.05 | TaiSHi | (He wont pay licences) |
| 14:06.24 | ManxPower | designdream: the SPA3000 does not support trunking |
| 14:06.46 | ManxPower | And in fact, there is pretty much no SIP endpoints that support trunking. |
| 14:07.44 | ManxPower | designdream: Most quality problems with SPAs are fixed by changing the voice packet size from .30 to .20. |
| 14:08.11 | designdream | ManxPower: will i notice a quality difference between SPA and wildcard? |
| 14:08.32 | ManxPower | designdream: Define "quality". Sound quality, reliability? |
| 14:09.28 | designdream | i just feel like using the SPA to bring in the POTS was hackish... |
| 14:09.35 | ManxPower | designdream: you will notice a big difference in audio quality between an incorrectly configured SPA and a correctly configured Wildcard. You won't notice much difference in audio quality between a correctly configured SPA and a Wildcard |
| 14:09.55 | designdream | the audio quality wasn't bad.. i think i might have the wrong settings relating to voltages, impendence, tones, etc |
| 14:10.02 | ManxPower | designdream: oh, using an SPA to bring in POTS is a horrid hack for anything except maybe a home user. |
| 14:10.25 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:11.20 | designdream | They are wanting to can our polycom501's and go back to a traditional system... I just got our managed switch in with QoS... so i wanted to request ordering the wildcard and giving voip one more shot |
| 14:13.02 | *** join/#asterisk AF-Slash (n=AF-Slash@216-161-216-130.hlna.qwest.net) |
| 14:13.42 | JT | designdream: don't think you can swing it towards ISDN? |
| 14:14.13 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:14.13 | designdream | JT for two lines? |
| 14:14.52 | JT | designdream: i guess it depends where you are, we have BRI here for same price as 2 * analogue lines |
| 14:15.49 | *** join/#asterisk wunderkin (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 14:15.51 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:16.00 | designdream | I just have enough experience when it comes to voip to decide which direction to head in... |
| 14:16.30 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:16.42 | jart | what should i eat for lunch today |
| 14:16.45 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 14:17.37 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 14:17.50 | vooduhal | Fixed my problem. Removed the r option from the Dial line and now they get Bell's messages. |
| 14:19.02 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 14:19.45 | JT | ah |
| 14:20.13 | SuPrSluG | is the wiki down? or just me? |
| 14:20.18 | E-bola | its down |
| 14:20.23 | sevard | <PROTECTED> |
| 14:20.23 | SuPrSluG | k |
| 14:21.45 | *** join/#asterisk friedrich| (n=friedric@e177244220.adsl.alicedsl.de) |
| 14:21.50 | ManxPower | vooduhal: don't EVER use the r option to dial |
| 14:22.01 | JT | r for retarded |
| 14:22.17 | sav_mcfly | got a problem, sometimes chan_jabber kills the asterisk 1.4.1 proc, does anybody got this problem too? |
| 14:23.05 | JunK-Y | sav_mcfly: please se backtrace.txt and report the bug. |
| 14:23.50 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 14:25.22 | *** join/#asterisk vasquez (n=vasquez@85.183.64.6) |
| 14:26.07 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 14:26.17 | vooduhal | I don't. FreePBX decided it wanted to. |
| 14:27.48 | Ch0Hag | How can I indicate on a handset what line a call is coming in on? |
| 14:27.54 | Ch0Hag | Other than setting CALLERID(name) |
| 14:28.05 | dj-fu | 'line'? |
| 14:28.25 | Ch0Hag | line, extension, device, whatever. |
| 14:28.48 | creativx | DID |
| 14:28.50 | creativx | er.. DNID |
| 14:29.00 | suma | how to increase the volume of the channel ? |
| 14:29.09 | ManxPower | Ch0Hag: Line, device, and DID are ALL DIFFERENT |
| 14:29.20 | creativx | Ch0Hag: callerid(dnid) will in some cases give you the dialed number |
| 14:29.23 | ManxPower | suma: on zap use rxgain and txgain |
| 14:29.52 | suma | the problem is the call is coming from zap, but i don't have control over the zap |
| 14:30.13 | suma | anyway to increase it on my end or like automatic gain control ? |
| 14:30.56 | Ch0Hag | *I* know what the dialled number is. |
| 14:31.02 | Ch0Hag | I need some way to tell the handset. |
| 14:31.46 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@83.223.131.104) |
| 14:31.50 | aiksa[LV] | Hi everyone |
| 14:32.10 | aiksa[LV] | does anyone knows whats happened to voip-info.org wiki? |
| 14:32.33 | aiksa[LV] | it's closed forever or what |
| 14:32.34 | creativx | Ch0Hag: set the callerid name to the dnid ;) |
| 14:32.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:32.45 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:32.55 | JT | my googness, it seems #asterisk is now voip.info.org tech support |
| 14:32.58 | JT | goodness |
| 14:33.02 | JT | someone should topic it |
| 14:33.06 | JT | it's down, not sure why |
| 14:33.10 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 14:33.29 | aiksa[LV] | JT: no offence - just quick search on google didnt return anything |
| 14:33.41 | Ch0Hag | < Ch0Hag> Other than setting CALLERID(name) |
| 14:33.42 | aiksa[LV] | I hoped that people here would know of what have happened |
| 14:33.57 | aiksa[LV] | does it have any mirrors... etc. |
| 14:34.07 | florz | Maybe we should put an article in the wiki stating that the wiki is down? |
| 14:34.14 | nays85 | maybe it's gone... the owner is going to take all the info and write a book |
| 14:34.14 | aiksa[LV] | florz :P |
| 14:34.19 | darkskiez | hopefully they are upgrading the wiki software to allow to be reorganised a bit |
| 14:34.27 | darkskiez | it was a tangled mess |
| 14:34.37 | darkskiez | with no mechanism to organise it |
| 14:34.46 | aiksa[LV] | darkskiez: yup, but still you could digg up tons of usefull info there |
| 14:34.56 | darkskiez | it will be missed indeed if its gone |
| 14:35.14 | TaiSHi | ManxPower: Is there still problems with autopause = yes / ringinuse = no ? |
| 14:35.21 | florz | sure, the very problem is that you had to dig =:-) |
| 14:35.38 | JT | it makes me uneasy that the site was the pet project of 1 company |
| 14:35.49 | JT | who could indeed take it away |
| 14:36.12 | florz | Yeah, and the copyright policy was such that mirroring basically wasn't allowed. |
| 14:36.38 | florz | basically "all you put in here is ours" |
| 14:36.46 | Ch0Hag | It wouldn't be a very accurate book. |
| 14:36.53 | florz | hehe :-) |
| 14:37.06 | aiksa[LV] | Ch0Hag: but it would have certain things which was not avaialble anywhere else |
| 14:37.09 | nays85 | especially the news chapter |
| 14:37.24 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 14:38.22 | aiksa[LV] | so what, we should play with waybackmachine now? recover all the posts and make a community owned wiki instead? |
| 14:38.31 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 14:38.41 | nays85 | we need a wiki that has it's backend mirrored to 2-3 different voip entities (digium, voicepulse, etc) whether they're commercial or not, so all the info isn't lost in one shot and they may have varying degrees of community involvement, etc |
| 14:39.06 | HarryR | has voip-info gone down or something? |
| 14:39.13 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 14:39.30 | aiksa[LV] | HarryR: you guessed it. |
| 14:39.42 | florz | in particular the contents should be under some free license, so that mirroring isn't a problem anyway |
| 14:39.46 | HarryR | ah, I like the mirrored idea.. i'll throw some ideas at "The Boss" |
| 14:39.52 | nays85 | if one entity wants to throw a bunch of ads on there, whatever, people have somewhere else to go |
| 14:40.12 | aiksa[LV] | florz, CC perhaps? |
| 14:40.22 | *** join/#asterisk mihinomenest (i=OnYg@cerebus.clandestineresearch.com) |
| 14:40.42 | mihinomenest | lawton? |
| 14:40.58 | florz | aiksa[LV]: Well, there are quite a bunch of pretty different licenses under the label of creative commons, but some certainly might be sensible |
| 14:41.05 | TaiSHi | Is there still problems with autopause = yes / ringinuse = no ? I dont kinda want my ops to get dropped if they're attending calls |
| 14:41.11 | aiksa[LV] | nevermind the ads, still a valuable info can be dug up in case it's needed |
| 14:41.11 | *** join/#asterisk qdk (n=qdk@193.164.155.44) |
| 14:41.19 | *** join/#asterisk shodan- (n=shodan@ip158.96-113-216.pppoe1.joliette.intermonde.net) |
| 14:41.23 | HarryR | is voipinfo just down temporarily? |
| 14:41.24 | creativx | damn i love click2dial |
| 14:41.27 | HarryR | or for good? |
| 14:41.42 | aiksa[LV] | HarryR: if I only knew. |
| 14:41.48 | JunK-Y | if it doesnt work, its probably cause its down? |
| 14:42.04 | aiksa[LV] | Thats why I joined this chan anyway (after a pause of half a year or so) |
| 14:42.19 | jart | yo junk |
| 14:42.34 | jart | sup dawg |
| 14:43.24 | aiksa[LV] | florz: still that wiki would have to be built from the ground up. a lots of usefull info would be lost |
| 14:44.04 | aiksa[LV] | like mine miserable comments as regards proper configuration of kernel for mISDN channel |
| 14:44.08 | JunK-Y | jart: not much |
| 14:44.10 | florz | aiksa[LV]: Yeah, that's true. But better now than later? Plus, it would make sense anyway ;-) |
| 14:46.07 | florz | aiksa[LV]: And actually I somehow doubt that the creator of some entry can be forbidden to relicense it if there hasn't even anything been paid for it. |
| 14:46.44 | JT | a lot of stuff is pulled from asterisk show application |
| 14:46.51 | JT | and other common documentation |
| 14:47.04 | JT | anecdtes are the most original content |
| 14:47.09 | JT | anectdotes |
| 14:47.51 | florz | JT: Any clue how that stuff got in there? |
| 14:47.54 | JT | if things look grim with voip-info i can put up web hosting for people to start adding stuff |
| 14:48.00 | JT | i have enough servers |
| 14:48.05 | JT | florz: people added it? |
| 14:48.52 | af_ | is there some document about zaptel interface development? |
| 14:48.53 | florz | JT: Yeah, sure, but what people and with which authorisation from the authors? Or just "illegally"? |
| 14:49.01 | florz | JT: Like Pirate-Pasted? |
| 14:49.04 | JT | who knows |
| 14:49.21 | JT | i don't think it really matters to be honest, unless someone copied and pasted the book |
| 14:49.51 | Mercestes | I don't think there is much legal limitation on posting a WIKI on asterisk configurations and "how tos." |
| 14:50.06 | Mercestes | unless Digium just wants the support abuse. |
| 14:50.48 | [TK]D-Fender | load chan_masochism.so |
| 14:50.48 | [TK]D-Fender | OK |
| 14:51.23 | florz | JT: Well, it depends from which perspective. I guess Digium doesn't care, but from a purely legal perspective, could they for example sue voip-info.org or the people who put the stuff in there? |
| 14:51.42 | Mercestes | on what grounds?? |
| 14:52.04 | florz | Mercestes: Because the texts are copyrighted by Digium, I'd say? |
| 14:52.14 | Mercestes | what texts? |
| 14:52.26 | Mercestes | the examples ppl posted of the configs or the break outs of the various commands? |
| 14:52.42 | florz | Mercestes: The show application texts |
| 14:52.43 | Mercestes | being Asterisk is open source, I don't see how anything would be copyrighted as far as asterisk is concerned. |
| 14:52.52 | Mercestes | that's all open source stuff. |
| 14:53.18 | florz | Mercestes: Ohh, you seem to have some serious misconception as to what "open source" means. |
| 14:53.21 | [TK]D-Fender | Mercestes: OSS != not Copyrighted. GPL is just a DISTRIBUTION rights license |
| 14:53.23 | Mercestes | as long as you copied whatever "copyright" texts that accompanied the show application stuff, I dont' think there would be any grounds for any legal crap. |
| 14:53.47 | [TK]D-Fender | Mercestes: Indeed, you = confused |
| 14:53.52 | Mercestes | *shrugs* I've seen online man pages about a billion times. |
| 14:54.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:54.06 | [TK]D-Fender | florz: And I can write a bok on how to use Excel, so why not *? |
| 14:54.27 | [TK]D-Fender | florz: Your question was a little silly. |
| 14:54.27 | Mercestes | I'm not confused....you can sue someone for anything you want, including just to be an ass.... |
| 14:54.33 | drfreeze | I am building a production Asterisk box, and am looking for advice on which OS to use. |
| 14:54.35 | florz | Mercestes: Asterisk is definitely copyrighted, and if you license it under GPL, you have to adhere to its requirements when redistributing it or parts of it. |
| 14:54.42 | drfreeze | Right now, I am considering FC5, FC6 or CentOS. |
| 14:54.53 | [TK]D-Fender | Mercestes: You can sue... and lose, and get countersued for defamation, etc ;) |
| 14:54.57 | Qwell[] | drfreeze: I'd reconsider |
| 14:55.00 | drfreeze | Any suggestions? |
| 14:55.02 | JT | drfreeze: you mean distribution |
| 14:55.08 | Mercestes | .... |
| 14:55.09 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-0cc80d5ecbc79c18) |
| 14:55.15 | drfreeze | JT: ? |
| 14:55.25 | Qwell[] | isn't that right Deeewayne? |
| 14:55.27 | florz | tklettke: Hmm? Sure you can write a book on how to use asterisk!?! |
| 14:55.33 | florz | hmpf |
| 14:55.38 | drfreeze | Qwell[]: gotta a OS of choice you would like to share with me? |
| 14:55.38 | florz | [TK]D-Fender: Hmm? Sure you can write a book on how to use asterisk!?! |
| 14:55.39 | [TK]D-Fender | drfreeze: Out of the 3, CentOS is teh most stable and popularly used so you'd have a good base of people to refer to in case of problems. |
| 14:55.44 | Qwell[] | drfreeze: Linux |
| 14:55.46 | Mercestes | ok, let's get into a six page discussion on what "copyright" means. The d20 system is "copyrighted" too but you can redistribute /reprint that freely as long as you include the original copyrights, as well as use it for free and variate on it as you wish... |
| 14:56.14 | drfreeze | Qwell[]: which distro |
| 14:56.24 | Mercestes | So if by "copyright" you mean intellectually claimed, fine, but if your trying to say copyright means "cannot be reprinted" then your wrong, unless the copyright specifies that specifically. |
| 14:56.29 | Deeewayne | qwell: isn't what right? |
| 14:56.56 | florz | Mercestes: "The copyright" is a law that says so explicitly. |
| 14:56.58 | eald | Mercestes: no, by default copyrighted materials means: you can't do anything with this |
| 14:57.06 | Deeewayne | qwell: nice job on making the news: http://www.digium.com/en/mediacenter/news/ |
| 14:57.06 | Mercestes | drfreeze, : Avoid all three, go Gentoo |
| 14:57.12 | drfreeze | [TK]D-Fender: What OS do you see with * most commonly? |
| 14:57.13 | Qwell[] | huh? |
| 14:57.18 | TaiSHi | Is there still problems with autopause = yes / ringinuse = no ? I dont kinda want my ops to get dropped if they're attending calls |
| 14:57.22 | Qwell[] | bahahaha |
| 14:57.39 | drfreeze | Mercestes: what's so special about Gentoo? |
| 14:57.41 | Qwell[] | awesome |
| 14:58.02 | Mercestes | drfreeze, It works. |
| 14:58.12 | drfreeze | Mercestes: so do the others |
| 14:58.19 | JT | drfreeze: stop saying OS when you mean distro, the great majority of installed OSes for asterisk is linux |
| 14:58.29 | aiksa[LV] | or bsd |
| 14:58.39 | drfreeze | JT: you're right. My bad. |
| 14:58.39 | JT | i said great majority |
| 14:58.41 | Mercestes | drfreeze, gentoo works better. |
| 14:58.44 | JT | bsd is a minority |
| 14:59.00 | aiksa[LV] | however * BE had some problems with running on free bsd if i remember correctly |
| 14:59.04 | drfreeze | Mercestes: better than CentOS how? |
| 14:59.07 | aiksa[LV] | JT, ok :) |
| 14:59.21 | florz | Mercestes: any texts, music, software and the like that are copyrighted the moment they are created in most jurisdictions - and that means that basically noone but the author may use them |
| 14:59.22 | JT | asterisk in general has problems under bsd |
| 14:59.29 | JT | interfacing with hardware.. forget about it |
| 14:59.32 | florz | s/that// |
| 14:59.36 | [TK]D-Fender | drfreeze: I use either CentOS or Slackware personally |
| 14:59.41 | drfreeze | I admire those who are working and deploying * on BSD, but I'm don't have that much time. :) |
| 14:59.42 | aiksa[LV] | I have had a number of * installations on slack, so far no problem |
| 14:59.49 | JT | drfreeze: my personal preference is debian |
| 15:00.23 | Mercestes | florz: This conversation can stop now....seeing has voip-info covered systems other than asterisk to try and speculate that it is offline due to a copyright infringement is...well just that, speculation. |
| 15:00.44 | drfreeze | Ok, so it looks like we have votes for CentOS, Slackware, Debian and Gentoo. |
| 15:00.50 | [TK]D-Fender | drfreeze: Debian, CentOS, Slackware are all very popular bases for which it is typically easy to compile * as all the dependencies are easily obtainable or pre-installed by default. |
| 15:00.51 | florz | Mercestes: Uh? Has anyone speculated anything like that? |
| 15:01.00 | drfreeze | It's like a typical Linux discussion. :) |
| 15:01.03 | Mercestes | florz: And your also very wrong. A copyright can have any limitations specified by the author from "freedom to use as you will" to "mine" and anything inbetween. |
| 15:01.09 | JT | Mercestes: because you've had enough of the conversation it should "just stop" as you deem necessary? |
| 15:01.16 | [TK]D-Fender | drfreeze: Gentoo is NOT on my recommend list unless you are already fairly comfortable in managing it. |
| 15:01.21 | Mercestes | TROLLS!!!!! |
| 15:01.27 | eald | Mercestes: no, that what a license do |
| 15:01.29 | Mercestes | Trolls the lot of you. |
| 15:01.42 | florz | Mercestes: No, a copyright is created by law, not by an author |
| 15:02.03 | drfreeze | [TK]D-Fender: Is there any problem on getting the latest * install with CentOS? |
| 15:02.13 | Mercestes | lol |
| 15:02.34 | [TK]D-Fender | drfreeze: I've used CentOS 4.2-4.4 without issue. Not sure about the latest Betas, but I'd avaoid them for now. |
| 15:02.35 | aiksa[LV] | I know, I know. GWBush ate the voip-info.org and the half of the moon togeather with that. |
| 15:02.51 | Mercestes | drfreeze: I find use="asterisk pri mysql zaptel" emerge -av asterisk mysql asterisk-addons asterisk-sounds zaptel to be disgustingly easy for me. |
| 15:02.55 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 15:03.04 | JT | maybe there'll be a conserva-voip-info.org |
| 15:03.19 | Mercestes | drfreeze: I've just had very good luck with Gentoo and Asterisk and very poor luck with CentOS/Redhat and asterisk. |
| 15:03.21 | drfreeze | [TK]D-Fender: yeah, I'm just looking fo rthe latest release. |
| 15:03.54 | eald | Mercestes: sure apt-get install asterisk isn't that hard right? or you need a howto for that? |
| 15:04.14 | *** join/#asterisk blackmousepad (n=blackmou@71-13-69-254.static.aldl.mi.charter.com) |
| 15:04.19 | blackmousepad | I've got asterisk setup behind a nat firewall with port 5060/udp forwarded to it's internal ip address. my soft phone (X-Lite) is also behind a nat firewall. so its asterisk - nat - internet - nat - xlite. I can register, and place calls but no audio in either direction. what am I missing? |
| 15:04.19 | rkeels | Does anyone here have any expertise in the MWI and voicemail department. I have a really big issue that I am not sure how to address |
| 15:04.22 | [TK]D-Fender | drfreeze: 4.4 should work without issue and has for me in every install of it I've done for clients. I currently use Slackware on 1 work server (test), and home (switching to CentOS as soon as I get off my ass) |
| 15:04.39 | [TK]D-Fender | blackmousepad: ... |
| 15:04.40 | [TK]D-Fender | ~sipnat |
| 15:04.42 | jbot | sipnat is probably for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:04.42 | drfreeze | :) |
| 15:04.53 | JT | blackmousepad: ports 10000-20000 udp depending on rtp.conf |
| 15:04.53 | [TK]D-Fender | lol... yeah THAT's going to work ;) |
| 15:04.53 | TaiSHi | Is there still problems with autopause = yes / ringinuse = no ? I dont kinda want my ops to get dropped if they're attending calls |
| 15:05.01 | JT | blackmousepad: you're only forwarding signalling, no media |
| 15:05.11 | Mercestes | eald: troll. :P |
| 15:05.15 | eald | :D |
| 15:05.16 | *** join/#asterisk boch (n=fran@190.48.192.47) |
| 15:05.25 | [TK]D-Fender | blackmousepad: You need a pile of settings in sip.conf as well... can't really detail it all for you now though.... |
| 15:05.31 | rkeels | About ten of my poly430's out of over 100 are not giving the proper indiction of new-voicemail until two hrs later aproximately |
| 15:06.01 | Mercestes | It's distro warring, s'what you get when you ask "what distro should I use." I like gentoo because I dno't get anything I didn't specifically ask for so I get a "clean system." |
| 15:06.07 | Mercestes | I also like being "under the hood" so to speak. |
| 15:06.10 | TaiSHi | Mercestes: Your M changed... |
| 15:06.23 | boch | could you please giveme a practical example of CUT() func usage, cause i cant get it work |
| 15:06.26 | blackmousepad | JT I've added in 10000-20000 (which is in the rtp.conf) file and still broke. |
| 15:06.31 | Mercestes | TaiSHi: what? |
| 15:06.45 | JT | "still broke" is not very descriptive |
| 15:06.55 | aiksa[LV] | so somebody should start cleaning jbot references to voip-info ?:P |
| 15:06.56 | rkeels | Message waiting anyone. |
| 15:07.01 | [TK]D-Fender | blackmousepad: There are a LOT more settings to make, and 5060 needs to be forwarded as well (all portsa are UDP) |
| 15:07.16 | blackmousepad | JT Call connects but no audio. |
| 15:07.17 | rkeels | Am I the only one that has this problem |
| 15:07.25 | seele_ | how can I make a otgoing call time limit into a context ?? |
| 15:07.50 | JT | blackmousepad: nat=yes externip= localnet= insecure=very |
| 15:07.51 | [TK]D-Fender | rkeels: in voicemail.conf you should be able to set the MWI check frequesncy and make sure you are doing "mailbox=#@context" in the sip.conf entry |
| 15:08.10 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 15:08.13 | blackmousepad | I've got 5060 and 10000-20000 forwarded. and I've put in nat=yes in both the extension and sip_nat.conf |
| 15:08.31 | *** join/#asterisk FinboySlick (n=Miranda@207.134.8.202) |
| 15:08.37 | blackmousepad | JT I didn't have insecure=very I'll that a shot |
| 15:08.48 | [TK]D-Fender | sip_nat.conf .... oh JOY |
| 15:08.50 | [TK]D-Fender | ~freepbx |
| 15:08.52 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:09.01 | JT | blackmousepad: what about the other 2? |
| 15:09.02 | rkeels | I have both of those set. MWI check freqeuncy is 10 but that affects all of the phones not just a handfull |
| 15:09.06 | FinboySlick | on your everyday canadian residential line, should switchtype = national? |
| 15:09.16 | [TK]D-Fender | rkeels: And your box format is OK? |
| 15:09.17 | blackmousepad | Already have them. |
| 15:09.21 | JT | FinboySlick: analogue lines don't have a switchtype |
| 15:09.21 | Ch0Hag | Does anyone know much about snom phones? |
| 15:09.23 | *** join/#asterisk lymers (n=lymers@pool-72-64-106-15.dllstx.fios.verizon.net) |
| 15:09.26 | Assid | hey tkd |
| 15:09.34 | [TK]D-Fender | FinboySlick: switchtype is for T1 connections, not normal analog |
| 15:09.39 | rkeels | What format |
| 15:09.39 | [TK]D-Fender | FinboySlick: its irrelevent |
| 15:09.51 | FinboySlick | [TK]D-Fender: Thanks. |
| 15:09.55 | rkeels | Centos with no Gui |
| 15:09.56 | [TK]D-Fender | rkeels: mailbox=123@vmcontextname |
| 15:09.59 | lymers | okay, if my telcom lines had a little static that wasnt really noticablt on a analog phone... would asterisk be more sensitive to it, and make it worse? |
| 15:10.03 | Ch0Hag | And specifically - how can I make a particular light flash without multiple accounts. |
| 15:10.25 | Ch0Hag | lymers: Maybe. Especially if the volume is increased. |
| 15:10.25 | JT | FinboySlick: hey, i was first :P |
| 15:10.51 | lymers | Ch0Hag: anything i can do to filter that? |
| 15:11.00 | Ch0Hag | Turn the volume down. |
| 15:11.11 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:11.13 | Dr-Linux | Qwell |
| 15:11.20 | lymers | ill try, its still at the default lvl |
| 15:11.31 | Ch0Hag | I had that on a line where rxgain=6, set =0 and it was gone. |
| 15:11.32 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 15:11.45 | aiksa[LV] | :) here you go: http://web.archive.org/web/20060822011730/www.voip-info.org/wiki/ |
| 15:11.52 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:12.02 | lymers | which config file? |
| 15:12.29 | Ch0Hag | zap |
| 15:12.34 | lymers | ty |
| 15:12.35 | Ch0Hag | I think it's zapata.conf |
| 15:12.47 | aiksa[LV] | Ch0Hag: what do you think by particular light? line? |
| 15:13.39 | *** join/#asterisk watchy (n=watchy@work.gwhsi.com) |
| 15:14.29 | TaiSHi | Mercestes: your M is cap now :P |
| 15:14.33 | watchy | hey tk you there? |
| 15:14.51 | watchy | is there a firmware out for polycoms to fix the time change issue? |
| 15:15.13 | Ch0Hag | aiksa[LV]: There are programmable function keys. If you have two accounts on different PBXs they can light up depending on which one a call is coming in on. |
| 15:15.19 | aiksa[LV] | exten=>10,hint,SIP/10 |
| 15:15.42 | Ch0Hag | I want to light them diffirently depending on what Asterisk tells it. |
| 15:15.45 | aiksa[LV] | Ch0Hag: ok, I see |
| 15:15.53 | Mercestes | TaiShi: I am on my home computer instead of my laptop so....Yea, it's Cap now. |
| 15:15.58 | aiksa[LV] | I guess you could do it with hint? |
| 15:16.08 | Ch0Hag | I don't know. |
| 15:16.09 | Ch0Hag | Could I? |
| 15:16.28 | TaiSHi | Mercestes: I have a question for you |
| 15:16.33 | aiksa[LV] | you assign a hint for a particular channel |
| 15:16.35 | TaiSHi | What's better, ninja or pirate? |
| 15:16.38 | TaiSHi | Ok, not that one |
| 15:16.43 | aiksa[LV] | SIP/10 in my sample |
| 15:16.53 | TaiSHi | I have a queue, with autopause = yes |
| 15:17.09 | aiksa[LV] | you could as well assign them to the channels you wish to "monitor" |
| 15:17.14 | TaiSHi | If I get more calls than operators, I've heard it will ring a busy operator |
| 15:17.36 | Ch0Hag | And how would the snom use the information? |
| 15:17.50 | Mercestes | TaiSHi, Why would you want that? |
| 15:17.57 | TaiSHi | I dont want to |
| 15:18.05 | aiksa[LV] | snom had a way to assign those hinted lines to the lights and buttons |
| 15:18.09 | TaiSHi | ringinuse = no is the way to |
| 15:18.20 | aiksa[LV] | through the web interface |
| 15:18.27 | TaiSHi | Is that working or still buggy ? |
| 15:18.32 | aiksa[LV] | at least 320 could do that |
| 15:18.33 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 15:18.36 | Mercestes | TaiSHi, : It should dump them into a queue and play MOH or ringing until a operator becomes available. |
| 15:18.48 | Assid | err.. is there a way to disable rtcp for a certain sip peer |
| 15:18.55 | aiksa[LV] | voip-info had a good tutorial on this. |
| 15:18.56 | TaiSHi | Mercestes: Ok |
| 15:19.02 | Mercestes | TaiSHi, : You'll have to try ringinuse because I'd never heard of it and Voip-info is down, so I can't read about it. |
| 15:19.10 | Mercestes | TaiSHi, Let us know in a wiki tho. |
| 15:19.35 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 15:19.43 | tutt9876 | hi |
| 15:19.58 | Ch0Hag | I see :) |
| 15:19.59 | TaiSHi | I will check *guru |
| 15:19.59 | genz | Has anyone experience, while using a T1, asterisk saying "all circuits are busy" when the error is really a number that no longer exists? |
| 15:20.04 | Assid | err.. is there a way to disable rtcp for a certain sip peer? |
| 15:20.18 | Ch0Hag | Google Cache saves the day. |
| 15:20.37 | tutt9876 | got a problem with a "SIP/2.0 407 Proxy Authentication Required" on incoming sip call and hangup follows |
| 15:20.54 | JT | Assid: err is there a way to keep repeating? |
| 15:21.18 | Assid | :P |
| 15:21.20 | TaiSHi | Ch0Hag: True, true |
| 15:21.24 | Assid | sowwy |
| 15:21.38 | TaiSHi | Mercestes: http://www.asteriskguru.com/archives/image-vp255522.html |
| 15:21.45 | lymers | whats the difference between rxgain and txgain ? |
| 15:21.54 | TaiSHi | rxgain = receive gain |
| 15:21.54 | genz | Its telling me HANGUPCAUSE is 41, you can dial 9527461720 to test it yourself |
| 15:21.56 | aiksa[LV] | Ch0Hag: - found it? |
| 15:21.57 | JT | genz: outbound? |
| 15:21.57 | bkruse_home | rx is incoming tx is outgoing |
| 15:22.00 | Ch0Hag | lymers: receive and transmit. |
| 15:22.02 | genz | JT: yes |
| 15:22.02 | TaiSHi | txgain = transference gain |
| 15:22.05 | TaiSHi | transmit * |
| 15:22.09 | tutt9876 | I guess it's a matter of sip.conf matter, but don't know more |
| 15:22.15 | JT | genz: in what country? |
| 15:22.19 | genz | JT: USA |
| 15:22.23 | JT | oh |
| 15:22.28 | JT | helps to say so :) |
| 15:22.31 | Ch0Hag | I think so. |
| 15:22.35 | tutt9876 | got a problem with a "SIP/2.0 407 Proxy Authentication Required" on incoming sip call and hangup follows |
| 15:22.42 | aiksa[LV] | Ch0Hag: - here some additional info http://www.snom.com/wiki/index.php/Interoperability/Asterisk hope it helps |
| 15:23.30 | genz | JT: Sorry. I guess I've been listening to Bush too much and have stopped realizing there are other countries in the world... (switches to BBC) |
| 15:23.31 | aiksa[LV] | on the avarage my clients are very satisfied with every snom phone i have ever sold them, and integrated with asterisk |
| 15:23.40 | b11d | genz.. nice :) |
| 15:23.43 | JT | lol |
| 15:23.58 | aiksa[LV] | the thing i could not say about BT |
| 15:23.59 | Ch0Hag | Cheers |
| 15:24.08 | Ch0Hag | Yeah snoms are nice. |
| 15:25.03 | genz | JT: Is this a problem I should take up with my telecom? |
| 15:26.03 | Ch0Hag | Well yes. They suck. |
| 15:26.14 | Ch0Hag | Some of their resellers are worse though. |
| 15:26.14 | tutt9876 | do you know something about insecure, permit, domain to avoid "407 proxy authentication required" |
| 15:26.18 | *** join/#asterisk chickendog (n=benseale@smtp.moderntech.com) |
| 15:27.18 | JT | genz: cause code 41 is somewhat ambiguous |
| 15:27.18 | nays85 | tutt9876 : who's your itsp? |
| 15:27.18 | JT | it usually means isdn L1 failure somewhere in the network |
| 15:27.20 | JT | or incompatibility |
| 15:27.23 | JT | depends on telco |
| 15:27.49 | JT | so "all circuits busy" is often accurate |
| 15:28.02 | genz | JT: So talk to the telco and ask them why they're sending that code for this number? |
| 15:28.12 | tutt9876 | utsp? |
| 15:28.12 | JT | it's not really the right cause code for a number permanently decommissioned |
| 15:28.14 | JT | yes |
| 15:28.14 | tutt9876 | itsp? |
| 15:28.22 | JT | ~itsp |
| 15:28.24 | jbot | itsp is, like, Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 15:28.40 | boch | how can i know the exit code of the System() app ? |
| 15:29.01 | tutt9876 | I do not have one: only making exchange between 2 proxy |
| 15:29.21 | tutt9876 | and one softphone connected to each proxy |
| 15:29.25 | Mercestes | TaiSHi, Looks good to me. |
| 15:29.44 | TaiSHi | Mercestes: So, autopause = yes and ringinuse = no |
| 15:29.54 | TaiSHi | It should not ring inuse peers ? |
| 15:29.58 | Mercestes | TaiSHi, S'waht the page said, give it a try. |
| 15:30.13 | TaiSHi | Will do |
| 15:30.19 | TaiSHi | Will report results to you aswell =) |
| 15:33.04 | tutt9876 | www.voip-info.org is down for 2 days now |
| 15:33.10 | TaiSHi | They just died |
| 15:33.16 | tutt9876 | died? |
| 15:33.40 | FinboySlick | Is it possible to have the same zapata channel in two different contexts? |
| 15:34.16 | tutt9876 | which are the best doc sites for asterisk doc? |
| 15:34.38 | Mercestes | They are being sued for copyright infrignement according to Florz. |
| 15:34.45 | Mercestes | By the KGB |
| 15:35.00 | JoNate | huh? |
| 15:35.01 | Mercestes | The israeli are threatening beheading with a kitchen knife. |
| 15:35.11 | JoNate | They should have used Telepathy... |
| 15:35.29 | giasai68 | hello, i installed mysql, when a start asterisk all work fine the connection to db is ok but when i received call or i make a caal nothing write in db why? |
| 15:35.30 | Mercestes | JoNate: I'm setting up a new mirror server with new, original information. telepathy-info.org |
| 15:36.57 | aiksa[LV] | Mercestes: could you point me to the article? |
| 15:38.41 | JT | what article |
| 15:38.46 | JT | Mercestes is talking shit |
| 15:38.58 | Mercestes | Mercestes is always talking shit. lol |
| 15:39.08 | sharp | voip-info down? |
| 15:39.16 | Mercestes | maybe I should start a telepathy-info.org tho...it seems to be collecting interest. |
| 15:39.28 | b11d | yueah |
| 15:39.29 | b11d | its down |
| 15:39.40 | b11d | telepathy.. is that a voip company? |
| 15:39.49 | b11d | :) |
| 15:39.51 | rkeels | Well I still can't fix my MWI problem |
| 15:39.53 | rkeels | I put checkmwi=10 in voicemal.conf and that didn't fix it |
| 15:39.54 | rkeels | I did the same thing for a specific channel in sip.conf |
| 15:39.56 | rkeels | because that is where according to the safari book that it should be (actually in the [general] context) |
| 15:40.00 | rkeels | Anyone? |
| 15:40.17 | b11d | what kind of phone? |
| 15:40.58 | Mercestes | b11d: It's a total voT solution. That's "Voice over Telepathy." |
| 15:41.19 | b11d | nice.. nice |
| 15:41.30 | b11d | didnt Radar from MASH get the patent for that? |
| 15:41.33 | aiksa[LV] | with additional savings over TCO and 101% improved ROI |
| 15:41.37 | b11d | what kind of licencing fees do you have to pay to Radar? |
| 15:41.39 | HarryR | VoIP - Voice Over Indigenous Pidgeon |
| 15:41.51 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:42.26 | mut | what if there are non indigenous? |
| 15:42.40 | b11d | Voice Over Ingenious Pigeons |
| 15:42.40 | b11d | ? |
| 15:42.58 | rkeels | More on MWI. The email comes on time. The light just doesn't turn on. |
| 15:43.03 | giasai68 | hello, i installed mysql, when a start asterisk all work fine the connection to db is ok but when i received call or i make a caal nothing write in db why? |
| 15:43.04 | HarryR | maybe naturally migrating birds would be better |
| 15:43.17 | b11d | i say we get Voice Over Whale Song. |
| 15:43.20 | b11d | VoWS |
| 15:43.20 | JT | b11d: indigenous != ingenious |
| 15:43.21 | b11d | nice |
| 15:43.27 | b11d | i know JT. |
| 15:43.28 | aiksa[LV] | bird flu alert! |
| 15:43.30 | JT | maybe bats |
| 15:43.34 | rkeels | Is there a module in Asterisk that I could reload that would clear up this problem |
| 15:43.40 | aiksa[LV] | giasai68: did you edit the cdr_mysql.conf ? |
| 15:43.56 | giasai68 | check i think no |
| 15:44.04 | Mercestes | VoVB I love it. |
| 15:44.15 | Mercestes | voice over vampire bats |
| 15:44.32 | aiksa[LV] | giasai68: you should. unless you specify it there asterisk has no hint of knowing that it should log something to mysql database. |
| 15:44.40 | b11d | what about the traditional Voice Over Human Voice? |
| 15:44.41 | b11d | VoHV? |
| 15:44.55 | aiksa[LV] | oooh. that HDVoice |
| 15:45.03 | Mercestes | What are the tree major forms of mass communication?? |
| 15:45.04 | HarryR | using the C&S codec |
| 15:45.09 | HarryR | Cup & String |
| 15:45.11 | Mercestes | Telephone, television, tell-a-woman. |
| 15:45.12 | b11d | haha |
| 15:45.22 | sharp | Voice Over Cups and String :) |
| 15:45.27 | sharp | VoCS |
| 15:45.29 | b11d | hehe |
| 15:45.32 | giasai68 | yes i insert all data incdr_mysql.conf |
| 15:45.51 | testdriver12500 | I have a simple question. How do I add a wait prefix to an outbound Zaptel channel. I think asterisk is firing off the dialed numbers before there is dial tone on the Zap channel. |
| 15:46.07 | testdriver12500 | leaving me with dial tone |
| 15:46.34 | aiksa[LV] | giasai68: is mysql module lodaded in asterisk? |
| 15:46.38 | Mercestes | testdriver12500, Wait(1) maybe? |
| 15:46.43 | giasai68 | my dubt is only on hostname=ip when database is hosted is it rigth? |
| 15:46.43 | E-bola | Do anybody here use grandstream gxp2000 with asterisk? Seems like a cheap and decent phone? |
| 15:46.55 | testdriver12500 | I tried that and it just waits before it bridges. |
| 15:47.06 | testdriver12500 | I need the wait between the bridge and the dial |
| 15:47.07 | Mercestes | Maybe "w" in the dial string, not sure. |
| 15:47.18 | Mercestes | check voip-info.org when it comes back for a "wait digit" |
| 15:47.25 | genz | E-bola: We're using it and quite happy with it. Mass config of it is tough (unless you're l33t with curl) and don't plan on using headsets with it. Other than that I love it. |
| 15:47.26 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 15:47.33 | aiksa[LV] | giasai68: what does "show modules like cdr_addon_mysql.so" from the CLI says |
| 15:47.35 | aiksa[LV] | ? |
| 15:47.48 | E-bola | genz: do you use the xml directory thing? |
| 15:48.00 | testdriver12500 | I saw reference to w, but no explanation unfortunately. |
| 15:48.17 | Mercestes | Should be w7133251234 then. |
| 15:48.18 | genz | E-bola: No. We setup the NerdVittles click-to-call stuff and there's been no need for it. |
| 15:48.23 | Mercestes | just use it as a digit for a 1 second wait. |
| 15:48.45 | Mercestes | Dial(ZAP/G1/w${EXTEN} |
| 15:48.52 | Mercestes | ) |
| 15:48.55 | Mercestes | >.> |
| 15:49.04 | giasai68 | i see only cdr_cvs.so cdr_custom.so and cdr_manager.so |
| 15:49.21 | genz | E-bola: We do have custom screens and did the custom ringtones. It works a lot better over http than tftp, and combined with some apache/php tweaks you can churn out some pretty cool stuff. How big of a deployment are you looking at? |
| 15:49.34 | aiksa[LV] | giasai68: you should install mysql module then |
| 15:51.03 | giasai68 | in odules appera app_addon_sql_mysql.so |
| 15:51.59 | aiksa[LV] | i am talking about 1.2.9.1, |
| 15:52.06 | aiksa[LV] | do you have a different version? |
| 15:52.35 | aiksa[LV] | like 1.4 ? or something. it could be different there. (dont know) |
| 15:52.40 | [TK]D-Fender | E-bola: ... |
| 15:52.41 | [TK]D-Fender | ~gs |
| 15:52.52 | jbot | gs is, like, South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 15:52.52 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:52.52 | giasai68 | <PROTECTED> |
| 15:52.56 | *** join/#asterisk shinux__ (n=shinux@41.222.64.203) |
| 15:52.59 | E-bola | genz: for starters only 30 phones |
| 15:53.09 | E-bola | [TK]D-Fender: ? |
| 15:53.10 | aiksa[LV] | giasai68: the used to be in asterisk addons package |
| 15:53.17 | aiksa[LV] | if i remember correctly |
| 15:53.24 | [TK]D-Fender | E-bola: Jbot's second comment |
| 15:53.27 | [TK]D-Fender | ~phones |
| 15:53.28 | jbot | rumour has it, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 15:53.30 | testdriver12500 | Mercestes: all I get is an 'Invalid timeout Specified: 'w5551212' |
| 15:53.34 | genz | [TK]D-Fender: It really all depends on the application and what's needed for the system. |
| 15:53.43 | giasai68 | ok i'm gonna to check and find these |
| 15:53.53 | giasai68 | tahnk you for now |
| 15:53.55 | genz | [TK]D-Fender: I still haven't found any good rhetoric for the gs hatred. |
| 15:54.03 | E-bola | ahh |
| 15:54.27 | [TK]D-Fender | genz: If you don't care about audio/manufacturing quality, and general usability, then sure... |
| 15:54.27 | E-bola | so far i have tried snom 320's and linksys spa's |
| 15:54.27 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:54.27 | aiksa[LV] | ok, bye, I am leaving |
| 15:54.31 | E-bola | the snom have better voice quality than the linksys |
| 15:54.35 | E-bola | but the snoms are dead ugly |
| 15:54.38 | [TK]D-Fender | genz: Plenty of accounts for their flakey ass firmware, ECHO issues, and other generic crappyness. |
| 15:54.44 | genz | E-bola: If you don't need full-duplex speakerphones, the Polycom 301s are nice too |
| 15:54.52 | E-bola | genz: ya i was considering them before |
| 15:54.57 | E-bola | but they are more expensive |
| 15:54.58 | Ch0Hag | Eh? |
| 15:54.59 | Ch0Hag | No application 'Hint' for extension |
| 15:55.04 | E-bola | especially here in denmark |
| 15:55.13 | Ch0Hag | Oh, no, ignore me. |
| 15:55.15 | Ch0Hag | I'm retarded. |
| 15:55.21 | aiksa[LV] | Ch0Hag: i guess hint was case sensitive but i might be wrong |
| 15:55.37 | aiksa[LV] | and its not aplication, its a priority |
| 15:55.47 | aiksa[LV] | thats it - bye |
| 15:55.57 | [TK]D-Fender | ~ygwypf |
| 15:55.59 | jbot | somebody said ygwypf was You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 15:56.06 | Ch0Hag | Yeah - that's what I got wrong. |
| 15:56.17 | Mercestes | testdriver12500, http://macnugget.org/projects/asterisk/page12 Booya |
| 15:56.19 | [TK]D-Fender | Ch0Hag: "hint" isn't an application, its a PRIORITY |
| 15:56.51 | Mercestes | The year is 1 AV, "after voip-info." The world is in chaos, but the peasants find a way to survive. |
| 15:57.07 | mog | lol |
| 15:57.38 | Dr-Linux | anybody tried cisco sip firmware 8.x ? |
| 15:58.45 | giasai68 | i insert cli command load cdr_addon_mysql.so and asterisk give me warnign already exist |
| 16:00.08 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 16:00.52 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 16:03.36 | TaiSHi | alaw or ulaw? That's the question... |
| 16:05.05 | Mercestes | TaiSHi, Where are you? |
| 16:05.15 | TaiSHi | Argentina |
| 16:05.39 | Mercestes | the u in ulaw means "United States." The a in alaw means "averywhere else." |
| 16:05.51 | af_ | heheheh |
| 16:06.02 | TaiSHi | Damn capitalist! |
| 16:06.11 | Mercestes | yea. We suck |
| 16:06.20 | TaiSHi | True, but you are helpful |
| 16:06.27 | Mercestes | :) |
| 16:06.37 | [TK]D-Fender | The road to Hell.... |
| 16:07.12 | TaiSHi | I just got argentinian-insulted o.o |
| 16:07.14 | Mercestes | is paved in Mercestes? |
| 16:07.21 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.203) |
| 16:08.56 | toombaloomba | anyone upgrade their Cisco 7941 to sip.8-0-4SR3A and find that the phone no longer registers or even sends any packets to their asterisk box? |
| 16:09.19 | [TK]D-Fender | Mercestes: Finally a foundation we can BUILD on! |
| 16:12.26 | Mercestes | lol |
| 16:13.06 | *** join/#asterisk [shodan] (n=shodan@ip199.99-113-216.pppoe4.joliette.intermonde.net) |
| 16:14.29 | Modcuts | anybody else having issues accessing http://www.voip-info.org/wiki/? |
| 16:14.42 | yonahw-work | modcuts: it's just you man |
| 16:14.44 | bkruse_home | Modcuts: its down |
| 16:14.51 | bkruse_home | yonahw-work: dangit, i ruined it |
| 16:14.53 | yonahw-work | the rest of us have been on all day long |
| 16:14.54 | bkruse_home | sorry ;[ lol |
| 16:15.01 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
| 16:15.05 | yonahw-work | nah its cool man i just had to go for it |
| 16:15.11 | Geert | how can I manually clear sip registration expiry? |
| 16:15.22 | Geert | I've restarted asterisk, but doesn't help |
| 16:15.29 | Geert | I need to un-register some phones |
| 16:16.12 | [TK]D-Fender | Geert: Manually delete their entry from astdb |
| 16:16.31 | Modcuts | bkruse_home: cheers |
| 16:18.03 | JunK-Y | some1 has an idea how * is dealing with multicast registers? |
| 16:20.24 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 16:20.47 | bkruse_home | Modcuts: np :D |
| 16:22.25 | Geert | [TK]D-Fender: thanks :) |
| 16:22.54 | [TK]D-Fender | Geert: Quite welcome |
| 16:23.07 | giasai68 | i instalkled mysql on atsrisk machine i installed all addons i modified file cdr_mysql.conf but when i received or i make a call noting write in db |
| 16:23.10 | giasai68 | plase help me |
| 16:24.46 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 16:24.48 | kink0 | hello |
| 16:24.48 | clyrrad | did you configure cdr_mysql.conf? |
| 16:25.07 | giasai68 | yes done |
| 16:25.16 | kink0 | I am unnable to compile zaptel in my new box, I got : /zttranscode.c:192: error: invalid use of undefined type 'struct page' |
| 16:25.19 | kink0 | any idea ? |
| 16:25.29 | clyrrad | run a 'reload' and see if you get errors towards the top of the output |
| 16:27.14 | giasai68 | nothing errors |
| 16:27.14 | clyrrad | Does it say it connected to the database? |
| 16:27.26 | *** part/#asterisk Geert (i=geert@irssi/staff/geert) |
| 16:27.34 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 16:27.49 | clyrrad | also did you create the cdr database? |
| 16:28.02 | blackmousepad | O.k. I've got x-lite sitting at 10.0.0.2 behind a nat router connected to the internet. And I've got asterisk at 10.254.0.2 behind a nat router connected to the internet that is port forwarding 5060 and 10000-20000 to 10.254.0.2 I've added the nat=yes and insecure=very to my xlite's context [210] and I've set nat=yes inside_net=10.254.0.0/255.255.255.0 and outside_addr=public_ip_here. when I place a call from xlite (or to |
| 16:29.27 | giasai68 | i insert command realtime mysql status and appear that Connected to asterisk@xxx.xxx.xxx.xxx, port 3306 with username xxxxxxxx for 1 minutes, 42 seconds |
| 16:30.24 | clyrrad | see my previous question |
| 16:30.47 | creature1 | blackmousepad: what version? |
| 16:30.53 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 16:30.55 | giasai68 | yes i created cdr database |
| 16:31.02 | blackmousepad | 1.2.9.1 |
| 16:31.12 | giasai68 | and it is connected to it |
| 16:31.26 | giasai68 | this is very strange |
| 16:31.39 | clyrrad | Can you insert a record into the database from the commandline using the same username and password as you specify in cdr_mysql.conf? |
| 16:32.01 | giasai68 | yes |
| 16:33.41 | clyrrad | do you have the /usr/lib/asterisk/modules/cdr_addon_mysql.so file? |
| 16:33.59 | giasai68 | yes and in show modules appear |
| 16:34.00 | wunderkin | Qwell: ? |
| 16:34.42 | clyrrad | your positive you can insert a record from the command line? And that the username and password are the same? |
| 16:35.27 | giasai68 | yes |
| 16:36.06 | clyrrad | I am not sure what else to suggest then |
| 16:36.33 | giasai68 | ok if i fix this problem i will let you know |
| 16:36.40 | giasai68 | thank you very much for yoiur time |
| 16:36.50 | clyrrad | np |
| 16:36.55 | [TK]D-Fender | blackmousepad: The values are "localnet=" and "externip=", not "inside_net=", and "outside_addr=" |
| 16:39.11 | b11d | i love you guys |
| 16:41.21 | Ch0Hag | Odd. Is there any good reason why my Snom won't update itself? |
| 16:41.30 | Ch0Hag | It sits on "Loading file:" |
| 16:41.53 | b11d | its because you touch yourself at night. |
| 16:42.36 | Ch0Hag | I knew I should have stopped that. |
| 16:42.36 | b11d | heh.. actually that sucks though.. |
| 16:42.36 | Ch0Hag | So am I now screwed? |
| 16:42.36 | b11d | i dont have any Snoms so I cant help.. |
| 16:42.36 | Ch0Hag | Figuratively. |
| 16:42.37 | b11d | nah.. theres a way to fix it.. guaranteed. |
| 16:42.39 | b11d | i just dont know it.. |
| 16:42.49 | Ch0Hag | Get a woman? |
| 16:42.52 | Ch0Hag | I'm working on that. |
| 16:43.10 | [TK]D-Fender | No, being "fixed" won't help you get a woman.... |
| 16:43.10 | b11d | heh.. my problem is im too afraid that the girl I like will say Yes. |
| 16:43.19 | b11d | no one cares about "no" -- its the "yes" that scares me :) |
| 16:44.48 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:44.56 | Ch0Hag | Ah maybe it's the proxy. |
| 16:48.47 | wunderkin | did misery crash or something? |
| 17:00.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 17:01.15 | christo | guys, I'm driving calls through the manager interface (using IO::Socket::INET) Is there any way to forcibly flush the remote end of the socket? For some reason it's being bad at providing real-time status on new calls. |
| 17:01.44 | christo | I have $| set |
| 17:02.38 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 17:06.09 | *** part/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:06.18 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:06.18 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:07.49 | testdriver12500 | In using x-lite 3.0, I am trying to dial out on a Zap1 interface. After dialing, all it does is connect me to dial tone on the PSTN. I have to re-dial the number, using the PSTN dial-tone. What am I missing? |
| 17:08.35 | [TK]D-Fender | testdriver12500: Paste the EXACT dial line as it is being called. |
| 17:08.45 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:12.42 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:12.42 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 17:12.43 | diclophis-work | y0 |
| 17:12.51 | diclophis-work | is there something like chan_oss for windows/osx? |
| 17:14.32 | wunderkin | Qwell: ping? |
| 17:14.35 | testdriver12500 | Fender: waiting for pastebin... |
| 17:14.51 | Qwell[] | ? |
| 17:15.09 | *** join/#asterisk rhino-james (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:15.22 | testdriver12500 | Fender: http://pastebin.com/898887 |
| 17:15.30 | blackmousepad | [TK]D-Fender thanks for pointing out my "localnet=" and "externip=", not "inside_net=", and "outside_addr=" it works now. |
| 17:15.59 | *** join/#asterisk lionel (n=lionel@ip-61.net-82-216-103.rev.numericable.fr) |
| 17:17.00 | lionel | Hello everybody. Does somebody know what happens to voip-info.org, it is unreachable for days here ? |
| 17:17.17 | testdriver12500 | voip-info.org is down. |
| 17:17.43 | [TK]D-Fender | testdriver12500: Those 2 things have nothing in common, and I didn't even ask for a pastebin. Its 1 line I want |
| 17:17.48 | wunderkin | Qwell: hey... any other ideas on what i should do about my hpec license key release problems and the kernel panic? |
| 17:18.08 | lionel | testdriver12500: definitely ? |
| 17:18.11 | [TK]D-Fender | testdriver12500: And THE BOOK covers the Dial app rather well as well. |
| 17:18.32 | wunderkin | is kevin the one that would investigate that? i'm trying to make sure who and that they will be able to look into it the next time we have the problem... because.. support.. obviously cant... |
| 17:18.40 | [TK]D-Fender | wunderkin: Doctor doctor, it ruts when I lift my arm like this! ;) |
| 17:19.21 | wunderkin | i'm afraid if kevin is the person that he would be too busy to look at it when it happens |
| 17:19.31 | testdriver12500 | Fender: Looked through the Dial() application in the book, but didn't see how to set a delay if necessary. Doesn't asterisk wait for Dial tone before sending the $EXTEN? |
| 17:19.53 | [TK]D-Fender | testdriver12500: Could you please just paste the line exatly as it is being called in your dialplan. |
| 17:20.16 | testdriver12500 | ahh.. |
| 17:20.57 | [TK]D-Fender | s/ruts/hurts |
| 17:21.11 | testdriver12500 | exten => _9NXXXNXXXXXX,1,Dial(Zap/1,${EXTEN}:}) |
| 17:21.17 | testdriver12500 | sorry about that... |
| 17:21.35 | [TK]D-Fender | testdriver12500: tHERES THE PROBLEM... BAD FORMATTING, BRACES, AND PARMS |
| 17:21.51 | [TK]D-Fender | exten => _9NXXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
| 17:22.22 | testdriver12500 | sorry, fat finger...what you had is what is in there... |
| 17:22.42 | testdriver12500 | except there is a comma not a / between the 1 and the $ |
| 17:22.50 | [TK]D-Fender | testdriver12500: Please paste the REAL line..... I don't work well with questionably recreated material :) |
| 17:23.11 | [TK]D-Fender | testdriver12500: Comma = basd, that is a parm seperator |
| 17:23.15 | [TK]D-Fender | bad* |
| 17:23.21 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:23.21 | testdriver12500 | ahhh |
| 17:23.46 | [TK]D-Fender | testdriver12500: the 2nd parm in Dial is the TIMEOUT, which means it will wait for a LONG time before giving up ;) |
| 17:23.56 | testdriver12500 | haha...thanks!!! |
| 17:23.59 | [TK]D-Fender | testdriver12500: np |
| 17:24.00 | testdriver12500 | let me give a shot |
| 17:27.01 | testdriver12500 | Fender: That did it. Thanks again. |
| 17:28.16 | [TK]D-Fender | testdriver12500: no prob |
| 17:31.59 | chickendog | hello |
| 17:33.34 | chickendog | i download AsteriskNOW and want to start playing with it |
| 17:33.43 | chickendog | I have a MP-114 Voip Gateway |
| 17:34.04 | chickendog | can i use this to get some analog lines to AsteriskNOW |
| 17:36.23 | chickendog | any one know? |
| 17:36.29 | chickendog | i want to start using Asterisk in our remote offices |
| 17:36.34 | chickendog | 1 to 7 people |
| 17:42.50 | awk | hi, will asterisk 1.2 users ever be fored to use 1.4 ? |
| 17:43.35 | Juggie | thats a silly question |
| 17:43.44 | awk | why is that? |
| 17:43.50 | Juggie | how could you be forced. |
| 17:43.59 | awk | not foced |
| 17:44.12 | rhino-james | asterisk developers with guns..... |
| 17:44.19 | awk | but will updates nstop being released for 1.2 |
| 17:44.20 | awk | like this henew sip dos vuln ? |
| 17:44.33 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:44.33 | Juggie | eventually |
| 17:44.36 | Juggie | but not right away |
| 17:44.37 | rhino-james | he is an obvious M$ user forced upgrades :) |
| 17:44.52 | Juggie | even so you could back port any security patches yourself you do have the source code |
| 17:47.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:50.18 | awk | rhino-james: you ahve no idea what im talking about |
| 17:50.18 | awk | im talking about alike apache has 1* and apache 2* |
| 17:50.19 | awk | completely deiifferent diesign |
| 17:51.42 | awk | design, have |
| 17:52.09 | rhino-james | check note to self no jokes :( |
| 17:53.01 | awk | you cant aply 1.2 patches to 1.4 can you ? |
| 17:53.45 | *** join/#asterisk waisky2 (n=waisky2@36.81.94.219.cbj02-home.tm.net.my) |
| 17:53.57 | Qwell[] | no |
| 17:54.03 | awk | so how are you supposed to backport any patches applied to 1.4 or the uture 2.0 or what ever. |
| 17:54.05 | Qwell[] | not without some work |
| 17:54.32 | Qwell[] | awk: How does Microsoft backport patches to Windows 3.1 from Vista? |
| 17:54.39 | Qwell[] | oh... they don't? |
| 17:55.09 | awk | err future |
| 17:56.02 | awk | I dont use windows so i wouldn't know, but i would presume they cant :) |
| 17:56.07 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 17:56.21 | awk | sorry im terribly lagged |
| 17:56.29 | Qwell[] | It's called a release cycle. A version of an application is only useful for so long. |
| 17:56.39 | b11d | Thats heresey! |
| 17:56.46 | b11d | :) |
| 17:56.55 | awk | Qwell[]: well not in the case of apache, etc |
| 17:56.58 | Qwell[] | b11d: When was the last time you used php version 1? :) |
| 17:57.01 | awk | as they have used version 1.* for how many years? |
| 17:57.02 | b11d | hahahah |
| 17:57.04 | Qwell[] | awk: I beg to differ. |
| 17:57.04 | chickendog | can i use voip Gateway (fxo) hardware to get analog lines to Asterisk? |
| 17:57.15 | b11d | well you could say that php 1 is in php 5.. so by using 5, im using 1. |
| 17:57.45 | awk | no what im talking about sticking my dialplan from 1.2 into 1.4 it does not work |
| 17:57.47 | Qwell[] | awk: what version of apache 1.x are you using? |
| 17:57.53 | Qwell[] | Surely you aren't using 1.0.0 |
| 17:58.11 | awk | so what im saying is current updates of 1.2 accepts the syntaxes, etc used |
| 17:58.12 | chickendog | What do i need to get analog lines to my Asterisk server? |
| 17:58.57 | awk | so you saying that you will in the near future be forced to restructure you entire dialplan just to keep up to date with where asterisk is heading? |
| 18:00.32 | chickendog | so what your saying is i have pigs flying out my butt? |
| 18:01.16 | awk | so how long is the end of life expectancy |
| 18:01.22 | awk | <PROTECTED> |
| 18:02.09 | awk | Qwell[]: and I was not the one talking about backporting patches |
| 18:03.17 | awk | anyone have any idea how long voip-info is going to be down for and is there no mirror? |
| 18:03.29 | Qwell[] | google cache |
| 18:03.32 | b11d | it will be down until its up |
| 18:03.37 | b11d | there is the google cache and the WayBack machine |
| 18:03.45 | waisky2 | no wonder I cant access till now |
| 18:04.38 | awk | Qwell[]: please can you answer my question |
| 18:04.49 | awk | wehen is the end of life cycle for 1.2 ? |
| 18:05.15 | Qwell[] | when 1.6 comes out |
| 18:05.21 | *** join/#asterisk wunderkin- (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 18:05.29 | awk | ok, great |
| 18:05.34 | b11d | yeah.. its driven by quality, not dates on a calendar. |
| 18:05.41 | b11d | i kind of hate "release dates" |
| 18:07.25 | chickendog | what does PSTN mean? |
| 18:07.39 | Qwell[] | ~pstn |
| 18:07.46 | jbot | it has been said that pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 18:07.46 | sysreq | public switched telephone network. |
| 18:07.46 | *** join/#asterisk friedrich| (n=friedric@e177243079.adsl.alicedsl.de) |
| 18:07.47 | awk | google it |
| 18:08.01 | b11d | Please Stop Talking Now |
| 18:09.33 | b11d | ... |
| 18:09.37 | b11d | Please Start Talking Now |
| 18:09.38 | b11d | ? |
| 18:10.42 | awk | please stfu |
| 18:14.18 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 18:14.40 | TaiSHi | DAmnit |
| 18:14.42 | TaiSHi | Since when >.< |
| 18:14.48 | TaiSHi | AgentCallBackLogin >.<!! |
| 18:18.37 | TaiSHi | Qwell[]: fyi, bought 3 G729 licenses |
| 18:18.46 | TaiSHi | And now I find out that AgentCallbackLogin is deprecated! |
| 18:19.14 | JunK-Y | this was announce few months ago. |
| 18:19.32 | TaiSHi | Few months ago I didn't even know what PSTN was :) |
| 18:19.47 | TaiSHi | JunK-Y: Is there any replacement ? |
| 18:20.32 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 18:20.52 | JunK-Y | in ur old install, when ya ran that app, there was a message, read it :) |
| 18:21.21 | TaiSHi | What old install ? :P |
| 18:22.28 | JunK-Y | never use that app? so how do ya know its deprecated? |
| 18:22.37 | TaiSHi | I google'd around |
| 18:22.41 | TaiSHi | But couldn't find out a fix |
| 18:23.03 | TaiSHi | Besides, it's not working :P |
| 18:26.09 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:26.19 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 18:27.07 | jart | ~sip |
| 18:27.14 | jbot | extra, extra, read all about it, sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 18:28.19 | TaiSHi | Why would I want to read SIP ? |
| 18:28.23 | TaiSHi | (again) |
| 18:31.11 | *** join/#asterisk acehunky (n=chat_jok@59.184.24.178) |
| 18:33.16 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:34.07 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:34.43 | anonymouz666 | chan_agent is deprecated? |
| 18:35.07 | anonymouz666 | can you do the same things in another way ? |
| 18:36.26 | dlynes_laptop | good morning, cp |
| 18:36.41 | CunningPike | Hey, dlynes_laptop - been a while. How's things? |
| 18:37.20 | dlynes_laptop | Yeah...it's been a while |
| 18:37.23 | dlynes_laptop | Things are good |
| 18:37.40 | dlynes_laptop | I'm really starting to hate programming now |
| 18:37.44 | CunningPike | lol |
| 18:37.44 | dlynes_laptop | but other than that :) |
| 18:37.55 | dlynes_laptop | I used to love programming many years ago |
| 18:38.02 | dlynes_laptop | but the years have sorta worn away on that |
| 18:38.15 | dlynes_laptop | having to write crap in php and java and all that kinda makes you not want to program anymore |
| 18:39.07 | dlynes_laptop | but on the upside |
| 18:39.15 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 18:39.21 | dlynes_laptop | I've been meeting about 2 or 3 new ladies every week :) |
| 18:39.32 | CunningPike | Heh |
| 18:39.46 | CunningPike | Plus ?a change...... ;) |
| 18:39.55 | dlynes_laptop | ?a means? |
| 18:40.32 | TaiSHi | ?a means ?a .. |
| 18:40.40 | dlynes_laptop | fer shi |
| 18:40.43 | TaiSHi | anonymouz666: deprecated as hell u_U |
| 18:40.54 | TaiSHi | tu mamá, por las dudas =D |
| 18:41.22 | CunningPike | Plus ca change, for those sad sacks without extended character support |
| 18:41.38 | dlynes_laptop | Well, no |
| 18:41.44 | dlynes_laptop | I have extended character support |
| 18:41.51 | dlynes_laptop | but what you wrote wasn't in unicode |
| 18:42.54 | anonymouz666 | TaiSHi: but there is some things that you can't do in other way? |
| 18:42.59 | *** join/#asterisk viperdude (n=chatzill@195.74.96.120) |
| 18:43.00 | CunningPike | Oh well...... |
| 18:43.31 | TaiSHi | anonymouz666: I haven't understood the 'dialplan way' |
| 18:43.38 | TaiSHi | So I'm stucked right now =) |
| 18:43.49 | TaiSHi | I've heard it can be done with addQueueMember |
| 18:44.01 | dlynes_laptop | CunningPike: write it again? |
| 18:44.17 | TaiSHi | "it again?" :P |
| 18:44.55 | dlynes_laptop | CunningPike: just realized I didn't have my unicode support enabled...I had upgraded kde, and so everything defaulted again |
| 18:44.56 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 18:44.56 | CunningPike | Plus ?a change |
| 18:44.56 | dlynes_laptop | nah....guess it's not in unicode |
| 18:44.58 | dlynes_laptop | cause it's still showing up as a question mark |
| 18:45.13 | dlynes_laptop | i'm using a unicode font now, though |
| 18:45.23 | CunningPike | Could be I'm not..... lol |
| 18:45.24 | Mercestes | I see a ?a too |
| 18:45.28 | dlynes_laptop | so you must be using la lingua franca |
| 18:45.51 | dlynes_laptop | CunningPike: i think irc only accepts unicode |
| 18:45.51 | TaiSHi | Mercestes: darling, any workaround for AgentCallbackLogin ? |
| 18:46.04 | dlynes_laptop | CunningPike: so if you send it something else, it converts whatever's not unicode to question marks |
| 18:46.23 | Mercestes | TaiSHi, : Static members. or just agent => in queues.conf |
| 18:46.47 | Mercestes | TaiSHi, : I think it's AgentLogIn now a days btw. |
| 18:47.04 | TaiSHi | Mercestes: Ok to agent => in queues.conf |
| 18:47.07 | TaiSHi | What about static members? |
| 18:47.16 | TaiSHi | AgentLogin needs to stay the call up or something |
| 18:47.18 | TaiSHi | Not sure |
| 18:50.09 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 18:50.11 | Mercestes | TaiShi: member => or agent => ...meh, wait for voip-info.org to be back up..I'm not real sure. I use statics. |
| 18:50.22 | markit | any mISDN guru? :) |
| 18:50.28 | Mercestes | TaiShi: And your right, agentlogin does require an open connection =/ |
| 18:52.46 | TaiSHi | Mercestes: It quite screwes it >.< |
| 18:53.01 | TaiSHi | agent => I think |
| 18:53.04 | TaiSHi | How's the static thingie ? |
| 18:53.44 | Mercestes | TaiShi: Just use member => Sip/device It's what I do. |
| 18:53.56 | Mercestes | TaiShi: Ppl never remember to log in /log out anyways |
| 18:53.56 | TaiSHi | So whenever they log, they become agents ? |
| 18:54.07 | Mercestes | TaiShi: No....the phones are permanent agents. |
| 18:54.22 | TaiSHi | Ok, but autopause works still ? |
| 18:54.31 | TaiSHi | Mmm, let me explain myself better |
| 18:54.39 | TaiSHi | If I log into * (SIP login) |
| 18:54.43 | TaiSHi | I become an insta-agent |
| 18:54.47 | TaiSHi | Right ? |
| 18:56.04 | Mercestes | The phone does. |
| 18:56.38 | TaiSHi | Ok, the phone and you are one |
| 18:56.50 | TaiSHi | The phone does not exist, try it Neo |
| 18:56.50 | Mercestes | I am one with the phones. |
| 18:57.07 | TaiSHi | :P |
| 18:57.24 | TaiSHi | btw, it isn't member => |
| 18:57.26 | TaiSHi | It's agent => |
| 18:57.33 | TaiSHi | (just checked example agents.conf) |
| 18:58.46 | TaiSHi | SIP/600 not logged in (musiconhold is 'default') |
| 18:58.48 | TaiSHi | >.< |
| 19:00.32 | Mercestes | It's member. |
| 19:00.46 | Mercestes | in queues.conf |
| 19:00.51 | Mercestes | forget agents. |
| 19:00.57 | TaiSHi | Roger |
| 19:01.32 | TaiSHi | And how do I ... |
| 19:01.33 | TaiSHi | erm... |
| 19:01.46 | TaiSHi | Mm, only way to check if it works is testing... right? |
| 19:02.08 | Mercestes | or you could trust me... |
| 19:02.10 | Mercestes | ...or hire me. |
| 19:02.14 | Mercestes | or test. |
| 19:03.20 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 19:06.17 | Sweeper | sigh. wiki still down. is there a mirror yet? and how bad is the outage? |
| 19:07.03 | Mercestes | It was a nuclear incident. |
| 19:07.04 | Qwell[] | kinda hard to mirror a server that's down |
| 19:07.25 | Sweeper | Qwell[]: depends on the sort of outage :P |
| 19:07.59 | TaiSHi | Mercestes: I trust you in this stuff |
| 19:08.03 | TaiSHi | Just, gotta run home |
| 19:08.09 | TaiSHi | I have a... 'bathroom emergency' |
| 19:08.13 | TaiSHi | Speak to you all tomorrow |
| 19:08.18 | Mercestes | hope everything comes out ok. |
| 19:08.19 | Sweeper | shit your pants again |
| 19:08.20 | Sweeper | ? |
| 19:08.21 | Sweeper | :V |
| 19:08.26 | TaiSHi | Thanks Mercestes =) |
| 19:08.28 | TaiSHi | bb all!! |
| 19:12.15 | dlynes_laptop | so what happened to voip-info, anyways? |
| 19:13.10 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 19:17.16 | JacksLivr | voip-info was hosted in valencia california where, as we all know, Jack Bauer did not stop the first suitcase nuke from exploding. CTU is feverously working to get voip-info back online as soon as Cloe can get to it. |
| 19:18.27 | *** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com) |
| 19:18.30 | [TK]D-Fender | chickendog: the MP-114 is available in 4 FXS or 4 FXO versions. If you have the FXO version then that is used to connect analog LINES and then you can use them with * |
| 19:18.58 | nays85 | we all know voip-info won't come back up until someone loses a finger |
| 19:19.00 | b11d | sweet.. -114 classic Dr. Strangelove reference.. |
| 19:19.04 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-106-203.dsl.irvnca.pacbell.net) |
| 19:19.05 | b11d | CRM-114 Discriminator's rock |
| 19:19.16 | MGSsancho | running through naked pardon me! |
| 19:19.23 | *** part/#asterisk MGSsancho (n=user@adsl-67-121-106-203.dsl.irvnca.pacbell.net) |
| 19:21.50 | nDuff | How do I set default values for sip.conf parameters? In particular, Asterisk 1.4 is behaving as if there's a default call-limit of 1, which breaks attended transfers and some other Good Stuff, and I'd rather not need to set it for every single phone. |
| 19:23.48 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:25.11 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 19:25.25 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:25.31 | l2cache | Is anyone else not getting to voip-info.org??? |
| 19:25.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:26.48 | *** join/#asterisk ta^3 (n=tacvbo@189.136.34.99) |
| 19:26.50 | Mercestes | l2cache: Works fine for me. Have you been banned? |
| 19:27.12 | l2cache | no..lol |
| 19:27.17 | Mercestes | are you sure? |
| 19:27.17 | Assid | doesnt open here either |
| 19:27.25 | nDuff | Mercestes: eh? It's been down for me too, for days now. |
| 19:27.27 | Assid | looks like some kinda routing issue |
| 19:27.35 | Mercestes | nDuff: Have you been banned? |
| 19:27.39 | quidpro | L2Cache: It's been down since yesterdya, yo might get sporadic spurts of it working.. but in genereal it's down |
| 19:27.42 | l2cache | Mercestes: maybe the page is cached for you |
| 19:27.48 | l2cache | thank you quid |
| 19:27.54 | nDuff | Mercestes: Absolutely not. I don't post at all. |
| 19:27.59 | Mercestes | Yea, it's down. lol |
| 19:28.12 | l2cache | asterisk administrators everywhere are panic -ing |
| 19:28.13 | nDuff | Mercestes: ...and there'v been a lot of other folks on IRC complaining. |
| 19:28.14 | l2cache | lol |
| 19:28.25 | b11d | also on the mailing lists.. |
| 19:28.26 | *** join/#asterisk multcom (n=multcom@201.75.108.20) |
| 19:28.37 | b11d | someone should just mirror it.. fuck permission from the voip-info guys :) |
| 19:28.57 | Mercestes | Yea, I had to quit my job because I don't really know anything. |
| 19:29.10 | multcom | hi everybody, can someone help me with attendend transfer?? |
| 19:29.19 | multcom | i just can't do it |
| 19:29.45 | multcom | i've already set everything in features.conf, but nothing |
| 19:29.56 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
| 19:29.57 | multcom | i just get blindxfer |
| 19:30.30 | Burgwork | my new polycom phone will not pick up the asterisk directory? how do I fix this? |
| 19:30.59 | multcom | by the way, i'm using Asterisk CVS-v1-0-09 |
| 19:31.04 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 19:31.10 | JunK-Y | pickup the asterisk dir? whatcha mean? |
| 19:32.47 | chickendog | hello |
| 19:33.10 | chickendog | i am new to AsteriskNOW and have 2 softphones trying to dial eachother |
| 19:34.28 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 19:34.33 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 19:34.35 | chickendog | the consol of AsteriskNOW is seeing the calls but rejecting them with chan_iax2.c:6754 socket_process: rejected connection attempt from x.x.x.x who was trying to reahch |
| 19:34.41 | chickendog | reach 600 |
| 19:34.49 | [TK]D-Fender | Burgwork: SIP servers don't just pass along directory info you know... Go download the Polycom Admin guides and lear how to set them up. |
| 19:35.13 | Burgwork | right |
| 19:35.14 | supjigatr | Anyone know how to have dial command relaibly do call hunting to the PSTN from a PRI? I get a few calls that just hang up on the end user when hunting. |
| 19:35.15 | christo | guys - I telnet to port 5038 and run the Status command. I then create another connection and originate a call (using a small perl script and IO::Socket). I expect to see the status events coming in thru the telnet connection, but they appear to be buffering - so I don't see the first 'Ringing' State change until the call is actually answered. Is there any way to make this more, er, eventful ? I'm using * version 1.4.0 |
| 19:36.16 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 19:36.36 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 19:37.00 | [TK]D-Fender | Burgwork: Ask your reseller to provide you the latest SIP & BootROM images for your phone (2.1.0 & 3.2.3 respectively) and get the admin guides, release notes & Amding guide addendum from Polycom |
| 19:37.04 | vooduhal | Sorry for asking this for a second day but no one answered yesterday. Can anyone point me in the direction of documenation explaining of that the status (xx ms) in "sip show peers" actually means. |
| 19:37.20 | Burgwork | [TK]D-Fender: my older sipura phones appear to be picking up the information directly from asterisk, but the new polycom doesn't |
| 19:37.36 | [TK]D-Fender | Burgwork: Rest assured they do NOT. |
| 19:37.39 | wunderkin | vooduhal, how long it takes for the phone to respond back from the sip options packet |
| 19:37.52 | wunderkin | it is a 'sip ping' not icmp |
| 19:38.27 | [TK]D-Fender | Burgwork: The most a phone can do is personally REMEMEBR who has called them, or they have called, or info entered directly into the phone. Nothing more. Not you could be running some OTHER piece of software that is doing some kind of work for you on that behalf.... |
| 19:38.30 | vooduhal | And what would cause one phone to take 400ms while another takes 20ms on the same system and same physical segment other than slow network stack? |
| 19:39.05 | [TK]D-Fender | vooduhal: Could be a really slow CPU & SIP stack on that phones. What model is the fast one, and which the slow? |
| 19:39.19 | [TK]D-Fender | vooduhal: Wifi = BAD btw |
| 19:39.20 | vooduhal | And would this in someone affect quality of service for a phoen with a 400ms delay? |
| 19:39.20 | Moobius | vooduhal: it is ping time. You can set the timeout with the "qualify=" option in sip.conf. |
| 19:39.35 | vooduhal | none if this is WiFi unfortunately. |
| 19:39.47 | Assid | okay anyone got a e61 working with asterisk ? |
| 19:39.54 | Assid | i wanna know what kind of ping responses you get on wifi |
| 19:39.54 | Burgwork | [TK]D-Fender: the sipura phones display the name of the called person in the lcd display, as where as the polycom will not, even with "Use Directory Names" checked |
| 19:39.58 | [TK]D-Fender | Assid: I've seen it working before... |
| 19:40.04 | Mercestes | vooduhal, Shoudn't in itself but it could be an indicator of another problem. |
| 19:40.09 | Assid | mine sounds like mr. robotto |
| 19:40.23 | [TK]D-Fender | Burgwork: Something else is royally screwed up then. |
| 19:40.34 | Assid | im guessing its the firmware.. i cant confirm until the end of this month/next month |
| 19:40.40 | Burgwork | [TK]D-Fender: it is likely the digitplan, but that appears to just work |
| 19:40.41 | vooduhal | The fastest phones I've seen are the SPA9xx Linksys (Sipura) phones with 10ms, the mids are the Polycoms x01s with 20-50, and a generic phone that we are looking at selling with 400ms. |
| 19:40.44 | [TK]D-Fender | Burgwork: the only time you won't see the name is on a call-waiting interrupting call |
| 19:40.55 | Assid | 20-50ms ? |
| 19:41.08 | [TK]D-Fender | Burgwork: wait.. the CALLED person? |
| 19:41.09 | vooduhal | Yep. Depending on the installation. |
| 19:41.17 | vooduhal | We have some on a linksys switch doing 20ms. |
| 19:41.18 | Assid | depends on your network |
| 19:41.22 | Burgwork | [TK]D-Fender: either, for internal numbers |
| 19:41.28 | vooduhal | And some doing 50 on a cisco 5500. |
| 19:41.59 | Assid | round-trip min/avg/max = 1.0/1.0/1.1 ms |
| 19:42.09 | Assid | 1ms is pretty decent |
| 19:42.13 | [TK]D-Fender | Burgwork: received calls should definately get the name. Not sure about CPID from directory. never tried. I know there is a CPID patch in the works though through SIP (and they're loking to make the patch channel-agnostic) |
| 19:42.26 | Assid | vooduhal : round-trip min/avg/max = 1.0/1.0/1.1 ms |
| 19:42.43 | Burgwork | [TK]D-Fender: right. However, I suspect something is amiss, as I can call internal within our office, 4xx, but not the other office, which is 2xx. |
| 19:42.45 | vooduhal | That's about what we get with ICMP but not with SIP. |
| 19:42.47 | b11d | that nick reminds me of Space Quest 2: vohauls revenge |
| 19:43.07 | vooduhal | So you have phones that show 1.0ms when you do a "sip show peers"? |
| 19:43.22 | Assid | er lemme check |
| 19:43.23 | b11d | mine average 34-41ms on that.. |
| 19:43.31 | b11d | my cisco vg-224 reports 12ms |
| 19:43.33 | [TK]D-Fender | Burgwork: Well Polycom = best these days... with our solitary lightning-rod here for "insurance" ;) |
| 19:43.37 | vooduhal | ICMP 1.184/1.491/1.936/0.267 ms |
| 19:43.57 | Assid | 33ms with that |
| 19:44.00 | vooduhal | Ah, ok. |
| 19:44.06 | vooduhal | Was starting to really worry. |
| 19:44.41 | Burgwork | [TK]D-Fender: the phone calls externally and internally great, it just doesn't deal with this internal extension issue |
| 19:45.18 | [TK]D-Fender | Burgwork: Well keep up on it, and if you seem really screwed, host your config files somewhere and I'll take a look later |
| 19:45.25 | Burgwork | [TK]D-Fender: ok, will do |
| 19:45.59 | Mercestes | Grandstream is the best..........doorstop ever! |
| 19:46.07 | Qwell[] | Mercestes: Not in my house |
| 19:46.23 | Assid | why does it drop to 33ms ? |
| 19:47.09 | christo | I telnet to port 5038 and run the Status command. I then create another connection and originate a call (using perl's IO::Socket). The status events come over the telnet connection, but they are buffered - so I don't see the 'Ringing' State change until later when the call answers. Is there a way to make this more real-time? (using v 1.4.0) |
| 19:48.13 | [TK]D-Fender | Qwell[]: Get lighter doors so the casing doesn't shatter so easily ;) |
| 19:48.41 | Qwell[] | ;) |
| 19:48.58 | christo | in fact I get the same problem if I just drive everything over telnet - I only start getting status back when the call is actually answered. Does this sound familiar to anybody? |
| 19:50.32 | Burgwork | [TK]D-Fender: the sipura phones. When I call an internal number, they are getting the name of the person from somewhere. Where is that coming from? |
| 19:50.51 | Assid | Burgwork: callerid(name) |
| 19:50.58 | Assid | check the caller id being set of the internal extension |
| 19:51.17 | Burgwork | Assid: on the phone? |
| 19:51.23 | [TK]D-Fender | Burgwork: CPID only comes from the phone directories last I checked, and that patch never made it "live". |
| 19:51.32 | Assid | or on the system |
| 19:51.33 | Burgwork | hmm |
| 19:51.34 | Assid | check both |
| 19:51.35 | [TK]D-Fender | Assid: No, on an OUTGOING call. |
| 19:51.40 | Assid | oh |
| 19:51.45 | Burgwork | it will show up on both phones |
| 19:51.53 | Assid | i thought providers dont carry name |
| 19:52.00 | Burgwork | mine says "calling XXX, at XXX" and their says "called by XXX, XXX" |
| 19:52.05 | Burgwork | this is only for internal extensions |
| 19:52.07 | vooduhal | Thanks guys. I think I got the information I needed. |
| 19:52.07 | Qwell[] | Assid: it isn't that it isn't carried |
| 19:52.34 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 19:53.37 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 19:55.17 | Burgwork | right, so the sipura phones are picking up the callerid field as the asterisk server is sending it to them? |
| 19:57.47 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 19:58.12 | [[blah]asfd | is the voip-info site having problems? or am i the only person who cannot access it? |
| 19:59.37 | Burgwork | [[blah]asfd: you are not the only person |
| 20:02.43 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:06.12 | b11d | voip-info is back up. |
| 20:06.27 | b11d | heh you all fell for it |
| 20:06.28 | b11d | :| |
| 20:06.53 | b11d | hah! you'll never find my hideout in /dev/urandom |
| 20:07.03 | Qwell[] | cat /dev/urandom > /dev/null |
| 20:07.07 | b11d | haha |
| 20:07.11 | b11d | such a great use of resources |
| 20:07.16 | Assid | hehe |
| 20:07.47 | b11d | actually wouldnt that cause urandom to overwrite /dev/null.. wouldnt cat /dev/urandom | /dev/null be better? |
| 20:08.19 | Corydon-w | No, because /dev/null is not an executable |
| 20:08.22 | *** join/#asterisk Stridernzl (n=neville@222-153-62-109.jetstream.xtra.co.nz) |
| 20:08.29 | b11d | it doenst need to be executable to be overwritten |
| 20:08.36 | b11d | the point is to pipe the data from urandom to null.. |
| 20:08.39 | b11d | so | is apt. |
| 20:08.39 | b11d | :) |
| 20:09.04 | Corydon-w | You'd only overwrite /dev/null if you tried to mv something to it. It's designed to take stdout/stderr |
| 20:09.11 | b11d | sigh.. |
| 20:09.16 | b11d | ok corydon.; |
| 20:09.26 | Corydon-w | Try it sometime |
| 20:09.34 | b11d | only if you break mercestes |
| 20:09.52 | Corydon-w | Done |
| 20:10.11 | Mercestes | what? |
| 20:10.16 | Mercestes | perposterous. |
| 20:10.32 | b11d | so what's everyone doing for St. Patricks day? |
| 20:10.54 | Mercestes | your mom. |
| 20:11.05 | b11d | sounds like a standard st. paddys day |
| 20:11.08 | b11d | she'll be at the AA meeting |
| 20:11.11 | Mercestes | lol |
| 20:11.23 | Mercestes | aren't we in the wrong chan? :P |
| 20:11.28 | b11d | I always wanted to start "Alcoholics Conspicious" |
| 20:11.32 | b11d | yeah usually |
| 20:11.40 | b11d | doh |
| 20:11.46 | Mercestes | Should switch before I end up in /topic. :P |
| 20:12.49 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 20:13.09 | b11d | shit.. we needed that cache. |
| 20:13.39 | *** join/#asterisk TaiSHi (i=WinNT@host221.190-30-53.telecom.net.ar) |
| 20:13.48 | TaiSHi | Mercestes, I've came from bathroom to torment you =_= |
| 20:15.08 | Mercestes | TaiSHi: Yay...tell me you washed your hands first. |
| 20:15.36 | TaiSHi | Of course |
| 20:15.42 | TaiSHi | I'm not that dirty ¬¬ |
| 20:15.43 | [TK]D-Fender | Qwell[]: Would you be so kind as to add a "WIKI Down" notice to the topic? |
| 20:15.57 | b11d | no one reads the topic anyway :) |
| 20:17.02 | TaiSHi | b11d: What's a topic ? |
| 20:17.21 | TaiSHi | Mercestes, I've already said I trust you, but added member => SIP/600 |
| 20:17.21 | [TK]D-Fender | b11d: No, thats only FreePBX users and simlar wretched villianous scum ;) |
| 20:17.22 | b11d | just a make believe place |
| 20:17.27 | b11d | hahaha |
| 20:17.30 | TaiSHi | And how do I check, erm... the status ? |
| 20:17.43 | b11d | just type /topic |
| 20:17.52 | b11d | ok maybe not |
| 20:17.58 | Mercestes | TaiShi Show queues. |
| 20:18.21 | Qwell[] | [TK]D-Fender: nah, I'm mean |
| 20:18.29 | [TK]D-Fender | :( |
| 20:18.37 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source PBX -=- Asterisk 1.4.1 (Mar. 2, 2007), Asterisk 1.2.16 (Mar. 2, 2007), Zaptel 1.2.15 (Mar. 2, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- Yes, voip-info.org is down. |
| 20:18.38 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com) |
| 20:18.49 | b11d | hahahaha |
| 20:18.51 | b11d | nice. |
| 20:18.51 | riddlebox | what is the cheapest polycom phone you can buy? |
| 20:18.57 | b11d | the 301 but you'll regret it |
| 20:18.59 | b11d | get the 430 |
| 20:18.59 | Qwell[] | riddlebox: used 301 |
| 20:19.01 | TaiSHi | Mercestes: queue show |
| 20:19.51 | tzafrir | What phones are capable of displaying messages sent with SendImage ? |
| 20:20.10 | b11d | heh i'd abuse that so badly ;) |
| 20:20.34 | tzafrir | soft phones, on Linux |
| 20:21.05 | Mercestes | TaiSHi: Uh, no...show queues |
| 20:21.05 | TaiSHi | Mercestes, erm, no, queue show |
| 20:21.05 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:21.05 | Mercestes | ... |
| 20:21.11 | TaiSHi | 1.4.1 ¬¬ |
| 20:21.17 | Mercestes | dd if=/dev/urandom of=/taishi's/face |
| 20:21.24 | Mercestes | oh...1.4.1 |
| 20:21.42 | Mercestes | core queue show verbose |
| 20:21.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:21.51 | Mercestes | or something rediculous like that. |
| 20:21.56 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:22.13 | TaiSHi | Nope |
| 20:22.17 | TaiSHi | queue show |
| 20:22.19 | TaiSHi | Just that |
| 20:22.19 | [TK]D-Fender | Qwell[]: Thanks |
| 20:22.45 | [TK]D-Fender | riddlebox: 301 is just fine, but perhaps you could describe the use you have envisioned for it... |
| 20:22.45 | b11d | no one will see that voip-info notice.. hell my topic cuts off after the 3rd 2007. |
| 20:23.24 | TaiSHi | b11d: client? |
| 20:23.33 | b11d | mIRC 6.something |
| 20:23.36 | riddlebox | [TK]D-Fender, is that the SoundPoint IP 301? |
| 20:24.01 | b11d | [15:25] [b11d VERSION reply]: mIRC v6.21 Khaled Mardam-Bey |
| 20:24.13 | [TK]D-Fender | riddlebox: Yes. |
| 20:24.19 | TaiSHi | Bleh, I'm using mIRC for windows and no inconvenientes |
| 20:24.44 | *** join/#asterisk topping (n=topping@ppp-67-124-90-162.dsl.pltn13.pacbell.net) |
| 20:24.54 | b11d | mIRC is one of the apps I still use from way back.. I cant think of another one.. maybe Winamp. |
| 20:24.59 | b11d | other than that i've replaced everything |
| 20:24.59 | Strom_C | xchat |
| 20:25.04 | Strom_C | :) |
| 20:25.08 | TaiSHi | b11d, agreed. |
| 20:25.19 | TaiSHi | Also... putty :D |
| 20:25.26 | b11d | thank god IRC is back "under the radar" again |
| 20:25.26 | riddlebox | [TK]D-Fender, I see it says it is SIP 2 line, so you can have two seperate sip accounts on that phone correct, that doesnt really mean that you can only have 2 CO's on that phone right? |
| 20:25.35 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 20:25.47 | *** join/#asterisk martin_harriss (n=martin@albion.Princeton.EDU) |
| 20:26.30 | b11d | ... |
| 20:27.11 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 20:27.38 | [TK]D-Fender | riddlebox: Means you can have 2 distinct registrations divided up between those lines keys, each key supporting 1-5 (if not 8) calls |
| 20:27.42 | [TK]D-Fender | 8* |
| 20:28.54 | b11d | "Oi! You're just some punk Kid isn't ya?" |
| 20:31.11 | b11d | ugh how can there be 315 idlers here.. |
| 20:31.12 | b11d | come on.. |
| 20:31.18 | b11d | yap it up |
| 20:31.37 | b11d | :) |
| 20:32.15 | mcab | we're not idle, we just talk real slow |
| 20:32.23 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:32.26 | mishehu | polycom spip 1.6.7 question - I'm getting |4|01|Edit|Parse error 9 with local cfg, is there any way to get it to tell me which file and what line number? |
| 20:32.29 | b11d | I thought this channel had a very specific anti-Ent policy. |
| 20:33.51 | mcab | mishehu: don't think so, no. try opening your config files in a validating XML editor, that might give you some clues |
| 20:34.30 | mcab | b11d: *harrump* just try and stop an Ent |
| 20:34.33 | b11d | :) |
| 20:35.04 | generalhan | anyone here familiar with setting up remote SIP Phones with Asterisk ? for some reason i cannot get this stupid phone to register. Here are my configs >> http://generalhan.pastebin.ca/395122 << can some one take a look and see if you notice anything incorrectly setup ? |
| 20:35.41 | *** join/#asterisk tinrsh (n=claudiu@81.181.94.112) |
| 20:35.41 | b11d | do we need to ask about firewalls inbetween? |
| 20:35.43 | generalhan | and let me know if there is something else that i need to check on, besides what i listed ... i really need to get this remote phone working and ive hit a road block that i cant get passed |
| 20:35.50 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 20:35.53 | tinrsh | hi all |
| 20:35.56 | generalhan | b11d: that Q to me ? |
| 20:36.12 | b11d | aye |
| 20:36.24 | generalhan | i listed the forwarded ports on my 2 routers in the paste |
| 20:36.35 | b11d | ah |
| 20:36.41 | generalhan | unless there is another step besides forwarding the ports |
| 20:36.49 | b11d | and nothing at all is coming up on the console related to it's failure? |
| 20:36.52 | b11d | it just doesnt register? |
| 20:37.05 | generalhan | b11d: nothing ... and yea, it just doesnt register |
| 20:37.18 | b11d | and both points are pingable? |
| 20:37.24 | generalhan | the Dlink isnt so great with firewall logs so i dont get much help there |
| 20:37.28 | b11d | ah |
| 20:37.30 | `p4r14h`work | hello, i have an asterisk box setup using VoIP and everything is running fine. I added a x100p to the server and configured all the zaptel stuff. I have a zap channel up and running and the x100p is hooked to a POTS line, but the alarm is red. can anyone help me out here? |
| 20:37.49 | generalhan | yes i can ping them both ... and i can use nmap to test port 5060 and it passes as well |
| 20:37.55 | b11d | thats bizarre eh |
| 20:38.08 | b11d | that x100p.. is it FXO or FXS? |
| 20:38.14 | `p4r14h`work | FXO |
| 20:38.19 | b11d | weird |
| 20:38.25 | generalhan | b11d: VERY. and it sux cause i swear im doing this correctly |
| 20:38.35 | b11d | hmm.. does tcpdump show anything hitting your pbx? |
| 20:38.37 | generalhan | but i get nothing in console about it even trying to register |
| 20:39.03 | generalhan | ?? |
| 20:39.19 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 20:39.21 | b11d | are you using a packet sniffer to ensure that the data is even reaching your PBX machine? |
| 20:39.28 | b11d | asterisk warnings or not.. |
| 20:39.31 | generalhan | no |
| 20:39.43 | b11d | i'll wager your issue is that dlink for one reason or another |
| 20:39.50 | generalhan | goodie |
| 20:39.57 | b11d | haha here we are on square 1 |
| 20:40.20 | `p4r14h`work | b11d, is there anything else besides the line that could trigger the alarm? |
| 20:40.55 | b11d | im actually not sure.. |
| 20:40.56 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 20:40.58 | `p4r14h`work | also i am using the line that was supplied with the x100p |
| 20:40.59 | irule | hi |
| 20:41.14 | `p4r14h`work | ive tried different lines etc. no workie :( |
| 20:41.33 | tinrsh | if I have an external peer to my telco company for which I have to register on using 6 phone numbers, for receiving calls from that peer do I also have to create 6 sip peers ? |
| 20:42.10 | b11d | well will you have all six being handled on one phone? or six phones? |
| 20:42.15 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:42.32 | b11d | `p4r14h`work.. might be a bad card.. out of the box issues with x100's arent rare. |
| 20:42.38 | b11d | but again. im really NOT sure. |
| 20:43.00 | *** join/#asterisk m0f0x (n=dsafsa@unaffiliated/m0f0x) |
| 20:43.06 | `p4r14h`work | b11d, ok well thanks for the help :D |
| 20:43.10 | b11d | :) |
| 20:43.14 | m0f0x | hi... anybody here getting good results with app_txfax ? |
| 20:43.29 | b11d | people seem to be 50/50 on that m0f0x |
| 20:43.50 | tinrsh | b11d: I will bridge them to variuous fxs media gateways depending on caller and called number |
| 20:43.55 | m0f0x | b11d mine works a little bit fine... but it holds the line after a sucessful transmission... |
| 20:44.06 | `p4r14h`work | anyone else help me with a red alarm on a X100P hooked to POTS? |
| 20:44.13 | m0f0x | b11d any clue about that? i can only release the line by stopping asterisk. |
| 20:44.16 | b11d | why not set a timeout and force a hangup? |
| 20:44.53 | b11d | `p4r14h`work.. you DID load the kernel drivers for your x100p right? |
| 20:45.06 | `p4r14h`work | yessir its all loaded on boot |
| 20:45.14 | b11d | ok, cool. Does it stay red when you disconnect? |
| 20:45.17 | m0f0x | b11d I already did, but asterisk keeps waiting TxFAX to give an exit code or something.... |
| 20:45.18 | b11d | and what about making a loopback? |
| 20:45.27 | `p4r14h`work | it does stay red when d/c |
| 20:45.28 | b11d | m0f0x.. yeah.. :) that sucks. |
| 20:45.51 | b11d | i'd send an email to digium or whoever makes the x100p.. |
| 20:45.53 | `p4r14h`work | i dont have a crimp tool for rj-11 :( |
| 20:45.53 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 20:45.56 | b11d | it might be a bad card.. |
| 20:46.05 | m0f0x | b11d mine is a TDM400P |
| 20:46.13 | `p4r14h`work | thats what i was sorta thinking |
| 20:46.14 | b11d | yeah I rock a TDM400P too |
| 20:46.18 | m0f0x | b11d yours work fine ? |
| 20:46.30 | b11d | hell yeah.. when I used it. I switched to sangoma a104d's and moved to a PRI. |
| 20:46.30 | irule | hi all, may I please get a little help? I installed a couple tiger3xx cards and got a compilation error on zaptel http://pastebin.ca/395141 thanks |
| 20:46.37 | `p4r14h`work | b11d, anyway to make a loopback without a crimp tool for rj11? |
| 20:46.42 | b11d | not that I know of :( |
| 20:47.08 | b11d | well you could possibly strip a length of an existing cable, and twist the right wires together |
| 20:47.13 | b11d | it'd be really crude though |
| 20:47.21 | `p4r14h`work | i have to take the telephone dood out to sushi if i want to use his tools or services :( |
| 20:47.26 | b11d | haha |
| 20:47.31 | b11d | im going to take that out of context :) |
| 20:47.37 | *** join/#asterisk Malph (n=chatzill@66-231-0-194.hosts.sdnet.net) |
| 20:47.52 | b11d | you just want access to his "tools" eh |
| 20:47.56 | b11d | sick bastard. :) |
| 20:47.57 | `p4r14h`work | LOLz |
| 20:47.58 | b11d | j/k |
| 20:48.16 | Malph | I understand voip-info is down any idea when it will be back up? |
| 20:48.17 | `p4r14h`work | not exactly... maybe his crimp tool, not any others =\ |
| 20:48.22 | `p4r14h`work | :) |
| 20:48.26 | b11d | Malph.. sooner or later. |
| 20:48.40 | b11d | heh.. like I said, you MIGHT be able to make one from an already-terminated cable |
| 20:48.46 | b11d | just cut it, strip it down, and connect the right wires.. |
| 20:48.52 | b11d | it'll be hard and crude, but it might work., |
| 20:48.55 | m0f0x | b11d can you send me one of your extension about tx_fax? |
| 20:48.59 | `p4r14h`work | ill search for a solution for that |
| 20:49.04 | quidpro | Malph: Just use Google's cache.. the more popular pages of voip0info are cahced there |
| 20:49.08 | `p4r14h`work | should the red alarm go away if unplugged? |
| 20:49.14 | b11d | no, im not using txfax in production.. |
| 20:49.19 | b11d | looking at it though |
| 20:49.48 | b11d | no but just a disconnect/reconnect might seat the cable right and make old RED go away. |
| 20:50.05 | Malph | quidpro: thanks the pages were just loading so low off google cahe that I decided to whine |
| 20:50.07 | m0f0x | yep. hard reset works fine |
| 20:50.16 | b11d | http://lists.digium.com/pipermail/asterisk-users/2004-September/056558.html |
| 20:50.20 | b11d | thats interesting.. |
| 20:50.26 | b11d | about x100p RED's |
| 20:50.32 | b11d | their RAID was causing it. |
| 20:50.58 | b11d | read that thread. |
| 20:52.08 | tinrsh | guys any ideeas why after starting asterisk, the chan_sip.so starts working after a few minutes, and until then if I issue sip <enter> at asterisk console I get "no such command " |
| 20:52.36 | Qwell[] | tinrsh: because sip isn't a command |
| 20:52.41 | b11d | yeah |
| 20:52.42 | b11d | type sip ? |
| 20:53.04 | b11d | I wish those error messages MADE sense :) |
| 20:53.09 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 20:53.18 | tinrsh | Qwell[]: sorry, I hurried, I have no sip <...> commands available |
| 20:53.33 | b11d | then you arent running chan_sip |
| 20:53.38 | tinrsh | and no sip peer or user can connect |
| 20:53.39 | b11d | I think |
| 20:53.43 | tinrsh | it is loaded |
| 20:53.47 | b11d | is sip in "show modules" ? |
| 20:53.52 | tinrsh | yes |
| 20:53.57 | b11d | you have a sip.conf ? |
| 20:54.10 | tinrsh | and after letting it like this for about 3-5 minutes, it start working |
| 20:54.21 | b11d | ok thats bizarre |
| 20:54.29 | tinrsh | yes I do have sip conf with 46 peers |
| 20:54.32 | Qwell[] | well, if you have really slow DNS resolution, that's normal |
| 20:54.43 | b11d | i'd fix that.. |
| 20:54.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 20:54.48 | b11d | run a local DNS external cache or something |
| 20:55.26 | tinrsh | I have dns server in the same switch with the asterisk box |
| 20:55.55 | tinrsh | and for the domains that asterisk works with that dns server is authoritar |
| 20:55.58 | b11d | is your resolv.conf correct? |
| 20:56.03 | tinrsh | yup |
| 20:56.11 | tinrsh | checked it more times |
| 20:56.14 | b11d | heh |
| 20:56.27 | b11d | what version of asterisk? |
| 20:56.42 | tinrsh | I've busted my head today with that |
| 20:56.45 | tinrsh | 1.2.13 |
| 20:56.57 | b11d | i'd consider upgrading to .16 at least |
| 20:57.01 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 20:57.03 | kippi | hey |
| 20:57.04 | kippi | how do I get into the menu for nortel handsets? |
| 20:57.10 | b11d | anything in the changelog about this between .13 and .16? |
| 20:57.16 | b11d | www.nortel.com i'd guess |
| 20:57.17 | b11d | who knows |
| 20:57.17 | b11d | :) |
| 20:57.29 | b11d | is this #asterisk-for-nortel-phones ? |
| 20:57.30 | b11d | :) |
| 20:57.50 | generalhan | hmm im still running 2.10 think upgrading may help my issue ?? lol. |
| 20:57.56 | b11d | 2.10 ? |
| 20:57.58 | b11d | i want a copy :) |
| 20:58.03 | kippi | just need to get into the configure so I can config the handsets |
| 20:58.14 | b11d | rtfm |
| 20:58.16 | [TK]D-Fender | kippi: Nortel makes all KINDS of handsets. Perhaps a model & central system info would help... |
| 20:58.24 | *** join/#asterisk Stridernzl (n=neville@222-153-62-109.jetstream.xtra.co.nz) |
| 20:59.04 | kippi | i2002 |
| 20:59.34 | b11d | http://www.congruentip.com/pages/Nortel%20i2002%20Manual1.pdf |
| 20:59.59 | generalhan | [TK]D-Fender: will you take a look at this paste ... i posted it JUST after you left and you had been helping me on this before. just want to verify that i have these steps correc at least ! http://generalhan.pastebin.ca/395122 |
| 21:00.13 | mihinomenest | how do I set the sipura PAP2's equivalent of "canreinvite"? |
| 21:04.45 | [TK]D-Fender | generalhan: How about telling me what I should be looking FOR... |
| 21:06.02 | generalhan | [TK]D-Fender: just something that looks wrong. like i know i checked this, but if you noticed that i was forwarding the complete wrong rtp port range or something. or that i had the the wrong lines in the sip.conf |
| 21:06.31 | generalhan | anything really ... ive done this by the book and still cant get the phone to register remotely. im working with a Cisco 7960 and an Aastra 9112i |
| 21:07.22 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:07.28 | [TK]D-Fender | generalhan: make sure to add "canreinvite=no" to [general] |
| 21:08.19 | generalhan | hmm |
| 21:08.20 | generalhan | alright |
| 21:08.23 | generalhan | will do |
| 21:08.28 | b11d | gotta run lads |
| 21:08.29 | b11d | ttyl all |
| 21:08.48 | [TK]D-Fender | mihinomenest: You don't set it in the device, you set it in * |
| 21:09.15 | generalhan | [TK]D-Fender: its weird the only requests i see from my remote location on my *-side router are these: http://generalhan.pastebin.ca/395177 and i have NO IDEA what those ports are, or why they are being used |
| 21:09.55 | [TK]D-Fender | generalhan: Show me some unfiltered SIP debug, output of your local servers firewall, etc. |
| 21:11.20 | mishehu | hmm can't get to voip-info, anybody know of a validator for polycom configs? |
| 21:13.56 | [TK]D-Fender | mishehu: Yeah... the error log on your provisioning server :) |
| 21:15.21 | mishehu | [TK]D-Fender: yeah, it's giving me an error of "Parse error 9 in local cfg"... no line number or filename... |
| 21:15.44 | [TK]D-Fender | mishehu: So start looking at every tag you changed.... |
| 21:15.46 | mishehu | I jacked up the log level in ipmid.cfg and sip.cfg... |
| 21:16.29 | mishehu | [TK]D-Fender: I did, since I only changed one registration in it, nothing else. that's why I'd like it to tell me where the parse error is, that'd help me find it. |
| 21:17.29 | [TK]D-Fender | mishehu: And I'd like $1,000,000... seems NOBODY is leaving happy today... |
| 21:17.41 | mishehu | [TK]D-Fender: apparently not. :-/ |
| 21:17.41 | [TK]D-Fender | mishehu: And you shouldn't HAVE an ipmid.cfg |
| 21:18.00 | mishehu | [TK]D-Fender: that depends on the firmware version if I'm not mistaken... |
| 21:18.16 | [TK]D-Fender | mishehu: 1.4 or earlier? MGCP? thats the only valid reason... |
| 21:19.46 | mishehu | [TK]D-Fender: configs were from 1.3.x, never changed them since it's *supposed* to be backwards compatible. didn't want to fix it until it's broke. now it's broke... maybe I'll fix it. |
| 21:20.15 | [TK]D-Fender | mishehu: All sorts of things ave changed in the YEARS that have passed... |
| 21:20.27 | [TK]D-Fender | mishehu: I take it you've just upgraded the firmware? |
| 21:21.32 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 21:22.41 | *** join/#asterisk Primus_ (n=primus@68.144.64.207) |
| 21:23.26 | mishehu | [TK]D-Fender: running 1.6.7 at the moment. |
| 21:24.18 | [TK]D-Fender | mishehu: Not a bad jump, but consider that jumping 3 majoy revisions won't just break, but more likely SHATTER your old configs. Time to start again fresh from scratch. |
| 21:24.25 | mishehu | at least according to the docs I read, it is supposed to be backwards compatible. seems that I was able to localize the parse error to the phone-specific config, though I still can't find where it is yet. |
| 21:24.47 | [TK]D-Fender | mishehu: And do yourself a favor : no stop-gap measures. Go get SIP 2.1.0 and BootROM 3.2.3 from your vendor and get them on the latest |
| 21:25.14 | mishehu | [TK]D-Fender: you know, that's exactly what I thought, but my home system still ticks away without being fixed. I'm planning on getting 2.1.x becaue of the DST crap at least. |
| 21:25.17 | mishehu | if nothing else. |
| 21:25.25 | mishehu | but I forgot who my vendor was heh. |
| 21:25.46 | mishehu | shame that freedomphones doesn't have that up yet. |
| 21:25.49 | *** join/#asterisk dasenjo (n=dasenjo@190.24.177.35) |
| 21:25.54 | [TK]D-Fender | mishehu: Plenty of other reasons. SIP 2.1.0 adds micro-browser support to the IP 430 & 501 |
| 21:26.09 | [TK]D-Fender | mishehu: Tables support, etc... lots of fixes. |
| 21:26.17 | mishehu | [TK]D-Fender: I never tried to use the microbrowser, how is it on the 501? |
| 21:26.35 | [TK]D-Fender | mishehu: Doubles the audio resolution on the 501 as well at a minimum |
| 21:26.43 | [TK]D-Fender | mishehu: its decent. |
| 21:27.06 | [TK]D-Fender | mishehu: Audio VOLUME that is. |
| 21:29.08 | mishehu | [TK]D-Fender: well it's definitely on my list of things to do, I just need to be able to sit down without everything else breaking on me as it has the past couple weeks and then I can update everything heh. |
| 21:29.43 | [TK]D-Fender | mishehu: well as I said, don't doa half-way fix... its just a long as the full-job |
| 21:30.14 | [TK]D-Fender | mishehu: Except more likely to need to be repeated on a shorter time-scale |
| 21:30.40 | *** part/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 21:34.01 | Strom_C | hey, here's a silly question |
| 21:34.07 | Strom_C | what's the difference between IAX and IAX2? |
| 21:34.10 | mcab | mishehu: you can make the DST config change to pretty much any version of the polycom software, but for god-sake, and the sake of your own sanity, upgrade the configs to ones compatable with your current S/W load |
| 21:34.13 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:34.22 | Strom_C | I haven't been around long enough to have used IAX1 :) |
| 21:34.39 | mishehu | I am, [TK]D-Fender talked me into it. |
| 21:34.58 | wunderkin | wow, i think i was, but i don't remember the difference |
| 21:35.00 | mcab | he is wise in the way of Polycom ;-) |
| 21:35.00 | mishehu | am making preparations now as we speak. |
| 21:35.09 | mishehu | yeah, that's why I'm doing it. |
| 21:35.22 | [TK]D-Fender | aaaaaoooouuummmmmmmm |
| 21:35.22 | mishehu | I respect his advice |
| 21:35.32 | mishehu | [TK]D-Fender: muuuuuuuuuuuuuuuuuuuuuuuu |
| 21:35.39 | mishehu | thats how I meditate. |
| 21:35.42 | mcab | I tend to isolate my config changes to my own file, then just overwrite the phone1.cfg and sip.cfg files with each new load |
| 21:35.49 | mishehu | and soon I'll need some drugs to medicate! heh |
| 21:35.55 | creature1 | hm, is the udp ports used for audio? |
| 21:36.02 | [TK]D-Fender | mishehu: "Moo" is what my beef is supposed to be doing 5 minutes before its served ;) |
| 21:36.19 | Strom_C | more like moo.gsm |
| 21:36.31 | [TK]D-Fender | phone1.cfg is a SAMPLE and should never be used with that name..... |
| 21:36.40 | mishehu | [TK]D-Fender: and it's supposed to be telling you about how it's been working out, and how it would really like it if you made a roast out of its rump... |
| 21:36.57 | wunderkin | i need some time to meditate more on my problems too :P setting registration timeout to 1 hour seemed to help greatly but we still have 2 people having problems.. are they doing it at the same time as the registration, is the 30 sec nat keepalive too much, or is there another problem? hmm |
| 21:37.00 | Strom_C | what is this, Milliway's? |
| 21:37.19 | mishehu | Strom_C: sure, why not? I'm starved. didn't have lunch today :/ |
| 21:37.49 | Strom_C | if you've done six impossible things this morning, why not round it out with breakfast at Milliway's? |
| 21:37.56 | [TK]D-Fender | wunderkin: Typical keep-alive is around 2s (qualify=yes). You normally DON'T have to set the phone to know its behind NAT, you jsut let * do its thing. |
| 21:38.07 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:38.07 | mcab | [TK]D-Fender: I set up my config line in <mac>.cfg like this: "<per-phonefile>.cfg,phone1.cfg,<per-systemfile>.cfg,sip.cfg". works well, and makes sure I don't miss any vital params |
| 21:38.51 | wunderkin | yeah, that is the only nat related thing i put in, i would loose the nat hole if i didnt put it in, because i have qualify off, i would rather have it off :P |
| 21:39.00 | [TK]D-Fender | mcab: I typically advocate sipcfg, then phoneXXXX.cfg where XXXX hold all the reg/phone specific stuff. |
| 21:39.17 | wunderkin | i'm not sure what else is special about us that would cause cpu usage problems |
| 21:39.23 | mishehu | shit, I think I missed a ups delivery too just now :-/ |
| 21:39.24 | [TK]D-Fender | wunderkin: I run a Polycom behind NAT here. turn it ON :) |
| 21:39.35 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 21:39.37 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376371.dsl.bell.ca) |
| 21:39.42 | wunderkin | qualify on? yeah that would be the next step, yucky though |
| 21:39.45 | [TK]D-Fender | wunderkin: My home IP 301 has 1 reg for home, 1 for a remote client's server |
| 21:40.10 | [TK]D-Fender | wunderkin: Feel free to up the qualify to 5s, but I would leave it around there. |
| 21:40.15 | mcab | [TK]D-Fender: right, I just like to make sure all my per-phone changes over-ride anything in sip.cfg. lets me give different people different idle-browsers, with a default one for everyone |
| 21:40.30 | tinrsh | any one why do I get error: dereferencing pointer to incomplete type when compiling asterisk-1.2.16 in codec_zap.c ? |
| 21:40.45 | [TK]D-Fender | mcab: Yeah I copy that tag section over to the phoneXXX.cfg for personalization. |
| 21:41.18 | [TK]D-Fender | mcab: And at which point, technically EVERYONE has their own MB pages as the ext is a parm on all lines calling it. |
| 21:41.38 | [TK]D-Fender | mcab: Helps with click-to-call scripts, etc. |
| 21:42.04 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 21:42.45 | mcab | aaaah, nifty :-) |
| 21:42.59 | mishehu | ugh, only thing I hate about polly com configs is that they hurt more than polly biting your finger off to make. too many settings heh ;-) |
| 21:43.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:43.07 | tzanger | heh |
| 21:43.10 | [TK]D-Fender | mishehu: Nowhere near so bad. |
| 21:43.19 | tzanger | I'm trying to get the damn phones to accept DST stuff and override the DHCP settings |
| 21:44.16 | nortex | tzanger, Where are you having problems with that? What firmware? |
| 21:44.23 | creature1 | Hm, i'm doing some tests to register a remote client with my asterisk. i have set RTP port range and SIP port.. he opened the same ports in his firewall and set the ports in his software. that should be it right? |
| 21:44.36 | tzanger | 1.6.something I think |
| 21:44.36 | creature1 | however, he can register |
| 21:44.39 | [TK]D-Fender | tzanger: You can cheat on jsut about any version at worst. |
| 21:44.43 | creature1 | but we cannot hear eachother at all |
| 21:44.51 | mcab | mishehu: once you script the config generation, it's no longer a headache :-) |
| 21:44.54 | *** join/#asterisk surfhai (n=surfhai@dslb-084-058-214-010.pools.arcor-ip.net) |
| 21:45.06 | [TK]D-Fender | creature1: His side should not have to do ANYTHING. |
| 21:45.23 | mishehu | [TK]D-Fender: another q about polly coms, what's the bootblock and why does it not seem to update? |
| 21:45.28 | [TK]D-Fender | creature1: YOURS may need a whack of settings if its behind nat itself |
| 21:45.31 | tzanger | [TK]D-Fender: parlez-vous francais? |
| 21:45.43 | creature1 | [TK]D-Fender: Yeah, it is. The ports are forwarded. |
| 21:46.00 | [TK]D-Fender | mishehu: Not 100% sure to tell you the truth. I think thats the "ultimate fallback" to prevent hard-bricking. |
| 21:46.02 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 21:46.21 | creature1 | [TK]D-Fender: but his side wouldn't work if he's behind nat using firewall etc..? |
| 21:46.28 | rhombus | So, who blew up the wiki, and what do we do about it? |
| 21:46.28 | [TK]D-Fender | tzanger: Non, mem-pas une seul mot! |
| 21:46.28 | creature1 | and blocked.. or would it? |
| 21:46.56 | *** part/#asterisk surfhai (n=surfhai@dslb-084-058-214-010.pools.arcor-ip.net) |
| 21:46.56 | nortex | tzanger, what settings are you using, can you pastebin them? |
| 21:46.56 | mcab | mishehu: the boot block is the part that gives the phones fail-safe bootrom upgrades. you rarely (if ever) should need to update it. It's also a manual process. |
| 21:46.56 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:46.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:46.57 | tzanger | [TK]D-Fender: what's the difference between using lui for 'hee' and il? |
| 21:47.03 | tzanger | (trying to help my stepson with his French homework) |
| 21:47.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:47.07 | [TK]D-Fender | creature1: pastebin your sip.conf masking ONLY passwords, and describe the full network path between * and him. |
| 21:47.19 | creature1 | [TK]D-Fender: will do |
| 21:47.24 | rhombus | tzanger: je parle francais |
| 21:47.44 | tzanger | nortex: haven't got htat far yet, I'm trying to find the dhcp overrides that someone else menitoned to me |
| 21:48.15 | [TK]D-Fender | tzanger: Not sure on your "hee" reference. Lui = him |
| 21:49.15 | tzanger | [TK]D-Fender: hmm... "lui c'est mon cousin" is saying is translated as "he is my cousin" |
| 21:49.17 | rhombus | Il is the subject and lui is object |
| 21:49.32 | tzanger | rhombus: man I suck at English let alone anothe rlanguage :_) |
| 21:49.42 | rhombus | tzanger: something is missing from that sentence |
| 21:49.48 | [TK]D-Fender | tzanger: Correct |
| 21:50.10 | rhombus | tzanger: what are we trying to do here, exactly? |
| 21:50.13 | tzanger | lui c'est mon cousin - shouldn't that translate to "him is my cousin" if lui = him? |
| 21:50.28 | rhombus | tzanger: the sentence is grammatically incorrect |
| 21:50.29 | [TK]D-Fender | tzanger: he/him depending on context. |
| 21:50.36 | tzanger | rhombus: the book says "lui c'est mon cousin" and then says that that means "he is my cousin" |
| 21:50.40 | *** join/#asterisk voipy (n=pirch@a81-84-60-131.cpe.netcabo.pt) |
| 21:50.52 | rhombus | tzanger: where in heaven's name did you get the book? |
| 21:50.58 | rhombus | tzanger: what part of the world are you in? |
| 21:51.03 | tzanger | rhombus: Ontario, Canada |
| 21:51.19 | [TK]D-Fender | tzanger: T'es pas incompetant comme LUI ;) |
| 21:51.34 | file | oh... that reminds me |
| 21:51.35 | tzanger | heh |
| 21:51.37 | mishehu | mcab: ah ok, just was curious to know why it wasn't in sync with the bootrom version |
| 21:51.40 | rhombus | tzanger: You see that structure in informal and slang styles, but it's not strictly correct |
| 21:51.43 | file | tzanger: you should grab an expo pass to IT360 |
| 21:52.00 | rhombus | tzanger: you can say "Il est mon cousin" |
| 21:52.04 | [TK]D-Fender | <- I'm an alien... I'm a legal alien... I'm an Englishman in Quebec! |
| 21:52.10 | tzanger | rhombus: that's what I'd have said |
| 21:52.17 | rhombus | tzanger: well, you're right |
| 21:52.19 | watchy | ur the man now dog |
| 21:52.22 | tzanger | file: :-) I'm not much for confs |
| 21:52.35 | rhombus | tzanger: and the textbook is dumb (and believe me, there are a lot of lousy textbooks out there) |
| 21:52.37 | [TK]D-Fender | rhombus: "lui" might still apply if you're pointing him out from far away :) |
| 21:52.40 | file | tzanger: fine! don't come party with blitzrage and me |
| 21:52.49 | mcab | mishehu: fair question, basically it doesn't need to be. Only worry about it if Polycom Support tells you to :-) |
| 21:53.05 | rhombus | [TK]D-Fender: yes, but then there's context and punctuation missing in the sentence |
| 21:53.18 | tzanger | rhombus: hmm okay, I appreciate both you and [TK]D-Fender's minor excursion from normal #asterisk talk :-) |
| 21:53.20 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 21:53.27 | [TK]D-Fender | rhombus: Don't blame me for his lack of expression! :) |
| 21:53.40 | rhombus | tzanger: It takes a village to raise a child, even if it's the global village |
| 21:53.57 | [TK]D-Fender | "PUNCTUATE!", he exclamed. |
| 21:54.16 | mut | exclamated! |
| 21:54.20 | rhombus | tzanger: just make sure your kid doesn't turn into a criminal delinquent, and it will have been worth it |
| 21:54.23 | mut | :O |
| 21:54.23 | tzanger | back to the polycom dhcp overrides though... I'm looking for 'dhcp' in my various configs |
| 21:54.25 | [TK]D-Fender | rhombus: And they can have their idiot back now ;) |
| 21:54.28 | tzanger | I see one line which may be helpful |
| 21:54.30 | nortex | tzanger, http://www.pastebin.ca/395239 The DHCP overisde is in the SNTP section. In my case I did not overide since the server specified is the same, but the daylight savings settings are there. |
| 21:54.30 | tzanger | <PROTECTED> |
| 21:54.38 | tzanger | that's a long one, holy hell |
| 21:54.41 | Qwell[] | heh, I was just looking at that... |
| 21:54.51 | [TK]D-Fender | tzanger: OVERRIDE <- |
| 21:54.54 | tzanger | yeah but the DHCP offset is "fixed" in the dhcp config |
| 21:55.12 | tzanger | er sorry the DHCP-supplied offset is fixed.. to change it I have to screw with the dhcp config. restart dhcp and cycle th ephone |
| 21:55.14 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:55.19 | tzanger | seems veyr sub-optimal |
| 21:55.21 | [TK]D-Fender | tzanger: Cheat and jsut set the month to "3" and override to true shoving in your SNTP server |
| 21:55.34 | tzanger | ahh another clue to the lui/il thing... the book is from Europe |
| 21:55.46 | Assid | wow.. my ringer is absolutely messed up |
| 21:55.49 | tzanger | perhaps Parisian French is different in this regard to Quebecois French |
| 21:55.49 | Assid | too low |
| 21:55.50 | Qwell[] | [TK]D-Fender: paste :p |
| 21:55.56 | rhombus | Who runs the wiki? How do we contact this person? |
| 21:56.00 | rhombus | Is this person on holiday? |
| 21:56.21 | rhombus | tzanger: I doubt it |
| 21:56.22 | voipy | Does anyone use Chan_cellphone and knows how to solve the bluetooth pairing prob on bluez-utils 3.7-1? |
| 21:56.31 | rhombus | tzanger: does it show a picture of a person pointing? |
| 21:56.31 | [TK]D-Fender | Qwell[]: paste what? |
| 21:56.31 | nortex | Qwell, I just pastebin a working config section http://www.pastebin.ca/395239 |
| 21:56.38 | Qwell[] | [TK]D-Fender: that |
| 21:56.48 | tzanger | rhombus: nope |
| 21:56.54 | [TK]D-Fender | Qwell[]: Why would I paste that... he just did ;) |
| 21:57.16 | rhombus | tzanger: it's just very context specific and the translation is not great |
| 21:57.31 | tzanger | [TK]D-Fender: hmm okay let me see... I don't want to override the server or the offset, but the daylightsavings I want it to use on its own |
| 21:57.36 | rhombus | tzanger: and textbooks in France and Québec both teach standard French |
| 21:57.37 | Assid | welcome sound doesnt work me either |
| 21:57.37 | Assid | damn |
| 21:57.41 | tzanger | rhombus: gotcha. I never was much good at French |
| 21:58.00 | tzanger | I know enough to ask for a coffee with cream and sugar please, ask where the can is and to apologize and say I don't speak French very well :-) |
| 21:58.16 | nortex | I used the polycom settings from 2.1.0 and it has been perfect. The overide was not needed since the server only provides the UTC time and the phone specifies its offset and then the DST settings. |
| 21:58.30 | rhombus | tzanger: being from Ontario, you might like "Bon Cop, Bad Cop" |
| 21:58.35 | tinrsh | any one why do I get error: dereferencing pointer to incomplete type when compiling asterisk-1.2.16 in codec_zap.c ? |
| 21:58.48 | tzanger | rhombus: oh yeah? I saw that at blockbuster |
| 21:58.57 | rhombus | tzanger: it's pretty funny |
| 21:59.29 | [TK]D-Fender | Apparently it rocks... I have a DivX of it w/o subtitles... now I can enjoy it more "live" :) |
| 21:59.42 | rhombus | tzanger: sorta encapsulates our petty domestic dispute over language and culture very well |
| 21:59.42 | tzanger | :-) |
| 21:59.54 | tzanger | oh yeah? I'm definitely going ot have to see it now |
| 21:59.55 | rhombus | it makes fun of both sides in equal measure |
| 22:00.21 | rhombus | and what's Canadian, if not doling out even the insults equally? :P |
| 22:00.43 | rhombus | wikiwikiwikiwiki |
| 22:00.45 | tzanger | rhombus: precisely! |
| 22:00.53 | rhombus | i'm jonesing for the wiki |
| 22:01.13 | tzanger | [TK]D-Fender: hmm so I guess I need to have the dhcp GMT override turned on if I want the phone to handle DST too... there doesn't appear to be a separate override for DST |
| 22:01.38 | *** part/#asterisk voipy (n=pirch@a81-84-60-131.cpe.netcabo.pt) |
| 22:01.48 | [TK]D-Fender | tzanger: DST is not part of SNTP |
| 22:01.55 | rhombus | wiki? |
| 22:02.06 | [TK]D-Fender | tzanger: Thats region and therefor device dependant |
| 22:02.06 | rhombus | i wonder... |
| 22:02.12 | rhombus | who runs the wiki. |
| 22:02.34 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:02.58 | JT | commpartners |
| 22:03.01 | nortex | All you Canadians have a good night and good luck with your polycom configs :) I'm going home. |
| 22:03.50 | Assid | damn.. the welcome sound never works for me |
| 22:03.53 | creature1 | what might be wrong when i get like 'scrambled' voice? the other peer can hear me just fine. should i suppose that it's his client that's weirdo? :D |
| 22:04.07 | tzanger | [TK]D-Fender: yes, but I have to override the GMT offset from DHCP, and then the phone should also take into account all of its lovely DST settings in that SNTP stanza |
| 22:04.24 | [TK]D-Fender | tzanger: No, you shouldn't actualyl have to override that... |
| 22:04.44 | [TK]D-Fender | tzanger: DST is the only thing NOT passed in DNTP |
| 22:04.48 | [TK]D-Fender | SNTP* |
| 22:05.08 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:05.35 | JT | creature1: can't you test it for yourself to hear the noise? |
| 22:05.42 | mcab | tzanger: your GMT offset should be set to Standard Time. DST is then added as necessary by the phone. |
| 22:05.44 | creature1 | JT: works great when calling pstn phones |
| 22:05.48 | tzanger | [TK]D-Fender: doesn't seem to be listening to me for some reason |
| 22:05.56 | tzanger | mcab: yeah, I understand that, and that's what I did have on the phones |
| 22:06.06 | tzanger | i.e. the dhcp server was set to -18000 for EST |
| 22:06.19 | creature1 | JT: this problem only occurs when i let someone outside my network register with my asterisk and make an internal call |
| 22:06.20 | tzanger | and the SNTP line up there seems to indicate that DST is in... april |
| 22:06.20 | tzanger | fuck |
| 22:06.21 | tzanger | that's why |
| 22:06.25 | tzanger | I'm an idiot |
| 22:06.32 | mcab | heh :-) |
| 22:06.44 | Qwell[] | wait, they pushed DST out a week at the end too? |
| 22:06.52 | Strom_C | Qwell[]: yes |
| 22:07.00 | Qwell[] | ugh |
| 22:07.23 | Strom_C | the HI/COCKS protocol (RFC 4373) |
| 22:07.45 | [TK]D-Fender | .... |
| 22:07.47 | [TK]D-Fender | no |
| 22:07.49 | [TK]D-Fender | COMMENT |
| 22:08.16 | xheliox | ... indeed. |
| 22:08.16 | tzanger | HI/COCKS? heh |
| 22:10.04 | tzanger | voip-info's got fixedDayEnable set... that shouldn't be right... start.month is 3, start.dayofweek=1 and satart.dayofweek.lastinmonth=0 should do it |
| 22:10.14 | rhombus | Qwell[] not just a week :\ |
| 22:10.33 | Qwell[] | that config file pasted earlier sure looked like a week :D |
| 22:10.45 | rhombus | it's two weeks |
| 22:10.49 | rhombus | on both ends |
| 22:11.21 | rhombus | wikiwikiwikiwiki |
| 22:11.31 | rhombus | Anybody remember Buck Rogers? |
| 22:11.35 | [hC] | Any of you guys used aastra phones much lately? I just met with sayson and they showed me a lot of the XML integration they're working on, and its got me really interested in their line up again |
| 22:11.47 | [hC] | Just curious of anyones complaints about their current state |
| 22:12.05 | Assid | WEIRD.. i cant get the welcome sound to play on my polycom anymore |
| 22:12.20 | Assid | and its enabled |
| 22:12.32 | xheliox | Sounds like a blessing in disguise. :p |
| 22:12.40 | rhombus | xheliox: lol |
| 22:12.43 | Strom_C | yes, i hate that sound |
| 22:12.56 | rhombus | Strom_C: the new one is much more pleasant |
| 22:13.17 | Strom_C | "Not only did you just wait five minutes for a reboot, but we're going to make you listen to this irritating sound!' |
| 22:13.22 | xheliox | What practical purpose does that thing serve, anyhow? |
| 22:13.26 | rhombus | ROFL |
| 22:13.39 | rhombus | xheliox: well, it is irritating. |
| 22:14.07 | rhombus | xheliox: Occasionally, you want that. Though I can't think of a specific example right now. |
| 22:14.11 | Assid | i want that sound! |
| 22:14.11 | Malph | I think it's so you can walk away from it while it takes forever to boot |
| 22:14.33 | mcab | Just buy a 650 instead, they boot real quick ;-) |
| 22:14.39 | Strom_C | do they? |
| 22:14.42 | rhombus | mcab: yep! Mighty impressed |
| 22:14.45 | rhombus | under 60 seconds |
| 22:15.18 | xheliox | Wow, that way you'll only miss 3 sales calls instead of 10... AWESOME! |
| 22:15.19 | rhombus | also, the welcome sound in the 2.1.0 release is much nicer |
| 22:15.21 | rhombus | xheliox: who cares? customers are annoying. |
| 22:15.31 | xheliox | I agree, but they like to send me checks. |
| 22:15.36 | xheliox | And that part isn't annoyinhg. |
| 22:15.37 | [TK]D-Fender | Yeah except why would a Polycom phone HAVE to reboot :) |
| 22:15.50 | mcab | tzanger: here's the Polycom DST tech-note - http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=10627&sliceId=SAL_PUBLIC_1_2&dialogID=2090583&stateId=1%200%202084980 |
| 22:15.51 | wunderkin | ... ! |
| 22:15.56 | rhombus | [TK]D-Fender: well, if it were running Windows, for example. |
| 22:15.57 | Assid | rhombus: it doesnt work for me :( |
| 22:16.18 | rhombus | Assid: when was the last time it worked? |
| 22:16.30 | tzanger | I just want normal ringtones, both for my polycoms and my treo |
| 22:16.33 | tzanger | mcab: danke |
| 22:16.37 | tzanger | rhombus: yes I remember buck rogers |
| 22:16.46 | tzanger | that robot with the dryer-vent arms freaked me out |
| 22:16.46 | rhombus | tzanger: wikiwikiwikiwikiwiki |
| 22:16.50 | Assid | dont know.. never really noticed |
| 22:16.52 | rhombus | tzanger: LOL |
| 22:16.59 | [TK]D-Fender | mcab: SMASING. |
| 22:17.07 | Assid | hrmm i wouldnt the thx sound |
| 22:17.09 | Assid | got the wav ? |
| 22:17.10 | [TK]D-Fender | mcab: and I use the Cisco "24" ring-tone :) |
| 22:17.17 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 22:17.38 | mcab | [TK]D-Fender: hah :-) |
| 22:17.41 | Assid | [TK]D-Fender: as the welcome ? |
| 22:18.16 | Strom_C | [TK]D-Fender: that's an old AT&T Merlin ring tone |
| 22:18.16 | rhombus | Assid: is it loading the welcome sound at boot time? how are you provisioning the phone? |
| 22:18.16 | xheliox | Nerds. All of you. |
| 22:18.16 | Assid | rhombus: ftp.. i see the logs in ftp of the file being loaded |
| 22:18.22 | mcab | Assid: don't really have anyplace to put it - pastebin take binaries? ;-) |
| 22:18.30 | mcab | xheliox: you're on IRC, QED |
| 22:18.33 | rhombus | xheliox: that's an occupational hazard. It's not like you're in #sexyblondes |
| 22:18.44 | rhombus | and even there... |
| 22:19.09 | rhombus | and you have welcome on warm boot enabled? |
| 22:19.20 | Assid | mcab: nah.. rapidshare? |
| 22:19.27 | Assid | rhombus: i think thats the diff in the new sip |
| 22:19.31 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
| 22:19.55 | rhombus | Assid: first piece of advice -- use the sip.cfg and phone1.cfg files unadulterated, as they come in the package |
| 22:20.05 | rhombus | Assid: and put all of your changes in the overrides file |
| 22:20.07 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:20.38 | tzanger | mcab: there. made the changes according to that techdoc. Thanks! |
| 22:20.39 | [hC] | yay, voip info is down again. |
| 22:20.41 | Assid | rhombus: was updating from the old one.. overlooked that change apparently |
| 22:20.55 | mcab | tzanger: np, hope that works |
| 22:21.07 | Assid | mcab: dcc? rapidshare? email ? |
| 22:21.13 | rhombus | Assid: making sure that it is set read-only, so that you can't bugger it up from the phone |
| 22:21.26 | markit | any mISDN guru? :) |
| 22:21.42 | rhombus | wunderkin: you are naughty, yes -- you also have a lot of free time on your hands, apparently |
| 22:21.50 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:22.22 | mcab | Assid: e-mail would work, /msg me |
| 22:23.00 | [[blah]asfd | i am getting thousands of errors... could anyone help me to stop this: http://pastebin.ca/395277 |
| 22:23.42 | k-man | how can i set the dial tone and other progress tones to sound australian in a sipura 941? (ie is there a list of the setting somewhere?) |
| 22:25.35 | ai-a | [[blah]asfd: your getting 1 error, repeated |
| 22:26.29 | [[blah]asfd | that is more accurate than what I stated... yes. How can I stop it? |
| 22:26.38 | *** join/#asterisk wunderkin- (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 22:29.10 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:31.52 | k-man | if you have a sip phone behind a firewall, do you have to do port forwarding to enable inbound dialing? |
| 22:32.38 | *** join/#asterisk AJaymn (i=AJaymn@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 22:32.42 | flenders | k-man: you have to allow traffic on 5060 and the rtp ports |
| 22:32.47 | flenders | to go through |
| 22:32.55 | k-man | flenders, ok |
| 22:32.58 | k-man | thanks |
| 22:33.04 | flenders | if you're nating, then you need to forward ports |
| 22:33.43 | flenders | if it's just a firewall between 2 network segments, and you're router knows how to get there, then just open up the ports |
| 22:34.04 | k-man | its nating |
| 22:34.35 | [TK]D-Fender | k-man: No. |
| 22:34.43 | flenders | I'm having a nightmare here to allow calls between users inside the office and users on the VPN |
| 22:34.46 | k-man | [TK]D-Fender, no? |
| 22:35.01 | k-man | [TK]D-Fender, how do you do it sans port forwading? |
| 22:35.03 | [TK]D-Fender | k-man: If you're talking a single phone behind a NAT, then NO you do not typically ahve to forward anything to it. |
| 22:35.15 | k-man | [TK]D-Fender, yes, that is what i'm talking about |
| 22:35.31 | k-man | [TK]D-Fender, DID is not working.. what should I check for then? |
| 22:35.42 | k-man | [TK]D-Fender, but outbound dialing is working |
| 22:35.45 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 22:35.48 | [TK]D-Fender | k-man: If you're talking about having it register to your * box, the * peer entry should jsut have "nat=yes", "canreinvite=no", and "qualify=yes" |
| 22:35.59 | *** join/#asterisk topping (n=topping@ppp-67-124-90-162.dsl.pltn13.pacbell.net) |
| 22:36.01 | flenders | the phones run on a different network segment, which doesn't have access to our VPN, so 'internal' calls are only sending audio one way |
| 22:36.15 | k-man | [TK]D-Fender, no, no asterisk involved.. just sip phone through NAT to sip provider |
| 22:36.25 | flenders | oh, k-man |
| 22:36.35 | flenders | don't worry about it then |
| 22:36.53 | flenders | as long as your phone has access to the outside world, you're set |
| 22:37.03 | [TK]D-Fender | k-man: Still should be no issue |
| 22:37.14 | k-man | flenders, i can make calls, but inbounc calls go straight to the providers voicemail system |
| 22:37.34 | flenders | k-man, I would say it's something on the settings on the phone |
| 22:37.44 | k-man | ok |
| 22:37.45 | k-man | checking |
| 22:37.56 | flenders | mynetphone? |
| 22:38.15 | k-man | mynetfone |
| 22:38.29 | k-man | why do providers insist on spelling spelling phone with an f? |
| 22:40.16 | Nugget | "fone" is like "drive thru" :) |
| 22:40.25 | k-man | and lite |
| 22:40.32 | k-man | stupid marketing spelling |
| 22:40.41 | tzanger | heh |
| 22:40.46 | tzanger | late nite, don't forget that |
| 22:40.56 | k-man | english spelling is hard enough without people inventing new stupid spellings of words |
| 22:41.07 | k-man | tzanger, yeah, good one |
| 22:44.37 | Mercestes | quik mart |
| 22:45.29 | flenders | k-man, did you follow their instructions on their website? |
| 22:45.41 | *** join/#asterisk wunderkin (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 22:45.44 | k-man | flenders, checking them as we speak |
| 22:45.56 | flenders | http://www.mynetfone.com.au/MNF202/ |
| 22:47.33 | *** join/#asterisk Skaag (n=skaag@80.178.76.62.adsl.012.net.il) |
| 22:47.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:48.10 | markit | how can I have pstn without a PCI board? I would like to build a very low power asterisk pc with sip phones and without PCI cards |
| 22:48.22 | Skaag | hey guys, my provider told me that my passwords are encrypted via md5, i changed my auth to plain but I still can not login to my provider, any ideas? |
| 22:48.23 | markit | something that can give me also callerid, possibly |
| 22:48.28 | JT | markit: using voip i guess |
| 22:48.29 | flenders | markit: SPA3000 |
| 22:48.33 | Skaag | (iax provider) |
| 22:48.55 | markit | JT using voip is a good idea, but I want to keep my current phone number |
| 22:48.57 | Skaag | my iax provider wants me to send the passwords in cleartext |
| 22:49.07 | JT | markit: use an ATA then |
| 22:49.08 | markit | flenders: can it be configured to "forward" calls? |
| 22:49.33 | flenders | markit: sort of |
| 22:49.36 | markit | JT evey ata is ok? even cheap ones? |
| 22:49.51 | flenders | markit: you need an ATA with FXO |
| 22:49.52 | JT | markit: sipura ones are usually ok for consumer use |
| 22:50.02 | flenders | markit: like the SPA3000 |
| 22:50.20 | JT | markit: analogue line i assume |
| 22:50.22 | markit | ok, thanks a lot |
| 22:50.31 | *** join/#asterisk X-Rob (n=rob-x@203.37.172.162) |
| 22:50.41 | Skaag | isn't there a law now that allows you to keep your phone number? |
| 22:50.46 | markit | JT yes, is for my home. My really small office has 1port bri instead, but I've a different pc for it |
| 22:50.56 | JT | markit: ok |
| 22:50.56 | Skaag | so like he could transfer the number to some voip provider? |
| 22:51.07 | *** part/#asterisk lionel (n=lionel@ip-61.net-82-216-103.rev.numericable.fr) |
| 22:51.15 | JT | Skaag: the merit of doing so is debatable though :P |
| 22:51.17 | markit | Skaag: in Italy I suppose they will hardly do |
| 22:51.28 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 22:51.42 | Skaag | ah, italy! |
| 22:51.42 | flenders | hmmm... pizza |
| 22:51.54 | Skaag | hehe |
| 22:52.03 | flenders | fuck, I've been on a stupid diet for a few weeks now |
| 22:52.12 | flenders | haven't had a pizza in a while |
| 22:52.15 | JT | how stupid is the diet? |
| 22:52.16 | Dovid | anyone know if voip-ifo is still down ? |
| 22:52.21 | Dovid | infi |
| 22:52.25 | flenders | JT: eating less |
| 22:52.26 | Dovid | info* |
| 22:52.30 | flenders | that's stupid enough |
| 22:52.35 | flenders | and gym everyday |
| 22:52.37 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-94.lsanca.dsl-w.verizon.net) |
| 22:52.39 | JT | flenders: eat less pizza? |
| 22:52.40 | flenders | which is fucking killing me |
| 22:52.52 | flenders | JT: lots of veggies and green stuff |
| 22:52.59 | JT | what about meat? |
| 22:53.04 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-174-162.ny325.east.verizon.net) |
| 22:53.06 | flenders | JT: meat is fine |
| 22:53.26 | flenders | at least my wife said so |
| 22:53.30 | JT | keep the protein levels up |
| 22:53.31 | JT | yeah |
| 22:53.38 | JT | otherwise your metabolism will go down |
| 22:53.41 | flenders | yeah, lots of fish |
| 22:53.48 | flenders | tuna specially |
| 22:53.51 | JT | carbs are the enemies to weight loss |
| 22:54.00 | JT | or steak, cheese, etc |
| 22:54.18 | flenders | yeah, I'm learning all that stuff the hard way |
| 22:55.31 | Dovid | anyone know if voip-info is still down ??? |
| 22:56.33 | JT | Dovid: instead of repeating, why don't you use a web browser? |
| 22:56.39 | JT | surely that'd be faster |
| 22:56.43 | Mercestes | JT: Omg! That's BRILLIANT. |
| 22:57.09 | flenders | haha |
| 22:57.29 | riddlebox | does anyone use the sonicwall vpn? |
| 22:57.34 | Dovid | hehe |
| 22:57.38 | Mercestes | Dovid: I'm guessing your not getting voip-info.org...and your starting to panic a little, and starting to doubt yourself, so your thinking to yourself "maybe it's just me.....maybe I'm doing something wrong. I'll just...ask these other guys." |
| 22:57.50 | [hC] | anyone know of a way to have asterisk beep to the transferee when the transferrer completes their attended transfer? (ie I transfer a call TO you, and when i hit transfer a second time to connect you to the call, you hear a beep) to know that the transfer has been completed? |
| 22:57.58 | Dovid | not working on my browser and i wante dto know if ev1 else had the issue as well as if anyone had any info what happend to the site |
| 22:58.04 | JT | Dovid: works fine here |
| 22:58.05 | JT | j/l |
| 22:58.07 | JT | j/k |
| 22:58.18 | DrukenLPY | Mar 14 17:55:37 NOTICE[22227]: chan_iax2.c:6907 socket_read: Rejected connect attempt from 67.69.139.242, who was trying to reach '6477463399@' |
| 22:58.24 | Dovid | Mercestes: thats what happens after a looooooooooooooooooooooooooooooooooooooooong day |
| 22:58.24 | DrukenLPY | WTF causes that god damn @ ? |
| 22:58.48 | flenders | JT: does it? my proxy is not getting there |
| 22:58.49 | Mercestes | DrukenLPY, Authentication failure |
| 22:58.56 | Mercestes | Dovid, lol. yea |
| 22:59.08 | flenders | Connection to 209.177.146.146 Failed |
| 22:59.40 | JT | flenders: missed the j/k bit? |
| 22:59.52 | flenders | oh, sorry mate |
| 22:59.58 | JT | still down |
| 23:00.09 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqut.cable.mindspring.com) |
| 23:00.18 | JT | if they were running a commercial web site or a phone system, their arse would have been fired |
| 23:00.21 | JT | 3 days downtime |
| 23:00.23 | JT | unheard of |
| 23:00.58 | Mercestes | Yea, as soon as they come back up I'm leaching those bitches. |
| 23:02.15 | flenders | not even my home adsl stays down for that long |
| 23:02.44 | JT | yeah |
| 23:03.07 | JT | my internal network uptime at home is close to 100% |
| 23:03.13 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:03.21 | DrukenLPY | JT: any idea what would cause an @ at the back of the DID on an incoming call ? |
| 23:03.22 | JT | the connection to the net is probably 99.9% up at least |
| 23:03.28 | russellb | JT: you're l33t |
| 23:03.35 | JT | DrukenLPY: provider sending it? |
| 23:03.40 | Qwell[] | DrukenLPY: <Mercestes> DrukenLPY, Authentication failure |
| 23:03.45 | DrukenLPY | nope... between my own boxen |
| 23:03.58 | JT | russellb: :P |
| 23:04.16 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:04.23 | DrukenLPY | Qwell: the box is recepient box is registered to the sending box |
| 23:04.39 | testdriver12500 | Has anybody run into an issue where an inbound IAX connection won't bridge to the s extension, but works with _N. |
| 23:04.47 | testdriver12500 | ? |
| 23:05.16 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 23:05.30 | Mercestes | testdriver12500, s is more for macros, afaik |
| 23:05.49 | Qwell[] | s == start |
| 23:05.51 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 23:05.57 | Qwell[] | if no extension is specified, it'll (usually) hit s |
| 23:06.08 | Dovid | JT: How long has voip-info been down ? |
| 23:06.15 | Dovid | i am willing to host it ;0 |
| 23:06.17 | Dovid | ;) |
| 23:06.26 | Mercestes | yea, but if your in an IAX with a context= recieving an incoming phone #, it takes the phone# as the extension I believe, which won't call "s" |
| 23:06.37 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 23:06.52 | Mercestes | again, afaik. I could be very wrong there. |
| 23:06.52 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:07.20 | testdriver12500 | Mercestes: Thanks. So is the _N. the way to go. There is a local and a 800 number on it and I want them both to follow the same context pattern |
| 23:08.00 | flenders | hey US guys, is HI = Hawaii? |
| 23:08.12 | testdriver12500 | HI = Hawaii |
| 23:08.37 | flenders | didn't know they had internet there |
| 23:08.39 | flenders | :D |
| 23:08.51 | testdriver12500 | with those beaches, they don't really need it. |
| 23:09.03 | flenders | that's what I thought |
| 23:09.21 | DrukenLPY | they do but it's flakey, lines are cut all the time from the glass... god damn dog making people break their pipes! |
| 23:09.39 | flenders | but voip-onfo.org is registered to a "James Thompson" in honolulu |
| 23:09.55 | flenders | they have internet AND voip! |
| 23:10.14 | testdriver12500 | Probably not hosted there..my guess |
| 23:10.36 | kink0 | good night |
| 23:10.49 | JT | over 120Gbit/s travels through Hawaii |
| 23:11.14 | JT | I'm not sure how many wavelengths they actually get for use in Hawaii itself |
| 23:11.27 | k-man | ah, DID is working now |
| 23:11.37 | kink0 | a simple question, if I use tdmoe, and setup several box with TE4xxx cards, that would not works ? right ? I mean because codec/decode is done just in one box, right ? ( I use g729 ) |
| 23:12.01 | flenders | k-man, was it the settings? |
| 23:12.21 | k-man | flenders, yeah, i think I hadn't enabled the NAT settings in the phone |
| 23:12.22 | JT | both legs of the southern cross cable network go to hawaii |
| 23:12.58 | k-man | any idea how i set the ringer on an SPA941 to something other than the ringing bell when receiving a call? |
| 23:13.01 | flenders | go to, or go through? |
| 23:13.41 | flenders | k-man, yeah, go to Ext 1, and set the "Default Ring" |
| 23:13.53 | flenders | 10 is the same as on 24. |
| 23:14.00 | tzafrir | kink0, how did g729 get involve here? |
| 23:14.10 | tzafrir | tdmoe is tdm |
| 23:14.32 | tzanger | I AM WE TODD IT |
| 23:14.35 | tzanger | I AM SOFA KING WE TODD IT |
| 23:14.39 | tzanger | dammitalltohell |
| 23:15.09 | *** join/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
| 23:15.35 | kink0 | tzafrir, yes, but if I code/decode g729 in the first machine that mount TDM over ethernet, all CPU load would burn this cpu, right ? |
| 23:16.07 | k-man | flenders, what do you mean by "10 is tha same as on 24"? |
| 23:16.08 | kink0 | tzafrir, I mean, that cpu load for coding would not able to add let's say 4 TE4xxx , one in each box |
| 23:16.20 | flenders | k-man: 24, jack bauer. |
| 23:16.25 | tzafrir | kink0, there is no strict rule here |
| 23:16.26 | *** join/#asterisk ta^3 (n=tacvbo@189.146.188.251) |
| 23:16.30 | flenders | k-man: never seen on channel 7? |
| 23:16.34 | tzafrir | but why do you need tdmoe? |
| 23:16.45 | tzafrir | as opposed to iax? |
| 23:16.47 | k-man | flenders, no, i don't watch commercial television |
| 23:16.56 | kink0 | tzafrir, large number of E1 ports |
| 23:16.59 | flenders | k-man: what do you watch? |
| 23:17.09 | flenders | k-man: no foxtel either? |
| 23:17.11 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 23:17.23 | k-man | abc and sbs almost exclusively |
| 23:17.30 | stuntdouble | Hello. I am using Asterisk 1.2.16 with ODBC. I get this error res_odbc: Error SQLConnect=-1 errorno =0 [unixODBC]Unable to connect to data source: library 'IscDbc' failed to load: IscDbc: cannot open sh. I can provide the config files separately. |
| 23:17.42 | flenders | k-man: I actually hate channels 7, 9 and 10 |
| 23:17.45 | k-man | flenders, and i use mythtv - so i virtually never watch live tv |
| 23:17.48 | flenders | but I have foxtel at home |
| 23:17.56 | flenders | k-man, I use mythtv too |
| 23:18.12 | k-man | flenders, cool |
| 23:18.18 | k-man | damn, did doesn't work any more |
| 23:18.19 | flenders | I downloaded the first 5 seasons of 24 |
| 23:18.21 | k-man | its weird |
| 23:18.28 | k-man | flenders, really? is it that good? |
| 23:18.38 | flenders | k-man: I got into it |
| 23:18.40 | k-man | flenders, are you watching bsg? |
| 23:18.52 | flenders | k-man: bsg? don't think so |
| 23:19.00 | kink0 | when calculations speaks about the half of channels with echo cancellation vs full channels capacity without echo c. , is refered to software echo cancelation ? |
| 23:19.00 | k-man | battle star galactica |
| 23:19.02 | JT | battlestar galactica |
| 23:19.07 | JT | much better than 24 |
| 23:19.10 | flenders | oh, no, haven't started yet |
| 23:19.12 | JT | i sort of got bored of 24 |
| 23:19.13 | k-man | flenders, its pretty good imho |
| 23:19.24 | kink0 | in other words, echo cancelation DSP also consumes a lot of CPU ? |
| 23:19.27 | orlock | i watched S3 E18 of BSG last night |
| 23:19.29 | JT | and the tv networks stuffed it up too much for me to continue watching the newer seasons |
| 23:19.31 | k-man | flenders, although the quality of episodes varies greatly |
| 23:19.35 | flenders | hmm, I'll download it then |
| 23:19.42 | flenders | I'm running out of content |
| 23:19.44 | flenders | :D |
| 23:19.54 | JT | kink0: you should buy a card with hardware ec |
| 23:20.07 | k-man | any other shows worth watching? |
| 23:20.11 | orlock | Heroes |
| 23:20.15 | orlock | Prison Break |
| 23:20.25 | flenders | yeah, guys here in the office are into heroes as well |
| 23:20.38 | orlock | heh |
| 23:20.42 | kink0 | JT yes, i am planning to do so, TE412P probably, but documents I saw does not says if calculations are with ec. dsp or software |
| 23:20.44 | JT | The West Wing |
| 23:20.45 | flenders | first season of prison break was pretty good |
| 23:20.48 | orlock | "he says you had a sword!" |
| 23:21.00 | kink0 | just exposes that with e.c. the number of channels goes to half |
| 23:21.16 | JT | kink0: what would the point of paying extra for hardware EC... if there was no hardware EC? |
| 23:21.51 | orlock | http://www.sso.ca/resources/rhsbl.html |
| 23:22.04 | orlock | yeah, great, give us a pile of useless information about your list |
| 23:22.14 | *** join/#asterisk blackmousepad (n=blackmou@71-13-69-254.static.aldl.mi.charter.com) |
| 23:22.15 | orlock | and dont give us a way to query it, or even find out what has been listed |
| 23:22.22 | kink0 | JT that is logic, but as info does not clarify if DSP or software... would be DSP also takes extra cycles from bus/cpu |
| 23:22.24 | designdream | so.. what trust mode is recommended.. COS or DSCP ? |
| 23:22.49 | stuntdouble | I have another question. What is recommended to connect multiple branch pbx servers running Asterisk? I came across the term "SIP Trunking". |
| 23:23.39 | blackmousepad | asterisk - nat(1) - internet - nat(2) - xlite_phone nat(1) is port forwarding 5060 and 10000-20000 extension is insecure=very nat=yes sip_nat.conf as nat=yes public host name, refresh interval, and internal network settings. I can place calls to sip extensions from xlite to the internal network where asterisk is but they can't call xlite (busy signal) audio works now (fixed earlier problem). any ideas? |
| 23:23.52 | JT | kink0: it's hardware EC, it's an octastic IC |
| 23:24.19 | orlock | grrr |
| 23:24.22 | orlock | blacklists. fuckem. |
| 23:25.12 | orlock | what a seriously useless POS site! |
| 23:25.30 | orlock | the link to more information brings up the same page, no way to query the service, etc |
| 23:25.51 | flenders | orlock, you've been blacklisted? |
| 23:25.55 | blackmousepad | orlock which site is that? |
| 23:26.07 | Mercestes | can anyone help me play with my PRIs? |
| 23:26.40 | JT | blackmousepad: isn't asterisk port forwarding? |
| 23:26.49 | kink0 | JT yes , I read octastic add best performace |
| 23:26.54 | JT | Mercestes: in what way? |
| 23:27.00 | kink0 | compared with older models from Digium |
| 23:27.11 | k-man | flenders, that ring 10 is aweful |
| 23:27.13 | Mercestes | Trying to add a span 3 adn it's not working...all 3 go down |
| 23:27.21 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:27.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:27.23 | blackmousepad | JT how do I find out? |
| 23:27.24 | Mercestes | everything shows ok but I can't call anything on any of hte PRis |
| 23:27.25 | flenders | k-man, they all are |
| 23:27.27 | kink0 | as this is for a large amount up to 64 E1's , I need to have in consideration all these details |
| 23:27.29 | Mercestes | tdm400p I believe. |
| 23:27.31 | k-man | flenders, true.. |
| 23:27.44 | JT | blackmousepad: i assume you'd remember whether you added port forwarding or not |
| 23:27.47 | k-man | flenders, ring 5 is bearbable |
| 23:27.55 | JT | tdm400p is analogue |
| 23:27.56 | JT | not pri |
| 23:28.10 | k-man | anyone here use optus cable? |
| 23:28.22 | Mercestes | the digium 4 port pri card then. |
| 23:28.30 | orlock | k-man: i think the guy behind me does |
| 23:28.31 | JT | they make a few |
| 23:28.55 | Mercestes | <PROTECTED> |
| 23:28.58 | Mercestes | lspci |
| 23:29.18 | flenders | TE410P is PRI, isn't it? |
| 23:29.32 | tzanger | flenders: can be, yes |
| 23:29.47 | k-man | orlock, can you ask him if all you need to do to use a router with optus is plug the router into the cable modem? |
| 23:29.51 | JT | Mercestes: so do you want to pastebin.ca zapata.conf and zaptel.conf, and tell us what it says when it fails? |
| 23:29.51 | blackmousepad | jt Port forwarding from the internet to the asterisk box behind nat? yes I added that. the sip extension behind that same firewall (on the same subnet as the asterisk box) can not call the softphone that's located behind a nat router out on the internet. |
| 23:30.13 | Mercestes | JT: Sure. |
| 23:30.27 | JT | blackmousepad: does the other softphone register to asterisk? |
| 23:30.46 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 23:31.17 | blackmousepad | yes, as far as I know. I can call from the xlite to the asterisk box and extensions on that network. but they can't call back to xlite. |
| 23:32.06 | JT | that has nothing to do with whether it registers or not |
| 23:32.09 | JT | you need to find out |
| 23:33.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:34.37 | orlock | k-man: it will all depend on the router.. i assume anything that does pppoe will be fine |
| 23:34.37 | *** join/#asterisk DJ-Pyro_ (n=abrezins@gateway.brevient.net) |
| 23:34.41 | Mercestes | http://pastebin.ca/395350 |
| 23:34.54 | Mercestes | Nice I see my problem. Zap status is RED on all three of my PRIs...but I don't know why... |
| 23:35.13 | k-man | orlock, ok, assuming the router supports optus, you don't need to put your optus login and pw in the router is what im asking |
| 23:35.39 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 23:36.01 | JT | you need a fixed MAC address for optus |
| 23:36.04 | JT | no other auth |
| 23:36.06 | JT | no ppoe |
| 23:36.10 | JT | uses dhcp |
| 23:36.14 | JT | unless it's changed |
| 23:36.22 | orlock | k-man: ok just asked |
| 23:36.25 | orlock | and its just DHCP |
| 23:36.25 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 23:36.26 | JT | oy maybe it was a fixed hostname |
| 23:36.34 | k-man | orlock, cool, thanks |
| 23:36.37 | lokkju_wrk | <PROTECTED> |
| 23:36.39 | orlock | not pppoe |
| 23:36.44 | lokkju_wrk | any suggestions? |
| 23:36.48 | blackmousepad | jt what's the asterisk cli command to show registed sip extensions? |
| 23:36.49 | lokkju_wrk | (running 1.2.16) |
| 23:38.38 | JT | Mercestes: yes red alarm bad |
| 23:38.47 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 23:38.47 | Mercestes | JT: A ny clues why? |
| 23:38.55 | BrianR___ | Is there a channel-independant jitter buffer in asterisk 1.2.x? |
| 23:39.37 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 23:41.30 | JT | Mercestes: not plugged in, wrong cable? |
| 23:41.43 | Mercestes | JT: all 3 plugged in. |
| 23:42.09 | JT | with what cables connected to what |
| 23:43.35 | Mercestes | JT: pri1 is to Logix router thorugh a T1 cable. PRI2 is to another T1card now, and Pri3 is to TWTC thorugh a t1 cable. |
| 23:43.58 | JT | "t1 cable" |
| 23:44.04 | JT | please be MUCH more specific |
| 23:44.22 | Mercestes | 1-2,4-5 straightthrough cables. No other wires pinned |
| 23:45.10 | blackmousepad | mercestes you mean an ethernet patch cord? |
| 23:45.25 | Mercestes | blackmousepad, With only 4 wires, yes. |
| 23:45.43 | *** join/#asterisk yakkop (n=yakkop@69.181.237.92) |
| 23:45.59 | Mercestes | Nice, now the PRI's aren't coming up at all under orig config |
| 23:46.04 | yakkop | hi...is asterisk able to do a wap push? |
| 23:46.05 | blackmousepad | ethernet only uses 1-2 4-5, T1 if I remember correctly uses 3-6 |
| 23:46.16 | JT | Mercestes: pin 1 to 2? |
| 23:46.38 | Mercestes | Yes |
| 23:46.46 | blackmousepad | (3-4 and 7-8 are for power over ethernet) |
| 23:47.49 | JT | so pin 1 on the first jack goes to pin 2 on the second? |
| 23:48.40 | Mercestes | JT: I'm pretty sure it's 1-1 2-2 4-4 5-5 |
| 23:48.47 | Mercestes | JT: But I can gladly check |
| 23:48.51 | JT | i'm pretty sure you're dead wrong there |
| 23:48.55 | JT | it's 1-4 2-5 |
| 23:49.15 | JT | and you use straight through cables for connections to the telco usually |
| 23:49.25 | Mercestes | It's a connection to a telco |
| 23:49.29 | Mercestes | through a pri interface. |
| 23:49.35 | JT | read: Standard ethernet cable |
| 23:49.40 | Mercestes | yoru pinout is what I call a "cross over." |
| 23:49.51 | JT | only the connection to ifax needs crossover cable |
| 23:49.54 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 23:49.58 | JT | Mercestes: well i did ask |
| 23:50.08 | JT | Mercestes: if pin 1 goes to pin 2 on the other? |
| 23:50.11 | JT | is this the case |
| 23:50.15 | JT | because if it is |
| 23:50.18 | JT | it's plain wrong |
| 23:50.25 | JT | not straight through, or crossover |
| 23:50.32 | blackmousepad | the connection to the telco is this cable going from the t1 interface card to the "smartjack"? |
| 23:50.36 | Mercestes | There is some serious problems here because my original working PRI isn't working now. |
| 23:50.40 | mmartinn | Does anyone know if there's a way to have the manager interface load after whatever adds "show version file..." ? |
| 23:50.43 | Mercestes | I've reloaded my orig configs and everything...let me check it out. |
| 23:51.14 | Mercestes | JT: It doesn't...it's straight through. But I've seen it go 1-4 2-5 as you said. It's usually kinda iffy depending on how they looop it up at their interface to begin with. |
| 23:51.14 | kink0 | good night guys !! |
| 23:51.26 | Mercestes | Let me go start yanking wires and get known good PRI 1 working. |
| 23:51.30 | Mercestes | and work up from there. |
| 23:51.49 | JT | Mercestes: well 1-2 is just wrong |
| 23:51.59 | JT | it should be 1-1 |
| 23:52.02 | JT | 2-2 |
| 23:52.03 | JT | 4-4 |
| 23:52.04 | JT | 5-5 |
| 23:52.12 | Mercestes | JT: Right...that's what I have, sorry. |
| 23:52.18 | JT | ok |
| 23:52.25 | JT | it's a normal ethernet cable |
| 23:52.31 | Mercestes | JT: The first "1-2 4-5" I had was merely referring to what pins I was using on one side.... |
| 23:52.37 | Mercestes | JT: which is...yes, quite useless information, I know. ;) |
| 23:52.39 | JT | ah ok |
| 23:52.52 | Mercestes | Let me go fix the wire I *know* worked before I touched anything and try to see what's going on here. |
| 23:53.17 | mihinomenest | since voip-info.org is down, how do I set the "from" username on a sip connection? |
| 23:53.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:53.33 | JT | fromuser? |
| 23:54.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:54.59 | blackmousepad | JT - on my issue. it looks like the xlite phone is registed (sip show peers lists the xlite extension and the public ip of the nat router it is behind) |
| 23:55.52 | JT | hmm |
| 23:56.06 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 23:58.11 | mihinomenest | oh that is just fucking beautiful. |
| 23:58.13 | mihinomenest | thanks jt. |
| 23:58.33 | JT | mihinomenest: does it work? |
| 23:58.45 | mihinomenest | yes. yes it does. |
| 23:58.52 | JT | sweet |
| 23:59.24 | tinrsh | cya |
| 23:59.36 | [hC] | nobody in here using aastra these days?? |
| 23:59.38 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 23:59.55 | mihinomenest | it was sending, "From: "Aphareus" <sip:3001@ipaddress>;tag=as34f7d005" |
