| 00:05.06 | ManxPower | C4away: an easy way is set up an alias in your local MTA |
| 00:14.52 | cvnet | if you got more than one phone number (did) from a provider, but want it to go to different context on extensions.conf how do you tell that? |
| 00:15.30 | _ShrikE | cvnet: goto |
| 00:16.17 | cvnet | _ShrikE no, it does go to that one context that i had for my first number, now i purchased another DID and i want it to go to different context in extensions.conf |
| 00:16.21 | ManxPower | cvnet: Usually you want all your calls from untrusted sources to land in the same context, then you can use Goto to route each DID to wherever you want to. |
| 00:16.37 | ManxPower | you would go to other contexts, of course. |
| 00:17.15 | ManxPower | [incoming] \n exten => 6665551212,1,Goto(customer-a,1212,1) |
| 00:18.13 | ManxPower | or even [incoming] \n exten => _66655512XX,1,Goto(customer-a,${EXTEN:6},1) |
| 00:18.18 | cvnet | hum i c now |
| 00:18.47 | cvnet | so where [incoming] \n exten => 6665551212,1,Goto(customer-a,1212,1) you create [customer-a] im in extensions.conf and tell it what to do correct? |
| 00:19.32 | ManxPower | correct. |
| 00:19.36 | cvnet | oo i c |
| 00:19.38 | cvnet | thanks a bunch |
| 00:19.56 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 00:20.09 | ManxPower | Contexts are both one of the hardest things to understand in Asterisk and as the same time one of the most important things to learn in Asteirsk |
| 00:20.59 | cvnet | Goto(customer-a,1212,1) <-- (Customer-A=NameOfContext,1212<-WHatDoesItRefferTo?,1=WhatDoesItmean?) |
| 00:21.27 | harry_v | Has anyone used the ramora a200 with up to 12 fxs cards before? Have a store that is wanting to upgrade and I dont feel like replacing there cat3 with cat5. |
| 00:21.56 | *** join/#asterisk dahunter3 (n=dahunter@pool-72-67-222-109.lsanca.fios.verizon.net) |
| 00:22.26 | dahunter3 | Any FAQ on best VOIP provider? SIP or IAX I don't care, I just care about quality |
| 00:22.29 | harry_v | this global finacial crunch is affecting just about everyone. |
| 00:22.41 | kornelak | cvnet: Goto(destination_context, destination_extension_in_context, destination_priority_in_extension) |
| 00:24.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:25.26 | cvnet | Can i use Goto(Destination_Context) the second, third paramter is it optionional or must ? |
| 00:27.26 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 00:31.59 | kornelak | cvnet: If you're going to a new context, yes, you need all 3 params |
| 00:35.08 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:42.18 | cvnet | ,Goto(customer-a,1212,1), im still confused on what the second parameter 1212 is ? |
| 00:45.05 | cvnet | if your number (did) is 416 811 1111 in extentions do you enter _4168111111 or _14168111111 ? |
| 00:49.45 | cvnet | exten => _5672579051,n,Goto(didx2,${EXTEN},1) <-- gives me busy signal |
| 00:49.48 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 00:49.54 | talntid | hi all. |
| 00:50.18 | talntid | i'm currently on a call. using a polycom phone... can i start recording the call somehow without restarting the call? |
| 00:53.07 | cvnet | exten => _5672579051,n,Goto(didx2,${EXTEN},1) <-- gives me busy signal (any help would be much appreciated) |
| 00:55.41 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-244-17-140.dsl.hrlntx.swbell.net) |
| 00:56.24 | *** join/#asterisk km2 (n=x@32.178.31.134) |
| 00:57.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7dcb346885000420) |
| 00:57.19 | Spirits-Sight | Can someone please tell me what I am doing wrong with my dailplain, all I am geting when I try and call 1-xxx-xxx-xxxx is a fast busy tone, here is the error and also the ext and sip files http://pastebin.com/mf2bc90f |
| 00:57.26 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 00:57.39 | ManxPower | do you have an exten => 5672579051,1,something in the [didx2] context? |
| 00:57.52 | ManxPower | also remember ALL extensions need a priority 1 |
| 00:58.11 | cvnet | ManxPower i got it fixed, i didnt know you could have to 1 in one context |
| 00:59.24 | cvnet | _5672579051,1,Goto(didx2,${EXTEN},1) \n _X,n,Goto(didx2,${EXTEN},1) <- didnt work, changed it to _5672579051,1,Goto(didx2,${EXTEN},1) \n _X,1,Goto(didx2,${EXTEN},1) <-- worked fine, note _X,1 <-- i didnt know you could have more tha one 1, i always use n, n didnt work changed it to 1 it worked |
| 00:59.26 | Spirits-Sight | ManxPower: I read alot of the book, I am still reading it, I have pick Voicepulse for my outgoing |
| 01:00.50 | talntid | nobody knows? :) |
| 01:01.08 | jaytee | hehe, this reminds me of the first month of my foray into Asterisk a couple years back and ManxPower saving my bacon because I'd rushed through most of the book :-) |
| 01:01.53 | jaytee | but after he and [TK]D-Fender beat me over the head with a cluebat I finally started getting it. |
| 01:02.06 | Spirits-Sight | LOL |
| 01:03.07 | harry_v | voicepulse seems to have a good reputation |
| 01:03.16 | Spirits-Sight | I hope my brain start to register it sooner then later, I understand a little through but just eithe to get me in touble I think |
| 01:03.29 | Spirits-Sight | I hope so :-) |
| 01:03.30 | ManxPower | start by reading extensions.conf.sample |
| 01:03.56 | jaytee | and then at least Chapters 3,4,5 and 6 of the book |
| 01:04.00 | Spirits-Sight | if you don't mind telling me, where is this? |
| 01:04.19 | jaytee | in /etc/asterisk if you compiled and did a make samples |
| 01:04.24 | ManxPower | /path/to/src/asterisk/configs |
| 01:04.33 | jaytee | there too |
| 01:05.04 | ManxPower | and /path/to/src/asterisk/doc is the official documentation |
| 01:05.34 | jaytee | Spirits-Sight, did you install by compiling or using packages from a repo? |
| 01:06.23 | *** join/#asterisk propellerhead (n=yogurt2u@209-38-17-190.fibertel.com.ar) |
| 01:06.23 | Spirits-Sight | I used Ubuntu package thing, I am going to be having it installed on CentOS, but I wanted to play with it now and so installed using the package thing |
| 01:07.11 | ManxPower | package docs are frequently in /usr/share/doc |
| 01:07.14 | jaytee | then go to www.asterisk.org and download the tarball of whichever version you're using and just extract the configs folder in it to your desktop as a reference |
| 01:08.21 | jaytee | last time I installed from a packaged version of * from the Ubuntu repos it didn't have the sample configs but that was back with 6.06 Dapper |
| 01:09.40 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 01:13.38 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 01:13.48 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:13.53 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 01:16.22 | *** join/#asterisk etfonhomey_ (n=chatzill@mobile-166-214-013-132.mycingular.net) |
| 01:18.26 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:19.17 | jaytee | ManxPower, I found out I passed the written part of the dCAP so I have a year to retake the lab portion if I choose. |
| 01:19.39 | *** join/#asterisk riddlebox (n=james@75.132.225.75) |
| 01:20.38 | harry_v | which is the better of the two echo canclers in system.conf mg2 or kb1 in most cases? |
| 01:20.54 | jaytee | mg2 usually |
| 01:20.58 | harry_v | k |
| 01:22.15 | harry_v | msg of echo cancller not on in cli after uncommenting the pound sign and reloading ast. |
| 01:23.00 | jaytee | well, first of all, it's spelled echocanceller, not canclers |
| 01:23.00 | harry_v | jatee, ever have experaince with the sangoma remora a200 series of cards? |
| 01:23.40 | jaytee | nope, never used sangoma products. hear they're pretty good but I'd rather put my money in a company that actually funds Asterisk open-source, not a leech |
| 01:24.58 | cvnet | hum, if you got a did that supports only ulaw (incoming), so calls comes in, and the terminiation provider only supports g729 (outgoing) would this work? would asterisk change the codec ? |
| 01:25.15 | jaytee | harry_v, how many ports on your sangoma card? |
| 01:25.50 | harry_v | I did a quick site survey and he has 12 analog phones. |
| 01:26.15 | harry_v | he is wanting to eventually have it replaced and wanted a price on the new system. |
| 01:26.32 | jblack | You can take advantage of the analog phones by getting an ata. |
| 01:26.37 | harry_v | and I want to know if you have had good experaince with the cards. |
| 01:26.44 | jblack | an 8 port ATA will run you.. about $250 iirc. |
| 01:27.05 | *** join/#asterisk andresmujica (n=andresmu@190.27.96.125) |
| 01:27.12 | harry_v | less then the the sangoma cars? who makes a 8 port ata? |
| 01:28.28 | ratmandu | digium? |
| 01:28.34 | jaytee | harry_v, my question about how many ports was for your error loading the echo canceller |
| 01:28.57 | harry_v | i think my echo cancler problems are resolved now. its not comming up in cli |
| 01:29.05 | jaytee | which needs a restart of dahdi |
| 01:29.11 | harry_v | right |
| 01:30.01 | harry_v | other party heard there side echo non on my side. |
| 01:30.04 | orkid | does anyone here use asterisk/asterisk14 on openwrt? |
| 01:30.13 | jaytee | say you had 2 fxo ports defined as channels 1 and 2 then the line would be the last line in /etc/dahdi/system.conf and would be "echocanceller=mg2,1-2" without the quotes |
| 01:30.30 | jaytee | gotta run an errand, bbiab |
| 01:30.35 | harry_v | jblack, who makes this 8 port ata and have you tested them in production? |
| 01:30.50 | harry_v | jaytee, thats what i used. |
| 01:32.01 | jblack | I use a linksys SPA-8K here at home. |
| 01:32.24 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:33.08 | harry_v | I never knew |
| 01:33.24 | *** part/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:34.31 | harry_v | only if it had four more ports. |
| 01:35.06 | orkid | wow lots of ports |
| 01:35.41 | cvnet | hum, if you got a did that supports only ulaw (incoming), so calls comes in, and the terminiation provider only supports g729 (outgoing) would this work? would asterisk change the codec ? |
| 01:35.50 | harry_v | that would be good for quick cheap setup for a small biz and then host the sip connections. |
| 01:38.56 | jblack | well, no erason you couldn't use two. |
| 01:39.27 | harry_v | depends on the cost and that of a remora a200 |
| 01:39.44 | jblack | anyways, those private messages you're rudely sending me, unasked.... It seems reliable as long as I stay away from the web configuration. |
| 01:40.39 | jblack | If I fiddle in the built in webserver very much, it seems unreliable. So I'm in the habit of flipping power, making changes in the web config, flipping power again. |
| 01:41.06 | jblack | Other than that, it's been solid for me. I've gone up to 3 concurrent calls. |
| 01:41.17 | jblack | And seen no indication it would have problems with 8 |
| 01:43.40 | Spirits-Sight | ManxPower: ok, I just looked at the sample and I don't see much that I didn't do, now I may of missed some thing but I would say I did not have three lines that look like I should of, here is my ext file again, I don't see what I am doing wrong, I am also geting the error that is also in the pastebin http://pastebin.com/d19f8edc8 my goal right now is just to be able to make calls right now with the most basic setup for maki |
| 01:44.31 | Spirits-Sight | don't forget that I am legally blind and my of missed it all-together |
| 01:44.45 | Spirits-Sight | my = may |
| 01:53.30 | unpaidbill | anyone here familiar enough with dahdi to lend advice here, http://pastebin.ca/1260904 ... for some reason i cant get the dahdi stuff to come up in asterisk, even though it appears that everything is working |
| 01:54.17 | unpaidbill | yeah it's asterisknow.. but maybe there is some dumb dahdi thing i am overlooking, it was all working yesterday |
| 01:54.55 | cvnet | if you got a did that supports only ulaw (incoming), so calls comes in, and the terminiation provider only supports g729 (outgoing) would this work? would asterisk change the codec ? |
| 02:00.54 | Spirits-Sight | Anyone using Voicepulse that can help me, I am trying to get it to work for outgoing calls as basic setup as can be, I want to build up on that and learn the stuff, if I have to much at once I don't learn as well and trying to break down their sample files that are on their website is not that easy as it uses macro and I don't know which ones I can get rid of to just do a basic setup |
| 02:04.31 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 02:05.07 | unpaidbill | oh |
| 02:05.08 | unpaidbill | i see |
| 02:05.24 | unpaidbill | dahdi doesnt throw an error when it fails to load |
| 02:05.32 | unpaidbill | even with asterisk -vvvvvvvvvvvvvvvvvvvvvgc |
| 02:05.40 | unpaidbill | arrr |
| 02:05.40 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 02:06.03 | Akiyuki | Anyone have experience with the Nortel 6812 or 6830? Or Vodavi 6812/6830? |
| 02:07.22 | riddlebox | Akiyuki, are those phones? |
| 02:07.45 | Akiyuki | Yes. I am having a hard time getting them working on my asterisk box. I know they work though, over mgcp because we use them at work w/ bandwidth.com |
| 02:18.29 | carrar | Akiyuki: http://www142.nortelnetworks.com/techdocs/IP6830O/pdf/LGN68126830-1224IG_01.05.pdf |
| 02:18.32 | carrar | read that yet? |
| 02:18.54 | carrar | konnichiwa |
| 02:19.07 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 02:19.25 | carrar | page 12 is what you need to set for dhcp so the phone knows where to go |
| 02:20.10 | carrar | those look like SIP phones (the 6812) not MGCP |
| 02:21.30 | Akiyuki | They are MGCP |
| 02:21.37 | carrar | that PDF says SIP? |
| 02:21.40 | Akiyuki | Yeah |
| 02:21.55 | carrar | maybe you can get sip firmware, or you need them to be mgcp? |
| 02:22.03 | Akiyuki | There are two models, one is a SIP the other is MGCP unfortunately. |
| 02:22.09 | carrar | ug |
| 02:22.25 | carrar | I'm sure they work the same probably |
| 02:22.29 | Akiyuki | I would prefer SIP. I tried TFTPing them a new firmware but the rejected it. |
| 02:22.51 | Akiyuki | Can you take a look at my sip.conf? I am getting time outs trying to connect to an external sip service. |
| 02:23.01 | carrar | ~pastebin |
| 02:23.02 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:23.03 | carrar | sure |
| 02:23.42 | Akiyuki | http://pastebin.ca/1261050 |
| 02:23.57 | Akiyuki | I am behind a router and have 5061 forwarded inward. |
| 02:25.32 | jaytee | Akiyuki, by default Asterisk listens on port 5060, not 5061 |
| 02:26.03 | Akiyuki | I changed that in sip.conf also |
| 02:26.31 | jaytee | did you do a module reload chan_sip.so? |
| 02:26.51 | Akiyuki | Yes |
| 02:27.00 | Akiyuki | Getting a time out error, as well |
| 02:31.31 | *** join/#asterisk ectospasm (n=ectospas@user-24-236-95-118.knology.net) |
| 02:31.41 | Akiyuki | When I do sip show registry, I get, sip1.sipdiscount.com:5060 jimisanchez 120 Request Sent |
| 02:35.08 | ectospasm | How do you specify a module parameter in DAHDI? I've tried adding wct4xxp_ARGS to /etc/dahdi/init.conf, but that doesn't seem to work. |
| 02:37.31 | cvnet | Can somoene please answer this question --> if you got a did that supports only ulaw (incoming), so calls comes in, and the terminiation provider only supports g729 (outgoing) would this work? would asterisk change the codec ? |
| 02:37.48 | carrar | MGCP is port 2727 |
| 02:37.59 | carrar | or you trying to make sip woro? |
| 02:38.00 | carrar | work |
| 02:38.01 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:38.04 | beek | cvnet: yes, Asterisk will do the transcoding. |
| 02:38.09 | Akiyuki | No, this is a seperate issue carrar |
| 02:38.19 | carrar | oh, sorry, was on the phone |
| 02:38.22 | cvnet | beek thanks alot |
| 02:38.24 | Akiyuki | :D its ok |
| 02:38.56 | carrar | try port 5060? |
| 02:39.19 | Akiyuki | Its in use by another device, thats why I used 5061 and forwarded that in |
| 02:40.12 | carrar | you get connected and no audio? |
| 02:40.17 | carrar | or not even connected |
| 02:40.42 | Akiyuki | not even connected |
| 02:40.51 | Akiyuki | Keep getting a message about it timing out |
| 02:41.00 | Akiyuki | http://pastebin.ca/1261050 |
| 02:41.02 | carrar | do you get registered? |
| 02:41.22 | Akiyuki | It just says "sip1.sipdiscount.com:5060 jimisanchez 120 request sent" |
| 02:42.03 | carrar | Why does it say 5060 in line 1? |
| 02:42.19 | carrar | oh thats the destination |
| 02:42.20 | carrar | nm |
| 02:42.31 | carrar | was just looking at the 19* |
| 02:42.35 | Akiyuki | ah ok |
| 02:43.09 | jaytee | Akiyuki, is there more to that sip.conf file? because you're missing a users section with peer and users defined |
| 02:43.21 | Akiyuki | i dont have any yet |
| 02:43.31 | jaytee | and you should never pastebin a sip.conf or other config without masking passwords |
| 02:44.03 | carrar | You try it not behind your router? |
| 02:44.18 | Akiyuki | Yeah, I can do it all day long from my home computer. |
| 02:44.22 | jaytee | is the phone MGCP? |
| 02:44.26 | *** join/#asterisk dmoldovan (n=tokey@titaniumsoft.net) |
| 02:44.46 | Akiyuki | Yes |
| 02:45.11 | Akiyuki | I am not going to be connecting a phone to this machine, just generating calls using .call files |
| 02:45.49 | jaytee | so you're using a server with a SIP provider account to send calls generated with call files to outbound numbers? |
| 02:45.51 | Akiyuki | the registration issue and the mgcp issue were/are on 2 differnet machines |
| 02:45.57 | jaytee | ok |
| 02:46.00 | Akiyuki | jaytee: yes |
| 02:46.06 | Akiyuki | sorry, i didnt realize that i hadnt mentioned that |
| 02:46.17 | jaytee | you still need to define your sip peer. A register statement alone won't cut it. |
| 02:46.51 | *** join/#asterisk mitcheloc (n=mitchel@adsl-163-36-50.hsv.bellsouth.net) |
| 02:46.56 | Akiyuki | ok, i will do that |
| 02:46.58 | jaytee | ~book |
| 02:46.59 | jbot | methinks book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:47.02 | Akiyuki | Is that what is keeping it from registering? |
| 02:47.11 | Akiyuki | I ordered the Oriely book from BNB.. its on its way |
| 02:47.19 | jaytee | Akiyuki, read pages 97-101 |
| 02:47.23 | dmoldovan | hi everybody, i was wandering if you could help with a few answers |
| 02:47.32 | jaytee | you can dowload the PDF from the link above |
| 02:48.04 | jaytee | Akiyuki, the PDF is free. |
| 02:48.10 | Akiyuki | oh |
| 02:48.14 | Akiyuki | I bought it for $45 |
| 02:48.23 | jaytee | pages 97-101 for SIP provider setup |
| 02:48.29 | Akiyuki | Thanks |
| 02:48.36 | jaytee | I just bought my third copy hardprint tonight |
| 02:48.48 | jaytee | I have one signed by Jared Smith |
| 02:48.50 | Akiyuki | waiting for it to open on this 486 :> |
| 02:48.55 | dmoldovan | does Asterisk use sendmail to send emails? and if it does, can it be configured as a service? |
| 02:49.01 | Akiyuki | Jared Smith from the subway commercials? |
| 02:49.09 | mitcheloc | lol |
| 02:49.17 | mitcheloc | no the guy from sokol & associates |
| 02:49.19 | jaytee | dmoldovan, yes it uses sendmail by default but that can be changed |
| 02:50.01 | jaytee | Jared from the Subway commercial's last name is Vogel or something like that. He lives here in Indiana. |
| 02:50.02 | dmoldovan | is there a way to configure it to avoid emails being lost if the server is down |
| 02:50.10 | jaytee | <PROTECTED> |
| 02:50.31 | Spirits-Sight | what does "exten => _1NXXNXXXXXX,n,NoOp(SIPCALLID: ${SIPCALLID})" do? |
| 02:50.32 | jaytee | He taught my Advanced Asterisk class last week at Digium in Huntsville. Excellent instructor |
| 02:51.14 | carrar | displays the variable of SIPCALLID to the console output |
| 02:51.26 | carrar | contents of ${SIPCALLID} |
| 02:51.31 | jaytee | Spirits-Sight, it just outputs to the console SIPCALLID: "whatever the value you assigned to the variable ${SIPCALLID} |
| 02:52.10 | carrar | Assuming you've set that someplace |
| 02:52.21 | Spirits-Sight | So when I make a phone call it should show this in the console |
| 02:52.35 | carrar | in the asterisk -r console |
| 02:52.42 | carrar | yeah |
| 02:53.26 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 02:54.08 | Spirits-Sight | ok then I was think correct, well it does not show it and also I don't see any where in the file that Voicepulse provides a var set to ${SIPCALLID} |
| 02:55.16 | Spirits-Sight | now I see a set(CALLERID(num)=xxxxxxxxxx) and one for name so could this just been a mistake in the file |
| 02:55.48 | jaytee | Spirits-Sight, check the channelvariables.txt file in your asterisk source directory. It lists all the system variables, ones that can be changed by you in the dialplan and "reserved" ones that can only be set or changed by Asterisk. |
| 02:56.17 | Akiyuki | jaytee: no luck, still getting the time out errors after adding peers and extensions |
| 02:57.12 | jaytee | Akiyuki, you're behind a router with nat, correct? |
| 02:57.35 | Akiyuki | yes |
| 02:57.54 | jaytee | ~sipnat |
| 02:57.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:58.04 | Akiyuki | :( |
| 02:58.06 | *** join/#asterisk rdgr (n=rich@82-33-200-190.cable.ubr01.aztw.blueyonder.co.uk) |
| 02:58.07 | Akiyuki | says unreachable now |
| 02:58.13 | jaytee | Akiyuki, better check that out ^^^^^^ |
| 02:58.34 | dmoldovan | is there a way to setup sendmail to queue the emails? |
| 02:58.48 | jaytee | dunno, never tried |
| 02:59.04 | jaytee | dmoldovan, google is your friend. sendmail is an ugly beast |
| 02:59.07 | Akiyuki | i read that earlier today. i defined nat=yes, and externip/localnet , etc |
| 02:59.38 | dmoldovan | jaytee, i thought i could find some help taming it in here... |
| 03:00.11 | jaytee | no, this is the #asterisk channel. try #sendmail |
| 03:00.58 | jaytee | although there might be a sendmail guru lurking who'd answer your question if you wait a few minutes. ya never know! |
| 03:01.37 | Akiyuki | jaytee: check this out, http://pastebin.ca/1261073 |
| 03:01.45 | dmoldovan | jaytee, thanks for advice, i'll try #sendmail too |
| 03:02.07 | *** part/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 03:02.08 | jameswf | look a ninja http://www.muslima.com/member_profile.cfm?ID=1084985&searchposition=200&searchtotal=1000 |
| 03:02.19 | jameswf | ninja dating omfg |
| 03:02.44 | neurosys | She's hot. |
| 03:03.19 | neurosys | Atleast I think it's a she. |
| 03:04.56 | dmoldovan | http://www.asterisk.org/ |
| 03:05.28 | jaytee | dmoldovan, good site! I've been there :-) |
| 03:05.51 | dmoldovan | sorry, wrong window, and yes, good site |
| 03:06.29 | jaytee | Akiyuki, you don't need the nat=yes for your [jimi] account |
| 03:06.37 | Akiyuki | i was desparate |
| 03:06.38 | Akiyuki | :D |
| 03:07.04 | Spirits-Sight | anyone know of a DID thats free to setup for incoming calls for testing reasons |
| 03:07.31 | jaytee | and how about pastbining the extensions.conf file so I can see what's in your default context? maybe a pastebin of the the CLI with "set sip debug on" would help too |
| 03:07.48 | neurosys | Spirits-Sight: les.netm 4 bucks a month. |
| 03:08.32 | Spirits-Sight | is it easy to config for incoming calls |
| 03:09.01 | neurosys | Thats not a DID. But yes. |
| 03:09.06 | neurosys | Sorry |
| 03:09.10 | neurosys | Yes it is |
| 03:09.40 | neurosys | It will even give you the asterisk config sample in accordance to your setup |
| 03:09.52 | Spirits-Sight | I don't want to setup a company yet for my incoming calls as I am going to port a number to them onces I can test and make sure things are the way they should be |
| 03:09.59 | Spirits-Sight | whats their website |
| 03:10.03 | neurosys | les.net |
| 03:10.13 | Spirits-Sight | thanks |
| 03:10.33 | neurosys | Spirits-Sight: np. That's who ive been using to learn on. |
| 03:11.19 | Spirits-Sight | see I can change to better company once I have things the way I want |
| 03:18.01 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 03:18.49 | Ritzerisk | am i able to see more of a Debug in the Cli i turned verbose on to 7 and debug on |
| 03:19.03 | Ritzerisk | i was trying to see more output on the call |
| 03:41.45 | *** join/#asterisk chendy (n=chatzill@121.34.152.233) |
| 03:42.38 | *** join/#asterisk enyawix (n=enyawix@68-114-138-145.dhcp.jcsn.tn.charter.com) |
| 04:00.32 | *** join/#asterisk ShaneAu (n=shane@203.56.250.52) |
| 04:01.31 | ShaneAu | Hi all... I have a queue setup with skip busy agents set to yes and a ring strategy of "rrmemory" |
| 04:02.01 | ShaneAu | I need to some members of the queue to be phoned before others |
| 04:02.25 | ShaneAu | I've played around with penalties but it doesn't seem to do what I want it to. |
| 04:02.34 | *** join/#asterisk ta^3 (n=tacvbo@190.154.36.86) |
| 04:02.37 | ShaneAu | For exmaple |
| 04:03.37 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 04:03.58 | ShaneAu | If I'm a member of the queue and have a penalty of 1 and another agent is a member of the queue with a penalty of 2, I take a call and I'm on the phone and another call comes through, it ignores the fact that I'm busy and starts calling on my phone's second line. |
| 04:04.21 | ShaneAu | I wish it would cascade over to the penalty 2 member. |
| 04:04.27 | ShaneAu | Is this possible? |
| 04:09.29 | *** join/#asterisk Ridgeback (n=jircii@65.120.140.163) |
| 04:09.53 | Ridgeback | anyone mess around with TDMoE? |
| 04:11.10 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
| 04:11.59 | *** part/#asterisk dmoldovan (n=tokey@titaniumsoft.net) |
| 04:12.11 | Ridgeback | anyone mess around with TDMoE? |
| 04:19.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:19.55 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 04:19.55 | joako | Yea |
| 04:20.18 | joako | Never worked right for me, seemed like the code wasn't maintained much after Asterisk 1.0 |
| 04:22.06 | Ridgeback | oh really? |
| 04:22.12 | Ridgeback | man I've had plenty of problems with it |
| 04:22.39 | Ridgeback | dahdi just not building channels. I can get the span up with no alamrs...it's just the dahdi side... |
| 04:23.34 | *** join/#asterisk ta^3 (n=tacvbo@190.154.36.86) |
| 04:30.32 | joako | I tried in the early 1.4.x days... looked at the source code and that sort of expalined it and I never tried again |
| 04:30.47 | joako | Any particular reason you wanted to use TDMoE? |
| 04:31.02 | drmessano | Why use TDMoE? |
| 04:31.29 | Ridgeback | oh i just wanted to play with TDMoE, thought it would be fun. but it's been painful! |
| 04:31.33 | [TK]D-Fender | Akiyuki: You should have your [sipdiscount] as "nat=no", and [jimi] as qualify=yes |
| 04:31.34 | *** join/#asterisk CrazyTux (n=brandon@user-vcauig4.dsl.mindspring.com) |
| 04:31.34 | drmessano | heh |
| 04:31.48 | [TK]D-Fender | Akiyuki: Also what have you forwarded to *? |
| 04:32.11 | Ridgeback | when i try to dial through one of my TDMoE channels i get: Unable to start channel: No data available |
| 04:32.51 | *** join/#asterisk workdraft (n=acxide@203.215.94.239) |
| 04:32.54 | [TK]D-Fender | Who really cares about TDMoE? Whats the point? |
| 04:32.56 | workdraft | yo |
| 04:33.30 | Ridgeback | TDMoE for me is for traininig on inexpsinve TDM stuff, without buying TDM cards. |
| 04:33.41 | workdraft | any idea how scalable asterisk is? does it scale up to 100 sip devices? |
| 04:33.51 | Ridgeback | workdraft: defintly! |
| 04:34.20 | [TK]D-Fender | Ridgeback: Umm.... "training" as in what exactly? |
| 04:34.27 | workdraft | ive read an ebook that says that it doesnt scale up to 100 SIP devices. |
| 04:34.32 | [TK]D-Fender | Ridgeback: it may as well be voip.... only less supported |
| 04:35.08 | [TK]D-Fender | workdraft: What else do your Rice Crispies say to you? |
| 04:35.27 | [TK]D-Fender | workdraft: Because I've seen 1000+ size deployments... |
| 04:35.38 | Ridgeback | [TK]D-Fender: for example, i may have to build TDM type channels across asterisk at work...but to learn about that stuff, I can practice at home on TDMoE and learn about the ins and outs. |
| 04:35.40 | workdraft | Asterisk, however, cannot act as a SIP Proxy. SIP devices can register with Asterisk, but |
| 04:35.41 | workdraft | as the number of SIP devices increases, Asterisk is not able to scale very well. Therefore, |
| 04:35.41 | workdraft | if we intend to use over about 100 SIP devices, Asterisk may not be appropriate. |
| 04:35.48 | workdraft | sorry for flood |
| 04:36.17 | [TK]D-Fender | Ridgeback: However TDMoE teaches you nothing of any practical application to any other tech. |
| 04:36.28 | Ridgeback | workdraft: there is avery good document on voip-info.org showing asterisk doing 400 calls of transcoding g729 to g711. |
| 04:36.42 | [TK]D-Fender | Ridgeback: its just "something to do", and if your system ends up depending on it just adds one more dependency |
| 04:36.48 | workdraft | great. thanks. ill follow that up; |
| 04:37.05 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 04:37.05 | file | it's the number of simultaneous channels and what they are doing that hurt |
| 04:37.08 | [TK]D-Fender | workdraft: Got a link to that doc? |
| 04:37.28 | file | inbound SIP registrations aren't that bad because the number of them happening at any given time are relatively low |
| 04:38.04 | workdraft | Building Telephony Systems with Asterisk by Packt Publishing |
| 04:38.22 | workdraft | Authors are David Gomillion and Barrie Dempster |
| 04:38.32 | Ridgeback | [TK]D-Fender: it's not to teach me other technoliges. But depending on customersm they still have T1's, with issues of SS7 and signalling. by learning about how asterisk uses these I can get the job done. |
| 04:38.47 | file | yeah that statement isn't true, even an unoptimized untweaked system can do 100 fine |
| 04:39.33 | *** join/#asterisk dynaguy (n=gao@d154-20-21-173.bchsia.telus.net) |
| 04:39.41 | [TK]D-Fender | workdraft: Whats the DATE on it?ridgTDMoE is TDMoE. The skills are not transportable, the protocols nothing alike. |
| 04:39.51 | [TK]D-Fender | wow... nice splt |
| 04:40.03 | [TK]D-Fender | workdraft: the DATE is...? |
| 04:40.09 | workdraft | havent checked it. wait a sec. |
| 04:40.18 | [TK]D-Fender | Ridgeback: TDMoE is TDMoE. The skills are not transportable, the protocols nothing alike. |
| 04:40.23 | Ridgeback | [TK]D-Fender: uh TDMoE does E&M, B8ZS, AMi etc.... you have to set all that stuff up.. so yes it is transportable. |
| 04:40.57 | workdraft | oopps. didnt read the date. it's dated 2005. |
| 04:40.59 | [TK]D-Fender | Ridgeback: Bleh... onlyt hing ont he other side is still *... not some oddball hardware |
| 04:41.04 | tzanger | Ridgeback: TDMoE has no concept of B8ZS or AMI, those are line codings and have no parallel when ethernet is the physical media. |
| 04:41.07 | [TK]D-Fender | workdraft: ANCIENT CRAP |
| 04:41.07 | workdraft | perhaps that time, it was true. |
| 04:41.17 | workdraft | i need a better ebook |
| 04:41.17 | Ridgeback | [TK]D-Fender: if you read the tDMoE docs they take tradiional TDM and just chop it into frames and shoot it across the LAN. Still normal TDM. |
| 04:41.31 | Ridgeback | tzanger: right I know that |
| 04:41.39 | file | tzanger knows of the TDMoE |
| 04:41.44 | jaytee | workdraft, anyone who can't scale Asterisk to over 100 sip devices either has a really crappy 10MB network of cascaded hubs instead of switches and a server with a 200mhz or slower CPU and 256MB of ram probably |
| 04:42.13 | Ridgeback | really all I want to know is if TDMoE is working in 1.6.x |
| 04:42.32 | file | jaytee: so practically speaking... your practical was ungood? |
| 04:42.54 | jaytee | file, it was 90 minutes. what can I say? I'm not a speed demon |
| 04:43.01 | jaytee | if I'd had two hours I would have passed |
| 04:43.13 | jaytee | I passed the written |
| 04:43.18 | file | I've sat in on a dCAPitation... it was interesting |
| 04:43.39 | Ridgeback | anyone know why dahdi says this: dahdi_call: Unable to start channel: No data available |
| 04:43.45 | jaytee | I think there were 7 or 8 of us taking it and I think only 2 or 3 passed the practical |
| 04:44.02 | file | jaytee: yeah. |
| 04:44.24 | drmessano | Ridgeback: I would suggest filing a bug report.. but I can tell you the outcome |
| 04:44.44 | jaytee | file, I've got up to a year to drill my ass off in speed configuration and I hope to be back in Huntsville within 6 months to take the practical again and pass it. |
| 04:45.06 | file | Ridgeback: I can tell you why that shows up, but no clue of the underlying reason from in dahdi |
| 04:45.14 | drmessano | "Like an appendix, this code is sitting there useless waiting to be removed through infectious bursting or evolution" |
| 04:45.21 | Ridgeback | drmessano: i guess TDMoE would be pretty low on the priority scale huh? |
| 04:46.05 | Ridgeback | file: hmmmm what is the general meaning of it? does it mean an audio or perhaps a control signal not getting to the dahdi module? |
| 04:46.06 | file | I think I've seen maybe 4 issues regarding TDMoE in my years and I do not think any of the current crew know it |
| 04:46.14 | drmessano | The last person to beta test TDMoE was Clayburg Wilmore of Baltimore, MD |
| 04:46.17 | drmessano | He died in 2002 |
| 04:46.20 | jaytee | I'm gonna pass it even though having it will be worthless. No one wants to hire someone over 50 for a good paying VOIP job anymore than someone wants a Charlie in The Box or a car with square wheels. I might as well just move to the Island for Misfit Techs and be done with it :-) |
| 04:46.24 | Ridgeback | drmessano: oh geeez |
| 04:46.37 | file | Ridgeback: chan_dahdi asked DAHDi to do something with hook start... and it failed |
| 04:46.38 | drmessano | RIP Clayburg |
| 04:46.44 | file | hook state... |
| 04:46.54 | Ridgeback | file: crap ok.... |
| 04:47.14 | file | so something in the dahdi kernel stuff returned an error |
| 04:47.31 | Ridgeback | so i guess I'm pretty much hosed for TDMoE experiments... |
| 04:47.53 | file | gives jaytee a muffin |
| 04:47.55 | jaytee | Ridgeback, talk to someone at Redfone and ask them for resources. |
| 04:48.31 | jaytee | file, ever see the Seinfeld episode where Elaine and Kramer start a business called Top O' The Muffin selling only muffin tops? |
| 04:48.37 | file | jaytee: sure have |
| 04:48.37 | Ridgeback | jaytee: does Redfone deal with TDMoE? |
| 04:48.53 | file | Ridgeback: they use a somewhat modified version of it |
| 04:49.15 | [TK]D-Fender | jaytee: You mean... we have a whole ISLAND now?!?! |
| 04:49.15 | Carlos_PHX | Ridgeback: Redfone pretty much is TDMoE for Asterisk, don't know of any others. |
| 04:49.19 | [TK]D-Fender | dances |
| 04:49.46 | Ridgeback | hmmm thats pretty interesting...got thier page up now |
| 04:49.48 | jaytee | Ridgeback, yeah. I uses TDMoE with zaptel to get T1 channels to Asterisk. don't know if they have added support for DAHDI but you might find config info on their site you could apply. |
| 04:49.51 | drmessano | IAX2 is the TDMoE of asterisk |
| 04:50.04 | drmessano | pwned |
| 04:50.07 | [TK]D-Fender | Ridgeback: Let me reiterate befoe you think its the Second Coming : nobody gives a shit about TMDoE. Investing infrastructure and effort down that road will lead to pain at some point. |
| 04:50.12 | jaytee | I'd rather stick with a Digium T1 card. |
| 04:50.32 | jaytee | YEAH!!! what he said ^^^^ |
| 04:50.44 | file | [TK]D-Fender: be nice or I'll replace you with an ice cream |
| 04:50.52 | drmessano | IAX2 FTWZOMGIFUARGUEUSUXOR |
| 04:50.55 | Ridgeback | [TK]D-Fender: ok thats fine no one cares about TDMoE. Its just my lab setup at home. for work we have the $$ for real gear..at home.. I don't.. |
| 04:51.07 | drmessano | I accidentally the whole IAX2 |
| 04:51.08 | Carlos_PHX | has been trying to go TDMoE for over a year, gave up at Astricon. |
| 04:51.29 | drmessano | I tried TDMoE once.. I didn't like it.. I didn't inhale. |
| 04:51.50 | Carlos_PHX | I drank the Kool-Aid and still wasn't convinced. |
| 04:51.54 | [TK]D-Fender | Ridgeback: and the "learning" factor is meaningless. to pass a call is 1 stupid dial just like any other. Since you are simulating FAILURE, there are plenty of other ways to test things |
| 04:52.00 | Carlos_PHX | Redfone does have a nice comet though. |
| 04:52.11 | jaytee | TDMoE stands for Totally Dumb Mutha*%#@!s on Ethernet |
| 04:52.24 | file | jaytee: so what did you think of the office? |
| 04:52.41 | jaytee | file, it was very nice. we got to take a tour of all the offices |
| 04:52.43 | drmessano | Total Dumbass Modulating of Ethernet |
| 04:52.49 | file | jaytee: I heard. |
| 04:52.52 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 04:52.53 | jaytee | drmessano, that's even better |
| 04:52.57 | Ridgeback | [TK]D-Fender: keep in mind people connect ASterisk via T1 cards to PBX's. I need to simulate that at a signalling level. TDMoE would have done that for me. |
| 04:53.57 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 04:54.01 | Carlos_PHX | Asterisk never fails. |
| 04:54.02 | jaytee | file, it was cool to meet several of the people I chat with in here. |
| 04:54.05 | Carlos_PHX | You don't need to test that. |
| 04:54.11 | [TK]D-Fender | Ridgeback: And since all you're doing is faking it at home... what does it PROVE? |
| 04:54.12 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 04:54.41 | [TK]D-Fender | Ridgeback: you either have G.711 equivalency or you don't. |
| 04:54.51 | [TK]D-Fender | Ridgeback: fake = worthless |
| 04:55.19 | Ridgeback | [TK]D-Fender: my configs, my ideas, TDM offers deterministic connections. VoIP doesn't. |
| 04:55.21 | jaytee | Ridgeback, if you really want to learn to "fake it" I can introduce you to my ex. |
| 04:55.44 | Ridgeback | jaytee: she faked it with you too????? |
| 04:55.59 | jaytee | she fakes it with everyone |
| 04:56.10 | Ridgeback | jaytee: too bad :( |
| 04:56.18 | Carlos_PHX | She told me it was real. |
| 04:56.18 | jaytee | too bad for her |
| 04:56.34 | drmessano | jaytee: She told me she was a man.. now I feel really stupid |
| 04:56.57 | Ridgeback | [TK]D-Fender: hey man, question for ya// how do you do E&M WInk over IAX2? |
| 04:57.53 | joako | You don't |
| 04:58.10 | Ridgeback | joako: I know :) |
| 04:58.19 | [TK]D-Fender | Ridgeback: What are you looking to test on it though? All * does is process calls so the only thing to "develop" is IVR's, dialplan, etc. for the few lines of config (which AREN'T the same), and the driver dependency, and potential hardware issues.... who cares what your call comes in on? |
| 04:59.28 | Ridgeback | [TK]D-Fender: uh yes it does. What if I have 4 T1 spans? what is the signalling? how do I handle alarm events. TDM is a different beast then jsut a DIAL/SIP |
| 04:59.57 | jaytee | my ex just started working for Microsoft 2 months ago. She thinks it's a good thing. I think it's her bad karma coming round and she's just too blonde to realize it. |
| 05:00.17 | drmessano | Using TDMoE to "simulate" T1 behaviour is like using "Days of Thunder" to simulate acting |
| 05:00.26 | jaytee | ROFL |
| 05:00.27 | [TK]D-Fender | Ridgeback: And how many of the real-world issues will be replicateable via TDMoE? There is no potential incompatibility within a given standard because it's * on both sides |
| 05:00.38 | Ridgeback | [TK]D-Fender: also a lot of my stuff i work on is PRI/BRI and ISDN |
| 05:00.56 | [TK]D-Fender | drmessano: "Days of Thunder is simulated acting.... and all BAD acting :) |
| 05:01.02 | Ridgeback | i'm not worrking abour the TDMoE outside frames. Thats from * to *. what I'm working on is the TDM portion. |
| 05:01.09 | drmessano | [TK]D-Fender: How do you simulate a bad NID with TDMoE? lol |
| 05:01.35 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-14-232.phlapa.east.verizon.net) |
| 05:01.42 | [TK]D-Fender | Ridgeback: Exactly. PRI can have tons of issues. I fake TDMoE isn't capable of the range of fuckups that can happen in the "real world". So useing it as a testing basis = useless |
| 05:02.46 | *** join/#asterisk phpboy (n=shane@196.36.108.18) |
| 05:03.21 | Ridgeback | [TK]D-Fender: well im not worring about fuck ups or major issues. What I'm working on is builind TDM/ DS0 calls, and a managing these calls in a certain way. The realworld system is using all T1's. We use signalling to monitor status. With TDMoE I can use two * boxes to build TDM channels between them to test out TDM functions. |
| 05:03.29 | jaytee | we need to post an April Fools joke on voip-info.org about the new "G.751" codec that supports Dolby 5.1 surround sound and see how many people come in here asking where to get the codec, what phones support it and how to configure Asterisk with it. |
| 05:03.37 | [TK]D-Fender | This is like racing RC cars as training for becoming a 747 PILOT. They're both vehicles right? |
| 05:03.48 | jaytee | ol |
| 05:03.51 | jaytee | lol |
| 05:05.33 | drmessano | HA |
| 05:05.38 | Ridgeback | [TK]D-Fender: ok so if TDMoE is bad. How do you build TDM channels from * to * with E&M signalling? |
| 05:05.45 | drmessano | G.733 |
| 05:06.06 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:06.12 | drmessano | G.733 = Quadrophonic HD Stereo @ 256k |
| 05:06.14 | jaytee | "Yeah! I'm ready to deploy to Iraq! I've played Gears of War 2 on my X-Box |
| 05:06.23 | [TK]D-Fender | Ridgeback: who WANTS to? You are trying to implement something nobody has seemed to find a valid reason to care about! We have TDM, SIP, IAX2, and so many other protocols. |
| 05:06.41 | Ridgeback | [TK]D-Fender: I need it. |
| 05:06.47 | [TK]D-Fender | Ridgeback: Ridgeback For what? |
| 05:06.50 | Spirits-Sight | [TK]D-Fender: I have good news, I am now able to make out going calls, I would like to ask you though is there a way to make it simpler then I have it or is it the best way it is now for voicepluse, I would like it to be as simple as can so I can build up on that and learn as I do changes http://pastebin.com/dd58720a |
| 05:06.52 | [TK]D-Fender | Ridgeback: Redfone? |
| 05:07.05 | joako | Ridgeback: T1/E1 cards... but if you want to just test E&M, why? I understand your curiosity but E&M is a very specific standard, either you implement it or you don't and Asterisk does... if you need to use it, it does and will work |
| 05:07.18 | Ridgeback | [TK]D-Fender: I want to run TDM between two boxes with E&M signalling. if theres a better way then TDMoE. then lets hear it. |
| 05:07.49 | Ridgeback | [TK]D-Fender: I need to build my lab up to support this. no need to go to a customer site and fumble around. |
| 05:07.55 | jaytee | Ridgeback, get * 1.4 and use zaptel and the setup guide for Redfone's equipment to setup the TDMoE span and have fun. There are examples for setting up the span in the zaptel.conf.sample files in earlier builds of zaptel for 1.4 |
| 05:07.58 | [TK]D-Fender | Ridgeback: what POINT is there to doing E&M between 2 * boxes? What does that offer you that any other channel can't? its a FRIGGEN CALL. |
| 05:08.10 | jaytee | but I don't think it's even supported at all in DAHDI |
| 05:08.35 | [TK]D-Fender | Ridgeback: * does E&M to interconnect to other people archaic CRAP PBX's. Why would anyone choose that is the link between to CAPABLE systems? |
| 05:08.44 | Ridgeback | [TK]D-Fender: becuse E&M is a sigalling protocal which must be understood and learned. Jsut like SIP is a siganlly protocol. |
| 05:08.45 | [TK]D-Fender | tw0* |
| 05:09.13 | [TK]D-Fender | Ridgeback: Funny... noone I know has any reason to care about E&M. |
| 05:09.17 | Ridgeback | [TK]D-Fender: Also TDM is different than a nondeterministic voip call. |
| 05:09.25 | jaytee | Ridgeback, are you going to study rotary pulse dial phones next? |
| 05:09.31 | joako | lol |
| 05:09.50 | Ridgeback | [TK]D-Fender: no one here uses t1's to interface to older PBX's anymore? |
| 05:10.00 | [TK]D-Fender | jaytee: WOAH buckaroo.... don't bypass the TELEGRAPH or the Smithsonian will be all over your ass! |
| 05:10.16 | Ridgeback | jaytee: plenty of people still use E&M |
| 05:10.16 | jaytee | I use T1 to interface to a Nortel switch but I use PRI not E&M |
| 05:10.22 | drmessano | I want to put a second NIC in two boxes, put a crossover between them, and set up an IAX trunk with 672 allowed calls between.. Just so I can have a DS3 over IAX2 |
| 05:10.26 | Ridgeback | jaytee: ok thats fine. |
| 05:10.50 | jaytee | well, thanks! Your approval means the world to me :-) |
| 05:10.58 | [TK]D-Fender | Ridgeback: Look at the %. How tiny? And you're looking to fake something whose channel contention will be the only possible point of comparison to the real thing because all the rest is a 100% idetical standard. |
| 05:11.05 | drmessano | OMFG |
| 05:11.11 | drmessano | I want telegraph over IAX2 |
| 05:11.25 | [TK]D-Fender | Ridgeback: thats the probloe... * talking to * = cooperation. Get * working with OTHER stuff is the problem and you can't fake that! |
| 05:11.42 | jaytee | dit dit dit dah dit dah dah dit dah dah dah dit |
| 05:12.30 | jaytee | I used to watch the guys in "Diddy Bop" wash out in tech school because they couldn't get their speed in Morse up to passing levels and they |
| 05:12.36 | jaytee | would crack under pressure. |
| 05:12.53 | jaytee | they usually dragged them out of their dorm rooms twitching and drooling |
| 05:13.09 | drmessano | * talking to * is like plugging an FXO ATA into an FXS ATA and calling that a simulation of the entire telco system. Now, put 10,000 miles of copper in line, throw a couple trees and buckets of water on top, then have your cat chew on the power cord for a few hours.. You're closer! |
| 05:13.09 | Ridgeback | [TK]D-Fender: you have to keep in mind the real issue isnt compatibility. i have a particular TDM need. by building TDM channels between two boxes with E&M signalling. I can get my ideas worked out if you don;t like that fine. regardless of your dislike for TDMoE, I still need TDM channels between two. |
| 05:14.11 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 05:14.16 | [TK]D-Fender | Ridgeback: Again all you'll succeed in faking out is channel availability contention because * will not have any problems in the standards it uses in communicating with itself. Therefor what is left? |
| 05:14.32 | [TK]D-Fender | Ridgeback: one a call is up, a call is up. |
| 05:14.48 | Ridgeback | [TK]D-Fender: thats fine. i want it to work. whats left are my ideas for channel mangement, scripting, remote controls etc.. |
| 05:15.08 | Ridgeback | [TK]D-Fender: fine fine, but like i sad TDM is different beast than a SIP call. |
| 05:15.13 | [TK]D-Fender | Ridgeback: Which has nothing to do with fighting to get a channel functional. |
| 05:15.21 | jaytee | Ridgeback, so then go get * 1.4 with zaptel and use the examples |
| 05:15.31 | drmessano | Simulating perfect conditions isnt teaching anything |
| 05:15.33 | Ridgeback | [TK]D-Fender: good, I dont want to fight to get a channel running! |
| 05:15.39 | drmessano | Its plug and play stupiditu |
| 05:15.40 | drmessano | Its plug and play stupidity |
| 05:15.44 | Spirits-Sight | [TK]D-Fender: by any chance did you see my post? |
| 05:15.45 | [TK]D-Fender | Which is what everyone else fights over learning *. Why can't my SIP call my other sip? I called on [1000] so why can't I dial it? |
| 05:15.52 | Ridgeback | drmessano: nope its not. but im not trying to simulate fault. |
| 05:16.00 | jaytee | I thought plug and play stupidity was Windows? |
| 05:16.00 | [TK]D-Fender | Spirits-Sight: It works right? |
| 05:16.11 | [TK]D-Fender | jaytee: "Plug&Pray" |
| 05:16.14 | joako | Anyone know of a SIP ATA with passthrough to a PSTN line? To get inbound calls from the PSTN but dial out via SIP? |
| 05:16.32 | [TK]D-Fender | joako: SPA-3102 |
| 05:16.36 | drmessano | Ridgeback: Youre not trying to simulate success either.. because it's guaranteed with a fake system that no better than calling DOS an IOS simulator |
| 05:16.45 | jaytee | damn Skippy, you're too quick! |
| 05:17.07 | Ridgeback | drmessano: im happy it would work. Once the TDM span is running, my stuff takes over which is exatcly what i want |
| 05:17.13 | jaytee | joako, what [TK]D-Fender said. It'll do that or FXS to FXO if you want. |
| 05:17.25 | [TK]D-Fender | drmessano: Ridgeback at which point what takes over doesn't give a crap WHERE the call came from |
| 05:17.30 | *** join/#asterisk chendy (n=chatzill@121.34.152.108) |
| 05:17.40 | Ridgeback | [TK]D-Fender: in my case it does. |
| 05:17.43 | Spirits-Sight | [TK]D-Fender: yes it works I am glad to say, but I would like to try and simplfie it say I can build up on it and learn as I do and not just use the code because it is said to be used that way, you know what I mean, I want to learn from doing as I read the book, I am rereading the area about dailplains and want to start from the most basic working point |
| 05:17.58 | [TK]D-Fender | Spirits-Sight: You whole config fits on my screen... |
| 05:18.01 | joako | Fender, so I can set that up like an SPA-2000, plug in the analog line to it as well and the inbound calls will get passed through, or does the call only go via SIP back to the server? |
| 05:18.07 | drmessano | Ridgeback: Then go for it.. The more that people get things wrong, the better prospects I have for future gainful employment |
| 05:18.10 | [TK]D-Fender | Spirits-Sight: How much smaller can you imagine? |
| 05:18.12 | drmessano | Carry on, my friend.. carry on |
| 05:18.26 | [TK]D-Fender | There'll be peace when you are done! |
| 05:18.49 | Carlos_PHX | Well, I'm going to go test drinking some shoe polish, it smells kinda like good whiskey so it should be close enough to see if I like whiskey. Then I'm going to go molest a pumpkin, that should be a good test of whether I'd like banging Carmen Electra. |
| 05:18.59 | [TK]D-Fender | Warning : the road les traveled has no service stations. |
| 05:19.06 | drmessano | I need to do a remake of the hollies "Write on".. call it "Wrong On" |
| 05:19.16 | Ridgeback | drmessano: thats fine, all I wanted know if 1.60.x supported TDMoE. didnt need a diatribe of opinions! i have a very specific need of which this would be perfect for. |
| 05:19.19 | drmessano | "Wroooong on.. even though there's no one listening to your calls" |
| 05:19.32 | [TK]D-Fender | Carlos_PHX: You had me at any prospect of banging carmen Elektra :) |
| 05:19.45 | Spirits-Sight | [TK]D-Fender: as small and as simple as it will work, as I said I only want to be in the simplest from so I can learn as I do each part, I know its already small but I am sure that there is stuff in there I don't need for handling out going calls |
| 05:20.10 | drmessano | Ridgeback: I know.. "I just want the fucking answers now, and I could care less what you people think" is about what IRC has turned into anyway.. I also left off the part about FREE and demanding. |
| 05:20.13 | Carlos_PHX | I once dated a girl who had kissed her at a party, so I'm only one degree away. |
| 05:20.35 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 05:20.46 | jaytee | hmmm, I ran cat extensions.conf|./astograph.py|dot -Tpng go.png and when I opened the go.png file most all of my contexts point to the [DEAD-END] context and that only points to [MORON]. I'm know I messed something up but I'm not sure what. :-) |
| 05:21.00 | Ridgeback | drmessano: simple questions get the most heat i guess |
| 05:21.10 | [TK]D-Fender | Spirits-Sight: exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|VPC_B) <- broken |
| 05:22.03 | drmessano | Ridgeback: No, coming into an IRC channel and querying the regulars like we're Google and getting pissy when opinions are returned with the search answers is what generates the most heat.. not that you would understand that. |
| 05:22.22 | [TK]D-Fender | Spirits-Sight: and your dialplan is almost as short as is possible |
| 05:22.43 | Ridgeback | drmessano: when did i get pissy? all i got was TDMoE is stupid remarks! geeez |
| 05:23.28 | Ridgeback | drmessano: i never askedfor an opion on TDMoE, all i asked was if someone know if TDMoE was supported. is that hard to answer? |
| 05:23.46 | Carlos_PHX | In our opinion, it is. |
| 05:23.54 | Spirits-Sight | what about the sip.conf file, also can you please explain why you said "exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|VPC_B)" is broken ? I am sorry that I may ask question when the answer is font of my eyes but I just may not see it (blind) :-) |
| 05:23.54 | drmessano | HA.. yeah.. thats all anyone said.. No one explained the lack of logic in your reasoning.. We just all busted out with "Duh, TDMoE IS ZOMG TEH DUMB".. we also didnt try to steer you in a better direction, or be helpful |
| 05:24.00 | drmessano | So yeah, you caught us. |
| 05:24.22 | Carlos_PHX | I hope he doesn't ask for a refund or give us a bad customer service rating. |
| 05:24.23 | Ridgeback | Carlos_PHX: thats fine. ok it is. great. i didn't ask that. |
| 05:25.04 | [TK]D-Fender | Carlos_PHX: If he doesn't like the free advice we'll give him DOUBLE his money back! |
| 05:25.21 | drmessano | Ridgeback: Sorry we didnt just answer yes or know.. This is a CHAT CHANNEL where people converse.. Not a Google proxy. |
| 05:25.27 | drmessano | or no* |
| 05:25.30 | Ridgeback | [TK]D-Fender: ill take it. |
| 05:25.32 | drmessano | GAH |
| 05:25.43 | *** join/#asterisk jjg (n=jjg@76.21.4.40) |
| 05:25.45 | [TK]D-Fender | Ridgeback: Just answer this : when is any call over TDMoE between 2 * ever going to fail? |
| 05:25.46 | Ridgeback | drmessano: ok google didnt say cap, no did the docs. |
| 05:25.57 | Carlos_PHX | Besides, it's late and many of us have been fixing broken shit all day and we'll just debate the finer points of whatever for the hell of it. |
| 05:26.00 | Ridgeback | [TK]D-Fender: you answer me, have you gotten it to work? |
| 05:26.09 | [TK]D-Fender | Ridgeback: Once the call makes it through, how is it any more functional than any other channel? |
| 05:26.39 | Ridgeback | [TK]D-Fender: becuase I'm not worried about the fact of the channel existance. I'm worried about the signalling. |
| 05:26.56 | drmessano | Ridgeback: great, so you come into a CHAT channel, and expect to use the lot of us like your google proxy.. and then get borderline insulting over "I didnt want your opinions, just an answer".. |
| 05:26.57 | [TK]D-Fender | Ridgeback: Ridgeback What are you going to do... try to set them up so they DON'T match? |
| 05:27.17 | Bad_Robot- | hi everyone |
| 05:27.17 | Ridgeback | [TK]D-Fender: have you ever worked on TDM before> |
| 05:27.42 | [TK]D-Fender | Ridgeback: Because barring that what kind of failure are you expecting to get? there are not protocol hiccups. it is the EXACT same on both sides. Unlike * + any other equipment. |
| 05:27.50 | Bad_Robot- | looking for advise on what to run for dns on phones with a pri? |
| 05:28.01 | Ridgeback | drmessano: all i aksed for was if anyone kne if it was supported.. a friendly yes or now or idk is fine. |
| 05:28.14 | Carlos_PHX | Bad_Robot-: Huh? |
| 05:28.17 | Spirits-Sight | [TK]D-Fender: what about the sip.conf file, also can you please explain why you said "exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|VPC_B)" is broken ? I am sorry that I may ask question when the answer is font of my eyes but I just may not see it (blind) :-) |
| 05:28.18 | [TK]D-Fender | Ridgeback: Again, you have a channel, or you don't. this is between 2 * which we know you can pass channels between any number of healthier ways. |
| 05:28.26 | Ridgeback | [TK]D-Fender: thats fine, i need to hanlde the signalling. |
| 05:28.28 | drmessano | Ridgeback: Ok, and your opinion doesnt fucking matter either, just like ours apparently |
| 05:28.38 | Bad_Robot- | my box has a pri t1 and curious what dns settings to put on phones |
| 05:28.39 | [TK]D-Fender | Spirits-Sight: Look where it jumps to |
| 05:28.44 | Ridgeback | drmessano: i never offered my opion |
| 05:28.50 | Ridgeback | opinion* |
| 05:28.53 | Carlos_PHX | Um, how does a PRI affect DNS? |
| 05:29.03 | Carlos_PHX | You mean data T1? |
| 05:29.13 | [TK]D-Fender | Ridgeback: But if you set them the same it will always work. What have you proved? that you can type E&M the same in 2 config stanzas? |
| 05:29.34 | Bad_Robot- | well my dhcp server is on a windows server so when our dsl went down the phones went down for some reason and yes i mean t1 sry |
| 05:29.40 | Carlos_PHX | I think he's proven he can keep #asterisk busy for hours, also. |
| 05:29.43 | drmessano | Ridgeback: If someone tells you "no" will you just leave? |
| 05:30.00 | Ridgeback | [TK]D-Fender:once they work I have aneed to handle the signalling for call control. outside of asterisk via E&M. |
| 05:30.30 | Carlos_PHX | Bad_Robot-: You might try a general network support channel. How to configure DNS is a pretty general networking issue. |
| 05:30.37 | Carlos_PHX | The phones use the same as the computers. |
| 05:30.42 | [TK]D-Fender | Ridgeback: ridbut that is NOT E&M between 2 * boxes. that you cannot emulate |
| 05:30.51 | drmessano | Carlos_PHX: This is why IRC is the way it is.. It used to be about DISCUSSION and some sense of community, now its about "LET ME COMES IN HERE AND DEMAND SOME ANSWER AND BITCH WHEN YOU GIVE ME WORDMOUTH TOO".. |
| 05:31.07 | [TK]D-Fender | Ridgeback: All this has done is added another ho that happens to use the same protocol. a hop which IS going to work as expected |
| 05:31.08 | Ridgeback | [TK]D-Fender: should be able to do E&M, it's in the dahdi config files |
| 05:31.15 | Carlos_PHX | The time from join to get answer to exit is increasingly shortened, isn't it? |
| 05:31.23 | drmessano | Carlos_PHX: The whole "I just wanted a yes or no, not your opinion" gives me the warm fuzzies |
| 05:31.24 | Spirits-Sight | [TK]D-Fender: if I am understand how it works, if VPC_OUT_P does not work then the next line tells it to go to VPC_OUT_B which it uses a name of VPC_B am I following it right, I hope I explained it right the way I am think it works |
| 05:31.33 | Ridgeback | [TK]D-Fender: I hope it works as expecte. once it does I can get to work on my ideas. |
| 05:31.41 | [TK]D-Fender | Ridgeback: In my 5 years of using * it has always supported E&M... |
| 05:31.47 | drmessano | Carlos_PHX: Ridgeback is a fine example of why people cant be bothered anymore.. or get an attitude |
| 05:31.55 | Bad_Robot- | that's how i have it the same as pc's but when dsl went down active directory dns couldn't resolve anything so phones stopped working and i was looking for advise on what to put. i put static ip's in phones and a live dns server and it all worked |
| 05:32.01 | [TK]D-Fender | ridbbut so far E&M between 2 boxes doesn't seem to add anything. |
| 05:32.12 | Ridgeback | [TK]D-Fender: i'm glad it does. once the calls are built I have my uses for E&M. |
| 05:32.31 | [TK]D-Fender | Ridgeback: between 2 *'s? |
| 05:32.49 | [TK]D-Fender | Ridgeback: What can one * signal to the other over it that is of any importance? |
| 05:33.13 | Ridgeback | [TK]D-Fender: E&M? E&M is how the calls are built off/on hook. |
| 05:33.32 | [TK]D-Fender | Ridgeback: Yes, but as a means of connecting 2 *'s. What does E&M add in value? |
| 05:34.06 | Carlos_PHX | Bad_Robot-: The phones should get DHCP just like anything else. From there your DNS server needs to be properly configured. There isn't one magic answer, it depends on your network. |
| 05:34.29 | Ridgeback | [TK]D-Fender: i'm not adding value to the system. I need to steal the E&M controls for out of band management |
| 05:34.31 | Carlos_PHX | Your DNS should know what the IP of your Asterisk server is. |
| 05:35.50 | Bad_Robot- | my dns server is 192.168.1.253 so when dsl went out the phones couldn't make calls thru the T1 which seemed weird |
| 05:35.57 | Spirits-Sight | [TK]D-Fender: did you get my response? by the way thanks for your help, I hope you don't mind that I ask question that may be dumb or whatever, I do thank you |
| 05:36.06 | Bad_Robot- | thx Carlos_PHX |
| 05:36.09 | drmessano | Bad_Robot-: What IP/hostname do the phones connect to? |
| 05:36.29 | [TK]D-Fender | Spirits-Sight>[TK]D-Fender: what about the sip.conf file, also can you please explain why you said "exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|VPC_B)" is broken ? I am sorry that I may ask question when the answer is font of my eyes but I just may not see it (blind) :-) |
| 05:36.39 | Bad_Robot- | the phones connect to asterisk and xo communcations laid the t1 with 23 lines |
| 05:36.45 | [TK]D-Fender | Spirits-Sight: what is ${EXTEN} doing in there? |
| 05:36.47 | drmessano | You didnt answer me |
| 05:36.49 | drmessano | Bad_Robot-: What IP/hostname do the phones connect to? |
| 05:37.17 | drmessano | I know its asterisk.. thats kinda silly |
| 05:37.20 | Bad_Robot- | 192.168.1.5 badphone.localdomain.local |
| 05:37.26 | drmessano | Ok, so internal |
| 05:37.32 | Bad_Robot- | sry this is the first setup for me |
| 05:37.52 | drmessano | Ah |
| 05:38.00 | drmessano | What DNS is asterisk using? |
| 05:38.04 | Bad_Robot- | but it's been alot of fun |
| 05:38.14 | Spirits-Sight | [TK]D-Fender: this is what was in the file from voicepule, I did not see or able to figure out how it was used as I did not see it asigned any where |
| 05:38.17 | Bad_Robot- | i have asterisk using 206.13.28.12 |
| 05:38.23 | drmessano | Ok, dont do that |
| 05:38.26 | drmessano | Use the AD DNS |
| 05:38.51 | [TK]D-Fender | Spirits-Sight: Go read the instructionfs for GotoIF and look at what it is doing. |
| 05:38.54 | Bad_Robot- | so it was asterisk that couldn't see the phones and nothing to do with resolution over the net? |
| 05:39.06 | drmessano | No it was asterisk that couldnt see DNS |
| 05:39.11 | drmessano | and SIP goes wonky |
| 05:39.15 | Spirits-Sight | ok thanks, will read now |
| 05:39.34 | drmessano | I normally install BIND and use 127.0.0.1 as a secondary DNS on boxes for that reason |
| 05:39.41 | Bad_Robot- | ok i'll will try that in the morning and add badphone to dns and point asterisks to 192.168.1.253 |
| 05:39.43 | drmessano | Otherwise I have seen weird crap |
| 05:40.33 | Bad_Robot- | so with a t1 there should be no dns involved other than phones to pbx but not over t1 i'd think |
| 05:41.04 | drmessano | Correct, if youre using a PRI |
| 05:41.28 | Bad_Robot- | ok :) i really appreciate it thx |
| 05:42.41 | Bad_Robot- | is it bad practice to put static ip's in the phones |
| 05:42.55 | drmessano | Its kinda silly.. |
| 05:43.01 | drmessano | DHCP should be fine |
| 05:43.18 | Bad_Robot- | it was a pain to login to each phone and change network settings |
| 05:43.37 | Bad_Robot- | lucky i only had 20 to do |
| 05:44.54 | Carlos_PHX | If you have a small network, bind isn't so hard to deal with. |
| 05:44.57 | Carlos_PHX | Worth doing. |
| 05:45.12 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 05:45.19 | Carlos_PHX | If your primary DHCP/DNS is on Windows, it helps to have a backup on a reliable OS. |
| 05:45.30 | Bad_Robot- | i want to do it right |
| 05:45.53 | Spirits-Sight | [TK]D-Fender: if I am understand it, it looks like it would have the exten I am calling from so in this case I believe if understand right it would be 100 so would look like this: exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?100|VPC_B) |
| 05:45.54 | Carlos_PHX | Two DNS servers is definitely right. |
| 05:45.55 | Bad_Robot- | hahah i could install dhcpd on centos box i guess |
| 05:46.00 | Carlos_PHX | You can do two DHCP also. |
| 05:47.18 | Bad_Robot- | i gotta get in early tomorrow and play before anyone gets in and needs a phone |
| 05:48.15 | [TK]D-Fender | Spirits-Sight: What is 100? |
| 05:48.46 | [TK]D-Fender | Spirits-Sight: Where do you get that # from? and that tells it where to JUMP TO. |
| 05:49.21 | Spirits-Sight | [TK]D-Fender: its a extion that is connected to a sip phone, no, the VPC_B tells it where to jump to I believe |
| 05:49.37 | Spirits-Sight | I am still reading about gotoIF |
| 05:50.11 | *** join/#asterisk pepperjack (n=happy@h204.180.29.71.dynamic.ip.windstream.net) |
| 05:50.15 | *** part/#asterisk pepperjack (n=happy@h204.180.29.71.dynamic.ip.windstream.net) |
| 05:50.28 | [TK]D-Fender | Spirits-Sight: " its a extion that is connected to a sip phone," <- no. |
| 05:50.42 | [TK]D-Fender | and a sip phone does not have an "extension". |
| 05:50.47 | *** join/#asterisk error404notfound (n=shoaibi@58-65-160-128.nayatel.pk) |
| 05:51.09 | error404notfound | I am trying to start asterisk on freebsd 6, before it worked fine and now I get "/libexec/ld-elf.so.1: /usr/local/lib/libh323_r.so.1: Undefined symbol "_ZN9PIPSocket17GetInterfaceTableER5PListINS_14InterfaceEntryEE"" |
| 05:51.36 | Spirits-Sight | its the content in the sip.conf which just happens to be the extion for the sip phone, sorry I am trying to userstand this |
| 05:52.06 | [TK]D-Fender | Spirits-Sight: ${EXTEN} is the number of the EXTENSION you are in the middle of PROCESSING. |
| 05:52.19 | [TK]D-Fender | Spirits-Sight: exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?100|VPC_B) |
| 05:52.32 | [TK]D-Fender | Spirits-Sight: its going to be the # you DIALED that matched that pattern |
| 05:53.34 | jaytee | time for some zzzzz's |
| 05:53.37 | jaytee | nite all |
| 05:53.45 | Spirits-Sight | so where ${EXTEN} in the second line is the number that was dailed, is this so it carries it to the next line? |
| 05:54.54 | Spirits-Sight | Sorry now I am confused |
| 05:55.45 | Spirits-Sight | so the ${EXTEN} is the extension in this case 100? |
| 05:59.34 | error404notfound | anyone? |
| 05:59.44 | [TK]D-Fender | Spirits-Sight: _1NXXNXXXXXX <- its the number you dialed that matched THIS |
| 06:00.05 | Spirits-Sight | that I understand |
| 06:00.29 | [TK]D-Fender | Spirits-Sight: Go read GotoIF again. You don't seem to understand what it does |
| 06:00.38 | Spirits-Sight | thats one thing I did understand |
| 06:03.04 | Spirits-Sight | GotoIF is if condition above failed in this case chanunavail then it tell it to goto next provider to try and make the call, and its doing this by telling it to got to VPC_B which is a backup, now it does this using a name instead of a content [content] thing, am I understand ? |
| 06:04.15 | error404notfound | I am trying to start asterisk on freebsd 6, before it worked fine and now I get "/libexec/ld-elf.so.1: /usr/local/lib/libh323_r.so.1: Undefined symbol "_ZN9PIPSocket17GetInterfaceTableER5PListINS_14InterfaceEntryEE"" |
| 06:04.34 | [TK]D-Fender | Spirits-Sight: No. It does not go to a PROVIDER. It goes to a specific PRIORITY, EXTEN & CONTEXT |
| 06:04.48 | [TK]D-Fender | Spirits-Sight: as in jumps somwhre else in your DIALPLAN |
| 06:05.15 | [TK]D-Fender | Spirits-Sight: what you do there is your job |
| 06:05.18 | Spirits-Sight | its like a if statement in php, if condition is met it finished and if not then it goes to what it tell it to go to as you said priority, exten |
| 06:06.41 | Spirits-Sight | So would it be safe to say there is no reason then to have ${EXTEN} on that line? |
| 06:06.48 | *** join/#asterisk dynaguy (n=gao@d154-20-21-173.bchsia.telus.net) |
| 06:07.12 | [TK]D-Fender | Spirits-Sight: I'm saying you're calling GotoIf as if you think it is a DIAL COMMAND. it is not. |
| 06:07.29 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 06:07.43 | [TK]D-Fender | Spirits-Sight: Go read Gotoif over again. |
| 06:08.04 | error404notfound | anybody.... |
| 06:08.21 | Spirits-Sight | I did not create that line, I am just trying to understand why its the way it is? Voicepluse has it in their setup like that |
| 06:09.25 | [TK]D-Fender | Spirits-Sight: Before using other people's code you should understandw hat it doesn. Or you might as well go stapling frisbees to your car's windshield wiper blades, but its just as applicable a theory :) |
| 06:09.41 | [TK]D-Fender | Wow, my typing skills are about shot for the evening... |
| 06:11.32 | Spirits-Sight | I don't drive (I am blind) and thats why I am asking what it is and how to make it simpler so I can build on that and learn to break that down and now get stuck trying to figure out things that my brian won't pick up pick up with out the simpler things first |
| 06:12.04 | Spirits-Sight | now = not |
| 06:13.02 | [TK]D-Fender | Spirits-Sight: I think you should look at each line in your dialplan. Yous hould understand how these apps work, and then look at the flow you want. Walk through it step by step seeing how each variable will be populated and evaluated. |
| 06:13.42 | [TK]D-Fender | Spirits-Sight: this is learning programming. * assumes you cabable of some of the most rudimentary programmers sens of logic |
| 06:13.58 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 06:17.38 | Spirits-Sight | I understand var and the flow, my problem is that if things are to confusing from the start its harder for my brain to follow the flow, like I was reading the book asterisk and I was able to follow the flow for creating a simple interactiive menu, but once I tryed to follow the extension file that voicepulse gave, it was tomuch for me to follow, this is why I was trying to shorten it with out losting the ability to make call |
| 06:19.33 | [TK]D-Fender | Spirits-Sight: then ignore the funky crap and come up with your own logic. Don't jsut cut & paste code. Doing that from the WIKI is a great way to FUBAR yourself as well... |
| 06:20.26 | [TK]D-Fender | Spirits-Sight: Many samples out there are just plain done wrong, poorly planned, or worse where you think they're on CRACK for coming up with that kind of junk |
| 06:20.32 | [TK]D-Fender | Anyway.... checkout time here. |
| 06:20.35 | [TK]D-Fender | later all... |
| 06:21.29 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 06:23.54 | *** join/#asterisk szallol (n=szallol@86.105.195.113) |
| 06:26.38 | drmessano | Its a league game, smokey |
| 06:48.27 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 06:52.41 | drmessano | http://photoput.com/viewer.php?file=8ykfi2bovcglvvjuldn1.jpg |
| 06:52.56 | drmessano | ^^^^^^^ FAIL |
| 06:53.59 | *** join/#asterisk dahunter3 (n=dahunter@pool-72-67-222-109.lsanca.fios.verizon.net) |
| 06:56.05 | *** join/#asterisk nagi (n=nagi@195.248.67.249) |
| 06:57.08 | demonist | lol |
| 06:57.12 | demonist | nice fail picture |
| 06:57.54 | demonist | not epic though |
| 06:58.11 | demonist | i like epic failures, such as pole vaults going wrong |
| 07:11.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 07:58.45 | *** part/#asterisk error404notfound (n=shoaibi@58-65-160-128.nayatel.pk) |
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| 08:00.20 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 08:00.25 | fcois93 | hello all ! |
| 08:04.37 | *** join/#asterisk apollonx (n=admin@92.47.26.29) |
| 08:06.25 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 08:08.02 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 08:08.47 | yidiyuehan | hi, any one knows why the IP phone couldn't hear remote party's keypad touch tone during a call? it's ok for analog phones and key phone system. |
| 08:11.17 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 08:11.21 | sosperec | hello |
| 08:16.53 | DarKnesS_WolF | tzafrir_laptop: i have a question about the rapidtunnle |
| 08:17.02 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.186) |
| 08:19.56 | *** part/#asterisk kornelak (n=karl@199.33.79.4) |
| 08:20.03 | yidiyuehan | hi, any one knows why the IP phone couldn't hear remote party's keypad touch tone during a call? it's ok for analog phones and key phone system. |
| 08:24.08 | *** join/#asterisk arnor001 (n=arossouw@dsl-146-31-195.telkomadsl.co.za) |
| 08:24.15 | *** join/#asterisk ghostknife (n=root@196.210.139.100) |
| 08:24.19 | szallol | somebody knows how to dial out on ~200 SIP lines in parallel? |
| 08:25.25 | ghostknife | I have a very weird problem.When I dial out, I get nothing but a quiet line, also, dialing in I hear ringing in the remote phone, but our phones don't ring. this is my verbose==5 output for dialing out: http://rafb.net/p/SX3pKL75.html |
| 08:25.59 | ghostknife | szallol: have ~200 trunk ports and lines? |
| 08:26.08 | ghostknife | szallol: plus a very strong machine |
| 08:26.52 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 08:27.54 | kaldemar | szallol: use chan local to divide into multiple dials |
| 08:28.26 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:29.09 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:30.47 | kaldemar | ghostknife: since you use freepbx, i suggest you ask in #freepbx. |
| 08:33.09 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 08:34.51 | *** join/#asterisk apollonx (n=admin@89.218.151.199) |
| 08:37.39 | szallol | what is the best method to initiate a call, by manager api or call file? |
| 08:38.39 | kaldemar | get to know how they work and decide what's best for you. |
| 08:43.26 | arnor001 | is there a way to configure asterisk dialplan, when calling a voip provider, to test audio quality over the line, then if its bad use the ISDN instead? |
| 08:46.02 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 08:52.04 | kaldemar | arnor001: no, unless you implement a way, but i'd say that's not feasible. you could test e.g. packet loss before dialing out before the call but that would take too much time and not provide useful results. |
| 08:52.40 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:53.47 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:54.34 | ghostknife | kaldemar: I just installed it like that for the basic configuration, beyond this I use pure asterisk configuration. |
| 08:54.47 | ghostknife | kaldemar: back then I didn't know asterisk and needed a headstart |
| 08:55.09 | ghostknife | kalbesdides , it's yet another digium failure |
| 08:55.40 | kaldemar | what is? |
| 08:56.34 | kaldemar | that you don't get ringing indication in your phone or that your phones don't ring? |
| 08:56.55 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 09:01.33 | ziram19 | hi can i view a chat that i made last night on irc? |
| 09:01.46 | ziram19 | on this channel i mean |
| 09:04.23 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 09:05.40 | drmessano | alt-f4 |
| 09:07.34 | C4away | ziram19: if your irc client was logging it |
| 09:07.37 | C4away | many don't by default |
| 09:07.58 | C4away | the other option is to look online for a site that posts logs |
| 09:08.08 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 09:09.19 | ziram19 | ok thanks |
| 09:11.24 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 09:12.02 | C4away | http://ibot.rikers.org/%23asterisk/20081118.html.gz |
| 09:13.45 | C4away | although you didn't say much that day ziram19 |
| 09:13.52 | C4away | this has a bit more conversation: http://ibot.rikers.org/%23asterisk/20081117.html.gz |
| 09:14.13 | C4away | not much though |
| 09:14.20 | neurosys | For what reason would Playtones() not make sounds on the channel? Just dead silence |
| 09:14.26 | C4away | what day are you actually looking for? |
| 09:14.34 | C4away | you are not giving it a proper tone? |
| 09:14.41 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-0074.yubc.net) |
| 09:14.45 | C4away | Playtones(sexsounds) wouldn't work |
| 09:14.55 | C4away | but Playtones(congestion) might |
| 09:15.07 | neurosys | C4away: I tried ring, busy, congestion, and made sure indications.conf was ok |
| 09:15.08 | dushantch | Hi, I get Unable to open pid file '/var/run/asterisk/asterisk.pid': Permission denied when I try to start asterisk 1.6. Any ideas? |
| 09:15.22 | C4away | that's just one reason it might not work |
| 09:15.40 | *** join/#asterisk ElDios (n=ElDios@85-18-35-21.ip.fastwebnet.it) |
| 09:15.42 | C4away | the other is that the channel isn't up, RTP ports could be blocked by a firewall ... |
| 09:16.01 | C4away | asterisk wasn't compiled properly maybe |
| 09:16.18 | ratmandu | dushantch, make sure that the user you are starting asterisk from has the ability to write to that directory |
| 09:16.30 | neurosys | C4away: But the channel does proceed normally outside of playtones(). Playback and background sound fine, authenticate() works, etc... |
| 09:16.35 | dushantch | ratmandu: I'm trying as root :) |
| 09:17.03 | C4away | what tone are you trying to generate? |
| 09:17.22 | neurosys | ring |
| 09:17.26 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-244-17-140.dsl.hrlntx.swbell.net) |
| 09:17.40 | C4away | exten => 1234,1,Ringing() |
| 09:18.18 | C4away | http://www.voip-info.org/wiki-Asterisk+cmd+Ringing |
| 09:18.19 | dushantch | ratmandu: found it, the folder asterisk had wrong permissions |
| 09:18.27 | ratmandu | ah |
| 09:18.27 | dushantch | ratmandu: thanks |
| 09:18.31 | ratmandu | np |
| 09:18.35 | C4away | odd that playtones isn't working though, haven't seen that before |
| 09:19.06 | neurosys | C4away: What does asterisk use to generate those tones? |
| 09:19.25 | C4away | probably indications.conf |
| 09:19.50 | C4away | curious if Ringing() will work |
| 09:19.53 | neurosys | C4away: I'm sorry. I mean hardware wise |
| 09:20.12 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:20.52 | neurosys | C4away: does the system require some sort of DSP or anything? |
| 09:20.57 | C4away | depends on the channel |
| 09:20.59 | arnor001 | kaldemar: would you test for packet loss using the System command in the dialplan , and does the System command return a result from the Console? |
| 09:21.20 | C4away | if it is a zaptel/dhadi thing then yea, it generates it and uses the analag card to "play" it |
| 09:21.37 | C4away | if it is sip/iax2 etc then it just mixes it into the stream and generates the binary data in the audio packets |
| 09:22.02 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |
| 09:22.39 | arnor001 | ok, thanks i'll do some research |
| 09:23.16 | kaldemar | arnor001: first off, i wouldn't do it. but you can always write results to a file and read it to a variable with app ReadFile. |
| 09:25.46 | arnor001 | kaldemar: the reason i'm asking is, ocassionaly when we call through our voip provider, after 1 minute, there is one way audio, i looked at voip-info wiki, but i couldnt find a way to diagnose the problem |
| 09:25.57 | C4away | dushantch is asterisk running? |
| 09:26.31 | dushantch | C4away: 'm still having troubles :), it doesn't start but I get no messages |
| 09:26.42 | C4away | killall asterisk |
| 09:26.47 | C4away | asterisk -dvvvvvvvvvvvvvvvvvvv |
| 09:26.54 | C4away | wait |
| 09:27.01 | C4away | asterisk -dfvvvvvvvvvvvvv |
| 09:27.02 | arnor001 | asterisk -gvvvvvvvvvvvvvcd works nice |
| 09:27.13 | C4away | I can't remember the foreground command |
| 09:27.17 | C4away | i think it is df |
| 09:27.34 | C4away | debug foreground very very verbose |
| 09:27.49 | dushantch | Segmentation fault (core dumped) |
| 09:28.08 | C4away | nice |
| 09:28.47 | C4away | recompile asterisk? |
| 09:28.56 | arnor001 | i'm considering switching to asterisk 1.4 from 1.2.27, are there any advantages or would it be a waist of time? |
| 09:29.04 | C4away | well |
| 09:29.05 | dushantch | C4away: I'll try :) |
| 09:29.12 | C4away | arnor001 I'd switch to 1.6 |
| 09:29.26 | C4away | if you have to go through and update your dialplan might as well do it once |
| 09:29.29 | arnor001 | but thats still a development release, right? |
| 09:29.33 | C4away | no |
| 09:29.41 | C4away | it has been released production |
| 09:29.56 | C4away | we have it running on our primary call router 15,000+ calls per day |
| 09:30.08 | arnor001 | but isnt it good practise to wait a while until software has , most bugs ironed out? |
| 09:30.16 | C4away | heh sure maybe |
| 09:30.25 | C4away | I installed it when it was still 1.6-beta9 |
| 09:30.37 | C4away | then upgraded to the first release recently |
| 09:30.40 | C4away | no issues yet |
| 09:30.45 | arnor001 | well, i cant afford to have problems on a production server |
| 09:30.47 | C4away | then again it is just a call router, pretty basic config |
| 09:30.57 | C4away | nor can I |
| 09:31.10 | arnor001 | i'll give it a try thanks |
| 09:31.14 | C4away | this is a call router for a phone company, hundreds of customers |
| 09:31.27 | C4away | has been in production for a few months now |
| 09:31.41 | C4away | no lockups or crashes at all |
| 09:31.43 | dushantch | hmm on gentoo only 1.2.27 or so is marked stable :) |
| 09:31.53 | C4away | well |
| 09:31.56 | C4away | that's gentoo |
| 09:32.13 | arnor001 | well yes, we use gentoo |
| 09:32.20 | C4away | I don't use precompiled asterisk anyway |
| 09:32.34 | C4away | we compile it from source for our needs |
| 09:32.43 | dushantch | I'm on gentoo currently, but using voip overlay, in it 1.6 is available :) |
| 09:32.51 | ratmandu | can I use # key as an extension in the extensions.conf? |
| 09:32.57 | C4away | yea |
| 09:33.11 | C4away | you can use 1234567890#*ABCD |
| 09:33.15 | ratmandu | thanks |
| 09:33.16 | neurosys | for a newbie, would you recommend 1.4 or 1.6? |
| 09:33.16 | C4away | as dialable digits |
| 09:33.29 | C4away | also you can use a-zA-Z as extensions too |
| 09:34.07 | C4away | but they can't be dialed from a DTMF keypad (A-D are only on 16 digit DTMF keypads) ... they can be called from other places in the dialplan though |
| 09:34.28 | C4away | like exten => joe,1,Dial(SIP/1234) |
| 09:35.08 | C4away | and somewhere else put exten 3031234567,1,Goto(my-extensions,joe,1) |
| 09:35.55 | dushantch | C4away: recompiled without some options, it starts now |
| 09:36.17 | dushantch | C4away: on amd64, but it can't connect to mysql |
| 09:36.32 | C4away | do you have the mysql client installed when compiling? |
| 09:36.38 | C4away | did you use make menuconfig |
| 09:37.09 | C4away | and check to make sure that the cdr_mysql module was available? |
| 09:37.15 | dushantch | C4away: mysql is installed, hmm maybe asteris user wasn't created |
| 09:37.20 | C4away | maybe |
| 09:37.36 | C4away | the database stuff takes some beating of one's head against the keyboard to get working |
| 09:39.11 | dushantch | hrmpf when I go to localhost:8088/asterisk/static/config/cfgbasic.html i get: The requested URL was not found on this serve |
| 09:39.21 | dushantch | for asterisk gui 2.0 |
| 09:40.23 | C4away | never used it |
| 09:43.06 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 09:46.15 | kaldemar | dushantch: check prefix in http.conf |
| 09:48.43 | dushantch | kaldemar: thanks. It says that when nothing is enabled it's asterisk, but I had to write asterisk and it started :) |
| 09:51.52 | dushantch | I must say that gui works and it's very nice, I just have to fathom why asterisk starts with asterisk -dfvvvvvvvvvvvvv but fails with /etc/init.d/asterisk start |
| 09:57.57 | *** part/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 09:58.18 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 09:59.15 | fcois93 | I dont find 'setglobalvar' in asterisk 1.6 !!??? |
| 10:00.45 | fcois93 | I find 'Set(GLOBAL(name)=value)' is it correct ? |
| 10:01.21 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 10:04.16 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 10:06.49 | tompaw | is there a way to set up a fallback route in asterisk? that way I wouldn't need to configure load balancing with SER |
| 10:07.00 | kaldemar | fcois93: SetGlobalVar was deprecated in 1.4 and removed in 1.6. |
| 10:07.10 | tompaw | example: I have 2 gateways, each has n channels. I want to hande 2n channels on my asterisk. |
| 10:07.29 | tompaw | so 50% of the time, gatewayA will return 503 or something similar (all chanels busy) |
| 10:07.43 | tompaw | is there a way to configure asterisk in a way it uses gatewayB instead? |
| 10:07.54 | kaldemar | yes there is. |
| 10:08.38 | kaldemar | you can dial the first server and then look into DIALSTATUS variable to define if it's necessary to take another route. |
| 10:11.03 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 10:12.32 | *** join/#asterisk itguru (n=p@82.108.189.20) |
| 10:12.57 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:13.18 | dushantch | hrmpf, is there some guide on how to make an asterisk database in mysql, to get it working? I get: res_config_mysql.c:637 load_module: MySQL RealTime: Couldn't establish write connection: Access denied for user 'asterisk'@'localhost' (using password: YES) |
| 10:14.39 | joobie | dush, check your mysql user/pass details.. make sure it can connect ot mysql and write |
| 10:15.43 | dushantch | joobie: sorry for my noobness, but where/how? |
| 10:15.43 | tompaw | kaldemar: good idea, thx. |
| 10:20.35 | joobie | dushantch, mysql -u <user> -p <pass> |
| 10:20.49 | joobie | that is the syntax from cli to connect to mysql.. start by trying to connect iwth your asterisk user and pass.. see if it works |
| 10:20.54 | joobie | then: use <database> |
| 10:21.01 | joobie | to try to select ur db |
| 10:21.05 | joobie | then u need to insert into the db.. |
| 10:21.09 | joobie | insert a row that is.. |
| 10:21.21 | dushantch | joobie: looks like instalation didn't make asterisk database |
| 10:27.27 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 10:27.33 | *** join/#asterisk protocols (n=protocol@p5791FEF4.dip.t-dialin.net) |
| 10:29.23 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-85.moldtelecom.md) |
| 10:29.38 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 10:29.44 | kotique | hey guys. How do I get a variable that's set in SIP INVITE dialog ? |
| 10:29.49 | fcois93 | I have that error, " Call from 'ser_sei-out' to extension '0170720000' rejected because extension not found. " but the extension is in the extensions.conf ans I have exten => _X.,1?Answer() |
| 10:29.56 | fcois93 | have you an idea? |
| 10:30.25 | kotique | Via: SIP/2.0/UDP xxxx:5060;branch=z9hG4bK1scdcjptogg709ii8h1lo2cok1 |
| 10:30.25 | kotique | X-UUID: e29e0575e79c4126a0b1ad007e7a8adb |
| 10:30.25 | kotique | X-DID: 011xxx |
| 10:30.31 | kotique | I need X-DID |
| 10:31.12 | fcois93 | kotique: a SIPHeaders you mean? |
| 10:31.17 | kotique | yea |
| 10:31.18 | kaldemar | kotique: core show function SIP_HEADER |
| 10:31.30 | kotique | great |
| 10:31.39 | kaldemar | in the future, use pastebin. |
| 10:31.41 | fcois93 | kotique: ${SIP_HEADER(your_header)} |
| 10:32.02 | fcois93 | <PROTECTED> |
| 10:32.12 | fcois93 | have you an idea? I run asterisk 1.6 |
| 10:32.29 | *** join/#asterisk shinao1 (n=shinao1@41.222.209.142) |
| 10:32.45 | kaldemar | fcois93: the extension has to be in the right context |
| 10:33.29 | kaldemar | show your dialplan and configuration for ser_sei-out |
| 10:33.44 | fcois93 | yes I know I saw everythings but dont understand |
| 10:42.02 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:44.40 | *** join/#asterisk ThoMe (i=tm@81.92.168.148) |
| 10:44.41 | ThoMe | hello |
| 10:45.13 | ThoMe | have a old 1.2.13 asterisk |
| 10:45.20 | ThoMe | and in my dialplan a line: exten => _0.,n,GotoIf($["${get-persoenliche-absender-rufnummer}" > "0"]?set-persoenliche-absender-rufnummer:set-default-absender-rufnummer) |
| 10:45.46 | ThoMe | ups |
| 10:45.46 | ThoMe | no |
| 10:46.08 | ThoMe | done ;) |
| 10:46.28 | disposable | i have ast 1.6 with imap voicemail storage. is there a way to have a backup for voicemail storage when that fails? e.g.: local files |
| 10:46.40 | kaldemar | ThoMe: you have a bunch of security holes in your asterisk. |
| 10:49.05 | ThoMe | kaldemar: ajo? |
| 10:49.44 | tompaw | kaldemar: would you mind having a quick look at my dialplan? http://pastebin.com/d5f527a3c |
| 10:50.27 | tompaw | does this make any sense at all? |
| 10:50.44 | tompaw | (assuming that my gateways send CHANUNAVAIL when all channels are busy) |
| 10:51.43 | kaldemar | i guess it does |
| 10:51.51 | ThoMe | 11:49:05 < ThoMe> kaldemar: ajo? |
| 10:53.42 | *** join/#asterisk rdgr (n=rich@82.33.200.190) |
| 10:54.11 | kaldemar | ThoMe: http://www.asterisk.org/security |
| 10:54.24 | *** join/#asterisk psykx-out (n=max@uberpussy.net) |
| 10:54.30 | psykx-out | Hi guys |
| 10:54.58 | tompaw | lovely domain name. |
| 10:55.15 | ThoMe | kaldemar: bunch or brunch? ;) |
| 10:55.30 | ElDios | hey guys.. I've this option on my grandstream phones |
| 10:55.33 | psykx-out | it's a friends server we use it to lurk on irc |
| 10:55.36 | ElDios | "Custom ring tone 1, used if incoming caller ID is" |
| 10:55.42 | ElDios | in your opinion, does it mean the "caller" in the exact meaning of the word or it could be a misuse, meaning the extension which I'm receiving the call on? |
| 10:55.47 | ElDios | (I'm mad... you can freely tell me so... -_-' ... but I'm desperate) |
| 10:56.08 | psykx-out | ElDios: it's what ever you set callerid too in your asterisk set up |
| 10:56.18 | ElDios | ah |
| 10:56.47 | kaldemar | ThoMe: bunch. :) you should consider updating to 1.2.30.2 if you're about to continue with the 1.2 branch. |
| 10:56.50 | ElDios | so this could be used to achieve the distinctive ringtones based on the account I'm receiveing the call on.. right? |
| 10:57.07 | psykx-out | should be |
| 10:57.51 | ElDios | psykx-out can you please make an example of a callerid? is it bound to some syntax of could be free as "support_line", "sales_line", "personal_line" ? |
| 10:58.12 | ElDios | s/of could/or could/ |
| 10:58.24 | ElDios | ^_^ |
| 10:58.34 | ElDios | nice bot |
| 10:58.56 | ThoMe | kaldemar: hm. you, should. but the customer give me no money more :-( |
| 11:02.25 | tompaw | kaldemar: one problem with my macro. ${EXTEN} in the macro == s, not the dialed number :> |
| 11:02.35 | tompaw | kaldemar: any way to extract original EXTEN from ARG? |
| 11:02.40 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:02.52 | kaldemar | tompaw: ${MACRO_EXTEN} |
| 11:03.06 | kaldemar | or use an ARG |
| 11:03.55 | tompaw | I like the 1st one ;) |
| 11:04.02 | tompaw | (if it does what I think it does) |
| 11:05.11 | *** join/#asterisk mRCUTEO (n=info@118.101.178.78) |
| 11:10.26 | tompaw | kaldemar: it does indeed, thanks! |
| 11:14.42 | protocols | is anybody experiencing problems with analog faxing and asterisk 1.4 and would recommend 1.6 instead? |
| 11:15.17 | psykx-out | ElDios: It can be anything |
| 11:15.51 | ElDios | thnx psykx-out |
| 11:16.41 | psykx-out | has anybody tunneld IAX over ssh or vpn (anything tcp) |
| 11:17.09 | psykx-out | I'm having packet loss but I have huge amounts of bandwidth to play with |
| 11:17.10 | *** join/#asterisk MRCUTEO (n=info@118.101.178.78) |
| 11:19.52 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:21.07 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 11:24.39 | mRCUTEO | anyone knows which GUI is easier to install for asterisk? |
| 11:32.19 | Maliuta | vim |
| 11:32.27 | Maliuta | no GUI support here |
| 11:33.24 | mRCUTEO | oh |
| 11:33.27 | mRCUTEO | sorry |
| 11:36.21 | tompaw | is there any kind on table describing which SIP error codes result in which DIALSTATUS values? |
| 11:38.16 | tompaw | or any way to debug DIALSTATUS? (like displaying its value in sip debug) |
| 11:44.22 | psykx-out | what does the console show? |
| 11:45.29 | tompaw | well, it shows sip headers |
| 11:46.56 | tompaw | strange |
| 11:47.03 | tompaw | exten => s,5,Set(IDX = $[${IDX} + 1]) ; << this doesn't seem to work |
| 11:48.59 | kaldemar | tompaw: you can NoOp variables |
| 11:49.26 | tompaw | kaldemar: thanks :) |
| 11:49.36 | tompaw | got it |
| 11:49.47 | tompaw | now wht the hell doesn't the line above increment IDX by 1? |
| 11:51.20 | tompaw | hm.. I tried changing Set to SetVar and I got: |
| 11:51.21 | tompaw | No application 'SetVar' for extension (macro-cidial, s, 5) |
| 11:51.25 | kaldemar | remove the spaces around = |
| 11:51.38 | tompaw | ok |
| 11:52.36 | tompaw | oh my god, it worked!! |
| 11:52.39 | tompaw | =) |
| 11:52.56 | tompaw | the whole thing works now, number portability lookup + failover route |
| 11:54.28 | *** join/#asterisk sysadmin-lb22 (n=asdf@87.236.144.35) |
| 11:54.39 | sysadmin-lb22 | hi |
| 11:55.13 | sysadmin-lb22 | I am trying to match extensions in extensions.conf using _X ..etc..however I need to match alphabetic names is this possible |
| 11:55.14 | sysadmin-lb22 | ? |
| 11:58.04 | kaldemar | yes |
| 11:58.25 | sysadmin-lb22 | kaldemar, can you please give an example..I would like to match John, Alex etc |
| 11:58.42 | kaldemar | exten => John,1,... |
| 11:58.52 | tompaw | ;) |
| 11:59.01 | sysadmin-lb22 | kaldemar, :) what about dynamic mathes |
| 11:59.04 | Maliuta | sysadmin-lb22: have you read the book? |
| 11:59.06 | sysadmin-lb22 | matches ** |
| 11:59.11 | Maliuta | ~book |
| 11:59.12 | jbot | [book] probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 11:59.26 | Maliuta | it's all there |
| 11:59.34 | kaldemar | read up on extension patterns |
| 12:00.13 | sysadmin-lb22 | well I did not read the book on this but I did check some links and from what I have found I can only match 1-9 0-9 etc |
| 12:00.40 | kaldemar | you found wrong. |
| 12:02.04 | sysadmin-lb22 | _. ? |
| 12:02.08 | kaldemar | it sure would be nice to be able to use regexp's. |
| 12:02.36 | kaldemar | don't use _., it will match to special extensions such as t and h also. |
| 12:03.26 | sysadmin-lb22 | http://tfot.leifmadsen.com/ch05s03s06.html |
| 12:03.34 | sysadmin-lb22 | This is where I should be looking right / |
| 12:03.43 | *** join/#asterisk tanacsdavid (n=david@office.axpnet.com) |
| 12:05.51 | kaldemar | yes. it doesn't explicitly say that you can use alphabets inside [] though. |
| 12:06.16 | *** join/#asterisk galeras (n=galeras@166.210.26.12) |
| 12:07.11 | Maliuta | sysadmin-lb22: I see the problem in your statement: "well I did not read the book ..."!!!! |
| 12:07.18 | Maliuta | RTFB!!! |
| 12:07.33 | *** join/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 12:07.59 | sysadmin-lb22 | Maliuta, thanks for being so nice |
| 12:08.19 | sysadmin-lb22 | Maliuta, no offense but the matching section is quite small and I just read through it using the html version of hte book |
| 12:08.34 | Maliuta | read docs before asking for help. It's simple |
| 12:08.54 | sysadmin-lb22 | Maliuta, and the matching for letters can only be done using _. |
| 12:08.55 | Maliuta | read the whole thing |
| 12:08.59 | Maliuta | there is more in there |
| 12:09.36 | sysadmin-lb22 | Maliuta, thanks for nothing |
| 12:09.36 | sysadmin-lb22 | bye |
| 12:09.36 | galeras | Hello, please tellme how can bypass dahdi driver downloading when running "make install" (server has not internet access) |
| 12:10.01 | *** part/#asterisk sysadmin-lb22 (n=asdf@87.236.144.35) |
| 12:10.35 | kaldemar | i guess he didn't bother to understand what i told him. |
| 12:11.14 | Maliuta | people really should read docs before asking basic questions |
| 12:11.23 | Maliuta | that's why they're there |
| 12:15.15 | *** part/#asterisk galeras (n=galeras@166.210.26.12) |
| 12:16.23 | *** join/#asterisk DarkRift (n=dark@65.92.170.221) |
| 12:18.59 | neurosys | Wellm atleast Im not the only newb who gets beat up around here ;) |
| 12:22.44 | *** join/#asterisk rcy`` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 12:28.10 | Yourname` | Damnit, my ebay posting sold but the guy paid me in cash and now I can't find "cash payment" somewhere |
| 12:31.08 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:40.08 | *** join/#asterisk rdgr (n=rich@82-33-200-190.cable.ubr01.aztw.blueyonder.co.uk) |
| 12:41.22 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-0074.yubc.net) |
| 12:41.29 | *** part/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 12:42.53 | dushantch | Hi, can only start asterisk 1.6 as root as it creates files in /var/run/asterisk as root, so that they can't be accesed as user asterisk. Is there some solution? |
| 12:44.01 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:59.30 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:02.53 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 13:10.14 | *** join/#asterisk stix_ (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 13:11.13 | stix_ | if I create an ivr and have a exten => t,1,Goto(s,1) can I limit the number of times an timeout is allowed? |
| 13:11.17 | beek | dushantch: Did you use: --localstatedir=$HOME/asterisk-bin/var and --sysconfdir=$HOME/asterisk-bin/etc when you ran ./configure? |
| 13:11.18 | stix_ | -n |
| 13:12.45 | *** join/#asterisk patrick-- (n=patrick@gate.devnull.biz) |
| 13:13.23 | patrick-- | Hey all, im planning to build an asterisk server for Home use with my 2 ISDN PSTN Lines. Would a simple AVM Fritz Card be sufficient? |
| 13:14.08 | *** part/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 13:14.37 | tzafrir_laptop | patrick--, yes |
| 13:14.50 | patrick-- | for the outgoing/incoming connection? |
| 13:15.07 | patrick-- | im planning on routing the ISDN calls via SIP Lines to VoIP phones |
| 13:23.12 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:23.12 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:24.39 | patrick-- | tzafrir_laptop: would a AVM card be sufficient? |
| 13:25.10 | tzafrir_laptop | yes (for one port) |
| 13:25.49 | patrick-- | doesnt the AVM Fritzcard support 2 B Channels? |
| 13:28.54 | tzafrir_laptop | each ISDN (BRI) wire can carry 2 B channels |
| 13:29.06 | tzafrir_laptop | that is: 2 calls |
| 13:29.23 | patrick-- | yupp i can only have 2 concurrenty calls anyway |
| 13:30.31 | patrick-- | im looking for a small setup.. possibly miniATX board, small case, low noise.. |
| 13:30.37 | patrick-- | any suggestions on that? :D |
| 13:30.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:30.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:31.18 | mort_gib | patrick--: Soekris 5501 |
| 13:31.57 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 13:32.59 | *** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-013-132.mycingular.net) |
| 13:33.05 | patrick-- | too small :D |
| 13:33.13 | patrick-- | i need PCI slots |
| 13:33.16 | mort_gib | -Are you 100% sure |
| 13:33.26 | mort_gib | It has one PCI slot |
| 13:33.52 | mort_gib | Enough for one card like Sangoma A500 |
| 13:34.25 | lmadsen | mort_gib: you can use a PCI card as long as it isn't full sized (I forget what they call them... but the low rise cards) |
| 13:34.35 | patrick-- | low profile |
| 13:34.40 | lmadsen | thats the word |
| 13:34.45 | patrick-- | well |
| 13:34.46 | mort_gib | That depends on the case |
| 13:34.54 | lmadsen | well, you can modify the case :) |
| 13:35.01 | mort_gib | I use them with the Rackmount case from Wim |
| 13:35.03 | patrick-- | it doesnt have to be that small.. i'd want to use it for other purposes too.. |
| 13:35.06 | lmadsen | or just put it in a static bag :D |
| 13:35.16 | mort_gib | Yeah |
| 13:35.20 | patrick-- | so 2 PCI Slots and room for a HD would be nice |
| 13:35.20 | lmadsen | you're probably better off with a micro-pc |
| 13:35.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:35.33 | mort_gib | 5501 comes with SATA support |
| 13:35.40 | patrick-- | Planning on combining an Asterisk, Fileserver and VDR |
| 13:35.40 | lmadsen | patrick--: that definitely doesn't match the soekris anymore |
| 13:35.57 | patrick-- | (vdr optional) |
| 13:36.01 | lmadsen | yuck... hope that fileserver isn't being used much |
| 13:36.06 | patrick-- | nah |
| 13:36.06 | mort_gib | True, Soekris is too small |
| 13:36.07 | patrick-- | "home" |
| 13:36.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:36.12 | lmadsen | ah |
| 13:36.15 | patrick-- | just for some storage |
| 13:36.38 | patrick-- | but im stuck with a FritzBox and im sick of not beeing in control of my calls properly.. |
| 13:36.42 | patrick-- | so used to my asterisk at work... |
| 13:37.00 | patrick-- | i even get jabber notifications in absence, etc :D |
| 13:37.11 | patrick-- | fritzbox sucks cock compared to all the ast features |
| 13:41.52 | ElDios | lil' trick (if possible) what is the registration string to have multiple numbers on one account? |
| 13:41.58 | *** join/#asterisk spiekey (n=mario@projekte.imos.net) |
| 13:42.00 | spiekey | hello! |
| 13:42.02 | ElDios | user:pass@sip.server:port/??? |
| 13:42.13 | ElDios | what do I have to put as ??? |
| 13:42.23 | spiekey | i am playign around with AsteriskNOW. I added a user but i am not able to edit it. |
| 13:42.36 | spiekey | when i select that user, then all the users are selected :-/ |
| 13:42.51 | spiekey | is this meant to be like this? |
| 13:49.33 | *** join/#asterisk etfonhomey_ (n=chatzill@74-143-196-254.static.insightbb.com) |
| 13:50.48 | *** join/#asterisk espent (n=espent@totem.fix.no) |
| 13:53.48 | lmadsen | ElDios: don't put anything after the / and the provider should send it generically. Whatever is after the / is what extension you request calls to come in as |
| 13:54.04 | lmadsen | spiekey: might want to check #asterisk-gui, not many in here use it |
| 13:54.20 | lmadsen | most people just use the CLI here, which is also why FreePBX, trixbox, etc... are not supported in this room |
| 13:55.13 | spiekey | lmadsen: thanks! |
| 13:56.57 | chazz | spiekey: that sounds like the issue with the old GUI (which would be in AsteriskNOW prior to 1.5) and firefox 3 and IE7 |
| 13:57.05 | chazz | use firefox 2.X or IE6 |
| 13:57.53 | ElDios | lmadsen nothing |
| 13:58.02 | ElDios | it says Auth send but it doesn't get registered |
| 13:58.19 | lmadsen | ElDios: then something else is wrong with your authorization because you don't need that last part usually |
| 13:58.37 | lmadsen | if you do, then that is very odd... but that would normally be the extension you would request incoming calls to come in on |
| 13:58.38 | ElDios | I will double check it =) |
| 13:59.47 | ElDios | you were right... typo :P |
| 13:59.52 | ElDios | sorry XD |
| 14:00.08 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:00.25 | ElDios | thnx lmadsen |
| 14:00.30 | ElDios | it's workin now |
| 14:01.46 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 14:02.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:03.57 | fcois93 | I have a problem when I use a global var in asterisk1.6 |
| 14:04.13 | fcois93 | I use ${codec} but nothing |
| 14:06.05 | fcois93 | after having done Set(GLOBAL(codec)=g711}) |
| 14:06.13 | fcois93 | have you an idea? |
| 14:06.33 | *** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 14:06.49 | Katty | morning |
| 14:08.38 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 14:10.06 | *** join/#asterisk LND (n=lee@nat66.mia.three.co.uk) |
| 14:10.31 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:11.30 | *** join/#asterisk telnettech (i=bsimpson@gw.percipia.com) |
| 14:15.04 | *** join/#asterisk Breal (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 14:15.35 | Breal | When making calls across a network ie 192.168.1.2 -> .3 it sounds metalic and garbled. Is there a setting that I can adjust to make this better? |
| 14:16.32 | ElDios | change codec Breal ? |
| 14:16.42 | SibRphrek | Breal: you are probably using the wrong codec |
| 14:16.52 | SibRphrek | darn ElDios beat me to it! |
| 14:17.14 | Breal | Oh ok, I wasn't sure if it was an error with the jitter settings or something.l |
| 14:17.17 | Breal | What codec should I use? |
| 14:17.33 | SibRphrek | which one are you using now? |
| 14:17.52 | SibRphrek | and where are you located? |
| 14:17.59 | SibRphrek | well |
| 14:18.05 | SibRphrek | i guess that doesn't really matter since it's internal |
| 14:18.06 | Breal | USA, NC |
| 14:18.21 | Breal | I am not sure since I do not see one specified in sip.conf |
| 14:18.28 | SibRphrek | most people use G.711 |
| 14:18.38 | SibRphrek | Breal: check this out |
| 14:18.38 | SibRphrek | http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 14:18.51 | SibRphrek | it tells you the commands to drop into the CLI |
| 14:18.57 | Breal | Thanks. Are all codecs supported by all hardphones? |
| 14:19.04 | ElDios | :P |
| 14:19.06 | SibRphrek | not that i remember |
| 14:19.09 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 14:19.13 | SibRphrek | it's been a long time since i've worked with this |
| 14:19.18 | SibRphrek | hence why i'm back in this chan |
| 14:19.36 | ElDios | wins |
| 14:19.45 | SibRphrek | but any device you buy tells you what codecs are supported |
| 14:19.51 | SibRphrek | ElDios 1 SibRphrek 0 :( |
| 14:19.55 | SibRphrek | anyway gotta jet |
| 14:19.56 | ElDios | XD |
| 14:19.57 | SibRphrek | will be back later |
| 14:20.00 | [TK]D-Fender | brNo |
| 14:20.04 | SibRphrek | ElDios: you around a lot? |
| 14:20.08 | ElDios | nope |
| 14:20.12 | ElDios | few days |
| 14:20.14 | SibRphrek | haha |
| 14:20.16 | ElDios | you? |
| 14:20.22 | SibRphrek | i usually idle |
| 14:20.34 | ElDios | eheh.. cya later then |
| 14:20.37 | SibRphrek | but i'm building a new asterisk system so i'll be in here asking questions |
| 14:20.39 | SibRphrek | later guys |
| 14:20.43 | SibRphrek | good luck Breal |
| 14:22.10 | Breal | Do I specify the codec in the [general] block? Or in the [user] block? |
| 14:22.19 | lmadsen | Breal: either |
| 14:22.29 | lmadsen | Breal: in the general block is the default, and the user block is the override |
| 14:22.41 | Breal | k |
| 14:22.43 | [TK]D-Fender | Breal: Apply the larger set to [general] and you should generally only have *1* codec per device |
| 14:22.49 | Breal | I'd rather make it default |
| 14:22.51 | Breal | yeah, thansk |
| 14:23.00 | [TK]D-Fender | Breal: "disallow=all" , "allow=alaw" for example |
| 14:23.25 | [TK]D-Fender | Breal: You should NTO be leaving it to default. Global choices lead to global screwups |
| 14:23.27 | [TK]D-Fender | NOT* |
| 14:24.30 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 14:25.22 | Karlitoo | hey guys I just installed addons for asterisk and tryed to go trough a ooh323 channel driver and I get an error http://pastebin.com/d6bf7ca32 |
| 14:25.27 | Karlitoo | any ideas |
| 14:26.13 | [TK]D-Fender | Karlitoo: Yeah, count your "o" 's |
| 14:26.48 | Karlitoo | yeah I know that in the extensions.conf it's OH323 not OOH323 |
| 14:26.54 | Karlitoo | and I put OH323 |
| 14:26.55 | Breal | ah, much better |
| 14:27.12 | Breal | Thanks guys, that made a world of differenced. |
| 14:28.10 | Karlitoo | <PROTECTED> |
| 14:28.10 | Karlitoo | <PROTECTED> |
| 14:28.10 | Karlitoo | <PROTECTED> |
| 14:28.49 | [TK]D-Fender | Karlitoo: FAIL :) |
| 14:29.07 | [TK]D-Fender | Karlitoo: its in the docs, its in your module load. Now try following it :) |
| 14:30.17 | Karlitoo | ... |
| 14:30.39 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 14:31.01 | Karlitoo | I'm sorry I'm as you can see kind of a newb at asterisk, so I have in the documentation on how to load a module |
| 14:31.08 | Karlitoo | is that what you wanted to tell me |
| 14:31.09 | Karlitoo | ? |
| 14:31.34 | [TK]D-Fender | Karlitoo: I can see you missed the big print somehow. Anyway, there you have it. Go play :) |
| 14:32.12 | neurosys | [TK]D-Fender: You can be very discouraging. :( |
| 14:32.21 | Karlitoo | :) yeah |
| 14:32.27 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 14:32.57 | *** join/#asterisk magumbade (n=magumbad@p5497FD1B.dip.t-dialin.net) |
| 14:33.21 | *** join/#asterisk magumbade (n=magumbad@p5497FD1B.dip.t-dialin.net) |
| 14:33.32 | Karlitoo | I know that the best way to learn something is just reading the docs and learn it by trial and error unfortunatly I'm at work and don't have much time to study and concentrate on 1 thing |
| 14:33.33 | [TK]D-Fender | Karlitoo: Useful tip : When things don't work, start by assuming that absolutely everything is wrong and trace its from the very start. The stuff we assume is correct is often not so and leads to digging around for nothing. |
| 14:34.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:35.12 | Karlitoo | ty [TK]D-Fender |
| 14:35.14 | Karlitoo | :) |
| 14:36.54 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:41.06 | jaytee | seems to be alot of people in here with an identity crisis |
| 14:41.13 | mark_csi | hi all, I've a problem with incoming pstn lines not hanging up on my asterisk box. I've checked that the dialplan is correct, anyone think of anything else? |
| 14:41.19 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 14:41.21 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 14:41.23 | *** join/#asterisk deStone_ (n=deStone@unaffiliated/destone) |
| 14:41.52 | neurosys | I never know who I am. Hence my lifelong handle :P |
| 14:42.33 | deStone_ | i'm wanting to setup an IVR --- does anyone have a moment to talk to me about steps to doing this? I currently have a host-based VOIP phone system (broadsoft) --- how can i utilize asterisk to help accomplish this? |
| 14:43.34 | Katty | [TK]D-Fender: bork, bork bork bork. |
| 14:44.10 | [TK]D-Fender | Katty: MEEP! |
| 14:44.22 | Katty | [TK]D-Fender: La la la LA! |
| 14:44.42 | [TK]D-Fender | deStone_: read up : |
| 14:44.42 | [TK]D-Fender | ~book |
| 14:44.43 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:44.45 | [TK]D-Fender | ^^^ |
| 14:44.54 | [TK]D-Fender | deStone_: Much to learn about configuring * |
| 14:47.50 | jaytee | Katty, is your webblog down? I had the link to the blacklisting stuff you'd posted last week and when I went there yesterday it was unavailable. |
| 14:48.41 | deStone_ | TKD: thanks |
| 14:48.46 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 14:52.34 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:53.57 | *** join/#asterisk websae (n=websaene@97-88-245-66.static.mdsn.wi.charter.com) |
| 14:56.01 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 14:56.13 | casix | hello |
| 14:56.51 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 15:00.13 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:03.52 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 15:06.17 | Katty | looks |
| 15:06.24 | Katty | looks like dydns hasn't updated. |
| 15:07.21 | Katty | jaytee: what info were you looking for? |
| 15:07.26 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:07.56 | Breal | What is no route to host on the CLI? Is that an improper user? |
| 15:07.56 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 15:08.03 | jaytee | I wanted to read your piece on blacklisting. You posted it while I was in class so I didn't have time to look at it in depth. |
| 15:10.16 | Karlitoo | [TK]D-Fender: how do I uninstall a module do I just remove it from the modlue dir or is there another method |
| 15:10.36 | jaytee | modules.conf |
| 15:10.37 | [TK]D-Fender | Karlitoo: Which? |
| 15:10.55 | Karlitoo | ooh323 |
| 15:11.01 | [TK]D-Fender | Breal: * has no idea how to contact them |
| 15:11.10 | Karlitoo | I need to remove ooh323 to put in oh323 |
| 15:11.28 | Karlitoo | cause I found out that ooh323 gives these error and that oh323 works |
| 15:11.35 | [TK]D-Fender | Breal: Can be numerous things. DNS error, routing error, calling a phone that needs to and has not regisitered |
| 15:11.56 | tzafrir_laptop | Karlitoo, oh323 is alive? |
| 15:12.01 | telnettech | jaytee: how did you do on your test |
| 15:12.05 | [TK]D-Fender | Karlitoo: I would do "noload => chan_oof323.so" in modules.conf |
| 15:12.37 | *** join/#asterisk mog (n=mog@nat/digium/x-8c04ba92f360eaf0) |
| 15:12.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:12.54 | jaytee | telnettech, hi!!! welcome to IRC. I passed the written. |
| 15:12.57 | tzafrir_laptop | Karlitoo, the in-tree h323 is maintained, though |
| 15:13.08 | telnettech | jaytee: congrats |
| 15:13.15 | jaytee | my score has been classified by the Department of Homeland Security :-) |
| 15:13.38 | jaytee | now I just have to retake the practical within a year. |
| 15:13.46 | [TK]D-Fender | MOG! Half-man, half-dog! He's his own best friend! |
| 15:13.46 | telnettech | jaytee: so you barely passed :) |
| 15:13.55 | mog | heh |
| 15:13.55 | jaytee | haha I got a 79 |
| 15:13.59 | [TK]D-Fender | jaytee: "Not a threat"? ;) |
| 15:14.05 | Maliuta | [TK]D-Fender: and he can lick his own balls |
| 15:14.13 | Bad_Robot- | lol |
| 15:14.22 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 15:14.30 | Karlitoo | tzafrir_laptop: you mean the one that comes with asterisk |
| 15:14.47 | Karlitoo | ? |
| 15:15.22 | tzafrir_laptop | that's h323 |
| 15:15.29 | jaytee | hey, I don't know what the big deal is. If I get finally get the dCAP it'll just drum up more consulting gigs for other Asterisk consultants to fix whatever I screw up :-) |
| 15:15.55 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-f619f6f7b37feb65) |
| 15:15.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.09 | [TK]D-Fender | tags-in jaytee |
| 15:16.15 | jaytee | hehe |
| 15:16.39 | [TK]D-Fender | jaytee: Thats a lot of my business... cleaning up after the previous incompetant consultant :) |
| 15:17.02 | jaytee | [TK]D-Fender, see! I'm tryin to help your bottom line here :-) |
| 15:17.59 | jaytee | plus I figure if what I do doesn't create a gig for you or someone else it might at least drive them to buy a copy of the book to help out Leif and Jared. |
| 15:19.50 | Katty | hugs jaytee |
| 15:20.23 | Bad_Robot- | [TK]D-Fender that's what the next guy is going to say lol |
| 15:20.37 | [TK]D-Fender | jaytee: If they got a nickel for every copy of the book we've sold for them... that'd be $.04 more each than they get now! Margins = suck! |
| 15:20.53 | [TK]D-Fender | Bad_Robot-: Lol... my clients are HAPPY :) |
| 15:21.00 | Bad_Robot- | :) |
| 15:22.06 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 15:25.01 | Breal | Is there a way to simplify this dialplan? http://pastebin.ca/1261427 |
| 15:25.07 | Breal | Like using regex or something else? |
| 15:26.03 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 15:26.05 | *** join/#asterisk Abydos313 (i=talkradi@linuxgeneration.net) |
| 15:26.58 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:27.30 | [TK]D-Fender | Breal: Not much. there are mocaros, but that wouldnot actually save yo any lines in THAT case |
| 15:27.33 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:27.36 | [TK]D-Fender | macros* |
| 15:28.17 | *** part/#asterisk am88b (i=siim@uba.linux.ee) |
| 15:31.47 | [TK]D-Fender | Breal: http://pastebin.ca/1261432 |
| 15:32.34 | [TK]D-Fender | Breal: And there isn't enough in your dialplan to run a pattern off of. your SIP device names don't match the dialplan extension so you have nothing really viable to run a variable exten off of. |
| 15:36.12 | jaytee | sorry, had to step out for a few |
| 15:36.21 | jaytee | hugs Katty back :-) |
| 15:37.07 | casix | Breal: may be you can store in db the relation of numbers - sip_user and then you just need to make a consult to the db |
| 15:37.50 | [TK]D-Fender | casix: because for 3 extens... AstDB is SO worth it... |
| 15:38.06 | [TK]D-Fender | casix: think of the pain if anything changes. |
| 15:38.14 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:38.19 | casix | yes yes for 3 yes, I mean if he have some more.... |
| 15:38.22 | [TK]D-Fender | casix: 1 line per exten to call a device in a generic way is GOOD |
| 15:38.33 | [TK]D-Fender | casix: maintaining AstDB is a PITA |
| 15:38.57 | ElDios | how do I match multiple numbers in DID? |
| 15:39.02 | casix | hehehe |
| 15:39.05 | ElDios | I need to match |
| 15:39.16 | ElDios | 02615345 [12345] |
| 15:39.25 | ElDios | what is the exact string that I need to put in the DID? |
| 15:40.55 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ed9cab03995b843d) |
| 15:40.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:41.04 | [TK]D-Fender | ElDios: What on earth does "matching a DID" mean? |
| 15:41.20 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:41.37 | Karlitoo | I need a lil help :) http://pastebin.com/d2567c1d9 |
| 15:41.56 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 15:43.26 | ElDios | [TK]D-Fender =) I mean that when someone call a certain number, I redirect it to the right extension depending on which number is |
| 15:44.05 | [TK]D-Fender | ElDios: What is the exact range? |
| 15:44.21 | ElDios | 026153451 026153452 026153453 026153454 026153455 |
| 15:44.33 | ElDios | 02615345 [12345] |
| 15:44.55 | ElDios | I have to slip it in two different groups |
| 15:44.58 | ElDios | 123 and 45 |
| 15:45.00 | [TK]D-Fender | ElDios: exten => _0261534[12345],1,Blah() |
| 15:45.07 | ElDios | oke |
| 15:45.08 | ElDios | aaaah |
| 15:45.10 | [TK]D-Fender | ElDios: exten => _0261534[123],1,Blah() |
| 15:45.12 | ElDios | could be the _ in front |
| 15:45.14 | [TK]D-Fender | ElDios: exten => _0261534[45],1,Blah() |
| 15:45.23 | ElDios | yeah. I didn't put the _ ahead |
| 15:45.33 | [TK]D-Fender | ElDios: Cardinal error |
| 15:45.40 | ElDios | ;) thnx a lot |
| 15:45.47 | ElDios | sorry for the awful question :P |
| 15:47.39 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 15:47.48 | Breal | Oho k, all of our extensions will begin w/ a 4... else, use a new dialplan |
| 15:48.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:49.01 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:50.50 | Breal | or, if the prefix is a 4, dial that # as a local ext... else if its a 9 use an outside line |
| 15:50.58 | ElDios | is it possible to restrict a particoular outbound route so that a specific extension use it when dialing? |
| 15:51.37 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 15:53.37 | [TK]D-Fender | Breal: No such thing as a "local extension" really. Every extension is jsut a number dialined in the dialplan. the fac that different patterns and more fiexed number do different things really isn't the matter. |
| 15:53.52 | [TK]D-Fender | ElDios: thats what CONTEXTS are for |
| 15:54.09 | [TK]D-Fender | ElDios: They separate what can and cannot dial |
| 15:55.12 | ElDios | thnx again [TK]D-Fender |
| 15:57.06 | casix | I have a problem comunicating users of two asterisks. I have created a sip configuration (http://pastebin.com/m50e941cc) but when a user of servidorA calls a user of ServidorB it says to me: chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from "user <sip:asdfa@asd>" |
| 15:57.13 | [TK]D-Fender | ElDios: You need to really sit down and read chapter 5 of the book. |
| 15:57.31 | casix | if I force the username with fromuser then I lost the original callerid |
| 15:57.41 | casix | how can I make it work? |
| 15:58.29 | ElDios | what book [TK]D-Fender ? |
| 15:58.49 | [TK]D-Fender | ~book |
| 15:58.50 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 15:58.53 | [TK]D-Fender | THE book. |
| 15:59.11 | *** join/#asterisk Segnale007 (n=Pietro@host21-255-dynamic.7-87-r.retail.telecomitalia.it) |
| 16:00.00 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 16:00.10 | ElDios | :) |
| 16:00.12 | bearded_blitz | should create a script that uses the amazon.com API's to restrict additional downloads beyond the first download without a review being written on amazon.com |
| 16:00.18 | ElDios | thnx for the 4th or 5th time |
| 16:00.48 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:03.38 | *** join/#asterisk CrazyTux (n=brandon@user-vcaurjr.dsl.mindspring.com) |
| 16:04.19 | [TK]D-Fender | bearded_blitz: /me writes "First downlaod failed and now I have to COMMENT on it? This book had better succed in downloading this time and be AWESOME or I'll axe-murder the authors!" |
| 16:04.33 | bearded_blitz | lol |
| 16:04.35 | bearded_blitz | that'd suck |
| 16:04.45 | [TK]D-Fender | :p |
| 16:06.17 | neurosys | Ok, I hope that I'm well prepared this time to ask my question. I am using a les.net DID to connect to my asterisk box, then trying to dial out and connect through another ITSP, quivoice. I am succeffuly regged to quivoice and when i connect thru a softphone, i can make calls from quivoice just fine. But when I try to dial out, i get an error. Here is my PasteBin with error, SIP and EXTENSIONS. http://pastebin.com/d793a9cd1 |
| 16:08.28 | [TK]D-Fender | neurosys: SIP/0013053315558@sip.quivoice.it <-- you have no peer to auth this call. I'd venture a guess that they don't like that |
| 16:08.46 | [TK]D-Fender | neurosys: Registering does NOT auth your calls. |
| 16:09.04 | bearded_blitz | registering just tells the other end where to send calls TO YOU when someone dials a DID |
| 16:09.17 | [TK]D-Fender | ~sipregister |
| 16:09.18 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 16:09.33 | neurosys | [TK]D-Fender: OHHHHH. So i need to enter a peer for quivoice into sip.conf with the proper credz |
| 16:09.46 | [TK]D-Fender | neurosys: Yes |
| 16:10.10 | neurosys | [TK]D-Fender: Thanks. Im on it :) |
| 16:10.15 | [TK]D-Fender | neurosys: When you set up a softphone is normally uses 1 set of credentials for both registering AND authing calls it palces via them |
| 16:10.57 | *** part/#asterisk bearded_blitz (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:11.26 | neurosys | [TK]D-Fender: The quivoice context will have to also be lesnet-incoming (or use an include) for the dial to work, correct? |
| 16:12.08 | [TK]D-Fender | neurosys: CONTEXT? its a peer... the context is for receiving calls from them and you can do whatever you feel like for this. |
| 16:12.42 | [TK]D-Fender | neurosys: And "context" has nothing to do with calling them. |
| 16:13.35 | neurosys | [TK]D-Fender: Guess i misunderstood that section of O'Reilly's asterisk :P Thanks :) |
| 16:14.53 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-37-217.w86-215.abo.wanadoo.fr) |
| 16:15.57 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:16.55 | Un1x | [TK]D-Fender could someone help me with music on hold please? |
| 16:17.22 | [TK]D-Fender | Un1x: try not to target people for support and rather jsut ask out into the channel. |
| 16:17.45 | [TK]D-Fender | Un1x: I'd rather not have to turn down the tons of stuff I'd rather not deal with on a 1-to-1 basis... |
| 16:18.10 | casix | anyone can help me with this sip trunk between two *. the configuration is here http://pastebin.com/m50e941cc but when a user of A pbx calls a user of B pbx, B pbx says that chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from "user <sip:asdfa@asd>", if I set a fromuser than the call goes ok but the callerid of the call is not the original extension is the fromuser of sip.conf. How can keep the callerid? |
| 16:18.16 | Un1x | Well, i was wondering this is my first time using musiconhold and was wondering if someone could just briefly, explain it i'm using Asterisk 1.4.22 with dahdi and a TDM400P |
| 16:18.45 | [TK]D-Fender | Un1x: Show us the failure. |
| 16:22.13 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 16:23.42 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 16:24.13 | rene- | hello, quick question guys, can anyone with an aastra phone could tell me what is the country of manufacture of it? |
| 16:24.27 | rene- | it is usually in a label in the bottom |
| 16:24.57 | [TK]D-Fender | rene-: China |
| 16:25.16 | rene- | thanks D-Fender |
| 16:25.28 | rene- | everything is made in china these days |
| 16:26.21 | Un1x | Sorry, im not getting how do i put it into extensions.conf so when i press the hold button it invokes it in the dialplan? |
| 16:26.22 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 16:28.29 | coppice | my kids were made in Hong Kong, but that's becoming sooooo old school now |
| 16:29.26 | [TK]D-Fender | un1MoH has nothing to do with the dialplan <- |
| 16:29.30 | [TK]D-Fender | Un1x: MoH has nothing to do with the dialplan <- |
| 16:29.45 | [TK]D-Fender | Un1x: It gets invoked by your DEVICE when it tells * to hold the call |
| 16:30.02 | Un1x | Ya, i just read that on voip-info it states, its only needed in dialplan if i was letting them listen to musiconhold rather then the plain old ringing |
| 16:30.24 | Un1x | but i cant find anythong on voip-info on how to go on about setting it up in 1.4 like such as adding the module to autoload |
| 16:30.30 | Un1x | aswell as configuration help |
| 16:30.49 | [TK]D-Fender | Un1x: MoH instead of ringing : "core show application dial" |
| 16:31.03 | Un1x | no i dont want to do MOH instead of ringing |
| 16:31.09 | Un1x | i only want them to hear music when i put them on hold |
| 16:31.20 | [TK]D-Fender | Un1x: then the sample config works, read it. |
| 16:31.53 | *** join/#asterisk CrazyTux (n=brandon@user-vcaurjr.dsl.mindspring.com) |
| 16:31.58 | Un1x | so i dont need to edit?, just load the module pretty much.. |
| 16:32.28 | [TK]D-Fender | Un1x: autoload should already be loading it |
| 16:33.08 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:33.49 | bijit | where can I change that after leaving a message on voicemail to break out to operator? |
| 16:33.52 | Un1x | well howcome music isn't playing then.. |
| 16:35.24 | bijit | or is it better to do it after voicemail app? |
| 16:35.40 | Un1x | this is my musiconhold conf file http://pastebin.com/d18e12de2 |
| 16:36.35 | *** join/#asterisk `paul (n=admin@125.252.70.126) |
| 16:36.39 | [TK]D-Fender | bijit: Once you begin leaving a VM the only shoice is to hit # and resume processing the dialplan |
| 16:37.08 | [TK]D-Fender | Un1x: Show me the failed attempt and could you please try to at least DESCRIBE what you are doing in the first place and whats involved... |
| 16:37.19 | `paul | how do i enable the colors on asterisk console? (right now its black and white everytime i do an asterisk -vvr) |
| 16:37.39 | Un1x | [TK]D-Fender i reloaded asterisk after editing my config file then i called a freind and pressed the hold button and he heard no music.. |
| 16:37.49 | Un1x | `paul -vvvvvc |
| 16:38.47 | casix | Un1x: try this command: show modules like res_musiconhold.so |
| 16:38.49 | [TK]D-Fender | Un1x: ... PASTEBIN |
| 16:39.26 | Un1x | [TK]D-Fender http://pastebin.com/d6a36887e |
| 16:39.41 | [TK]D-Fender | Un1x: What "hold button", on what PHONE? |
| 16:39.54 | SuPrSluG | hello |
| 16:39.59 | [TK]D-Fender | Un1x: You seem to not understand the what is meant when we ask you for "details" |
| 16:40.07 | Un1x | [TK]D-Fender its an analogue phone. |
| 16:40.16 | [TK]D-Fender | Un1x: then what you want is IMPOSSIBLE |
| 16:40.18 | bijit | [TK]D-Fender: So after I hit # all the options its gives after that is from the dialplan and not the function of the vm? |
| 16:40.30 | [TK]D-Fender | Un1x: ther is no such thing as "signalling hold" on ana analog phone. |
| 16:40.51 | Un1x | i see is it possible perhaps where i can enter an extension during a call like *blah so it signals hold? |
| 16:41.26 | [TK]D-Fender | bijit: After # you may have the review option (if you set it for that box), if you do and complete that, or don't have ti at all, then the dialplan continues. |
| 16:41.45 | SuPrSluG | polycom behind nat trying to register w/ server on public. 1 will the others get 401. exact same configs for all. any ideas? |
| 16:41.50 | [TK]D-Fender | Un1x: Depends what your phone is plugged into <- |
| 16:41.57 | Un1x | TDM400P |
| 16:42.50 | [TK]D-Fender | Un1x: TDM channels do not offer "hold". You can either park the call, or let them sit in dead-air with the phone's "hold" (which is well.. dead air) |
| 16:43.13 | Un1x | How, can i park the call and then unpark it when i want to speak to the person? |
| 16:43.26 | [TK]D-Fender | Un1x: By reading up on call parking. |
| 16:43.32 | Un1x | alright |
| 16:43.54 | Un1x | but, its kinda weird telcos can somehow read signals from analogue phones for musiconhold but asterisk cant. |
| 16:44.05 | [TK]D-Fender | Un1x: No, they can't |
| 16:44.19 | [TK]D-Fender | Un1x: there is no such thing as a "hold signal" from an analog phone |
| 16:44.33 | Un1x | i see |
| 16:44.44 | Un1x | so what your saying is i can do it if i get an IP phone? |
| 16:45.10 | [TK]D-Fender | Un1x: Yes because they offer this signalling |
| 16:45.37 | [TK]D-Fender | Un1x: Most ATA's can do this as well. Good added reason why ATA's are better than TDM FXS. |
| 16:45.44 | bijit | [TK]D-Fender: this may sound stupid but where can I look up for this (if you set it for that box)? |
| 16:46.06 | *** join/#asterisk Yourname`_ (i=Yourname@unaffiliated/yourname/x-837320) |
| 16:46.07 | Un1x | I see |
| 16:46.07 | [TK]D-Fender | bijit: voicemail.conf sample |
| 16:46.17 | Un1x | well callparking here i come :) |
| 16:46.30 | bijit | [TK]D-Fender: thanks. |
| 16:48.55 | Un1x | [TK]D-Fender when the call is parked how do i tell it to play music |
| 16:49.23 | [TK]D-Fender | Un1x: taht is what it does. That is its nature |
| 16:49.34 | [TK]D-Fender | Un1x: GO READ |
| 16:49.52 | Un1x | oh ok |
| 16:50.46 | casix | anyone can help me with this sip trunk between two *. the configuration is here http://pastebin.com/m50e941cc but when a user of A pbx calls a user of B pbx, B pbx says that chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from "user <sip:asdfa@asd>", if I set a fromuser than the call goes ok but the callerid of the call is not the original extension is the fromuser of sip.conf. How can keep the callerid? |
| 16:51.13 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 16:53.03 | Un1x | [TK]D-Fender, here is my features.conf http://pastebin.com/d6449a596 |
| 16:53.13 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 16:53.21 | Un1x | i did include => parkedcalls in extensions.conf aswell |
| 16:53.28 | Un1x | but when i press #700 it doesn't work |
| 16:53.47 | [TK]D-Fender | Un1x: but you're not reading the big print and you're not showing the call. |
| 16:54.03 | [TK]D-Fender | Un1x: You don't seem to be learning from your mistakes here... |
| 16:54.21 | Un1x | [TK]D-Fender, big print? |
| 16:54.25 | Un1x | lol dude i read it |
| 16:54.31 | Un1x | http://www.voip-info.org/wiki-Asterisk+call+parking |
| 16:54.50 | Un1x | scroll down half way it tells you a quick way to get it up and running and it doesn't work |
| 16:54.53 | [TK]D-Fender | un1Yeah... over the span of what, *5* whole minutes? Why do you even think "#700" will park your call? |
| 16:55.22 | [TK]D-Fender | Un1x: And WIKI's are NEVER wrong... and YOU aren't either I guess.... |
| 16:55.32 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:55.38 | [TK]D-Fender | Un1x: Who us what you're DOING and we'll show you why its failing. |
| 16:56.12 | Un1x | well i pasted the features.conf above and i called my cell and pressed #700 |
| 16:56.17 | Un1x | and its not parking the call as it should.. |
| 16:56.26 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:57.45 | [TK]D-Fender | un1there's more to it than features.conf |
| 16:58.07 | jameswf | should be my new signature: No venti is twenty. Large is large. In fact tall is large and grande is spanish for large. Venti is the only one that doesn't mean large. It's also the only one that's italian. Congratulations you're stupid in three languages. |
| 16:58.32 | Un1x | yes i also did the include => parkedcalls in extensions.conf... |
| 16:58.47 | *** join/#asterisk dhill (i=dhill@dhcp-222.iserv.net) |
| 16:58.47 | [TK]D-Fender | Un1x: What part of "sho us what you're doing" don't youg et? |
| 16:59.11 | Un1x | What would you like me to show you.. i showed you my configs i told you what i was doing, what else can i show you? |
| 16:59.32 | *** join/#asterisk Breal (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:59.37 | Breal | What does this mean? SIOD ERROR: unbound variable : tts_textasterisk |
| 17:00.00 | dhill | I have using odbc/postgres for Realtime extensions. In the extension, I have it do Local/${EXTEN}@customers. It seems like Asterisk does not do another psql query for the customers context. or do i have something wrong? |
| 17:00.22 | [TK]D-Fender | Un1x: the CALL. your dialplan. |
| 17:00.58 | [TK]D-Fender | dhill: What do you have in that context in extensions.conf? |
| 17:01.20 | dhill | that context is also in psql |
| 17:01.52 | dhill | oh wait |
| 17:01.55 | dhill | i need a |
| 17:01.58 | dhill | [customers] |
| 17:02.00 | [TK]D-Fender | dhill: wrong answer... try again... |
| 17:02.01 | dhill | switch => Realtime |
| 17:02.03 | dhill | :P |
| 17:02.05 | dhill | right? |
| 17:02.05 | [TK]D-Fender | dhill: YES. |
| 17:02.06 | dhill | haha |
| 17:02.08 | dhill | god i suck |
| 17:02.21 | Un1x | [TK]D-Fender http://pastebin.com/d634faf4f |
| 17:02.41 | [TK]D-Fender | Un1x: Good now read the instructions : "core show application dial" |
| 17:02.54 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.197.131) |
| 17:03.33 | dhill | since my sip users are also in psql.. any way from asterisk to make "sip show users" work? |
| 17:03.52 | *** join/#asterisk mark_csi (n=mark@host217-41-18-3.in-addr.btopenworld.com) |
| 17:03.53 | [TK]D-Fender | dhill: IIRC there are realtime dump options from CLI |
| 17:04.50 | mark_csi | hi all - anyone have any experience in uk zapata.conf? PSTN stays up even after inbound caller hangs up. |
| 17:05.16 | [TK]D-Fender | Grabbing lunch, back in a few minutes |
| 17:07.24 | Un1x | [TK]D-Fender tell me if this example would work or can i have the context say a number such as 700 |
| 17:07.25 | Un1x | exten => _X.,1,Set(CALLERID(number)=8884829892) |
| 17:07.25 | Un1x | exten => _X.,2,Dial(${splitinfinity}/${EXTEN}) |
| 17:07.26 | Un1x | exten => _X.,3,Goto(parkinglot,${ARG1},1) |
| 17:08.17 | bmoraca | what are you trying to do? |
| 17:08.49 | Un1x | bmoraca, trying to get the calls into parking |
| 17:09.47 | *** join/#asterisk CrazyTux (n=brandon@user-vcaurjr.dsl.mindspring.com) |
| 17:10.05 | bmoraca | features.conf dictates your park extension, which is typically 700 |
| 17:10.37 | bmoraca | if you are using an analog phone, you can use (i believe) *2 while in a call to initiate a transfer |
| 17:10.57 | bmoraca | once again, that * command is going to be defined in features.conf |
| 17:11.27 | bmoraca | there's also an option for blind transfer, though that won't tell you what the parking space number is |
| 17:12.12 | bmoraca | so you'll want to do an attended transfer to whatever your parking lot is, listen to the parking space, hit #, and then your line is free and they are parked |
| 17:12.29 | *** join/#asterisk xorl (n=xorl@li30-130.members.linode.com) |
| 17:12.35 | dhill | [tk]d-fen: thanks again |
| 17:12.38 | bmoraca | i have several boxes configured with this through the use of ATAs. never tried it with an FXS port |
| 17:12.49 | xorl | hey, quickq Q, why would audio ingoing/outgoing be completely muted without modifying any of the confs |
| 17:13.00 | bmoraca | i believe the fxs port needs some extra zapata.conf config, though |
| 17:13.07 | Un1x | hrmp, well |
| 17:13.14 | xorl | I get the calls incoming outgoing |
| 17:13.21 | xorl | but can't hear anything, we are using an external PBX |
| 17:13.26 | xorl | (through vitelity) |
| 17:13.39 | psykx-out | xorl: did you reboot? do a hard reboot allowing for a anycards to power down properly |
| 17:13.47 | bmoraca | xorl, are you using a TDM card? if so, adjust your gain. if it's a SIP trunk, you're not NATing properly |
| 17:14.14 | xorl | No cards local, asterisk just routes to the IP Phones (cisco), directly to our line provider Vitelity |
| 17:14.29 | bmoraca | then you're not NATing properly |
| 17:14.41 | xorl | No nat at all. |
| 17:14.50 | *** join/#asterisk esdrasbeleza (n=esdras@sisyphus.dreamhost.com) |
| 17:14.53 | xorl | It was all working yesterday and *nothing* has changed. |
| 17:14.56 | xorl | Literally. |
| 17:15.18 | psykx-out | xorl: which version of asterisk? |
| 17:15.21 | xorl | I diffed my configs from 3 days ago from my rdiff backups and kept going farther back, not a single thing has changed |
| 17:15.36 | xorl | 1.4.21.2, |
| 17:15.40 | psykx-out | module reload and if that doesn't work reboot |
| 17:15.48 | xorl | reboot the entire system? |
| 17:16.21 | xorl | i restarted asterisk itself but that didn't seem to solve anything |
| 17:16.46 | xorl | maybe my provider is screwing up |
| 17:16.58 | Qwell | somebody want to kick tmobile for me, and have them send my phone out already? |
| 17:17.11 | [TK]D-Fender | Un1x: that pattern is BAD. You should not be using anything so generic and you are faining to understand what todo to park a call. |
| 17:17.31 | xorl | Qwell: G1? |
| 17:17.39 | esdrasbeleza | hello. I'm developing some scripts to take care of Asterisk that's running at my job's network. The local Asterisk administrator told me that I couldn't delete the voicemail messages manually or Asterisk would get crazy with the number of messages. Is this right? How can I avoid this? |
| 17:17.58 | Un1x | [TK]D-Fender can you show me a parked call example because the examples, on the voip-info are not so accurate |
| 17:18.08 | xorl | reloaded the modules, no go. |
| 17:18.15 | xorl | restarted asterisk completely, no go. |
| 17:19.08 | [TK]D-Fender | Un1x: You do an ATTENDED TRANSFER to an exten that calls "ParkCall" (natively 700 in the [parkedcalls] context) |
| 17:20.00 | xorl | ugh |
| 17:20.03 | xorl | this is so frustrating |
| 17:20.51 | [TK]D-Fender | xorl: And proportionately you have shown us nothing. |
| 17:21.11 | xorl | [TK]D-Fender: Well, if I *knew* what it was to show you, i'd have fixed it now wouldn't I have? lol |
| 17:21.21 | [TK]D-Fender | xorl: Show us your server & its condition in detail along with your configs and MAYBE we'll have enough info to actually help you. |
| 17:21.21 | xorl | I have maximum verbosity on my logging, watching sip debug. |
| 17:21.41 | casix | anyone can help me with this sip trunk between two *. the configuration is here http://pastebin.com/m50e941cc but when a user of A pbx calls a user of B pbx, B pbx says that chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from "user <sip:asdfa@asd>", if I set a fromuser than the call goes ok but the callerid of the call is not the original extension is the fromuser of sip.conf. How can keep the callerid? |
| 17:21.52 | [TK]D-Fender | xorl: If you're not sure what to show us then show use EVERYTHING. pastebin is your friend |
| 17:21.53 | [TK]D-Fender | ~pb |
| 17:21.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 17:21.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:22.02 | xorl | [TK]D-Fender: will do, give me a minute |
| 17:22.12 | Qwell | xorl: naturally |
| 17:22.27 | xorl | which configs do you want? |
| 17:22.34 | xorl | sip.conf exten.conf logger.conf? |
| 17:22.38 | Qwell | xorl: you have one? |
| 17:22.39 | [TK]D-Fender | xorl: sip |
| 17:22.50 | [TK]D-Fender | xorl: and the SIP debug for your failed calls. |
| 17:23.07 | xorl | [TK]D-Fender: Well, they all fail, call comes through, no audio. |
| 17:23.17 | xorl | How do I cut out the sip debug logs from the console |
| 17:23.27 | [TK]D-Fender | xorl: cut&paste |
| 17:23.43 | xorl | works for me |
| 17:23.48 | [TK]D-Fender | xorl: Should be easy to provide a failed call if they all fail |
| 17:24.38 | xorl | Well fail in the audio sense, all works pretty well. |
| 17:24.52 | xorl | call comes in, acknowledges the pick up, and the hang up, just no audio in/out |
| 17:26.00 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 17:26.18 | xorl | http://209.112.245.34/sip.conf |
| 17:26.41 | [TK]D-Fender | xorl: Your * is behind NAT? |
| 17:26.50 | xorl | the phones are, the * is not. |
| 17:27.06 | Un1x | how can i add a third command under this |
| 17:27.06 | Un1x | exten => _X.,1,Set(CALLERID(number)=8884829892) |
| 17:27.06 | Un1x | exten => _X.,2,Dial(${splitinfinity}/${EXTEN}) |
| 17:27.15 | Un1x | err nvm |
| 17:27.19 | [TK]D-Fender | SMRT |
| 17:27.38 | xorl | ? |
| 17:27.52 | SuPrSluG | polycom behind nat trying to register w/ server on public. 1 will the others get 401. exact same configs for all. any ideas? |
| 17:27.57 | [TK]D-Fender | xorl: That wasn't for you.. |
| 17:28.03 | xorl | oh heh |
| 17:28.07 | [TK]D-Fender | xorl: So now for the failed call please... |
| 17:28.42 | xorl | give me two seconds |
| 17:30.39 | xorl | [TK]D-Fender: you get that notice? |
| 17:31.33 | [TK]D-Fender | SIP/2.0 401 Unauthorized |
| 17:31.43 | [TK]D-Fender | xorl: Lokos liek your phones are authing wrong. |
| 17:31.49 | Un1x | would anyone have an example dialplan where i can press #77 in a call and dial a number for the call to be transfered to |
| 17:32.02 | xorl | That's weird, they worked fine yesterday |
| 17:32.10 | *** join/#asterisk ccesario_ (n=ccesario@linux.unialco.com.br) |
| 17:32.28 | [TK]D-Fender | xorl: Well you should have noticed that you did not show me a failed CALL. There is no CALL in three <- |
| 17:32.32 | [TK]D-Fender | there* |
| 17:32.45 | xorl | That's not the call, i copy and pasted it when I called in heh |
| 17:33.03 | [TK]D-Fender | Un1x: FORGET the "#" for a second as something you can ASSUME and realize what I told you you need to do. |
| 17:33.23 | [TK]D-Fender | [12:19]<[TK]D-Fender>Un1x: You do an ATTENDED TRANSFER to an exten that calls "ParkCall" (natively 700 in the [parkedcalls] context) |
| 17:33.46 | [TK]D-Fender | [12:02]<[TK]D-Fender>Un1x: Good now read the instructions : "core show application dial" |
| 17:33.53 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 17:33.58 | Un1x | no im not trying to park the calls anymore im trying to lets say someone calls for me and im out they can press #77 and enter the number where i can be reached and asterisk bridges the call |
| 17:35.07 | [TK]D-Fender | Un1x: Show us the dialplan for your call. |
| 17:35.40 | Un1x | [TK]D-Fender, http://pastebin.com/d634faf4f |
| 17:36.02 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:36.33 | telnettech | can someone tell me if the asterisk -vvvr /tee/tmp/log is useable on 1.2 version |
| 17:36.36 | bmoraca | that's not exactly a simple thing to explain here...heh |
| 17:36.53 | xorl | [TK]D-Fender: ok, i got an even bigger sip log |
| 17:36.54 | [TK]D-Fender | Un1x: So what do incoming calls land on? |
| 17:37.31 | Un1x | no this is for an outgoing call im trying to do the xfer like i can call a freind then tell him hold i'll transfer u and then do #77 and it transfer to the phone number |
| 17:38.30 | bmoraca | Un1x, that's a simple attended transfer. you can do that form any phone. you don't need to do anything special in the dialplan for it. (perhaps a t dial option) |
| 17:38.42 | [TK]D-Fender | [12:02]<[TK]D-Fender>Un1x: Good now read the instructions : "core show application dial" |
| 17:39.47 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com) |
| 17:40.03 | xorl | man, I am stumped |
| 17:40.51 | Breal | <PROTECTED> |
| 17:41.07 | telnettech | new to *......can someone tell me how to dump my SIP messages to a log from the CLI. I am using 1.2 version |
| 17:41.30 | xorl | not even echo is working |
| 17:41.34 | xorl | somethings totally fubar |
| 17:42.33 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 17:42.46 | [TK]D-Fender | xorl: Make sure 'canreinvite=no" is under [general] and every section. Also all your phones should be "nat=yes", "qualify=yes", and yuor ITSP peers should be "nat=no" |
| 17:43.16 | casix | telnettech: you can do that with ngrep: ngrep port sip_port and host ip_of_user_want_to_watch |
| 17:43.46 | sdaniels | Im trying to record incoming calls, it is working except when the call is forwarded to an outside number, can someone take a look at this? thanks http://www.pastebin.ca/1261522 |
| 17:44.01 | telnettech | casix: this is from the CLI or the linux prompt |
| 17:45.02 | xorl | [TK]D-Fender: Forgive my cannon fodder brain lol, ITSP just flew over my head. |
| 17:45.19 | casix | telnettech: linux prompt |
| 17:45.50 | telnettech | casix: thanks |
| 17:46.15 | [TK]D-Fender | xorl: your provider's entries |
| 17:46.19 | xorl | got it |
| 17:46.56 | xorl | It's so weird. |
| 17:46.59 | xorl | I hear a dial tone, |
| 17:47.07 | xorl | but if I dial a number out, i don't even hear it ringing |
| 17:47.18 | xorl | but the number I call gets reached. |
| 17:48.25 | Breal | <PROTECTED> |
| 17:48.44 | *** join/#asterisk hansin (n=eric@c-67-173-251-102.hsd1.co.comcast.net) |
| 17:48.49 | jameswf | new startrek :) http://link.brightcove.com/services/link/bcpid1562587978/bctid2541780001 |
| 17:50.58 | *** join/#asterisk mark_csi (n=mark@host217-41-18-3.in-addr.btopenworld.com) |
| 17:52.46 | mark_csi | hi all, I've found a patch that fixes an issue I'm having with my system. Does the version of the patch matter? I'm using 1.4.22 and the patch is 1.4.19 |
| 17:53.08 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 17:57.03 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 17:58.07 | [TK]D-Fender | mark_csi: the patch you're looking for was not integrated into the mainline? |
| 17:58.09 | *** join/#asterisk rcahilig (n=sysad@202.78.75.246) |
| 17:58.17 | rcahilig | hello guys, I would like to ask if it is possible to send and receive fax in Asterisk |
| 17:58.30 | Carlos_PHX | Yes |
| 17:58.40 | Carlos_PHX | You will need some add-ons. |
| 17:58.52 | xorl | I have seriously never been this frustrated with a phone system in my life. |
| 17:59.08 | Carlos_PHX | That's because you think Asterisk is a phone system, and it's not. |
| 17:59.09 | rcahilig | what particular addons do I need? |
| 17:59.31 | Carlos_PHX | spandsp for one. You should read the fax docs on voip-info.org. |
| 17:59.36 | casix | bye |
| 17:59.52 | Carlos_PHX | A lot depends on how you will connect to the fax devices and PSTN. |
| 18:00.19 | sdaniels | when using mixmonitor app, if the call is forwarded to an outside line I dont get any audio.. any ideas? http://www.pastebin.ca/1261531 |
| 18:01.35 | xorl | Carlos_PHX: lol, well, my VoIP system has no voice. |
| 18:01.44 | xorl | I can hear the Asterisk voice mail message play |
| 18:01.46 | [TK]D-Fender | sdaniels: For one, that is not FORWARDING, next you should call mixmonitor before EACH dial. |
| 18:01.53 | xorl | so asterisk Does transmit audio. |
| 18:01.58 | xorl | Could it be my third party PBX provider? |
| 18:03.38 | Carlos_PHX | xorl: You have a NAT issue. |
| 18:03.44 | Carlos_PHX | ~nat |
| 18:03.44 | jbot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:03.48 | Carlos_PHX | Oops |
| 18:03.52 | xorl | Carlos_PHX: * == not behind nat. |
| 18:04.12 | xorl | No fireawll either, disabled that to 100% be positive of all routing/firewall/nat issues. |
| 18:04.14 | Carlos_PHX | Then you have an RTP port issue. |
| 18:04.23 | Carlos_PHX | Who is the service provider? |
| 18:04.32 | xorl | vitelity |
| 18:04.50 | Carlos_PHX | Well, they don't do NAT, so it sounds like RTP port problems. |
| 18:05.00 | Carlos_PHX | Did you change rtp.conf? |
| 18:05.13 | xorl | Haven't touched it. |
| 18:05.24 | *** join/#asterisk andresmujica (n=andresmu@190.25.103.139) |
| 18:05.38 | Carlos_PHX | Somehow, your RTP packets (voice) are not being delivered. |
| 18:05.41 | xorl | diff'd vs. the stock -dist |
| 18:05.47 | Carlos_PHX | Is there one-way audio, or neither side gets audio? |
| 18:05.53 | xorl | neither side. |
| 18:06.09 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:06.11 | Carlos_PHX | I have never seen this happen without NAT. |
| 18:06.16 | xorl | well, i can hear the asterisk voicemail deal |
| 18:06.29 | sdaniels | [TK]D-Fender: I added another mixmonitor before the second dial, but there is still no audio. by no audio I mean that when it dials the 2nd number (my cell) i do not hear the caller or vice-versa. it works as expected if I answer the call on ext 6000 |
| 18:06.34 | Carlos_PHX | Right, that's phone > Asterisk, but Asterisk > Vitelity fails. |
| 18:06.59 | xorl | no no i mean external call -> asterisk i can broadcast. |
| 18:07.01 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 18:07.05 | Katty | zomghai |
| 18:07.06 | Carlos_PHX | A CODEC failure should fail the call, but what CODEC are yo uusing? |
| 18:07.22 | [TK]D-Fender | sdaniels: Well you'd better provide some mroe details about who each leg of the call is an how all the networking involved is set up |
| 18:07.37 | [TK]D-Fender | Katty: O HAI |
| 18:07.38 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 18:07.39 | xorl | Not even sure, I was dumped with administrating this asterisk server a month and a half ago |
| 18:07.47 | xorl | Haven't touched it, backed up the configs nightly etc. |
| 18:07.48 | *** join/#asterisk ccesario_ (n=ccesario@linux.unialco.com.br) |
| 18:07.50 | Katty | [TK]D-Fender: i have something terribly bad for me. |
| 18:07.52 | Katty | [TK]D-Fender: a Whopper |
| 18:07.53 | xorl | But just left it alone cause it's worked. |
| 18:07.56 | Katty | [TK]D-Fender: from Burger King |
| 18:08.14 | Katty | puts hospital on speed dial in case of heart attack |
| 18:08.17 | [TK]D-Fender | Katty: ZOMG <3 a tack! |
| 18:08.19 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:08.24 | Katty | [TK]D-Fender: tack? |
| 18:08.30 | Katty | [TK]D-Fender: attack? |
| 18:08.31 | Katty | oh |
| 18:08.32 | Katty | duh |
| 18:08.33 | Katty | soryr |
| 18:08.33 | Carlos_PHX | xorl: You're saying it used to work and stopped? |
| 18:08.35 | Katty | also, sorry |
| 18:08.45 | xorl | Yes |
| 18:08.49 | xorl | Worked yesterday. |
| 18:08.51 | Katty | brain spastic, apparently. took me a second |
| 18:08.51 | Carlos_PHX | Whoa |
| 18:08.51 | xorl | Stopped working today. |
| 18:08.52 | [TK]D-Fender | Katty: http://i239.photobucket.com/albums/ff293/Sharpstar/20jfmky.jpg |
| 18:08.53 | Qwell | Katty: 75 7/8 :( |
| 18:08.57 | Carlos_PHX | You checked with Vitelity for issues? |
| 18:08.59 | Katty | Qwell: ! |
| 18:09.06 | xorl | I have contacted them but no word back yet. |
| 18:09.10 | Katty | Qwell: 73 and 1 bar |
| 18:09.17 | xorl | Carlos_PHX: So it is most likely an RTP issue then. |
| 18:09.18 | Katty | [TK]D-Fender: lawl. cute. |
| 18:09.20 | Carlos_PHX | Wow, that's a strange one. |
| 18:09.23 | Carlos_PHX | Certainly RTP |
| 18:09.29 | Carlos_PHX | That's where the voice is. |
| 18:09.29 | xorl | Indeed. |
| 18:11.30 | [TK]D-Fender | Katty: Glad I got a laugh for it... "Mission Accomplished" |
| 18:12.27 | Breal | Where does asterisk store its sound ifles? |
| 18:12.28 | Breal | files |
| 18:12.48 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 18:13.09 | sdaniels | [TK]D-Fender: please have another look at the modified pastebin when you have a moment, thanks. http://www.pastebin.ca/1261546 |
| 18:13.26 | xorl | is there other providers of end point PBX solutions other than Vitelity |
| 18:13.48 | Katty | [TK]D-Fender: ;) |
| 18:13.48 | xorl | well trunks |
| 18:14.05 | Katty | well shucks? |
| 18:14.17 | Katty | that sounded better in my head. |
| 18:14.27 | [TK]D-Fender | sdaniels: and I told you to show use what you are dilaing and descrbin how its all set up |
| 18:14.47 | [TK]D-Fender | xorl: ... |
| 18:14.49 | [TK]D-Fender | ~itsp |
| 18:14.49 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:14.51 | [TK]D-Fender | ~itsplist-us |
| 18:14.52 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:14.54 | [TK]D-Fender | ~itsplist-ca |
| 18:14.54 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 18:15.03 | xorl | [TK]D-Fender: Thank you. |
| 18:15.05 | xorl | Sorry, i don |
| 18:15.23 | xorl | i don't know all the proper */phone system terminology :P |
| 18:15.30 | xorl | I code in perl and C all day, not usually my deal. |
| 18:15.41 | jer_ | does not recommend babytel. they flip out over higher than normal call volumes, think you're running a company off a residential line -- when in fact, you have kids. (most calls incoming, not outgoing) |
| 18:16.00 | [TK]D-Fender | jer : I concure... clients of mine lokoed at them and get the runaround. |
| 18:16.20 | jer_ | i do use unlimitel now, happier than a pig in shit |
| 18:16.32 | [TK]D-Fender | jer_: thats where mine ended up going as well |
| 18:21.28 | bmoraca | NVFaxDetect relies on precise timing from zaptel hardware (or ztdummy) doesn't it? |
| 18:26.30 | *** join/#asterisk legis (n=wadsack@unaffiliated/legis) |
| 18:27.01 | *** join/#asterisk rhousand (n=ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:27.27 | legis | Hi, any ideas why would I get one way audio when doing transcoding on a call? I'm doing ulaw -> g729 |
| 18:27.52 | legis | If I do ulaw -> ulaw or g729 -> g729 it works fine. |
| 18:28.09 | [TK]D-Fender | legis: Describe each leg in detail |
| 18:28.50 | legis | [TK]D-Fender: SIP/ATA -> * -> ITSP |
| 18:29.04 | [TK]D-Fender | legis: Now the networking... |
| 18:29.24 | legis | [TK]D-Fender: SIP has a public IP, * too. |
| 18:29.33 | legis | no NAT. |
| 18:29.42 | rhousand | I am using record in my extension.conf file. when i dial the extension how do i end the recording? |
| 18:30.02 | [TK]D-Fender | rhousand: "#" |
| 18:30.32 | rhousand | I tried that first but it does not end the recording. I have to hangup |
| 18:30.57 | rhousand | i'll try again |
| 18:31.24 | [TK]D-Fender | rhousand: If it doesn't, then you have a DTMF configuration problem. |
| 18:32.19 | rhousand | [TK]D-Fender: well what seem to happen is when i enter "#" it hangsup |
| 18:33.30 | rhousand | o i found the issue |
| 18:33.42 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 18:33.45 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 18:33.56 | rhousand | thanks anyways |
| 18:34.46 | [TK]D-Fender | rhousand: after "#" the app quits at your dialplan continues to do whatever is next. If that happens to be "nothing, then well... there goes your call. |
| 18:36.48 | legis | [TK]D-Fender: you know if a problem with g729 could cause that? |
| 18:36.57 | legis | a problema with g729 license |
| 18:37.31 | [TK]D-Fender | legis: possible... but prove it by preventing reinvites first |
| 18:38.02 | [TK]D-Fender | legis: Test each, debug each, compare |
| 18:38.19 | sdaniels | [TK]D-Fender: I updated http://www.pastebin.ca/1261572 I dont know what else to explain. |
| 18:38.30 | jameswf | someone just called me and asked for a fax tone so I screeched in their ear... |
| 18:39.03 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:39.23 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-16d5a6fcce14bb8d) |
| 18:40.22 | [TK]D-Fender | sdaniels: You are not showing a failed call with SIP debug, you are not describing the networking involved for this call END-TO-END, you are not showing us your configs. So basically, you offer nothing for us to assist you with. |
| 18:40.58 | sdaniels | [TK]D-Fender: a simple I dont know would have been fine. thanks. |
| 18:41.18 | legis | [TK]D-Fender: what do you mean by 'preventing reinvites first' ? |
| 18:42.13 | [TK]D-Fender | sdaniels: Don't know? You haven't shown us anything. Do you drive up to the mechanic, poitn a finger at your car, and ask him whats wrong without letting him look under the hood? |
| 18:42.54 | [TK]D-Fender | legis: set for same codecs and force RTP through *. See if it works. then do the same for transcoding, etc |
| 18:45.52 | legis | [TK]D-Fender: yeah ulaw/ulaw works, g729/g729 too, the problem is ulaw/g729 |
| 18:46.18 | [TK]D-Fender | legis: confirm while assuring that rtp on identical codec filters through *. This part is crucial |
| 18:47.44 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:51.26 | legis | [TK]D-Fender: ah ok, I get it :D, let me check |
| 18:52.09 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 18:53.12 | *** join/#asterisk ManxPower (n=manxpowe@204.sub-70-222-220.myvzw.com) |
| 18:53.38 | [TK]D-Fender | legis: Its either a reinvite / networking issue or a broken codec issue.... the latter is much less likely. |
| 18:54.43 | ManxPower | Use the Digium codecs, stronger than all the others -- now with titanium! |
| 18:55.00 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 18:55.01 | *** join/#asterisk ecrist (n=ecrist@chunk.ip6.secure-computing.net) |
| 18:55.46 | ecrist | quick question - does asterisk have the ability to change caller ID on an incoming call based on with SIP trunk it comes in on? |
| 18:56.00 | Carlos_PHX | I could have sworn I'd seen this before but can't find it... How to pass the PIN for a conference to the MeetMe app so the user doesn't have to enter it? |
| 18:56.16 | Carlos_PHX | ecrist: Ys |
| 18:56.17 | Carlos_PHX | Yes |
| 18:56.41 | ecrist | thanks Carlos_PHX |
| 18:57.06 | Breal | I can not hear audio if my sip device is on the same machine as my asterisk box |
| 18:57.44 | ManxPower | Breal: not all that surprising. |
| 18:58.27 | *** part/#asterisk ecrist (n=ecrist@chunk.ip6.secure-computing.net) |
| 18:59.26 | Breal | What would cause that? |
| 18:59.46 | Breal | Or how can I fix it rather |
| 19:00.03 | [TK]D-Fender | ManxPower: Ti does not alloy with C ;) |
| 19:00.27 | ManxPower | Breal: two devices trying to use the same RTP or SIP ports on the same machine. Really nothing different than trying to run 2 web servers or 2 smtp servers on the same machine. |
| 19:00.52 | Breal | Is there anything that can be done? |
| 19:00.57 | *** join/#asterisk [netman] (n=netman@238.Red-88-23-119.staticIP.rima-tde.net) |
| 19:00.59 | Breal | Like use different ports..etc |
| 19:01.03 | ManxPower | Breal: do you want to spend a few hours trying to fix this or do you want to just run the softphone on a different system? |
| 19:01.54 | ManxPower | Breal: configure your softphone to use a port other than 5060/UDP for the source port and use ports other than 10,000 - 20,000/UDP for Audio. |
| 19:02.10 | Breal | ok |
| 19:02.30 | ManxPower | Softphones give me hives so I don't use them. |
| 19:03.01 | ManxPower | Breal: If you have NAT or firewall on the asterisk box that could also be a problem |
| 19:03.42 | Carlos_PHX | Huh, I thought I was the only one with that disease. |
| 19:04.49 | ManxPower | Carlos_PHX: It's a VERY common condition among experienced Asterisk people. |
| 19:04.49 | ManxPower | Some might say you are not an experienced Asterisk person until you develop an allergy to softphones. |
| 19:05.11 | Corydon76-dig | I must not be an experienced Asterisk person, then |
| 19:05.34 | Corydon76-dig | However, I would NEVER deploy one in production |
| 19:05.45 | ManxPower | Corydon76-dig: Well, that is close enough. |
| 19:06.24 | ManxPower | What I have found that learning how to configure and use a softphone doesn't help much with hardphones or Asterisk. So I think it's mostly a waste of time. |
| 19:06.45 | *** join/#asterisk hansin (n=eric@c-67-173-251-102.hsd1.co.comcast.net) |
| 19:06.48 | Corydon76-dig | It's not a waste of time for business travelers, for example, though |
| 19:07.05 | Corydon76-dig | That's the one place that I would still advocate a softphone |
| 19:07.51 | [TK]D-Fender | yup.. pretty much jsut for remote laptop user. |
| 19:07.55 | Corydon76-dig | Then you can implement things like VPN tunnels across which your calls are sent and all sorts of network traversal issues, not to mention the problem of carrying around a bulky phone |
| 19:08.13 | ManxPower | <rant>Much like this miserable excuse for an SDK for Novatel EVDO devices. Looks like it was built by a bunch of drunken college kids with epilepsy. I'll end up spending 10x the mount of time trying to get the SDK to work than it will ever save me.</rant> |
| 19:08.42 | ManxPower | Corydon76-dig: even when I traveled I used an ATA rather than a softphone |
| 19:09.05 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 19:09.14 | Corydon76-dig | ManxPower: you also manage the networks, though |
| 19:09.23 | ManxPower | Corydon76-dig: Yes. |
| 19:09.39 | Corydon76-dig | ManxPower: I dunno about even an ATA in a hotel room, for example |
| 19:10.02 | ManxPower | Corydon76-dig: worked fine for me at all the hotels with stable internet access. |
| 19:10.24 | Carlos_PHX | I've tried to encourage our customers to use softphones, they just don't. |
| 19:10.32 | Carlos_PHX | They take their physical phones with them to travel. |
| 19:11.29 | xorl | Carlos_PHX: Yeah got ahold of our itsp, they were saying RTP appears to be blocked and we don't have a firewall, so they have their network guy looking into it |
| 19:12.39 | ManxPower | I never really liked VoiceOverIPOverInternet |
| 19:15.18 | Katty | did someone just twitter follow me? |
| 19:15.58 | Carlos_PHX | xorl: Cool, good luck. |
| 19:16.04 | ManxPower | Katty: Not in public, I hope! |
| 19:16.15 | Carlos_PHX | You previously asked about other ITSPs, yes, there are plenty. Vitelity is well respected though. |
| 19:20.27 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 19:20.37 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 19:22.52 | mikealeonetti | with my SIP service, the fromuser field has my caller id. But ONE internal extension needs to show up as another caller ID. What's the best way to set that? Is there such a thing as a dynamic fromuser? |
| 19:23.05 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 19:23.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:26.10 | Carlos_PHX | mikealeonetti: http://www.voip-info.org/-index.php?page=Asterisk cmd SetCallerID |
| 19:26.16 | [TK]D-Fender | mikealeonetti: Don't set FROMUSER. change your dials. |
| 19:26.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:27.18 | mikealeonetti | [TK]D-Fender: As in use SetCallerID as Carlos_PHX pointed out? |
| 19:27.18 | Carlos_PHX | Yeah, that's the way to do it. |
| 19:27.25 | [TK]D-Fender | mikealeonetti: Yes, only using the proper functions. |
| 19:27.38 | mikealeonetti | you're the man Carlos_PHX (and [TK]D-Fender) |
| 19:27.44 | mikealeonetti | lots of <3 |
| 19:28.30 | Carlos_PHX | mikealeonetti: Keep in mind your ITSP may or may not respect that. |
| 19:28.51 | Carlos_PHX | So if it fails, don't assume you did it wrong. |
| 19:29.55 | [TK]D-Fender | mikealeonetti: No.... still assume you did it wrong... just know that you might even be wrong about that too ;) |
| 19:30.04 | Carlos_PHX | Heh |
| 19:31.08 | mikealeonetti | so I'll just use an IF I guess to set a variable |
| 19:31.11 | mikealeonetti | and use that |
| 19:31.13 | *** join/#asterisk km2 (n=x@mobile-166-217-255-036.mycingular.net) |
| 19:31.36 | Carlos_PHX | We have a SIP variable for each user which is externalid= and then do a set on the dial. |
| 19:31.55 | mikealeonetti | ah |
| 19:32.01 | mikealeonetti | that works too |
| 19:32.06 | Carlos_PHX | So there's an account-wide global for the company main number, and one if needed for each DID. |
| 19:32.19 | Carlos_PHX | Some companies like the main number shown. |
| 19:32.37 | mikealeonetti | so, on sip.conf I can set a default variable, and then a custom one for just her |
| 19:32.55 | mikealeonetti | I can just make up variables in sip.conf? |
| 19:32.57 | Carlos_PHX | Sure, and then your dial just sets every call to $externalid |
| 19:33.01 | Carlos_PHX | Sure |
| 19:33.12 | mikealeonetti | and use them as normal variables in extensions.conf? |
| 19:33.37 | Carlos_PHX | Right |
| 19:33.56 | mikealeonetti | that's pretty intuitive |
| 19:34.11 | Carlos_PHX | We have lots of variables in sip.conf that get used in outbound call processing. |
| 19:34.21 | Carlos_PHX | Quite easy, powerful. |
| 19:34.24 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 19:35.47 | *** join/#asterisk ccesario_ (n=ccesario@189.20.219.10) |
| 19:37.32 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 19:39.39 | rene- | Carlos_PHX: How are u? |
| 19:42.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:46.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:46.43 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 19:53.43 | cesar_CR | guys I am seen this in the cli :chan_sip.c:16869 handle_request_invite: Call from '' to extension '013525551690000' rejected because extension not found. |
| 19:53.58 | cesar_CR | I have blocked the IP via iptables.. |
| 19:54.07 | mikealeonetti | Carlos_PHX: so I can use ${externalid} in extensions.conf? |
| 19:54.16 | cesar_CR | I am hacked ??? |
| 19:54.51 | seanbright | i have been hacking into someone's machine all day |
| 19:55.05 | seanbright | anyone else wants in the poor bastard's IP is 127.0.0.1 |
| 19:55.42 | stintel | lmao |
| 19:55.47 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 19:56.12 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:58.13 | [TK]D-Fender | cesar_CR: so far that doesn't prove that its from this person you claim to have blcoked, nor that your block was functional in the first place. |
| 20:01.09 | carrar | Just unplug your box from the internet |
| 20:01.16 | carrar | that solves it |
| 20:01.36 | [TK]D-Fender | carrar: Not enough... remove POWER from the computer! |
| 20:01.40 | carrar | hahah |
| 20:01.48 | [TK]D-Fender | HACK THIS BIOTCH! |
| 20:01.59 | mikealeonetti | I'm still doing somethign wrong |
| 20:02.00 | [TK]D-Fender | <NO CARRIExzzzzzzzzzzzzzzzzzzz> |
| 20:02.01 | mikealeonetti | big surprise |
| 20:02.06 | carrar | If I got a $1 for everytime someone did a SWAP of extensions from 1000 to 9999 trying to register I wouldbe rich!! |
| 20:02.16 | [TK]D-Fender | mikealeonetti: 1 step down, 11 to go! |
| 20:02.19 | carrar | SWEEP |
| 20:02.52 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 20:03.07 | cesar_CR | [TK]D-Fender, I have blocked the IP that made that, I got the IP from a sip set debug on |
| 20:03.38 | carrar | You need a security policy |
| 20:03.42 | carrar | heh |
| 20:03.57 | mikealeonetti | well, I set externalid=+1516xxxxxxx in the sip.conf but when I set SetCallerID("ID Name" ${externalid}) the number comes up as "asterisk" |
| 20:03.58 | cesar_CR | [TK]D-Fender, I have a lot of those to different numbers from the same ip |
| 20:03.58 | mikealeonetti | quite funny |
| 20:04.31 | mikealeonetti | actually, it says "terisk" |
| 20:06.58 | cesar_CR | [TK]D-Fender, is it serious ? |
| 20:07.04 | *** part/#asterisk Mark17 (n=mark@vnc.tt.streamservice.nl) |
| 20:07.23 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 20:07.24 | carrar | w00t |
| 20:08.22 | mikealeonetti | I wonder what I set wrong |
| 20:10.47 | *** join/#asterisk Aces12 (n=ric@65.208.182.66) |
| 20:11.07 | Aces12 | hey anyone here have experience getting a cisco 7960 to work with asterisk? |
| 20:11.36 | cesar_CR | carrar, so you have this type of things allways ? |
| 20:11.48 | cesar_CR | Aces12, only a 7911G |
| 20:11.59 | cesar_CR | I have 12 of them |
| 20:12.11 | Aces12 | cesar_cr is it required to use a tftp server with cisco phones? |
| 20:12.15 | Aces12 | do you use them with yours? |
| 20:12.48 | cesar_CR | yes, very mucho without it the phones does not work |
| 20:12.55 | Aces12 | cesar_cr i see. |
| 20:13.18 | Aces12 | was it hard to get the phones to work with sip? |
| 20:13.20 | cesar_CR | even the softphone from cisco needs it |
| 20:13.52 | cesar_CR | Aces12, the hard thing was to get the oficial a latest sip firmware |
| 20:14.09 | Aces12 | cesar from cisco? or from asterisk? |
| 20:14.10 | cesar_CR | and get the xml file well done |
| 20:14.16 | Aces12 | where do i get that at? |
| 20:14.22 | carrar | cesar_CR, yeah, welcome to the internet |
| 20:14.55 | carrar | cesar_CR, security needs to be high on your list if you put something on the internet |
| 20:14.56 | cesar_CR | carrar, thanks |
| 20:15.11 | carrar | it will get scanned and hit at |
| 20:15.24 | cesar_CR | Aces12, Cisco Phones means cisco firmware |
| 20:15.26 | carrar | and exploited if there are holes |
| 20:15.48 | *** join/#asterisk VoIPDontCry (n=seba0606@adsl190-28-133-98.epm.net.co) |
| 20:15.55 | VoIPDontCry | hi everybody |
| 20:16.11 | VoIPDontCry | <PROTECTED> |
| 20:18.10 | Aces12 | cesar what is the latest sip firmware for their phones? and where did you find yours? |
| 20:19.45 | *** join/#asterisk Anggelus (n=a@189.16.236.1) |
| 20:22.02 | cesar_CR | Aces12, SIP 11.8.4 and you need to have an account in cisco... or something like that to be able to download it |
| 20:22.24 | cesar_CR | Aces12, but my xml file is on the wiki |
| 20:24.01 | VoIPDontCry | anybody knows a good link to study how to configure elastix/asterisk to obtain better sound in calls ? |
| 20:24.40 | LoRez | three dups in 4 seconds? seriously? |
| 20:24.52 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.90) |
| 20:25.43 | Carlos_PHX | Damn, got SlimJim grease on my handset. |
| 20:26.22 | mikealeonetti | I'm definitely doing somethign wrong with this variable |
| 20:27.17 | mikealeonetti | "Name" <${externalid}> should work no problem, right? |
| 20:27.49 | Carlos_PHX | Hmm, you are setting the name, not the number? |
| 20:27.55 | Carlos_PHX | You know that won't make it to the PSTN right? |
| 20:28.24 | mikealeonetti | should I just do <${externalid}> then? |
| 20:28.26 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 20:28.41 | Carlos_PHX | Do you want to set name or number? |
| 20:28.47 | mikealeonetti | just the number |
| 20:29.32 | VoIPDontCry | just the number |
| 20:29.41 | VoIPDontCry | when a call arrives from a PSTN trunk |
| 20:30.15 | Aces12 | ceaser what do you think of these directions? |
| 20:30.18 | Aces12 | http://www.plumbersurplus.com/Blog/post/2008/05/Cisco-760-and-Trixbox-Problems-in-our-VOIP-Implementation.aspx |
| 20:30.21 | Carlos_PHX | exten => s,n,Set(CALLERID(number)=${EXTERNALID}) |
| 20:31.32 | Carlos_PHX | Cisco phones are great. They sell fast on eBay to people who don't know any better and you can use the money to buy a decent phone. |
| 20:31.51 | carrar | Cisco 7900 are great phones |
| 20:32.00 | Carlos_PHX | Yes, to eBay, I agree. |
| 20:32.05 | carrar | to use |
| 20:32.06 | mikealeonetti | Carlos_PHX: the caller id number says "terisk" |
| 20:32.48 | Carlos_PHX | I didn't know you could put characters in the number field, interesting. |
| 20:32.50 | Carlos_PHX | Sounds useful. |
| 20:32.59 | VoIPDontCry | :/p |
| 20:33.10 | mikealeonetti | that's what I said! |
| 20:33.21 | Carlos_PHX | Do a NoOp to display the externalid variable just before the set. |
| 20:33.26 | Carlos_PHX | See if it's right in the CLI |
| 20:35.39 | Breal | Can asterisk register to a remote MGCP asterisk box? |
| 20:35.46 | Carlos_PHX | Yes |
| 20:35.47 | [TK]D-Fender | Breal: No |
| 20:35.54 | Carlos_PHX | If... |
| 20:36.04 | [TK]D-Fender | Carlos_PHX: * cannot act like an MGCP server |
| 20:36.10 | [TK]D-Fender | (sorry, PHONE) |
| 20:36.19 | Carlos_PHX | Ah, well, hmmm |
| 20:36.34 | Carlos_PHX | I'd have to go look at the MGCP notes I have beyond "it sucks." |
| 20:36.42 | Breal | We want to have an asterisk server inhouse use a remote mgcp server as our voip trunk |
| 20:36.44 | Carlos_PHX | I defer to [TK]D-Fender's knowledge on this one. |
| 20:36.45 | [TK]D-Fender | Breal: You can connect MGCP phones to *, but not MGCP servers |
| 20:37.11 | Breal | Oh, we want to do asterisk<->asterisk |
| 20:37.13 | Carlos_PHX | The remote MGCP server is Asterisk or a media gateway like a Cisco router? |
| 20:37.20 | Breal | Carlos_PHX: Asterisk |
| 20:37.25 | Carlos_PHX | Then why MGCP?? |
| 20:37.37 | Breal | Managed by bandwidth.com |
| 20:37.43 | Breal | Its cheaper to use MGCP then SIP |
| 20:37.48 | [TK]D-Fender | Breal: Don't use MGCP |
| 20:37.53 | Carlos_PHX | Bahahahaha |
| 20:37.59 | [TK]D-Fender | Breal: CHEAPER? Thats retarded |
| 20:38.02 | Carlos_PHX | Seriously? |
| 20:38.04 | Carlos_PHX | Yeah |
| 20:38.04 | [TK]D-Fender | COMPLETELY |
| 20:38.18 | *** join/#asterisk Howie69 (n=Owner@host-12-173-142-207.nctv.com) |
| 20:38.21 | Carlos_PHX | is happy to have heard the funniest thing all week this early in the week |
| 20:38.35 | Breal | They charge us $30 per sip account, but we can unlimited "seats" on the mgcp side |
| 20:39.13 | Carlos_PHX | [TK]D-Fender: What's the jbot code for cheap VoIP? |
| 20:39.30 | Carlos_PHX | ~cheap |
| 20:39.30 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:39.58 | Breal | We are looking at $30 per line, at 100 lines |
| 20:40.22 | Carlos_PHX | And how much for MGCP? |
| 20:40.43 | Breal | Thats $3000 for 100 sip accounts per month. Or, we can have unlimited seats that are MGCP included in our T-1 |
| 20:41.05 | Carlos_PHX | If you are behind a server, you don't buy "seats" |
| 20:41.14 | Carlos_PHX | You buy SIP channels or what some call trunks. |
| 20:41.29 | Breal | This is a hosted solution that is not in our building. |
| 20:41.46 | [TK]D-Fender | its all just channels.. completely retarded |
| 20:41.48 | Carlos_PHX | So then you're not doing asterisk to asterisk. |
| 20:41.52 | Howie69 | I have a asterisk box, and was using IAX2. But when I switch to SIP, It works as well. However, if I remove the router ( box straigh to cable modem ), IAX2 still works but all SIP registrations fail |
| 20:41.58 | Breal | We just want to setup an inhouse server that would ast as a mgcp client and make calls through it |
| 20:41.59 | Howie69 | if I put the router back, SIP works again |
| 20:41.59 | Carlos_PHX | You're doing MGCP phone to MGPC gateway? |
| 20:42.03 | Howie69 | any ideas? |
| 20:42.04 | Breal | No |
| 20:42.05 | [TK]D-Fender | ]a call is a call is a call. Do you charge more for talking to them in SPANISH? |
| 20:42.20 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 20:42.24 | Carlos_PHX | Breal: If you have a server, you no longer have hosted. |
| 20:42.34 | Breal | Its how bandwith.com bills us |
| 20:42.36 | Carlos_PHX | BTW, that's a really high per seat price. |
| 20:42.45 | Breal | We are not hosting our own service, they are |
| 20:42.57 | Carlos_PHX | If you run a server, you are not hosted. |
| 20:43.01 | Breal | We just want to make an asterisk server connect in and pretend to be a phone so that we can make some calls through one of the accounts. |
| 20:43.04 | Carlos_PHX | If you are hosted, you do not have a server. |
| 20:43.15 | Carlos_PHX | Dude, then you do the same with SIP |
| 20:43.28 | [TK]D-Fender | Breal: Maybe you can run FreeSWITCH as a media gateway or something... |
| 20:43.43 | Carlos_PHX | Or shop for someone with a better seat price. |
| 20:44.06 | Breal | I dont think you understand. We are just trying to make asterisk act like a mgcp phone and register and an mgcp device on an mgcp account on an asterisk box |
| 20:44.23 | Carlos_PHX | Right. You don't understand that you can do the same with SIP. |
| 20:44.26 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 20:44.34 | *** join/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net) |
| 20:44.40 | Breal | They dont offer SIP with this package. Its an addon |
| 20:44.56 | Breal | And its extremely expensive. We already have seats we arent using, and this would be a better solution for us. |
| 20:44.56 | Qwell | get a different package |
| 20:44.56 | Carlos_PHX | You have a SIP account now, right? |
| 20:45.05 | Breal | No |
| 20:45.11 | Breal | no |
| 20:45.16 | Carlos_PHX | You said you have seats you are not using? |
| 20:45.19 | Carlos_PHX | SIP seats? |
| 20:45.20 | Carlos_PHX | No? |
| 20:45.24 | Breal | MGCP seats |
| 20:45.29 | mikealeonetti | lawn chairs |
| 20:45.35 | mikealeonetti | or sofas? |
| 20:45.48 | Carlos_PHX | At that price, Aeron |
| 20:46.16 | mikealeonetti | oh word |
| 20:46.29 | neurosys | Are aerons really that nice? |
| 20:46.42 | Carlos_PHX | Not really. I like my high-end Steelcase chairs better. |
| 20:48.47 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 20:49.27 | Qwell | Breal: If the calls are asterisk<>asterisk, how does bandwidth even get involved? |
| 20:50.11 | [TK]D-Fender | Breal: We do understand. You seem not to have registered what I sai earlier. * cannot act like an MGCP PHONE. |
| 20:51.17 | awk_r | ~book |
| 20:51.18 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:52.06 | neurosys | Good book too ;) |
| 20:52.37 | awk_r | very, but jbot doesn't exist in #asterisk-gui :-) so i had to spam #asterisk and copy/paste |
| 20:52.42 | [TK]D-Fender | mikealeonetti: Raw-Cat Lawn Chairs |
| 20:53.10 | mikealeonetti | [TK]D-Fender: did, I don't know what those are |
| 20:53.13 | mikealeonetti | dude8 |
| 20:53.32 | awk_r | [TK]D-Fender, that sounds too close to ClawFoot Tub |
| 20:53.37 | awk_r | related? |
| 20:53.40 | [TK]D-Fender | mikealeonetti: Say it out load a few times until you feel silly... |
| 20:53.57 | mikealeonetti | [TK]D-Fender: haw haw haw haw |
| 20:54.20 | Breal | Qwell: bandwidth.com hosts an asterisk server for us |
| 20:54.32 | Qwell | but you own the server? |
| 20:54.32 | mikealeonetti | speaking of which |
| 20:54.35 | mikealeonetti | I need to call them |
| 20:54.42 | mikealeonetti | I can't make outgoing calls |
| 20:54.48 | Qwell | and manage it? |
| 20:54.50 | Breal | no |
| 20:54.52 | Breal | they own the server |
| 20:55.04 | Qwell | do you manage it? |
| 20:55.04 | mikealeonetti | I'm going to be here forever tonight |
| 20:55.09 | Breal | we just want to make an inhouse server connect to it to place calls... pretend to be a phone, which apparently cant be done |
| 20:55.12 | Breal | no we dont manage it either |
| 20:55.14 | [TK]D-Fender | mikealeonetti: No you're not... |
| 20:55.34 | mikealeonetti | [TK]D-Fender: until 6 probably |
| 20:55.38 | mikealeonetti | that's forever enough |
| 20:55.43 | [TK]D-Fender | mikealeonetti: See, looking better already |
| 20:56.00 | mikealeonetti | I could be doing tons better things, though |
| 20:56.15 | mikealeonetti | like working out |
| 20:56.19 | mikealeonetti | or watching Buffy the Vampire Slayer |
| 20:57.47 | [TK]D-Fender | mikealeonetti: I'd like to work her over... |
| 20:58.16 | mikealeonetti | lol |
| 21:00.45 | *** join/#asterisk saftsack (n=oliver@e179059052.adsl.alicedsl.de) |
| 21:01.03 | saftsack | hi is there any channel for linux wireless? |
| 21:01.18 | saftsack | im talking from an irc channel |
| 21:01.37 | [TK]D-Fender | saftsack: ##networking ##linux |
| 21:02.08 | VoIPDontCry | HI |
| 21:02.14 | VoIPDontCry | ANYBODY CAN HELPME ¿? |
| 21:02.24 | jblack | 3Woot. Looks like DOW closed below 8K. |
| 21:02.38 | neurosys | WOW! |
| 21:03.23 | jblack | VoIPDontCry: Your capslock is on. Try asking the question directly. |
| 21:03.42 | VoIPDontCry | I asked yet, but I will try again |
| 21:04.29 | VoIPDontCry | How can I configure my Elastix for to see CallerID in my X-Lite ? |
| 21:04.37 | VoIPDontCry | I have a PSTN trunk onlu |
| 21:04.39 | VoIPDontCry | only |
| 21:05.00 | [TK]D-Fender | VoIPDontCry: GUI's are NOT supported here. |
| 21:05.07 | [TK]D-Fender | VoIPDontCry: Go ask in #freepbx |
| 21:05.48 | mikealeonetti | if I try to restart the Cisco 7960 phones with the *6+settings key and they are still retaining the old settings, is there a way to make them use the TFTP? |
| 21:11.17 | VoIPDontCry | well, how can I configure asterisk for that |
| 21:11.26 | VoIPDontCry | ? |
| 21:13.33 | [TK]D-Fender | VoIPDontCry: if you're using a zaptel/DAHDI compatible card the options are "usercallerid=yes", "callerid=asreceived". |
| 21:14.08 | SuPrSluG | what causes a phone that worked earlier to give a 401 unauthorized. |
| 21:14.13 | *** join/#asterisk bluregard (n=matt@66.251.248.24) |
| 21:15.35 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 21:15.53 | [TK]D-Fender | SuPrSluG: Changing the auth |
| 21:16.16 | *** join/#asterisk c4t3l (i=rcallico@equinox.alluvium.com) |
| 21:17.01 | VoIPDontCry | what config file is for zaptel/DAHDI ? |
| 21:17.03 | VoIPDontCry | zaptel.conf ? |
| 21:17.27 | SuPrSluG | same username/secret |
| 21:17.29 | hansin | My work has a load of Polycom SoundPointIP 500 phones. One problem is that they have the MGCP firmware loaded. I know I can update to SIP, but because of limited memory, I can only go to bootrom 3.2.2 and SIP 2.1.3 (though switching from MGCP to SIP I guess is not encouraged by Polycom). What are peoples take on these phones? Is there a used market for any of these, given the memory limitations (I am sure the 501 is more popular)? |
| 21:18.15 | *** join/#asterisk ManxPower (n=manxpowe@70.sub-70-223-105.myvzw.com) |
| 21:18.25 | Howie69 | still |
| 21:18.31 | Howie69 | no SIP registrations work |
| 21:18.32 | SuPrSluG | hansin: only way to get money out of 'em is sip. |
| 21:18.35 | [TK]D-Fender | hansin: Is that a metric load, or an imperial load? |
| 21:18.36 | Howie69 | if i put the router back, they work fine |
| 21:18.44 | Howie69 | IAX works fine too |
| 21:18.46 | Howie69 | any ideas? |
| 21:19.13 | VoIPDontCry | [TK]D-Fender: what config file is for zaptel/DAHDI ? zaptel.conf ? |
| 21:19.16 | ManxPower | Howie69: sounds like the classic problem is skipping some step in the ~sipnat page. |
| 21:19.20 | ManxPower | ~sipnat |
| 21:19.21 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:19.40 | hansin | [TK]D-Fender: Probably a couple hundred of them maybe, so I guess that would be metric ;) |
| 21:19.43 | [TK]D-Fender | VoIPDontCry: for Zaptel its /etc/asterisk/zapata.conf |
| 21:19.58 | [TK]D-Fender | hansin: Well consider them worthless w/o SIP |
| 21:19.59 | Howie69 | ManxPower: you missed the point :) I removed the sip nat stuff, and the SIP registration says 'No Nat' |
| 21:20.08 | VoIPDontCry | thx D-Fender! |
| 21:20.25 | ManxPower | Howie69: so there is no nat involved? |
| 21:20.26 | hansin | SuPrSluG: If they can be upgrade to SIP in-house, would it be worth the effort? |
| 21:20.31 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 21:20.39 | hansin | [TK]D-Fender: I guess same question to you... |
| 21:20.47 | Howie69 | ManxPower: not when I remove the router, no |
| 21:20.51 | SuPrSluG | if ya plan on ebayin em |
| 21:20.56 | Howie69 | ManxPower: NIC -> Cable Modem |
| 21:21.09 | hansin | SuPrSluG: Okay, I suppose I can check going rates on eBay. Thanks. |
| 21:21.23 | [TK]D-Fender | hansin: Yes it'd be worth it. Harder to sell something the client has to clean up after |
| 21:24.03 | SuPrSluG | [TK}D-Feender:what's this auth you speak of |
| 21:25.01 | SuPrSluG | how did I change the auth? 2 phone w/ exact same config regiter |
| 21:25.43 | SuPrSluG | man my typing sux |
| 21:25.49 | SuPrSluG | register |
| 21:26.34 | root52 | Hey all, when i do #sip show peers i notice that in the status column my two SIP trunks are monitored but none of my phones are. Is that a config problem with the phones? |
| 21:27.46 | Howie69 | ManxPower: I'm trying to weed through some sip logs for you |
| 21:29.01 | Howie69 | too many |
| 21:29.07 | Howie69 | too many sip extensions registering |
| 21:32.08 | *** join/#asterisk holos (n=cosmond@209.167.131.35) |
| 21:32.28 | holos | Anyone have any ideas how to test a toll free in Singapore? or anyone here from Singapore? |
| 21:33.24 | neurosys | How do you pause output in the CLI? |
| 21:34.19 | mikealeonetti | if the Cisco 7960 phone is just reset, is downloading the SIPDefault.cnf and the ./SIPmac.cnf but not fetching the firmware and it keeps restarting, is it broken or did I do something wrong? |
| 21:37.30 | *** part/#asterisk holos (n=cosmond@209.167.131.35) |
| 21:38.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:40.10 | [TK]D-Fender | ok, checkout time. Later all |
| 21:40.35 | *** part/#asterisk VoIPDontCry (n=seba0606@adsl190-28-133-98.epm.net.co) |
| 21:42.30 | *** join/#asterisk dippo (n=cwage@209.149.57.26) |
| 21:42.33 | M1s3ry | neurosys, exit the CLI, otherwise set your terminal to not scroll along if you are looking through previous information. |
| 21:42.53 | dippo | hi. according to this page (http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2) when using GSM you should see roughly 64kbps for one call and ~14kbps for each additional call |
| 21:43.03 | dippo | but I am definitely seeing it double up to 120Kbps for the second call.. |
| 21:43.11 | dippo | is that page wrong or does this indicate a problem in my configuration? |
| 21:43.22 | dippo | this is bandwidth being used by the IAX2 trunk to a provider, btw |
| 21:44.20 | dippo | so, sip handset -> (g.711) -> PBX -> (gsm over IAX2) -> trunking provider |
| 21:46.18 | Howie69 | ManxPower: yes, I double and triple checked, I have nat turned off everywhere |
| 21:47.15 | Qwell | dippo: that's when using IAX2 trunking, I believe. |
| 21:47.24 | Qwell | though, those figures seem way off |
| 21:47.32 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-a61d36e5d527d381) |
| 21:48.31 | talirk81 | when i use exten => s,n,Record(${FILE}:wav,10,30) in a dial plan (${FILE} is just a file name. Why does the record app immedaitly finish and after the beep and not let me record a message? |
| 21:49.12 | dippo | i am using IAX2 trunking |
| 21:49.21 | dippo | it's not a big deal, but i was just curious |
| 21:49.33 | dippo | 64Kbps + only 14 for each additional call would be nice, but i am definitely not seeing it |
| 21:49.44 | dippo | someone told me GSM uses a delta vs. the original signal for additional calls, hence the minimal increase |
| 21:55.25 | *** join/#asterisk puppet (n=iriche@c83-251-23-219.bredband.comhem.se) |
| 21:55.53 | puppet | Hmm, I got a problem here, Asterisk answers a call, but my cellhpone don't start ticking seconds = It don't answer and I don't get any sounds at all |
| 21:56.43 | puppet | http://pastebin.ca/1261739 |
| 21:57.04 | *** join/#asterisk ix33 (n=ix@7b.85.b6.static.xlhost.com) |
| 21:57.52 | giovani | eww, it's trixbox |
| 21:58.02 | giovani | why is the peer unknown, did you not define it? |
| 21:58.03 | puppet | giovani: yeah but im swapping it out for pbx in a flash like now |
| 21:58.17 | giovani | why not just run asterisk plain? |
| 21:58.38 | puppet | giovani: did that before, but i want a easy way to add IVR menu, fax and all thoose stuff |
| 21:58.44 | puppet | giovani: without spending 20h on that |
| 21:58.48 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 21:58.50 | giovani | heh |
| 21:58.55 | giovani | shouldn't take 20 hours to write in an IVR |
| 21:58.56 | giovani | but, ok |
| 21:59.24 | puppet | giovani: well recording and all that, want it easy easy eay ;) i know it dont take 20h but fax takes a while longer to get it good, i have set it up before, but still, easier to have a interface |
| 21:59.38 | giovani | but for "easy easy easy" you get odd problems like this |
| 21:59.52 | puppet | giovani: true |
| 21:59.53 | giovani | go to #trixbox |
| 22:00.08 | giovani | I have no clue what's going on in this log, they have 20 checks going on |
| 22:00.13 | giovani | did you not define the peer? |
| 22:00.17 | giovani | because it says it's an unknown peer |
| 22:00.23 | puppet | yeah i know its strange |
| 22:00.33 | giovani | well ... did you define it? |
| 22:00.37 | puppet | yeah i did |
| 22:00.41 | puppet | according to trix |
| 22:00.47 | giovani | well ... talk to them |
| 22:00.49 | puppet | ehh sc-wage this im uninstalling it |
| 22:00.55 | giovani | or roll asterisk yourself -- it's not hard |
| 22:01.05 | puppet | nah it's easy ;) |
| 22:01.10 | puppet | did some labbing with realtime mysql before |
| 22:02.42 | puppet | giovani: is there ANY dist that offers good done packages? or is it still best to compile self |
| 22:02.49 | mikealeonetti | for variables to be used in extensions.conf do I have to use setvar=EXTERNALID=number in sip.conf? |
| 22:03.03 | giovani | puppet: I use the ubuntu package, works perfectly fine |
| 22:03.07 | puppet | cool |
| 22:03.15 | giovani | no idea about other distros |
| 22:03.20 | puppet | well i love ubuntu |
| 22:03.27 | giovani | I literally have a system up and running in under 25 minutes |
| 22:03.29 | puppet | so it's fine, you sue freepbx or anything like that or just straight off? |
| 22:03.35 | giovani | absolutely not |
| 22:03.38 | giovani | don't use that crap |
| 22:03.44 | puppet | sucks that bad? |
| 22:03.48 | giovani | yes |
| 22:03.54 | giovani | I clean out all of the configs as well |
| 22:04.03 | giovani | the comments are lengthy, makes it difficult to read |
| 22:04.15 | puppet | Got old configs but they are like 1.1 or 1.2 |
| 22:04.18 | giovani | ubuntu also has the zaptel driver in a package |
| 22:04.25 | giovani | just gotta compile it using m-a |
| 22:05.33 | ix33 | my 40-extension 1.4.19.2 install was up & running at one point for 10 weeks straight under moderate load |
| 22:05.51 | ix33 | is there a recommendation out there against running so long? |
| 22:06.02 | puppet | giovani: cause all I really need is a simple IVR, Voicemail with E-mail, Fax recieving, sure fax sending would be nice but yeah |
| 22:06.07 | *** part/#asterisk dippo (n=cwage@209.149.57.26) |
| 22:06.19 | giovani | puppet: I haven't done any fax work |
| 22:06.32 | giovani | how simple is the ivr? |
| 22:06.40 | M1s3ry | ix33, you're running linux... not really |
| 22:06.53 | puppet | Very, Press 1 to leave a message, press 2 for contact information press 3 for english.. kinda ;p |
| 22:07.08 | giovani | puppet: should be able to write it in under 20 minutes |
| 22:07.09 | tzafrir_laptop | giovani, well, sort of. I sometimes even get some bug reports from ubuntu users regarding the package |
| 22:07.11 | giovani | (and troubleshoot it) |
| 22:07.13 | puppet | giovani: 10 min ;P |
| 22:07.20 | puppet | that is easy the fax is harder |
| 22:07.24 | tzafrir_laptop | But ubuntu bug reports remain largely unfixed |
| 22:07.39 | giovani | tzafrir_laptop: it's in universe, it's an unsupported package from debian, that's why |
| 22:07.40 | ix33 | M1s3ry: thanks |
| 22:07.54 | giovani | you'd need to know how ubuntu packages applications and supports them before criticizing the package |
| 22:08.05 | tzafrir_laptop | Ubuntu asterisk and zaptel packages are universe. That is: they just take whatever there was in Debian Sid at the time of the release |
| 22:08.17 | giovani | ... heh, yes, I know |
| 22:08.17 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:08.36 | Akiyuki | Can someone recommend a high quality sip trunk line for a call center, based in the USA? |
| 22:08.49 | puppet | doing that now reinstalling to ubuntu :) thanks for setting me straight gio ;D why do i try to take the easy road when i know how to do it |
| 22:09.16 | giovani | puppet: it's not easier in the long run with trixbox -- their configs are messy, and difficult to hand-edit |
| 22:09.24 | Carlos_PHX | Akiyuki: Well, we're an ITSP and server a few call centers, with strong uptime and call quality. |
| 22:09.25 | ix33 | followup: so during the course of that 10 weeks apparently a call got stuck in a continually 'open' state where 'core show channels' would show one guy dialed into VoiceMailMain() the whole time. is there a way to force a connection like that down? |
| 22:10.35 | Carlos_PHX | ix33: I've heard of this from many people and there doesn't seem to be a good solution. We have our servers reboot on a weekly basis and that cleans out whatever is going on, including the occasional Asterisk memory leak. |
| 22:10.35 | Akiyuki | Carlos_PHX: What is ITSP? |
| 22:10.41 | Qwell | ~itsp |
| 22:10.41 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:10.47 | Carlos_PHX | ITSP = te |
| 22:10.48 | bkruse | Qwell: beat me to it. |
| 22:10.49 | Carlos_PHX | Well, there |
| 22:10.52 | Akiyuki | ty |
| 22:10.57 | Akiyuki | What's the name of the company? |
| 22:11.02 | Carlos_PHX | www.televolve.com |
| 22:11.26 | Carlos_PHX | If you want to chat on a private channel let me know, rather not do sales here. But we do provide what you're looking for. |
| 22:11.49 | ix33 | Carlos_PHX: i had my * 'restart when convenient' every morning at 3, but this one call kept it up for 10 weeks |
| 22:12.05 | Carlos_PHX | Yes, that and the memory leaks are why we do full reboots. |
| 22:12.21 | Carlos_PHX | I know, Linux doesn't need to be rebooted blah blah you're doing it wrong. But...it's what works. |
| 22:12.24 | Aces12 | i have a sangoma a200 it looks like i have all the drivers installed correctly and the zap show channels shows all my ports.. i have these ports connected to my pots lines.. i have setup a default route, but everytime i go to dial i get "the number is not in service, please check your number and try again message" |
| 22:12.30 | ix33 | i guess it's not likely that i'll lose an important call at 3 AM |
| 22:12.36 | Carlos_PHX | Exactly. |
| 22:12.49 | ix33 | ok i'll omit the 'when convenient' |
| 22:12.52 | Carlos_PHX | We don't have any 24x7 customers yet, so no issue for us. |
| 22:14.21 | ix33 | it was still rock solid for those 10 weeks, but it was spinning one CPU 100% when i found it |
| 22:14.49 | Carlos_PHX | Yeah, someone I work with tells me he sees that a lot. |
| 22:14.54 | Carlos_PHX | We don't, luckily. |
| 22:16.15 | ix33 | so i saved my old company $14k on a PBX and all i got was a jar of M&M's |
| 22:16.47 | ix33 | thanks *! |
| 22:17.50 | Akiyuki | Carlos_PHX: Ok, what channel? |
| 22:23.37 | Carlos_PHX | Anybody here doing direct CNAM dips? |
| 22:24.37 | Carlos_PHX | Akiyuki: Sent you a private message, but saw no reply, did you get that? |
| 22:26.45 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.43) |
| 22:31.06 | Akiyuki | damn |
| 22:31.13 | Akiyuki | i gotta leave, we can do this tomorrow, will you be in here? |
| 22:31.19 | Akiyuki | emergency w/ my wife and kid and daycare issue |
| 22:31.46 | Akiyuki | email me, jfreeman@homeinsurance.com |
| 22:31.47 | Akiyuki | thanks |
| 22:33.13 | *** join/#asterisk telecos (n=sergio@84.166.219.87.dynamic.jazztel.es) |
| 22:34.40 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:37.13 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 22:43.46 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 22:53.40 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 22:56.38 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 22:58.35 | beek | [TK]D-Fender: Good evening. |
| 22:58.45 | [TK]D-Fender | beek: Hello |
| 22:58.56 | beek | [TK]D-Fender: Would you believe the major issues I've been having with caller ID are the fault of the telco? |
| 22:59.10 | beek | [TK]D-Fender: It finally hit me to try a callerid box and see if it worked -- it didn't. |
| 22:59.17 | Qwell | beek: A telco at fault? I'm SHOCKED! |
| 22:59.30 | beek | Qwell: Did I note a hint of sarcasm? |
| 22:59.32 | [TK]D-Fender | UN-POSSIBLE! |
| 22:59.52 | xorl | So I have these Cisco 7960's here, any idea on what logo_url's bitmap requirements are bit wise? |
| 22:59.57 | [TK]D-Fender | beek: More like an entire concerto ;) |
| 23:00.04 | ManxPower | "The Catholic Church is never wrong!" "What about the policy of non-involvement in the Holocust?" --Dogma |
| 23:00.14 | beek | In four parts. |
| 23:00.31 | xorl | [TK]D-Fender: Figured out my issue earlier. |
| 23:00.39 | xorl | wasn't asterisk, or vitelity. |
| 23:01.01 | xorl | My ISP was dropping our RTP packets for some reason. |
| 23:01.20 | ManxPower | xorl: Welcome to the world of voice on the internet |
| 23:01.32 | xorl | Yeah.. |
| 23:01.38 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 23:01.42 | [TK]D-Fender | alksjdhjkdlasjdhf sadf l NO CARRIER |
| 23:05.13 | Carlos_PHX | Did your ISP say...why... |
| 23:05.26 | Carlos_PHX | Is it Airband by any chance? |
| 23:06.21 | xorl | oh not now, we had it all routing through Cox |
| 23:06.25 | scooby2 | man why do I get the crap jobs. this client wants their queue to setup to ring for 60 seconds if no one is available but to ring indefinitely if someone is available. |
| 23:06.34 | xorl | Carlos_PHX: But oddly enough we do *use* airband |
| 23:06.40 | xorl | <PROTECTED> |
| 23:06.45 | Carlos_PHX | Shocker |
| 23:06.55 | xorl | They known for blocking RTP? |
| 23:06.58 | Carlos_PHX | They don't like people using their pipes to get to other service providers. |
| 23:07.09 | xorl | ah nice. |
| 23:07.20 | Carlos_PHX | One of our customers had their RTP de-prioritized by them. |
| 23:07.39 | Carlos_PHX | So like, Sirius streaming was higher priority than voice. |
| 23:08.04 | bmoraca | scooby, i had one request to have calls roll through 5 different queues of people...3 seconds, then 5 seconds, then 10 seconds, then 5 seconds, then 5 seconds...then go to autoattendant. how's that for suck? |
| 23:08.08 | Carlos_PHX | It was fun to troubleshoot. |
| 23:08.11 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-176-170.phil.east.verizon.net) |
| 23:09.18 | [8none1] | xorl: http://www.google.com/search?q=cisco+7960+bmp top link |
| 23:09.32 | [8none1] | xorl: Google is your friend |
| 23:09.39 | xorl | Yes, yes it is. |
| 23:09.44 | xorl | I was googling but failed. |
| 23:10.01 | xorl | Still pretty anoyed about my whole phone system going down for hours for no reason what so ever. |
| 23:10.04 | bmoraca | 7940 and 7960 BMPs are NOT fun. |
| 23:10.12 | bmoraca | and very unforgiving |
| 23:10.21 | Carlos_PHX | xorl: Did they have an excuse? |
| 23:10.25 | xorl | Carlos_PHX: No. |
| 23:10.30 | xorl | They didn't even get back to us lol |
| 23:10.38 | xorl | I have it tunneling right now. |
| 23:10.50 | Carlos_PHX | Sheesh |
| 23:11.38 | ManxPower | hugs his PRIs |
| 23:12.03 | Qwell | an ISP not admitting fault? I am SHOCKED! |
| 23:12.05 | Qwell | smirks |
| 23:12.06 | xorl | ManxPower: We're going to get that setup in here soon. |
| 23:12.15 | Carlos_PHX | PRIs are for sissies. |
| 23:12.45 | Carlos_PHX | xorl: Just use the right ISP, issues like this shouldn't be common. |
| 23:12.45 | ManxPower | Carlos_PHX: And VoIPoInternet are for people that want to be castrated by their users. |
| 23:13.09 | Carlos_PHX | Heh, right, well, as an ITSP I need not express my disagreement. |
| 23:13.15 | xorl | So, gimp can't save 8-bit bmps? |
| 23:15.02 | [8none1] | ManxPower: have you ever heard of a MMPRI? |
| 23:15.13 | [8none1] | MMPRI = Multi-Market PRI |
| 23:15.13 | denon | just use photoshop, it's so much easier. |
| 23:15.34 | ManxPower | [8none1]: I can imagine what it would be. Normally provided by CLECs |
| 23:15.52 | Carlos_PHX | Is that like an MMRPG |
| 23:16.16 | xorl | denon: Don't have any machines with photoshop here. |
| 23:16.16 | [8none1] | ManxPower: You can create site diverse PRI trunk groups |
| 23:16.25 | [8none1] | Carlos_PHX: yeah, exactly |
| 23:16.41 | ManxPower | [8none1]: why not just route that stuff over a QoS'd lan. |
| 23:16.43 | bmoraca | a PRI is far cheaper than a data connection over which you could run 23 simultaneous calls... |
| 23:16.58 | Carlos_PHX | Not in the US. |
| 23:17.01 | [8none1] | We have 6 total 3 in each of our primary offices. |
| 23:17.16 | voxter | I can run 23 simultaneous calls over DSL, and DSL is cheaper than PRI |
| 23:17.30 | bmoraca | i'd like to see that. no, really |
| 23:17.32 | Carlos_PHX | T1 is cheaper than PRI, and that's good for nearly 70 calls. |
| 23:17.34 | [8none1] | If will fill or have an outage on one site the calls roll to the other site will all the DID info intact |
| 23:17.44 | Carlos_PHX | bmoraca: Seriously? Because we do it all the time. |
| 23:17.47 | [8none1] | Then we route to the proper site via VoIP trunks |
| 23:17.51 | voxter | I have multiple customers doing more than 23 calls on DSL. |
| 23:17.54 | Carlos_PHX | You're not in the US, I have to assume? |
| 23:18.17 | Carlos_PHX | We have a happy customer doing 15 on a $65 cable connection. |
| 23:18.23 | bmoraca | i'm having a hard time believing you can get 10 calls, let alone 70 calls over a 1.5mbit connection. |
| 23:18.39 | Carlos_PHX | bmoraca: Um, why? |
| 23:18.47 | bmoraca | uhm, because of the bandwidth required. |
| 23:18.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:18.56 | voxter | bmoraca: Two very important things to consider: G729 and IAX Trunking |
| 23:18.58 | Carlos_PHX | The bandwidth required is 22k per call, so... |
| 23:19.15 | [8none1] | thinks bmoraca is watching YouTube on his T1 while making phone calls |
| 23:19.23 | Carlos_PHX | Man, I hate it when I find out something I've been doing for years is impossible. |
| 23:19.31 | voxter | Carlos_PHX: me too. |
| 23:19.32 | ManxPower | bmoraca: Do the math. UDP overhead for RTP is about 16Kkbps per channel, GSM is about 16kbps per channel. |
| 23:19.34 | bmoraca | 22k per call? on what codec? IAX2 only works for trunking. |
| 23:19.38 | Carlos_PHX | I think he's watching redtube |
| 23:19.54 | bmoraca | or softphones |
| 23:20.00 | ManxPower | with iax2 TRUNKING overhead is about 16k for ALL the channels combined, GSM at 16k |
| 23:20.26 | voxter | I spoke about this at astricon this year, i never knew i was lying to everyone! |
| 23:20.35 | ManxPower | with G729 the codec is even smaller |
| 23:20.44 | bmoraca | if I have a hosted PBX with remote phones, even connecting with g.729, there is NO WAY i am getting 10 simultaneous calls over a T1. |
| 23:20.49 | Carlos_PHX | I guess I should turn off that call center on the T1, we can't possibly handle their 60 concurrents. |
| 23:20.56 | voxter | I can cram about 85 calls safely onto a 768k dsl connection with g729 and iax2 trunking. |
| 23:21.03 | Carlos_PHX | bmoraca: You're wrong. |
| 23:21.08 | Carlos_PHX | Period. |
| 23:21.09 | ManxPower | voxter: The fact you have customers doing 23 calls on a DSL means you are crazy, not that you are wrong. |
| 23:21.34 | Carlos_PHX | "This is probably going to suck, but feel free to try the cheap way first." |
| 23:21.36 | ManxPower | bmoraca: Just how much bandwidth do you think a call takes??????????????????? |
| 23:21.42 | voxter | ManxPower: local circuit DSL, never hits the internet :) |
| 23:21.49 | bmoraca | voxter, that defies math. there is no way you are getting <10k per call bandwidth. |
| 23:22.03 | Carlos_PHX | bmoraca: We have hosted customers with 70+ phones on T1. |
| 23:22.05 | voxter | bmoraca: why? g729 uses 8kbit |
| 23:22.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:22.21 | Carlos_PHX | Call centers with 60 concurrent on T1. |
| 23:22.59 | ManxPower | bmoraca: then show us your math |
| 23:23.09 | ManxPower | We will show you where you are wrong. |
| 23:23.53 | bmoraca | g.729 is ~30kbps for two-way communication, by conservative estimation. at least from everything i've ever read and every benchmark i've ever seen. |
| 23:24.12 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:24.17 | Carlos_PHX | thinks bmoraca learned math from Verizon. |
| 23:24.26 | ManxPower | you count bandwidth usage in ONE direction |
| 23:24.28 | jblack | harsh words |
| 23:24.47 | Carlos_PHX | bmoraca: Can you explain why RIGHT NOW I have 47 concurrents on a T1? |
| 23:25.11 | voxter | has 39 currently on a DSL customer |
| 23:25.19 | voxter | right now. |
| 23:25.33 | bmoraca | show me the core show channels. |
| 23:25.37 | bmoraca | i'm curious. |
| 23:25.38 | jblack | And if he's using local dsl, he could be goign full 10 megabit, bidi. |
| 23:25.48 | ManxPower | bmoraca: G.729 uses 8k, if you are running that on SIP then you have about 16k of overhead. |
| 23:26.04 | jblack | You can fit a lot of 8 kilobit channels into 10 megabits. |
| 23:26.07 | ManxPower | Now if you want to count BOTH directions of the call then you need to count BOTH directions of a T-1. |
| 23:27.40 | jblack | Theoretically, one could get 1280 concerrent calls out of 10 megabit. Actual practice would be in the few hundred. |
| 23:27.40 | ManxPower | I'll paste a popular bandwidth calculator for you a in a momenty |
| 23:27.40 | bmoraca | jblack, i'd like to see where you're getting this magical 10mbit DSL circuit. |
| 23:27.45 | jblack | bmoraca: Any two dsl modems talking directly, back to back. |
| 23:28.25 | bmoraca | and that happens in production exactly when? in that instance, theoretical bandwidth is far, far higher (we use some that get nearly 100mbit) |
| 23:28.29 | jblack | You'd want to keep the cat-3 cable down to well less than a mile. |
| 23:28.30 | Carlos_PHX | We have 7mb symmetrical here. |
| 23:28.42 | ManxPower | bmoraca: : http://www.voip-info.org/wiki-Bandwidth+consumption |
| 23:28.50 | Carlos_PHX | And cable is 22 down / 4 up. |
| 23:29.14 | ManxPower | With MPLS you can pretty much get any bandwidth you want. |
| 23:29.15 | jblack | bmoraca: That's what he's using now, and I've implemented it on rare occasion for businesses, that have sites that are close together. |
| 23:29.21 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 23:30.05 | Carlos_PHX | bmoraca: I have to wonder why you are fighting people that are already doing what you say is impossible. |
| 23:30.48 | ManxPower | Carlos_PHX: especially considering the fact several of the people talking on this subject have been using Asterisk for more than 5 years |
| 23:30.51 | jblack | I think he considers consipiracy as more likely than him being wrong. |
| 23:31.08 | bmoraca | i'm not fighting. i'm saying that your estimates defy the mathematics that i've always held to be true. namely that 1.5mbit / 32kbps != 85. |
| 23:31.31 | jblack | 1.5mbit what? Who's talking 1.5 mbit? |
| 23:31.46 | bmoraca | they're talking about pushing 85 concurrent calls over a T1 |
| 23:31.46 | [TK]D-Fender | bmwho said 32kbps? |
| 23:31.46 | ManxPower | well more correctly 1544kbps / 32kbps |
| 23:32.09 | saftsack | so g.729 isnt as good as announced if no iax2 trunking is used? |
| 23:32.09 | bmoraca | 32kbps is one-way g.729 bandwidth. |
| 23:32.23 | ManxPower | saftsack: what has been announced? |
| 23:32.32 | ManxPower | bmoraca: You. Are. Wrong. |
| 23:32.50 | Carlos_PHX | I hate it when theory beats reality. |
| 23:32.51 | [TK]D-Fender | bmoraca: Really? thats news... |
| 23:32.56 | ManxPower | go look at the damn SPECS. 8kbps is G.729 |
| 23:33.05 | bmoraca | 8kbps is the bitrate, yes |
| 23:33.08 | [TK]D-Fender | bmoraca: Where I come from its < 10kbps... |
| 23:33.11 | bmoraca | but that's NOT the only calculation. |
| 23:33.14 | ManxPower | I am sorry your numbers are not correct. |
| 23:33.26 | [TK]D-Fender | bmoraca: whos "math"? ;) |
| 23:33.32 | Carlos_PHX | Verizon |
| 23:33.36 | [TK]D-Fender | :D |
| 23:33.39 | ManxPower | bmoraca: no, it's not the only calculation. You would have about 16kbps of UDP overhead for RTP audio |
| 23:33.42 | saftsack | ManxPower, everbody says that g.729 is the best codec for low bandwith consumption. but g.729 has more bandwith than g.726 |
| 23:33.49 | saftsack | digium announced it |
| 23:34.01 | ManxPower | saftsack: CITE YOUR URL |
| 23:34.02 | jblack | A t1 is also symmetric. You get 1.54 megabits in each direction. |
| 23:34.05 | [TK]D-Fender | Carlos_PHX: There's your .02 CENTS worth ;) |
| 23:34.24 | saftsack | ManxPower, what do you mean? |
| 23:34.51 | ManxPower | "(5:33:49 PM) saftsack: digium announced it" OK. Show me the announcement |
| 23:35.55 | ManxPower | For bandwidth calculations it is usually good to separate the codec usage and the UDP overhead |
| 23:36.04 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000374.dsl.bell.ca) |
| 23:36.46 | saftsack | they sell it ... |
| 23:37.06 | ManxPower | expecially because that overhead can vary with SIP .vs. IAX2 .vs. IAX@ Trunking |
| 23:37.11 | ManxPower | saftsack: SHOW ME |
| 23:37.22 | saftsack | yes i mentioned it before that it does vary |
| 23:37.33 | saftsack | but on a common sip connection it doesn't vary |
| 23:37.39 | saftsack | ManxPower, why do you shout? |
| 23:37.58 | Carlos_PHX | It might be interesting to compare all the wild connections we all use. My current favorite is Wi-MAX, not so crazy but some people say it doesn't work. |
| 23:38.12 | ManxPower | saftsack; for one thing you are not listening. You said "ManxPower, everbody says that g.729 is the best codec for low bandwith consumption. but g.729 has more bandwith than g.726" and "digium announced it". |
| 23:38.35 | ManxPower | Well I say if Digium announced that g.729 uses more bandwidth than g726 then there should be a URL for you to show me where Digium said that. |
| 23:38.47 | saftsack | http://www.digium.com/en/products/g729codec.php |
| 23:39.16 | ManxPower | looking now. |
| 23:39.19 | saftsack | first chapter. just g.729 is shown as the best choice if g711 is too large |
| 23:39.52 | ManxPower | saftsack: but you said g726. |
| 23:40.27 | saftsack | yes i said that g726 seems to be better than g.729 which is shown on the above calculator link |
| 23:41.17 | ManxPower | I'm obviously not understanding what you are trying to say. |
| 23:41.59 | saftsack | at the beginning i mentioned this: "saftsack> so g.729 isnt as good as announced if no iax2 trunking is used?" |
| 23:42.17 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 23:42.18 | saftsack | i want to say that g.726 seems to be the best codec for quality and bandwith going in common |
| 23:42.29 | Carlos_PHX | LPC10 beats them all. Sounds best too. |
| 23:43.26 | saftsack | voip-info.org says that it sounds robotic |
| 23:43.32 | saftsack | http://www.voip-info.org/wiki-LPC10 |
| 23:43.50 | Carlos_PHX | Yes, I think robots are hot, so it sounds best to me. |
| 23:44.01 | Carlos_PHX | When I call phone sex lines using LPC10... |
| 23:44.39 | Carlos_PHX | I'm wearing nothing but a Titanium thong and drinking 10w30... |
| 23:45.57 | *** join/#asterisk kjs (n=kjs@mx1.vm.bytemark.co.uk) |
| 23:47.05 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 23:51.17 | [TK]D-Fender | As long as we're talking Tricia Helfer or Grace Park, I'm game ;) |
| 23:51.26 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:52.34 | Carlos_PHX | I had to google it. I don't see no robots, but otherwise very nice. |
| 23:53.29 | [TK]D-Fender | Carlos_PHX: ... BattleStar Galactica.... they're CYLONS |
| 23:53.35 | Carlos_PHX | Ah |
| 23:53.55 | Carlos_PHX | Google just had human photos. Not that they were bad. |
| 23:54.05 | Carlos_PHX | http://hellcrazy.net/hella/ass/images/Grace-Park-7.JPG |
| 23:54.08 | [TK]D-Fender | Carlos_PHX: Significantly "not bad" |
| 23:54.21 | *** join/#asterisk Sgeo (i=897d6932@gateway/web/ajax/mibbit.com/x-bfca9c2ab095162b) |
| 23:54.25 | Carlos_PHX | I've done worse. |
| 23:54.40 | [TK]D-Fender | Carlos_PHX: You and 99% of the planet :) |
| 23:55.06 | Sgeo | I know this sounds inappropriate, but what OSI layer is call processing a part of? Looking through Wikipedia, it looks like Layer 3, but I don't know how helpful Wikipedia is in this case |
| 23:55.29 | ManxPower | Sgeo: layer 4 -- application layer |
| 23:55.41 | ManxPower | just like FTP or HTTP |
| 23:55.56 | xorl | bah bastard 7960 phone |
| 23:56.08 | Sgeo | Erm, isn't application layer layer 5 on the Internet model, and 7 on OSI? |
| 23:56.15 | Sgeo | ty though |
| 23:56.30 | [TK]D-Fender | It runs ON TCP & UDP. Do the math |
| 23:56.38 | Carlos_PHX | xorl: Those words just go together. Bastard and Cisco. |
| 23:57.16 | Carlos_PHX | wants layer 1 SIP, get rid of the overhead. |
| 23:57.25 | Sgeo | So cdmaOne uses layers 1, 2, and 7? |
| 23:57.26 | xorl | Carlos_PHX: Indeed. |
| 23:57.30 | ManxPower | Sgeo: I would actually have to look it up, but "layer 4 switches" are "switches" than handle the higher level protocols like HTTP, FTP, etc. Layer 3 switch would be something that understand IP and a layer 2 switch just understands the low level protocol like ethernet |
| 23:58.41 | xorl | Carlos_PHX: Damned 7960 is ignoring my tftpboot file |
| 23:59.39 | Sgeo | I don't know if "call processing" is a higher level protocol or not |
