00:00.25 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net) |
00:04.03 | SpaceBass | lol |
00:04.03 | SpaceBass | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5788092558&rd=1 |
00:05.39 | _DAW | Im bidding now!!! |
00:05.39 | SarahEmm | that link has been pasted in here like 8 times :P |
00:06.06 | SpaceBass | yeah, sounds great right? good luck getting new firmware! |
00:06.12 | _DAW | :) |
00:06.20 | opus__ | is anyone having problems dialing out with bubbahtel?? i can't make a call |
00:06.35 | opus__ | :) |
00:06.38 | SpaceBass | lol |
00:06.48 | SarahEmm | opus__: what error message are you receiving? |
00:06.56 | _DAW | too much tarnish |
00:09.33 | *** join/#asterisk doughecka (~Miranda@doughecka.user) |
00:09.40 | Jinxy | Jul 11 20:09:02 NOTICE[22827]: chan_sip.c:7733 handle_request: Registration from 'John <sip:200@192.168.0.254>' failed for '192.168.0.2' |
00:09.52 | Jinxy | but, I've got all the information right in sip.conf & the softphone |
00:14.02 | Nugget | I'll bet you don't. |
00:16.18 | Jinxy | well, I just uncommented the xlite config in sip.conf, and put the info from there in xlite softphone |
00:16.31 | Jinxy | executed reload on * console |
00:16.44 | Jinxy | I keep getting the registration error, but it doesn't state why it is failing |
00:16.47 | Jinxy | username/etc |
00:17.39 | gravemind | ergh, there are two mpg123 instances running |
00:17.44 | gravemind | one spawned the other apparently |
00:19.54 | Jinxy | ah ha, forgot a critical thing |
00:20.04 | Jinxy | [] has to be same as extension name |
00:20.10 | Malthus | :) |
00:20.29 | Malthus | no |
00:20.38 | Jinxy | no? why did it work then? |
00:20.43 | Malthus | magic |
00:20.57 | Jinxy | please be serious, I am trying to learn |
00:21.01 | Malthus | I have xlite/asterisk set here |
00:21.02 | Jinxy | the last thing I want is a detour |
00:21.20 | Malthus | the username is not the same extension |
00:21.47 | Malthus | I have a numeric extension but alphanumeric username |
00:22.03 | Malthus | [name] is the username by the way |
00:22.56 | Nugget | you two are both saying the same thing. |
00:22.58 | Jinxy | Malthus, ah ha |
00:23.05 | Jinxy | I was using extension as the username |
00:23.25 | Jinxy | so by naming xlite1 to extension, which I had defined as the username in xlite, I fixed my problem |
00:23.25 | Nugget | there is no direct relationship between a sip entry and an extension. |
00:23.52 | Jinxy | ok, so now I am logged in, what can I do to test if things are working? |
00:24.05 | Jinxy | any preconfigured automated extension that has some sexy lady on the other end? |
00:24.11 | Malthus | 800 |
00:24.12 | Jinxy | I've not specified any external things yet |
00:24.12 | Malthus | oops |
00:24.14 | Malthus | 600 |
00:24.26 | Jinxy | nothing happens |
00:24.42 | Malthus | oh |
00:24.52 | Malthus | thats part of the demo |
00:24.59 | Malthus | look at your extensions.conf |
00:25.07 | Malthus | search for demo |
00:25.18 | Malthus | good starting point |
00:25.22 | Jinxy | ya, I am going through the file right now, was hoping some of those things were already running |
00:25.45 | Malthus | is the demo in there? |
00:25.52 | Malthus | search for [demo] |
00:26.02 | Jinxy | done |
00:26.08 | Malthus | none? |
00:26.48 | Jinxy | nada |
00:26.55 | Jinxy | working |
00:27.19 | gravemind | would a system using sip phones and asterisk that works somewhat like a traditional PBX where each phone has an "extension" line, voicemail line, conference line, etc? if so, how do phones indicate to asterisk which line is being used... |
00:27.22 | Jinxy | that was just an extension I tried |
00:27.37 | gravemind | maybe the extension number passed along when dialing? |
00:28.07 | Malthus | gravemind: each SIP phone registers with asterisk |
00:28.12 | Jinxy | holy jeebus |
00:28.22 | gravemind | yes, but do sip phones come with multiple line buttons? |
00:28.28 | Jinxy | you have to press the dial button on xlite 2x |
00:28.36 | Malthus | haha |
00:28.56 | gravemind | like if you want to get your voicemail, you can press the voicemail button and pick up the receiver, and it'll dial that out |
00:29.10 | Druken | that's just a speed dial |
00:29.14 | Malthus | yeah |
00:29.33 | gravemind | what about if your phone is rang both by external lines and an internal extension, distinguishing between those etc |
00:29.35 | Malthus | you configure the phone to call a specific number for voicemail etc. |
00:29.53 | gravemind | you know, you've seen phones that have multiple lines calls can come in on, and pick which one to pick up right on the phone |
00:29.57 | Druken | gravemind: that's something you have to do in your dialplan |
00:30.15 | Malthus | gravemind: you're typing pretty fast btw |
00:30.20 | gravemind | i do that :P |
00:30.24 | *** join/#asterisk HaHaOok (~norman@60-240-240-66.tpgi.com.au) |
00:30.40 | gravemind | would that kind of thing be implemented as a speed-dial for the *8<exten> thing? |
00:31.16 | Malthus | basically asterisk can do all that and more |
00:31.19 | gravemind | sorry, im just trying to grapple with how asterisk would replace a traditional PBX system |
00:31.25 | Druken | your thinking of voip phones like network phones, and they don't work the same way |
00:31.27 | gravemind | particularly the "lines" coming into each phone |
00:31.36 | gravemind | hrm. |
00:31.59 | *** join/#asterisk mansing (~mansing@250-116.customer.cloud9.net) |
00:32.01 | harryvv | gravemind you can interface asterisk to databases and show grapically what it can do. But whats your intentions? |
00:32.29 | gravemind | i could see how it _could_ be done, but i doubt it is |
00:32.33 | *** join/#asterisk Cybertoy (~Cybertoy@ool-457852fa.dyn.optonline.net) |
00:32.55 | *** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
00:33.06 | Malthus | gravemind: it is |
00:33.16 | Malthus | eg with xlite |
00:33.19 | bdunn | EVERYTHING IS WORKING! Wow. Except one thing. Music on Hold is completely silent. Any ideas? |
00:33.27 | gravemind | something like when a call comes in on line X, you ring the phone with /X as the extension, and it flashes the light for line X |
00:33.36 | Malthus | mpg123 problems |
00:33.38 | gravemind | bdunn: have you enabled a source in musiconhold.conf? |
00:33.52 | gravemind | Malthus: is xlite a windows softphone? |
00:33.56 | Malthus | yes |
00:34.02 | gravemind | oh, too bad then |
00:34.05 | bdunn | gravemind - Ahhh... yes, but wrong path I think. I'll try that. |
00:34.06 | gravemind | im stuck with kiax for now :) |
00:34.16 | Malthus | and the sip hardphones are very similar |
00:34.22 | Malthus | kiax is nice |
00:34.25 | gravemind | bdunn: also note that it takes a full restart for musiconhold changes |
00:34.33 | Malthus | if kde is your thing |
00:34.36 | Malthus | I use iaxcomm |
00:34.39 | gravemind | its not, but it works. |
00:34.43 | Malthus | yes |
00:34.46 | gravemind | and looks good |
00:34.53 | Malthus | and it supports multiple lines |
00:34.56 | gravemind | only annoying thing is the dialpad being on a seperate tab |
00:35.24 | Malthus | so you can accept and make multiple calls at the same time |
00:35.34 | Malthus | callerid on each incoming call |
00:35.38 | gravemind | right |
00:35.41 | Malthus | oh |
00:35.50 | Malthus | there is an xlite for linux |
00:36.00 | gravemind | ... i don't think i want it though |
00:36.02 | gravemind | sounds like it'd suck. |
00:36.14 | iCEBrkr | Does EAGI even work? |
00:36.16 | Malthus | its as good as the windows version |
00:36.27 | Malthus | but I stick with iaxcomm |
00:36.46 | bdunn | OKay... music on hold... can I pick any MP3 for that? |
00:37.01 | Malthus | only reason I don't use kiax is because it would be out of place with my gone desktop |
00:37.07 | gravemind | bdunn: yes, but unless you have the real mpg123(not mpg321 as many distros have) it wont downsample corretly |
00:37.11 | gravemind | so you'll get a bunch of static |
00:37.22 | Druken | bdunn: you don't use a mp3, but a directory containing mp3s |
00:37.26 | gravemind | Malthus: purist :P |
00:37.29 | gravemind | i use fluxbox, nothing matches |
00:37.33 | bdunn | gravemind- I built this from cvs. Should it have the right stuff? |
00:37.41 | gravemind | bdunn: mpg123 isnt part of asterisk |
00:37.58 | Druken | gravemind: actually it is on the cvs |
00:37.59 | bdunn | gravemind - Ahh.. okay.. Could also be the problem then. :-) |
00:38.13 | Malthus | look at the wiki |
00:38.17 | gravemind | Druken: really? its nto even proper open source... |
00:38.31 | Malthus | http://voip-info.org |
00:38.34 | Druken | gravemind: pretty sure it is.... |
00:38.45 | gravemind | Druken: its open source, but its not /proper/, licensing issues to be had |
00:38.47 | Malthus | there is a lot of info on making mp3s work with asterisk |
00:39.13 | gravemind | i chose to reencode MOH at 8khz ahead of time to save cpu |
00:39.27 | Malthus | :) |
00:39.51 | Malthus | I chose to hum along in my head |
00:40.03 | Malthus | saved me configuration and even more CPU |
00:41.08 | gravemind | Malthus: long story short, im just wondering if hardware phones come with "line buttons" that light up when a given extension is ringing you |
00:41.08 | gravemind | like one lights up for an internal call, another for external line N, so on |
00:41.08 | gravemind | and you press that button to answer that specific line |
00:41.22 | gravemind | but im guessing sip phones dont normally have that functionality so asterisk cant implement it |
00:41.39 | mansing | gravemind: hardphones -- look at the Sipura |
00:42.00 | Malthus | some do |
00:42.07 | mansing | also Grandstream has a multiple SIP appearance phone |
00:42.18 | Malthus | the ones at work do |
00:42.29 | gravemind | i wasnt thinking an entire seperate registration is needed |
00:42.35 | Malthus | but most of the better ones include a proper LCD |
00:42.46 | gravemind | just a way that asterisk could send a DTMF code with the ring to indicate which line |
00:42.52 | Malthus | so you can get more info that just internal/external |
00:42.56 | gravemind | and in turn, dial out a DTMF to pick a line |
00:43.22 | Jinxy | how can I tell if I am connected to an IAX2 trunk? |
00:43.31 | Malthus | you can configure asterisk to do what you want with the callerid it sends |
00:43.47 | gravemind | Malthus: say you have a receptionist who has access to 10 incoming lines. if two lines are ringing at once, and you want to pick up a specific one, how would you go about doing that? |
00:44.21 | mansing | with a multiple appearance phone |
00:44.30 | Malthus | her phone would show both incoming |
00:44.36 | mansing | each button is a different extension |
00:44.42 | mansing | asterisk manages the buttons |
00:44.43 | Malthus | button beside each to answer |
00:44.50 | mansing | yes |
00:44.50 | bdunn | Can someone please tell me where mphmp3 is supposed to go in the file system |
00:44.55 | gravemind | fair enough |
00:45.21 | mansing | I did that with a Sipura |
00:45.22 | gravemind | out of curiosity, could that be done with a single extension, assuming its a pure sip phone? |
00:45.42 | mansing | on incoming calls, asterisk checks each of the four exts on the sipura. |
00:45.47 | mansing | each ext == one button |
00:45.50 | Druken | gravemind: you should be able to use the SIP stuff for that.... |
00:46.17 | gravemind | hm |
00:46.18 | gravemind | okay, thanks. |
00:47.31 | Druken | SetVar(_ALERT_INFO= |
00:47.41 | Druken | take a look at that... might help ya out |
00:48.11 | *** join/#asterisk cabronsito (~awefawe@201.240.45.194) |
00:48.12 | Katty | mew |
00:50.03 | *** join/#asterisk justinks (~kennedy10@machine76.Level3.com) |
00:52.57 | *** join/#asterisk int19h (~Miranda@PC00019127.eng.monash.edu.au) |
01:00.33 | Jinxy | do I need to do anything special if my * box is behind a linux firewall so my AIX2 trunk works? |
01:00.48 | Jinxy | *IAX2 trunk rather |
01:02.28 | *** join/#asterisk Mistil (Twister@24-179-94-064.dhcp.chtn.wv.charter.com) |
01:03.01 | *** join/#asterisk w0w0 (~apardo@159.Red-83-41-3.pooles.rima-tde.net) |
01:03.03 | bdunn | OKAY... Music on Hold is working... BUT VERY VERY VERY VERY quietly. Any suggestions? |
01:03.44 | harryvv | there is a loud setting |
01:06.54 | bdunn | When I set it to loud I don't hear anything at all. :-( |
01:09.02 | Malthus | maybe it caused you to go deaf |
01:09.08 | bdunn | Ahh... music! |
01:09.12 | iCEBrkr | Blah. |
01:09.19 | Malthus | :) |
01:09.29 | bdunn | I had to go to default.... now it's working at the right volume. No idea what I did. |
01:09.33 | iCEBrkr | EAGI() doesn't appear to be working correctly. |
01:09.40 | bdunn | That's everything... it's ALL WORKING! Holy crap! |
01:09.46 | iCEBrkr | bdunn: hehe |
01:10.13 | darwin35 | E911 can go to hell |
01:10.20 | iCEBrkr | darwin35: lol |
01:11.04 | darwin35 | I want to shoot the people who voted for this bill and then move up the line from there |
01:11.08 | darwin35 | gfrrrrrr |
01:11.33 | Nugget | got the ill communication |
01:11.36 | darwin35 | no easy answer |
01:11.54 | darwin35 | and down to less then 90days to implament it |
01:12.36 | harryvv | and the phone companies wont give up the info ? |
01:12.39 | justink | darwin35: whats up with the E911 grief? Most ppl want their phones, even if VoIP, to function like a phone. As another phone geek put it "if it walks like a dog and wags like a dog, then ppl expect it to be a dog." |
01:12.40 | sivana | darwin35: we're working on an affordable solution |
01:13.13 | darwin35 | open ss7 still has alot of work |
01:13.41 | darwin35 | and it is going to put alot of new startups down and existing voip providers down |
01:13.43 | Mistil | any1 else have nufone and have an 800 number that isnt working? |
01:13.47 | harryvv | how is vonage tackeing the e911 problem? |
01:13.58 | darwin35 | they wont say |
01:14.06 | Jinxy | so guys, do I need to forward any ports to get an IAX2 trunk to work if my * box is behind a linux firewall? |
01:14.24 | Nugget | I think you're confusing "firewall" and "nat" |
01:14.32 | harryvv | e911 cannot be enforced..if thay do thay need to hire alot more field agents. |
01:14.50 | Jinxy | Nugget, the * box actually has internet ip, but it is behind a linux firewall |
01:15.00 | sivana | harryvv: here in Canada, they are enforcing it through the ILECs and CLECs |
01:15.06 | Jinxy | which filters most of the unrelated traffic coming to it, so I need to know if I should allow in a certain port |
01:15.07 | Nugget | "port forward" is a nat concept, though. |
01:15.12 | iCEBrkr | Did 911 always have ANI? |
01:15.23 | sivana | harryvv: the CLEC providing my PRI has threaten to shut it down if I don't comply |
01:15.32 | Jinxy | Nugget, I understand, but I need to know if I need to allow in any unrelated connection? |
01:15.47 | harryvv | silvana, wow really. |
01:15.52 | Nugget | http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
01:16.04 | darwin35 | they can inforce by making it the t-1 e-1 dsl/cable providers right to cut you off if you have anything running on a voip port |
01:16.09 | sivana | harryvv: ya, they're forcing the CLEC/ILECs to enforce their networks |
01:16.21 | Nugget | third hit on google for "asterisk firewall iax", for what it's worth |
01:16.47 | harryvv | I think thats rediculios. I use sixtel to call into the states and I dont need no fricken 911 from them. |
01:17.03 | sivana | that's different though |
01:17.11 | harryvv | Well I hope so! |
01:17.31 | darwin35 | 800/866/877 providers dont have to worry |
01:17.41 | harryvv | I guess the only way around it is be a country to county voip carrier. Or sell voip to bussiness that have pstn. |
01:17.46 | darwin35 | just did/pstn providers |
01:17.56 | sivana | only providers that serve local residential customers |
01:17.57 | sivana | ya |
01:18.32 | harryvv | seems to me residential is the most profitable because I suspect thay wont use as much bandwith vs say a call center |
01:18.33 | darwin35 | they would have to have 1 line that you could map to thier system for 911 |
01:19.01 | *** join/#asterisk NatRH (~Nat@dargo.trilug.org) |
01:19.03 | harryvv | yea, and splice off that line to a red phone in a red box on the wall in the event the pbx goes down. |
01:19.12 | SarahEmm | redbox? ;) |
01:19.14 | harryvv | its extra security. |
01:19.19 | darwin35 | have them have a sipura or something mapping the call to thier pbx service |
01:19.41 | *** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
01:19.50 | darwin35 | that would work |
01:20.20 | harryvv | have the door triger a ringer microswitch when its opened.. and ringer switch stays on when the door is closed untill its reset. :) |
01:20.44 | darwin35 | I want a red phone |
01:20.53 | darwin35 | with a flashing blue light on it |
01:20.57 | harryvv | That way, if thay do shut it by accident and 911 calls back, it will still ring. |
01:21.07 | darwin35 | yes |
01:21.20 | harryvv | its just good insuance. |
01:21.25 | darwin35 | I think it should be a pbx box at every company |
01:21.26 | harryvv | peace of mind |
01:21.27 | harryvv | :) |
01:22.02 | darwin35 | well we where looking into being a virtual pbx company |
01:22.09 | harryvv | One federal company I have worked in and i suspect its req by the fire department in all buildings is a red phone box on every floor. |
01:22.19 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
01:22.20 | darwin35 | for small/medium companies |
01:22.49 | darwin35 | that would be the answer |
01:23.01 | *** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
01:23.06 | darwin35 | but if the pbx goes down then they cant take calls anyways |
01:23.07 | bdunn | What is the command to add a voice mail box? |
01:23.23 | darwin35 | ee voicemail.conf |
01:23.25 | harryvv | look at the bottom of voicemai..conf |
01:23.26 | SarahEmm | vi voicemail.conf |
01:23.28 | darwin35 | addd the box |
01:23.38 | darwin35 | then reload asterisk |
01:23.43 | harryvv | examples at the bottom of voicemail.conf |
01:23.53 | harryvv | if you have samples installed |
01:23.56 | darwin35 | the first time someone calls the exten it will creat the box and prompts |
01:25.01 | darwin35 | but with the red phones would they dial 911 only or have a full keypad |
01:25.10 | bdunn | Seems like there was a command line way to create a voice mailbox. |
01:25.17 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
01:26.02 | ManxPower | Asterisk will now automagically create the voicemailbox when the first message is lefst |
01:26.14 | darwin35 | you could write a bash script to add a new vm boox to the end of the file |
01:26.27 | bdunn | When I try to access the voicemail, I get silence, so I can't leave a message. |
01:26.38 | *** join/#asterisk hans (fugalh@falcon.fugal.net) |
01:27.26 | hans | single-port fxs ATA - recommendations and antirecommendations? |
01:28.04 | ManxPower | hans: SIPura, always SIPura |
01:29.08 | hans | why? |
01:30.17 | *** join/#asterisk emp (~emp@70.57.239.37) |
01:30.21 | ManxPower | They make good stuff for a decent price. |
01:30.46 | ManxPower | Their hardphone is not all that great, but good for the price if you accept it's limitations |
01:31.50 | Qwell | <rant> |
01:32.12 | Qwell | You know what I love? When you're told on Monday night at 6:30, that the entire network is being upgraded on Friday night. |
01:32.30 | Qwell | and, that you'll need to come in at midnight or so, and wait for them to finish, so you can fix shit and test it |
01:32.44 | *** join/#asterisk jm_dot_com (~jmdotcom@adsl-67-119-233-167.dsl.sndg02.pacbell.net) |
01:32.47 | Qwell | ...I'm leaving my rant tag open |
01:33.01 | emp | is it possible to have an asterisk system use the call transfer feature on a my POT ? |
01:33.08 | Jinxy | interesting, I can make a call through voipjet if I use iaxcomm utility on my windows box, but it won't work through asterisk |
01:33.22 | Qwell | Jinxy: Then your setup is wrong in asterisk |
01:33.59 | JerJer | mmm POT |
01:34.08 | hans | ManxPower: thanks for the recommendation. any chance on finding one locally (normal size city USA) or am I stuck with online shipping costs? |
01:34.29 | Jinxy | Qwell, I've looked everywhere |
01:34.35 | Qwell | Jinxy: Well, what happens? |
01:34.44 | Jinxy | I dial, I get an error |
01:34.57 | Qwell | "an error". yeah, THATS helpful |
01:35.02 | Jinxy | I am copying it |
01:35.03 | Jinxy | <PROTECTED> |
01:35.03 | Jinxy | <PROTECTED> |
01:35.25 | Jinxy | I've got the setup as suggested by voip |
01:35.28 | Jinxy | *voipjet |
01:35.29 | SarahEmm | lol jerjer |
01:36.30 | jm_dot_com | hello guys i have basic telco questions trying to find the best solution for my office setup. Ive, been activly useing asterisk for about year half and this is my first time on a irc list. |
01:38.14 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
01:38.43 | *** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
01:39.02 | SarahEmm | well it's not a list, but ask away jm_dot_com |
01:40.14 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
01:40.43 | jm_dot_com | I an asterisk box colocated in our dc. And 2 remote offices with dsl so far this has worked out great, however useing SIP providers for incoming lines however with our call volume incressing have not been happy with the way the sound qualy has been |
01:41.32 | jm_dot_com | so looking at possibly having a dids or pri coming directly into office and haveing asterisk plug into that. but wanted to konw if this would be most cost effective solution. also trying to stay within a montly budget. and |
01:42.14 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
01:42.41 | jm_dot_com | i like how now we can have unlimite calls come in and it just opens another line which is why i dont want to go with regualr pots lines as the cost adds up just to get a new number. |
01:43.13 | SarahEmm | well, if you want to leave SIP/IAX providers, you're moving to a PRI, so.. i'm not sure what other options you have :) |
01:43.58 | jm_dot_com | Will this help to incress my voice clarity this is the realy concern, of ours as clients are begging to complain about sound quality. |
01:44.22 | SarahEmm | well, it could be an issue of quality between the office and your * box too. |
01:45.40 | *** join/#asterisk iq (~iq@70-59-160-156.omah.qwest.net) |
01:45.40 | jm_dot_com | would useing hardware sip phones over the soft phones make any diffrence. our office to dc is usualy in the 30ms and from * box to SIP providers around 80 as it goes back east we are in west coast. |
01:46.06 | iq | hi |
01:46.20 | SarahEmm | it might, i'm not sure jm_dot_com :) |
01:46.29 | SarahEmm | i'm likely one of the worst people in the channel to talk to about sound quality.. |
01:46.36 | SarahEmm | so someone else may be better to help you on this :) |
01:46.47 | Katty | SarahEmm: i'm the worst, me thinks |
01:47.00 | SarahEmm | Katty: oh? |
01:47.22 | Katty | SarahEmm: i don't know hardly anything |
01:47.28 | SarahEmm | Katty: oh. |
01:47.30 | jm_dot_com | txs. do you happen to know aprox how much pri did lines run. are they cheaper if went with someone like xo vs a local bell. |
01:47.35 | SarahEmm | Katty: i know stuff, i just can't hear so good :) |
01:47.44 | SarahEmm | not sure jm_dot_com |
01:47.44 | Twister | bout how much of an internet connection you all think id need for say..a max of 8-10 outgoing and 2-3 internal calls at once, also what would be the best codec that would give me a good balance of quality/low bandwidth use |
01:47.52 | Katty | SarahEmm: but i did set up samba and let a co-worker have access to the master.cvs |
01:48.04 | Katty | and then i figured out how to do a ssh tunnel |
01:48.07 | Katty | so i guess i'm not all that dumb |
01:48.13 | SarahEmm | *nodnods* you're not :) |
01:48.21 | Katty | yay :> |
01:49.04 | Twister | meow! :) |
01:49.16 | SarahEmm | mew! |
01:49.26 | *** part/#asterisk mansing (~mansing@250-116.customer.cloud9.net) |
01:51.25 | Druken | moo... |
01:52.07 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
01:52.08 | Twister | baaaaaaa! |
01:52.31 | Druken | damn sheep! |
01:52.36 | JerJer | woof |
01:53.05 | Katty | mew |
01:53.19 | Druken | who has a phone game in the dialplan ? |
01:54.12 | SpaceBass | game? |
01:54.30 | SpaceBass | i like to send people into a never ending IVR... does that count? |
01:55.11 | Katty | like blackjack |
01:55.15 | Katty | press one for another card |
01:55.17 | Katty | press 2 to stay |
01:55.26 | Katty | your score is 20, my score is 18 |
01:55.27 | Katty | you win |
01:55.32 | Katty | press 3 to play again |
01:55.34 | Jinxy | any iax cli commands? |
01:56.01 | JerJer | gotta have a really bad impresssion of Sean Conerry |
01:56.05 | Twister | help iax2 |
01:56.11 | Druken | nah.... a game... |
01:56.36 | *** part/#asterisk hans (fugalh@falcon.fugal.net) |
01:56.38 | int19h | has anyone heard of any games where callers can compete for queue position? |
01:56.53 | *** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
01:57.14 | bdunn | Outgoing calls show caller ID with a new of Wireless Caller. Any fix to that? |
01:57.27 | SarahEmm | int19h: ROFL!! |
01:57.30 | SarahEmm | int19h: that would rock! |
01:57.47 | Twister | hmmm |
01:57.50 | Twister | *gears turn* |
01:57.51 | int19h | SarahEmm: yea, I thought so :) |
01:58.47 | Jinxy | does voipjet use aix2 or aix? |
01:59.01 | Jinxy | according to their config, it seems like it is aix2 |
01:59.06 | SarahEmm | Jinxy: it's IAX. |
01:59.07 | Jinxy | but damn, I can't get it to work ugh |
01:59.09 | SarahEmm | AIX is an operating system. |
01:59.12 | SarahEmm | a nice one. |
01:59.15 | SarahEmm | but still. IAX. |
01:59.25 | Jinxy | SarahEmm, not iax2? |
01:59.34 | SarahEmm | err |
01:59.39 | SarahEmm | i meant the acronym is IAX, not AIX |
01:59.43 | SarahEmm | i have no idea what the answer to your q is |
01:59.47 | Jinxy | ya, I keep confusing it |
01:59.54 | Twister | aix...mmmm..drooooool |
02:00.17 | SarahEmm | then again, i'm Weird ;) |
02:00.53 | int19h | sheesh.... you people.... |
02:00.57 | int19h | AIX is a designer label |
02:01.13 | Jinxy | nothing wrong with my voipjet account, works just fine if I use iaxcomm windoze program |
02:01.13 | SpaceBass | thats A|X (pipe not I) |
02:01.17 | int19h | damn |
02:01.19 | int19h | good point |
02:01.22 | SpaceBass | :) |
02:01.31 | int19h | :) |
02:02.13 | darwin35 | Mr President please pick up the blue phone. Mr President please pick up the blue phone |
02:02.40 | int19h | I was major into asterisk about 6 months ago, but my interest has declined substantially in recent months... tell me... is there an app for modifying the audio stream within calls yet? |
02:02.42 | Nugget | Alright, give me Hamm on five, hold the Mayo. |
02:02.52 | darwin35 | SpaceBass pls pick up the white curtisy phone |
02:02.55 | int19h | i.e. filter the audio? to, lets say, cause a reverb effect? |
02:03.15 | SpaceBass | darwin35: give me the mayo on white and hold the red phone |
02:03.46 | SpaceBass | damn Nugget beat me to it... couldnt quite remember it |
02:04.13 | darwin35 | all those attending cluecon pls pickup the yello com phone |
02:04.18 | *** join/#asterisk Blake0PS (~blake@blakeops.com) |
02:04.23 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
02:04.40 | Nugget | SpaceBass, don't start with that red zone/white shit again. We all know what this is really about. |
02:04.56 | SpaceBass | i was telling someone about that scene just last night |
02:04.59 | darwin35 | no thats black and white |
02:05.08 | darwin35 | red and white are phones |
02:05.15 | SpaceBass | looks like I picked the wrong day to quit sniffing glue |
02:05.24 | *** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com) |
02:05.25 | *** join/#asterisk florinm (~florin@host-84-9-255-27.bulldogdsl.com) |
02:05.29 | florinm | hi guys |
02:05.40 | SpaceBass | is there a way to make an ATA call something as soonn as it goes off hook? if so I may have to get an old school red phone |
02:05.41 | SarahEmm | darwin35: i'd love to, but.. mew. |
02:05.41 | florinm | anyone knows what is with goldenpbx ? |
02:06.07 | darwin35 | when my tdm 40b gets here I plan to have a red/blue/yello/green line setup |
02:06.25 | florinm | ?? |
02:06.49 | darwin35 | yes its called autodial and its in the web config |
02:07.04 | litage | `sip show peers` shows a registered device (softphone), but the softphone was shutdown 2 minutes ago. how can i fix this? |
02:07.05 | SpaceBass | darwin35: what web config? |
02:07.21 | darwin35 | ata's have a web interface |
02:07.28 | *** join/#asterisk Qorky (~Pooa@202.173.160.26) |
02:07.33 | Druken | is there a good web based config for asterisk yet? possibly a free one? |
02:07.37 | darwin35 | if you login to it you can configure it |
02:07.45 | darwin35 | amp |
02:07.47 | SpaceBass | Druken: amp |
02:07.50 | Qorky | can anyone help me please. when I call my meetme conference. Asterisk crashes out. |
02:07.54 | darwin35 | is about the only one I have tested |
02:07.54 | Qorky | anyone else seenthis ? |
02:08.03 | Druken | url ? |
02:08.09 | darwin35 | you dont have a timing device |
02:08.24 | _DAW | Qorky - you have zaptel card or ztdummy? |
02:08.28 | Qorky | hmm but i do |
02:08.31 | darwin35 | you7 need to compine zaptel and ztdummy |
02:08.40 | Qorky | i have a z100p in the box |
02:08.47 | darwin35 | did you run ztcfg on it |
02:08.55 | Qorky | Channel map: |
02:08.55 | Qorky | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
02:08.55 | Qorky | 1 channels configured. |
02:09.06 | darwin35 | ok then stop asterisk |
02:09.09 | Qorky | i ment x100p |
02:09.11 | darwin35 | type ztcfg |
02:09.20 | darwin35 | and then start asterisk |
02:09.22 | Jinxy | worky worky |
02:09.28 | darwin35 | and the timing should work |
02:09.35 | Jinxy | stupid firewall had ip for voipjet off |
02:09.36 | Jinxy | :( |
02:09.44 | darwin35 | and if not read /var/log/asterisk |
02:11.29 | Qorky | the messages file |
02:11.29 | Qorky | ? |
02:11.35 | Qorky | it doesnt have much in there. |
02:11.41 | Blake0PS | hi, if i start in.tftpd from inetd and specify -v (verbose logging) where does it log to? |
02:11.52 | Qwell | Blake0PS: probably /var/log/messages |
02:12.59 | litage | what can i do about asterisk thinking a softphone is connected/registered when the softphone isn't running? |
02:13.14 | *** join/#asterisk Twister (Twister@24-179-94-064.dhcp.chtn.wv.charter.com) |
02:13.40 | darwin35 | reset the time out |
02:13.46 | darwin35 | in the conf file |
02:14.07 | darwin35 | and make sure the client logs out not just turns the client off |
02:14.29 | litage | shouldn't turning the client off (ie: shutting down the software) log off automatically? |
02:14.40 | litage | darwin35: there isn't any way to manually log a client off from asterisk? |
02:15.18 | darwin35 | not that I know of other then the time out will clear the db |
02:15.31 | darwin35 | if it fails to reg in the time required |
02:15.37 | litage | thanks darwin35 |
02:16.29 | *** part/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca) |
02:16.42 | darwin35 | np |
02:16.44 | Jinxy | I need some suggestions from the pros here |
02:16.51 | Katty | take a nap |
02:16.54 | emp | is a X100P a good way to get started with an * setup? Where are good places to buy? |
02:16.56 | Katty | you'll feel better in the morning |
02:17.07 | Jinxy | emp, fleabay |
02:17.11 | darwin35 | x100p are no longer made |
02:17.23 | darwin35 | get a tdm 400 |
02:17.33 | Jinxy | emp, you can get a clone, and a small mod to the driver can get you started |
02:17.34 | darwin35 | it has 1 fxo 1 fxs |
02:17.37 | Jinxy | but, they have echo issues |
02:17.55 | Jinxy | darwin35, you have to buy modules for it |
02:18.02 | Katty | what should i do now? |
02:18.09 | Katty | 2 hours till bed time and nothing to do |
02:18.13 | SpaceBass | you can get an x100p (clone) from ebay for like $10 |
02:18.15 | Qorky | how can i check the timing ? |
02:18.23 | Qorky | is working from the x100p ? |
02:18.29 | darwin35 | it comes with 2 modules if you buy the dev kit for 175 |
02:18.33 | Jinxy | anyway, I've got 4 POTS lines and about 8~12 analog phones |
02:18.43 | Jinxy | I don't want to replace all the phones with ip ones, gonna cost me a lot |
02:18.50 | Jinxy | and I also want to give each phone its own extension |
02:18.53 | darwin35 | then thet the 12 line card thats out there |
02:18.54 | Jinxy | what is my best option? |
02:19.13 | darwin35 | 3 tdm40b |
02:19.22 | darwin35 | for about 900 |
02:19.26 | emp | darwin35, where is a good place to buy online? |
02:19.49 | Jinxy | darwin35, with 4 FXS modules each? |
02:19.54 | darwin35 | http://www.telephonyware.com/telephonyware/scan/fi=products/st=db/co=yes/sf=category/se=Analog CTI Cards/op=eq/tf=category,description.html?id=kTSeGr8d |
02:20.03 | darwin35 | there is all the cards |
02:20.09 | darwin35 | and pricing |
02:20.47 | emp | nice, thanks |
02:21.32 | Blake0PS | How do I get the in.tftpd server to log? in.tftpd -vvvvvvvvv /tftpboot/ |
02:21.36 | Blake0PS | that doesn't work |
02:21.43 | darwin35 | the voicetronix card with 12 fxo/fxs ports is nice you jumper it for fxo or fxs on each port |
02:21.48 | pfn | use tcpdump or tethereal |
02:22.12 | Blake0PS | which one is easier to use and view the output of |
02:23.21 | justink | has any company made a 12port ATA using SIP? An external solution for fxs ports? |
02:23.45 | iCEBrkr | Blake0PS: I think it logs to /var/log/messages |
02:24.08 | Qwell | Didn't I say that earlier? |
02:24.41 | iCEBrkr | (10:11)< Qwell> Blake0PS: probably /var/log/messages |
02:24.42 | iCEBrkr | Yup. |
02:24.44 | iCEBrkr | :D |
02:25.17 | iCEBrkr | EAGI just isn't ment to be. |
02:25.20 | *** join/#asterisk ToR\L (toril@cpe-24-58-23-240.twcny.res.rr.com) |
02:25.26 | ToR\L | hi folks |
02:25.32 | JerJer | moo |
02:25.43 | darwin35 | not that I have seen |
02:26.06 | ToR\L | I have an asterisk server (1.0.9) for voip at home, using a sip ata... how do I stop it from sending mwi info? |
02:26.26 | Qwell | ToR\L: remove the mailbox= from sip.conf? |
02:26.29 | ToR\L | I have these cordless phones that say "msg waiting off" every few seconds |
02:26.31 | Qwell | or whatever that option is |
02:26.32 | ToR\L | I commented it out |
02:26.40 | Qwell | and reloaded? |
02:26.41 | ToR\L | mr qwell |
02:26.42 | ToR\L | yup |
02:26.49 | Qwell | Then your phone sucks. :D |
02:26.52 | iCEBrkr | lol |
02:26.55 | ToR\L | lol |
02:26.57 | ToR\L | doh |
02:27.00 | iCEBrkr | ToR\L: listen to your messages. That turns it off. :) |
02:27.01 | ToR\L | my 5 phones suck! |
02:27.05 | ToR\L | no messages |
02:27.10 | iCEBrkr | oh. that problem |
02:27.11 | iCEBrkr | haha |
02:27.13 | litage | voicetronix says this about their openswitch12: "Standard-sized PCI form factor that easily fits into any regular PC chassis" how? the card looks like it's 2 feet long |
02:27.13 | iCEBrkr | hrrm. |
02:27.16 | darwin35 | the netweb401 rock |
02:27.18 | iCEBrkr | I forget how I fixed that |
02:27.27 | Qwell | litage: most machines are 2 feet long |
02:27.27 | ToR\L | I set checkmwi=3600 |
02:27.34 | ToR\L | commented out the mailbox line |
02:27.38 | ToR\L | still it insists |
02:28.00 | ToR\L | vonage forums insist its in the ata |
02:28.08 | Malthus | maybe that's why it says 'messae waiting off' |
02:28.08 | iCEBrkr | haha qwell is the master of the obvious |
02:28.09 | Qwell | litage: put it in one below all of the HDs... |
02:28.10 | litage | errr, giraffe would be better |
02:28.11 | ToR\L | I have a linksys rt31p2 |
02:28.17 | Qwell | litage: You must've never seen old ISA cards man... |
02:28.22 | Qwell | those things were fucking SCARY |
02:28.31 | Qwell | iCEBrkr: I try. :) |
02:28.55 | Qwell | iCEBrkr: remember those old video boards, with the ide controllers builtin? |
02:29.03 | iCEBrkr | Hrrm, you could build a ceiling fan with those old ISA cards |
02:29.14 | iCEBrkr | VLB baby!! |
02:29.22 | Qwell | vlb? |
02:29.31 | Qwell | video logic board? |
02:29.32 | Malthus | vesa local bus |
02:29.32 | iCEBrkr | Yea, the full length Vesa Local Bus cards |
02:29.34 | Qwell | oh |
02:29.34 | ToR\L | a search for mwi disable on voip-info.org and google came up with nothing really good |
02:29.35 | Qwell | heh |
02:29.39 | Qwell | close enough... |
02:29.57 | Malthus | VLB gotten beaten by its competitor 'PCI' |
02:30.03 | iCEBrkr | That the 'fast' 'enhanced' IDE controller on it |
02:30.11 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
02:30.14 | Qwell | iCEBrkr: ata33? heh |
02:30.16 | iCEBrkr | haha |
02:30.32 | litage | what's the difference between `sip show peers` and `sip show users`and `sip show registry`? |
02:30.36 | iCEBrkr | lets not forget the 16550 UART I/O chip set! |
02:30.41 | Malthus | lol |
02:30.46 | Qwell | litage: users != peers != registered users |
02:30.48 | iCEBrkr | BLAZE'N |
02:31.38 | iCEBrkr | eep, I showing my geekness. and age.. 16550 should be a unknown number. |
02:32.23 | ManxPower | No, the 16550AF was good, it was the older models were the bane of my tech support life for a year. |
02:32.27 | litage | Qwell: what does asterisk consider to be a user, peer, and registered user? |
02:32.44 | iCEBrkr | eeww! vic |
02:32.47 | ToR\L | hehe |
02:32.59 | ToR\L | I missed acoustic couplers by | | that much |
02:33.25 | ToR\L | but a 300 baud vicmodem still needed a rotary dial phone for manual dialing |
02:33.39 | Malthus | later people |
02:33.40 | iCEBrkr | Eh, my 300 baud modem had tone dial. |
02:36.13 | iCEBrkr | <PROTECTED> |
02:36.42 | file | eh? |
02:36.45 | file | why do you want bkw |
02:36.55 | drumkilla[work] | everybody should want file |
02:36.56 | MikeJ[Laptop] | cuz he;s sexy |
02:36.59 | iCEBrkr | I dont' 'want' him.. he might like that. |
02:37.04 | file | yeah - everyone should WANT me! |
02:37.07 | iCEBrkr | lol |
02:37.11 | file | in a sexy way |
02:37.14 | Qwell | file: should...but don't |
02:37.17 | MikeJ[Laptop] | me? |
02:37.19 | file | ;( |
02:37.19 | iCEBrkr | WTF is up with this EAGI crap?! |
02:37.31 | iCEBrkr | My Asterisk box is looking at me like Helen Keller. |
02:37.33 | MikeJ[Laptop] | ummmm |
02:37.38 | MikeJ[Laptop] | wow... |
02:37.38 | iCEBrkr | ..as in, it can't F'n hear me. |
02:37.47 | MikeJ[Laptop] | that's wrong |
02:37.50 | ToR\L | hmm the mailbox line I commented out was in my section of sip.conf |
02:37.53 | ToR\L | sound about right? |
02:37.59 | Qwell | yesh |
02:38.01 | Qwell | yeah |
02:38.06 | darwin35 | how did helen kellers parents punish here |
02:38.10 | iCEBrkr | oh no |
02:38.13 | iCEBrkr | look what I started |
02:38.17 | darwin35 | put door knobs on all the walls |
02:38.22 | MikeJ[Laptop] | introduced her to you? |
02:38.22 | ToR\L | lol |
02:38.23 | *** join/#asterisk cybyc (~cyb@Ottawa-HSE-ppp269044.sympatico.ca) |
02:38.38 | MikeJ[Laptop] | wow... sorry.. that was wrong too... |
02:38.53 | darwin35 | how did helen keller burn her face ... she tried to answer the iron |
02:39.01 | iCEBrkr | According to the so called documentation, the audio file descriptor in EAGI is stderr+1 |
02:39.30 | darwin35 | I wear a hearing aid I can tell the jokes |
02:39.30 | iCEBrkr | darwin35: oh, so you're allowed to poke fun? |
02:39.44 | Qorky | so is there a way to test me timing? meetme still bombs out |
02:39.53 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
02:40.05 | darwin35 | yes look at the tools included |
02:40.12 | darwin35 | in the zaptel dir |
02:40.19 | iCEBrkr | aye, buy me pint o'ale and ye timing shall be fixed, mate |
02:41.20 | Qorky | you mean this? |
02:41.20 | Qorky | Opened pseudo zap interface, measuring accuracy... |
02:41.21 | Qorky | 99.987793% 99.987793% 99.987793% 99.987793% |
02:41.23 | darwin35 | did you edit the meetme.conf |
02:41.41 | darwin35 | and map meetme right |
02:42.04 | Qorky | erm i have edited it. |
02:42.13 | Qorky | under rooms it has this |
02:42.13 | Qorky | conf => 50 |
02:42.19 | Qorky | i have no password for it. |
02:42.28 | Damin | It's fucking ICEBRKR |
02:42.34 | iCEBrkr | Oh god. |
02:42.43 | iCEBrkr | It's that beer drinker.. Damin |
02:42.48 | Damin | Whats' up dude? |
02:42.51 | iCEBrkr | Not a damn thing. |
02:42.54 | iCEBrkr | Playing in hurricanes already |
02:43.24 | iCEBrkr | But, right now, I'm bored off my ass, so I'm hacking around with EAGI stuff |
02:43.48 | ToR\L | iCEBrkr: you said you've turned off mwi? |
02:44.08 | iCEBrkr | ToR\L: I think I just reset my ATA and the MWI reset |
02:44.11 | _DAW | boooo Emily :( |
02:44.47 | ToR\L | hmm |
02:45.13 | iCEBrkr | I had a remote phone do that.. All the messages were erased but the MWI was still blinking |
02:46.02 | file | MWI! that's what I was going to look at |
02:46.12 | *** part/#asterisk cybyc (~cyb@Ottawa-HSE-ppp269044.sympatico.ca) |
02:46.18 | ToR\L | reset the ata |
02:46.51 | Damin | iCEBrkr: Are you up for an N2Net Alumni OUting? |
02:47.06 | Damin | iCEBrkr: With Mac The Hack as the guest of honor? |
02:47.06 | iCEBrkr | Damin: If you fly me up there LOL |
02:47.08 | ToR\L | doh no that didn't do it |
02:47.11 | iCEBrkr | Mac!!!! |
02:47.13 | ToR\L | it was maybe 90 seconds |
02:47.33 | iCEBrkr | Damin: When were you thinking? |
02:47.51 | Damin | iCEBrkr: Late august.. |
02:48.05 | iCEBrkr | Damin: I was thinking of maybe visiting Labor Day |
02:48.15 | Qorky | darwin35: that look ok ? |
02:48.37 | darwin35 | dmesg | less and look for the config of the card |
02:48.38 | ToR\L | actually 30 seconds |
02:48.41 | ToR\L | hmm |
02:48.49 | darwin35 | cut and paste in a pvt window |
02:49.37 | litage | when using x-lite, how do you dial a phone number that contains letters? |
02:49.53 | Twister | just like you do on any other phone |
02:50.01 | Twister | translate the letters into numbers |
02:51.24 | Damin | iCEBrkr:4 |
02:51.59 | *** join/#asterisk vuvie (~vuvie@bb219-74-45-31.singnet.com.sg) |
02:52.01 | Damin | iCEBrkr: That would be cool. A Labor Day Weekend NACS Outing.. Maybe Steve and Martin can re-enact the Summer Sausage incident.. |
02:54.07 | *** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net) |
02:55.57 | litage | twisted: what if userA and userB have "williamA" and "williamB" as their extensions, though? if you translate letters into numbers, there's ambiguity.. |
02:56.14 | iCEBrkr | hahahahahhahahaaha |
02:56.18 | Qwell | When would their extensions be letters? |
02:56.37 | Qwell | s/when/why/ |
02:56.47 | litage | because they're easier for users to remember |
02:57.06 | litage | Bob: "what's joe's extension?" Alice: "joesmith" |
02:57.31 | Qwell | users who can't remember 4 digit extensions should be shot |
02:57.35 | iCEBrkr | haha |
02:57.48 | litage | Qwell: and if you have 30 extensions? |
02:57.50 | iCEBrkr | Easier to remember, but harder to spell on the phone. |
02:58.11 | Qwell | "Bob: Whats Pams extension?" "Alice: piyawan-patcharaprasertsook" |
02:58.11 | litage | iCEBrkr: if you use a softphone, it's not harder |
02:58.24 | *** join/#asterisk nifter (~nifter@190-141.SPEEDe.golden.net) |
02:59.02 | Qwell | I'd smack our phone lady if she made extensions names... |
02:59.12 | litage | Qwell: why? |
02:59.27 | Qwell | that was the real name of one of our employees. She would have either had to break the standard (first name + last name), or actually make a 30 digit extension |
02:59.47 | ToR\L | in high school |
02:59.53 | ToR\L | we had a networked system |
03:00.11 | ToR\L | and your username was first three letters of your last name, then first 3 of your first name |
03:00.19 | ToR\L | so john smith was smijoh |
03:00.22 | ToR\L | we had a don hart |
03:00.31 | litage | Qwell: then you limit it to something like "first letter of first name + lastname, max X chars" |
03:00.43 | Qwell | litage: Then you hit ambiguity |
03:00.48 | Qwell | John Smith vs Jack Smith |
03:01.22 | iCEBrkr | It's called an Excel Spreadsheet... |
03:01.27 | iCEBrkr | Print and distribute. |
03:01.31 | iCEBrkr | Pin to your cube wall. |
03:01.35 | iCEBrkr | k/thx/bye |
03:01.39 | litage | many ISPs/network admins/etc used the old "first letter of first name + lastname, max 8 chars" rule, and it worked fine |
03:01.49 | Qwell | litage: bzzt |
03:01.52 | Jinxy | anybody know the vista 350 time/date procedure to update adsi information? |
03:02.16 | litage | Qwell: if jsmith already existed and jack smith came along, he'd get something like jsmith2 |
03:02.19 | Qwell | litage: Then you have to add numbers. Let's use something like msmith |
03:02.28 | Qwell | msmith2, msmith3, msmith10, msmith95 |
03:02.31 | Qwell | Thats 8 chars |
03:02.45 | Qwell | You have enough employees/customers, and you're toast |
03:02.56 | iCEBrkr | yea, so whats Jack Smiths number?? 3 or 95? |
03:03.06 | Qwell | iCEBrkr: in which order did he join? |
03:03.09 | iCEBrkr | jsmith3? jsmith95? |
03:03.11 | Qwell | oh, and that |
03:03.12 | iCEBrkr | I dunno man. |
03:03.19 | iCEBrkr | we just assign them that way :D |
03:03.29 | Qwell | "Whats Jacks number?" "jsmith...uhh...46?" |
03:03.37 | iCEBrkr | Qwell: exactly |
03:03.41 | Qwell | yeah, I got you |
03:03.55 | litage | iCEBrkr: asking "what's jack smith's number?? 3 or 95?" is the same as asking "what's jack smith's number? 123456 or 234567?" |
03:04.09 | litage | iCEBrkr: it's just that "jsmith3" and "jsmith95" are more descriptive |
03:04.18 | iCEBrkr | I dunno man. It's pretty simple to just have a printed piece of paper for people to reference. Matter of fact, it's quite common. |
03:04.23 | Qwell | "Whats Jacks last name?" "How do you spell that?" |
03:04.27 | Qwell | or, 4 digit numbers |
03:04.44 | ToR\L | iCEBrkr: can you show me an extension line from extensions.conf (minus any phone #s)? |
03:04.50 | iCEBrkr | litage: Ever see those inserts on the phones? |
03:04.55 | iCEBrkr | hehe |
03:04.57 | litage | if you have a doc with the extensions, then spelling doesn't matter |
03:05.12 | iCEBrkr | I don't know what you're asking me fore? |
03:05.13 | iCEBrkr | err |
03:05.13 | iCEBrkr | for |
03:05.31 | litage | iCEBrkr: i'm just responding to your statements, that's all |
03:05.37 | iCEBrkr | ; 2101 / Sipura:Port1 - Florida |
03:05.37 | iCEBrkr | exten => 2101,1,Macro(extension,2101) |
03:05.42 | iCEBrkr | ; 2102 / Sipura:Port1 - Tim |
03:05.42 | iCEBrkr | exten => 2102,1,Macro(extension,2102) |
03:05.46 | iCEBrkr | ; 2103 / Sipura:Port2 - Andrea |
03:05.46 | iCEBrkr | exten => 2103,1,Macro(extension,2103) |
03:05.49 | ToR\L | hmm |
03:05.52 | iCEBrkr | PRetty simple |
03:06.34 | litage | Qwell: whether your extension sheet says "joe smith => 1234" or "joe smith => jsmith95" doesn't matter, except that 1234 is harder for an avg user to remember |
03:06.51 | Qwell | and jsmith95 is harder to dial |
03:07.04 | ToR\L | exten => xxxxxxxxxx,1,Dial(SIP/xxxxxxxxxx,30,tr) |
03:07.06 | ToR\L | where the x |
03:07.10 | iCEBrkr | litage: I dunno man, I've been working with 3-4 digit extensions since I was 7.. Not that big of a deal |
03:07.11 | ToR\L | x's are the # |
03:07.18 | iCEBrkr | litage: you remember the people you call the most. |
03:07.19 | litage | Qwell: so what's more important? ease of dialing, or ease of remembering? |
03:07.34 | iCEBrkr | s/7/17 |
03:08.09 | iCEBrkr | And at my last job, we had a sweet Call Manager interface. |
03:08.13 | litage | iCEBrkr: i agree, 3-4 digit extensions are easy; basic. i'm just wondering why letters are translated into #s rather than being treated as valid, separate chars |
03:08.18 | iCEBrkr | Which I wanna code up for Asterisk. |
03:09.21 | iCEBrkr | Ok, yeah, I'm getting nohwere with this EAGI crap |
03:09.23 | *** join/#asterisk dysan (~ack@202.37.224.27) |
03:10.25 | dysan | im getting make[1]: *** [chan_zap.o] Error 1 errors and chan_zap.c error: dereferencing pointer to incomplete type errors, how do i get around these? |
03:10.40 | litage | anyways, so i got x-lite working on linux, speakers and mic and all. why might x-lite work properly, but kphone not even bother to touch /dev/dsp? |
03:10.57 | iCEBrkr | kphone always blew up for me as soon as the call was completed |
03:11.34 | litage | iCEBrkr: kphone kicked the bucket if it lost focus for me, so you're faring better with it than i =P |
03:11.37 | iCEBrkr | err, connected. |
03:11.38 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
03:11.45 | iCEBrkr | haha |
03:11.57 | iCEBrkr | it'd ring, I go to answer it and it'd blow up |
03:12.04 | litage | strange |
03:12.17 | iCEBrkr | I haven't had much luck with softphones |
03:12.27 | litage | mine connects fine, but doesn't play or record audio |
03:12.35 | iCEBrkr | oh wait. I got that fwd/pulver.communincation thing working |
03:13.05 | litage | x-lite has worked perfectly on 2 win and 1 mdk machine so far for me |
03:13.41 | iCEBrkr | x-lite's interface is F'n GAY |
03:13.45 | litage | aye |
03:13.53 | iCEBrkr | Could they make it any more bulky? |
03:14.28 | litage | yeah they could. i've seen a couple other softphones that took up like 1/3 of your desktop and had like 5mm of padding around each button |
03:14.37 | litage | it was like they were expecting you to use a broom on a touchscreen |
03:14.42 | iCEBrkr | haha |
03:14.53 | litage | the name slips my mind hto |
03:14.54 | litage | tho |
03:15.03 | iCEBrkr | it was some java basd one, I think.. |
03:15.14 | iCEBrkr | or maybe it was one of the 'better' looking ones for Linux.. just a binary |
03:15.20 | iCEBrkr | I think I know which one you're talking about |
03:15.44 | litage | =P you just have to shake your head when you see things like that. i mean, WHAT are the developers thinking?? |
03:16.19 | litage | some people undervalue desktop realestate :( |
03:16.33 | iCEBrkr | They're thinking they have 1600x1200 dual displays |
03:17.20 | iCEBrkr | WinAmp is the only interface that has it right.. I'm talking the original.. Not the 'modern skin' |
03:18.23 | litage | yeah i agree. i'm not a fan of those new-fangled winamp5 skins that are funky shapes and whatnot |
03:18.35 | emp | my ISP uses Interleave on their DSL lines... so my first hop can range between 40-100ms lag. Would this pose a problem setting up a * box? What is acceptable lag? |
03:18.45 | ToR\L | iCEBrkr: how about the mailbox line in sip.conf? |
03:18.51 | Qwell | emp: less then 100 for sure |
03:18.52 | ToR\L | then I won't bug ya anymore ;) |
03:19.06 | Twister | im creating a menu that brings a call in on one trunk then exits it on another |
03:19.14 | Twister | here is that the cli outputs |
03:19.17 | Twister | Executing Dial("IAX2/NuFone@outbound-nufone/2", "SIP/broadvoice/13045905796") in new stack |
03:19.17 | Twister | <PROTECTED> |
03:19.33 | iCEBrkr | Wow. My latency is lower than I thought |
03:19.41 | iCEBrkr | OK (30 ms) |
03:19.45 | iCEBrkr | OK (23 ms) |
03:19.49 | Qwell | 30 to your provider? |
03:20.14 | iCEBrkr | Well, that's 'sip show peers' |
03:20.22 | iCEBrkr | My asterisk box is remote.. my sipura's are here |
03:20.41 | emp | is DSL suitable? should I be looking at T1 or colo? |
03:20.48 | iCEBrkr | emp: It's doable. |
03:20.58 | Druken | emp for how many calls? |
03:20.58 | iCEBrkr | emp: You can get skiping and dropout tho |
03:21.23 | emp | Or maybe I just need to find the tech that can turn off Interleave on my line :) heh |
03:21.24 | iCEBrkr | I ran my asterisk setup off DSL for awhile. |
03:21.41 | iCEBrkr | emp: If you're just building a glorified answering machine for home it should be ok |
03:21.43 | Twister | http://pastebin.com/311643 |
03:22.00 | Twister | theres the pastebin of my menu |
03:22.27 | emp | I would like to be able to add remote extensions down the road... but that might be a while |
03:22.39 | iCEBrkr | ToR\L: I have the mailbox=2101 in my sip.conf |
03:23.07 | *** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au) |
03:23.13 | iCEBrkr | emp: oh, I had more issues with using voicepulse for LD |
03:24.36 | iCEBrkr | I did a phone interview over voicepulse and my DSL was saturated. Could have been that I didn't tinker with the codecs |
03:25.14 | spoot_nick | i'm looking for a working FOSS CDR rating/billing system. does anobody has any advice on the subject? |
03:25.28 | *** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz) |
03:25.50 | ToR\L | hmm |
03:25.53 | spoot_nick | tried trabas (everybody says it sucks). got it to work, didn't understand how i can pull records from asterisk's cdr table. CDRTool is free for personal use, so it doesn't suit |
03:26.05 | litage | how fast does telephonyware.com ship orders? |
03:26.40 | iCEBrkr | spoot_nick: I don't have a complete system, but I log all my calls to MySQL |
03:26.43 | file[laptop] | you can't get everything for free. |
03:26.53 | iCEBrkr | file[laptop]: who asked you? |
03:27.01 | iCEBrkr | :D |
03:27.07 | file[laptop] | I asked myself. |
03:27.09 | iCEBrkr | oh |
03:27.10 | file[laptop] | I count as two people |
03:27.13 | file[laptop] | see? |
03:27.17 | iCEBrkr | file[laptop]: Proceed. |
03:27.27 | JunK-Y | file: u and u double personality? :P |
03:27.32 | file[laptop] | exactly |
03:27.34 | spoot_nick | fine, but the question applies to FOSS =) |
03:27.34 | iCEBrkr | lol |
03:27.35 | JunK-Y | hehhe |
03:27.52 | iCEBrkr | file[laptop]: get back to work!! |
03:27.59 | file[laptop] | meh work ended awhile ago |
03:28.04 | file[laptop] | I just wanted to clean up my config files |
03:28.11 | file[laptop] | and debug a problem |
03:28.18 | iCEBrkr | Ooooooooh, personal stuff. I see. |
03:30.29 | file[laptop] | uh my cellular provider like... billed me for something they shouldn't have |
03:30.57 | MikeJ[Laptop] | what was it.... a puppy? |
03:31.02 | JunK-Y | erotic lines are bad for phone bills u know it file! |
03:31.18 | file[laptop] | nah I called a friend... local |
03:31.20 | file[laptop] | but they charged me |
03:31.27 | file[laptop] | my evenings/weekends thing didn't kick in |
03:31.53 | JunK-Y | ya, shits like that happens so often, that why im checking really carefully my bills. |
03:32.49 | *** join/#asterisk jaike (~a@203.131.137.76) |
03:33.51 | MikeJ[Laptop] | tzafrir, you around/ |
03:33.55 | file[laptop] | they charged me $25 |
03:34.03 | file[laptop] | billed my local calls as long distance for a night |
03:34.22 | jaike | anyone know the cause of choppy asterisk calls? were currently using asterisk and vonage and we dont seem to have problems with voice quality with vonage like we have with asterisk |
03:35.14 | jaike | were using sipura phones and cisco 186 atas with asterisk |
03:37.54 | iCEBrkr | My calls are only choppy when they're important ones. |
03:38.05 | iCEBrkr | I can talk to my parents back home all night and day without a problem. |
03:39.43 | jaike | am just wondering why vonage quality is ok while asterisk isnt ... what protocol is vonage using? |
03:39.49 | Twister | im assuming you are using a zaptel |
03:39.55 | Twister | interface for vonage? |
03:40.09 | jaike | were using sip |
03:40.17 | Twister | got a softphone account? |
03:40.27 | *** join/#asterisk fugitivo (~ajf@201.255.104.90) |
03:40.28 | fugitivo | hello |
03:40.45 | Jinxy | hello fugitivo |
03:41.18 | Jinxy | guys, would I be limited in my ability to use * features on a analog phone compared to an ip phone? |
03:42.00 | emp | oh, great... I'm in one of 3 states I can't switch my # to vonage :) |
03:43.01 | iCEBrkr | Jinxy: I prefer ATA+Analog phones |
03:43.03 | Qwell | emp: Can't really use vonage with asterisk anyways |
03:43.17 | *** join/#asterisk brenda (~nnnnn@c-67-182-205-227.hsd1.ut.comcast.net) |
03:43.18 | fugitivo | Jinxy: it depends, a cisco 79xx ip phone? :) |
03:43.26 | Qwell | jaike: probably g729 |
03:43.28 | emp | Qwell, what are some providers that are compatible? |
03:43.41 | Jinxy | fugitivo, say grandstream 101 |
03:43.46 | iCEBrkr | emp: there's a ton listed up on www.voip-info.org |
03:43.58 | Jinxy | fugitivo, are you saying some features are phone specific? |
03:44.22 | iCEBrkr | Jinxy: The feature you're losing with a IP-Phone is taking your friends out for beer after you've set it all up. Cuz they cost too damn much!! :D |
03:44.34 | fugitivo | Jinxy: well, cisco has those ip phones with big screens where you can run xml applications :) |
03:44.50 | Jinxy | fugitivo, I am talking features provided by * |
03:45.00 | darwin35 | anyone have the line for musiconhold for madplay |
03:45.04 | Jinxy | call transfer, music on hold and such |
03:45.10 | fugitivo | darwin35: yes |
03:45.22 | darwin35 | fugo can you pvt me with it |
03:45.28 | fugitivo | holdon |
03:45.51 | fugitivo | darwin35: default => custom:/var/lib/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000 --output=raw:- --attenuate=-4 |
03:46.25 | fugitivo | Jinxy: no, you're not limited |
03:46.38 | Jinxy | thank you very much |
03:46.51 | darwin35 | thnks |
03:46.53 | mishehu | hmm... was there a post-1.0.9 clid bug? I noticed I'm not getting clidname with my pri anymore. |
03:46.54 | Jinxy | fugitivo, I've got 12 analog phones, and replacing them won't be cheap |
03:47.03 | darwin35 | I was missing the attenuate |
03:47.08 | Jinxy | so, I was thinking of just getting a channel bank |
03:47.25 | Jinxy | and on top of that, I've got 4 POTS lines |
03:47.26 | fugitivo | Jinxy: hmm, check the wiki about the channel banks |
03:47.35 | Jinxy | fugitivo, I know about that, no biggy there |
03:47.56 | fugitivo | Jinxy: ok, just be sure to get a channel bank reported to work ok with asterisk |
03:52.16 | Jinxy | fugitivo, say features like call waiting, what if my analog phone doesn't support it? |
03:55.31 | darwin35 | that fixed it |
03:55.51 | iCEBrkr | Jinxy: Why wouldn't an analog phone support call waiting? |
03:55.52 | Nugget | call waiting does not require cooperation from the phone on an analog channel. |
03:55.55 | iCEBrkr | it's a flash-hook |
03:56.04 | Nugget | it's impossible for an analog phone to "not support" call waiting |
03:56.15 | Jinxy | oh, I was thinking visual call waiting |
03:56.21 | iCEBrkr | Nugget: well, it might not have the 'flash' button which might confuse the user :D |
03:56.22 | Jinxy | that the phone needs to support |
03:56.39 | darwin35 | i have a flash hook setup |
04:00.15 | |Vulture| | Any Sangoma users around? |
04:00.24 | file[laptop] | blitzrage!!! |
04:00.27 | blitzrage | file[laptop]: !!! |
04:00.38 | blitzrage | SwK_: hey |
04:00.53 | SwK_ | hey |
04:00.54 | jaike | 1.0.9 has a queue bug...some agents logged in to a queue do not receive calls from the queue..had to go back to 1.0.7 |
04:00.56 | SwK_ | what |
04:01.02 | SwK_ | heh |
04:01.18 | blitzrage | SwK_: you msg'd me |
04:01.27 | SwK_ | yeah |
04:01.53 | SwK_ | i was gonna ask about a bug you found on RTP path |
04:02.10 | SwK_ | but I figured it out... wasnt a SIP bug but a libpri bug |
04:03.45 | |Vulture| | can I bring a PRI into a TSU750, then output it to a PRI and a signle FXS? |
04:04.17 | *** join/#asterisk SwK (buteod@12-219-156-206.client.mchsi.com) |
04:04.37 | SwK | damned client |
04:05.16 | drumkilla | jaike: care to gether details on that so that we can get it fixed? |
04:06.09 | jaike | drumkilla: ok..will try to replicate problem again |
04:06.21 | drumkilla | sweet :) |
04:12.00 | *** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
04:12.14 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net) |
04:12.16 | darwin35 | ok in cvshead when I do help database it shows it using put and get yet head examples say to use set ${db |
04:12.40 | darwin35 | did they go back to put and get |
04:12.55 | drumkilla | no, that example just didn't get updated |
04:13.09 | darwin35 | ok |
04:13.45 | darwin35 | drum have a min I have issues them things not getting put into the db |
04:13.59 | SwK | damn it |
04:14.11 | SwK | where is that setting for app_voicemail on which sound file to attach? |
04:14.26 | darwin35 | voicemail.conf |
04:14.26 | iCEBrkr | voicemail.conf? |
04:14.31 | SwK | yeah |
04:14.37 | SwK | in voicemail.conf |
04:14.44 | darwin35 | read it |
04:14.48 | SwK | tI am |
04:14.54 | SwK | i dont see it tho... |
04:15.08 | SwK | i guess its one of the RTFSC config options |
04:15.41 | darwin35 | exten => password,username,email addy,file |
04:15.42 | *** join/#asterisk znoG (~gs@200.115.216.109) |
04:15.46 | darwin35 | its all there |
04:16.27 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
04:21.12 | *** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net) |
04:27.15 | *** part/#asterisk jaike (~a@203.131.137.76) |
04:30.21 | darwin35 | http://pastebin.ca/17514 |
04:30.34 | darwin35 | its not putting infor into the db |
04:30.43 | darwin35 | when I set the functions |
04:31.31 | *** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
04:34.17 | darwin35 | if more is needed I can paste the full exten.conf |
04:37.05 | darwin35 | what did everyone jump ship |
04:37.15 | file[laptop] | yes. |
04:38.28 | Hmmhesays | it's late |
04:39.14 | file[laptop] | it's 1:38AM, yes |
04:39.20 | file[laptop] | I should have said no, someone's waiting for me! |
04:39.53 | Hmmhesays | 1:38 huh? are you off in canadia? |
04:40.02 | file[laptop] | Atlantic Canada |
04:40.13 | file[laptop] | just wanna tell you don't worry... I will be late, don't stay up and wait for me |
04:40.18 | file[laptop] | say again you're dropping out my battery is low |
04:40.19 | file[laptop] | so you know |
04:40.22 | file[laptop] | we're going to a place near by |
04:40.23 | file[laptop] | gotta go |
04:40.56 | *** part/#asterisk OzJames (~opera@203.208.64.29) |
04:42.24 | *** join/#asterisk viking667 (viking@203.184.23.240) |
04:42.33 | viking667 | whew. Here at last. |
04:42.50 | file[laptop] | so you think. |
04:43.14 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:43.24 | viking667 | anyone here using the irssi script "speak.pl" that feeds the festival text-to-speech engine? I'm not managing to get it to go. |
04:44.08 | viking667 | I was told to come here - or at least, I think I was... |
04:44.42 | *** join/#asterisk dano_ (~bismantux@216.147.160.17) |
04:45.55 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
04:46.54 | johnh51 | :) |
04:48.25 | darwin35 | man fixiing this is a pain |
04:50.12 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
04:54.34 | *** join/#asterisk jgenender (~blah@209.181.65.238) |
04:54.57 | jgenender | Hello...I am have a bit of an issue. I am doing sip-to-sip... |
04:55.08 | jgenender | Basically Teliax to my asterisk box |
04:55.19 | jgenender | and I am getting duplicate digits with DTMF |
04:55.24 | jgenender | i.e. if I dial 1010 |
04:55.36 | jgenender | It sometimes understands 110110 |
04:55.39 | jgenender | or 10010 |
04:55.40 | jgenender | etc |
04:55.49 | jgenender | any ideas on what is wrong or how to fix this |
04:55.55 | jgenender | There is no Zaptel in this config BTW |
04:58.32 | darwin35 | http://pastebin.ca/17517 |
04:58.42 | darwin35 | does this look correct |
04:59.16 | darwin35 | i know part of a macro i s missing it has not ben writen yet |
05:03.12 | jgenender | Hmmm...time to build from the HEAD...looks like my issue is a bug: http://bugs.digium.com/view.php?id=4659 |
05:05.06 | darwin35 | check that you set your dtmfmode= and dtmf= rfc2833 |
05:05.20 | darwin35 | in your sip.conf and other fonf files |
05:05.31 | drumkilla | fonf files!!! |
05:05.38 | jgenender | I believe I did... |
05:05.52 | darwin35 | dbl check |
05:06.08 | Qwell | drumkilla: There were like 4 other typos you could have chosen ;p |
05:06.39 | jgenender | Ok...the dtmfmode=rfc2833 but not dtmf...both need to be set? |
05:07.16 | jgenender | Ok...both set...same problem |
05:07.21 | jgenender | Looks liek its that bug |
05:07.28 | jgenender | like^^ |
05:07.34 | darwin35 | what phone |
05:07.53 | darwin35 | I set both |
05:07.59 | jgenender | I am calling an inbound number for Teliax...no matter what phone I use I get this problem |
05:08.15 | jgenender | I tried from Cell, from standard PSTN, and from Cisco 7940 |
05:08.25 | darwin35 | hmm |
05:08.40 | darwin35 | what day of head you on ? |
05:08.50 | jgenender | I am on standard 1.0.7 |
05:09.21 | jgenender | Everything works fine from SIPPhone and FWD (which are dtmfband=inband) |
05:09.27 | jgenender | Teliax is rfc2833 |
05:10.39 | *** join/#asterisk MustDie (~voip@ool-18b91f29.dyn.optonline.net) |
05:10.42 | *** join/#asterisk cfrank_ (~cfrank@wsip-24-234-137-140.lv.lv.cox.net) |
05:13.48 | WilliamK | evenin ya'll |
05:16.15 | MikeJ[Laptop] | jgenender, did you try head, with and without that patch? |
05:16.37 | MikeJ[Laptop] | testing is appretiated |
05:16.39 | jgenender | No...that is next...I just wanted to see if others are having similar issues...this really is an annoying issue |
05:16.48 | MikeJ[Laptop] | what are you seeing |
05:17.19 | jgenender | I am getting duplicate or double DTMF (somewhat random) for dtmfmode=rfc2833 |
05:17.22 | MikeJ[Laptop] | that one should go in pretty quickly as soon as we get a disclaimer fromthe guy, and he has been actively responding soo.. |
05:17.26 | jgenender | Which what Teliax uses |
05:17.40 | MikeJ[Laptop] | inbound, to asterisk? |
05:17.50 | jgenender | Yes |
05:18.03 | MikeJ[Laptop] | I thought that bug was about how we send dtmf |
05:18.24 | jgenender | Hmmm...let me re-read it |
05:18.41 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
05:19.12 | jgenender | MikeJ[Laptop]: No...i tlooks like its reading DTMF |
05:19.32 | MikeJ[Laptop] | ummmm |
05:19.56 | jgenender | "If rtp rfc2833 packets arrive out of sequence for a single DTMF event, |
05:19.57 | jgenender | Asterisk may report double or triple digits received |
05:19.58 | jgenender | " |
05:20.20 | jgenender | Thats the same issue I am having (or so it seems) |
05:20.36 | MikeJ[Laptop] | ok, it is both sending and reciving maybe.. I am lookin at thte patch |
05:21.14 | MikeJ[Laptop] | ok.. looks like both sending and reciving... |
05:21.37 | MikeJ[Laptop] | well.. give it a try, and post results to that bug to say how it affected you... |
05:21.45 | jgenender | I will..thanks. |
05:23.13 | *** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
05:24.12 | jgenender | MikeJ[Laptop]: Do you know if the rtp.c becomes part of the asterisk binary or one/many of the libs? |
05:24.30 | jgenender | I just want to determine what I need to update from a binary perspective |
05:24.44 | MikeJ[Laptop] | yes, part of the bin... |
05:24.52 | jgenender | ok thanks |
05:25.00 | jgenender | Patching now... |
05:25.00 | MikeJ[Laptop] | but if you are going from stable to head, you need to do it all regardless |
05:25.07 | jgenender | ok |
05:25.37 | MikeJ[Laptop] | make clean, remove all the .so modules, then make, make install |
05:25.45 | jgenender | doing it now |
05:25.47 | MikeJ[Laptop] | ok.. .sleep time for me. |
05:25.50 | MikeJ[Laptop] | night all |
05:25.58 | jgenender | nn...and thanks |
05:27.31 | DA-MAN | how often should one rebuild head |
05:28.30 | Qwell | DA-MAN: hourly |
05:29.30 | DA-MAN | ok! |
05:29.30 | DA-MAN | hehe |
05:29.36 | DA-MAN | i do mine weekly |
05:33.36 | MustDie | seimhourly |
05:36.23 | *** join/#asterisk cj (~cjcollier@CJ.monthly-bronze.supporter.pdpc) |
05:38.10 | *** join/#asterisk jaike (~a@203.131.137.76) |
05:38.27 | jaike | question guys...will setting the jitterbuffer too high have any negative effects? |
05:40.48 | jgenender | big delays |
05:41.18 | mosty | i'm having trouble getting a linksys pap2 to work from behind NAT, it won't connect to an asterisk server outside the NAT (but will to one inside). i've tryed playing with the NAT settings, but nothing seems to work. what could be wrong? |
05:41.45 | cfrank_ | Anyone built meetme2 w/o mysql? (postgres only) |
05:41.48 | DA-MAN | mosty, pap2-na |
05:41.50 | DA-MAN | ? |
05:42.01 | mosty | da-man, yes |
05:42.27 | jgenender | MikeJ[Laptop]: Don't knwo if you are asleep but the patch worked ;-) |
05:42.32 | DA-MAN | do you have nat=yes, qualify=yes on sip.conf |
05:43.24 | mosty | da-man: i had nat=yes, qualify=xxx in the general section. i've changed qualify to yes now |
05:44.19 | DA-MAN | no you have to have it in the [ext] stanza, not in general |
05:45.36 | opus__ | holy JESUS SON OF GOD |
05:45.49 | opus__ | i'm getting cvs spam |
05:46.02 | opus__ | (chris farley voice) |
05:47.02 | Math` | cvs spam? |
05:47.11 | Qwell | cvs commit spam? |
05:47.26 | mosty | da-man: ok, i created an [ext] secion, and put the nat and qualify settings in there, restarted asterisk. still doesn't register (how can i force the unit to attempt to register again?) |
05:47.31 | jaike | jgenender: big delays? choppy calls? |
05:47.56 | DA-MAN | ensure that register is et to yes under the line |
05:48.28 | jgenender | jaike: This bigger the buffer the longer the delays in when the voice gets to you |
05:48.39 | jgenender | You may be waiting for voice...its annoying |
05:49.39 | mosty | da-man: register is already set to yes. i got it to reboot via ivr, but still no connection |
05:50.21 | DA-MAN | whats sip debug say |
05:50.56 | infinity1 | hola folks |
05:51.11 | opus__ | hey MATH |
05:51.20 | opus__ | yo QWELL |
05:51.47 | opus__ | infinity1 - sveasoft forums has a good iptables rate limitor script for that |
05:51.48 | mosty | da-man: there are a bunch of messages flying past, some mention the pap2's private lan address, others mention my nat machines public ip address |
05:52.03 | opus__ | infinity1: do both |
05:52.19 | infinity1 | opus__: already got QoS setup. |
05:52.24 | opus__ | really |
05:52.28 | infinity1 | opus__: and graphed ;) |
05:52.42 | opus__ | what. |
05:52.54 | infinity1 | http://prelude.brendon.com/~brendon/cacti/graph.php?rra_id=all&local_graph_id=54 |
05:52.59 | opus__ | no way |
05:53.31 | infinity1 | i'm first a network engineer. then a voip experimenter. heh |
05:53.34 | opus__ | haha that awesome |
05:53.47 | infinity1 | obviously * is in purple |
05:53.56 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
05:54.30 | jaike | bye guys..thanks jgnender..will try lower settings.. |
05:54.32 | opus__ | Thats pretty unique |
05:55.22 | infinity1 | thanks. |
05:55.22 | Math` | fuck just got an * crash on a call transfer |
05:56.16 | Math` | (cvshead) |
05:56.56 | |Vulture| | Math`: how many times did you transfer? just 1nce? |
05:57.03 | Math` | |Vulture|: for that call, yeah |
05:57.07 | opus__ | where do you download movies |
05:57.16 | infinity1 | bit torrent sites. |
05:57.18 | |Vulture| | hmm there is an issue with transfering multiple times |
05:57.21 | Math` | opus__: www.torrentspy.com / wwww.mininova.org |
05:57.22 | opus__ | ever since btenet died with rss feeds i'm been dailyshowless |
05:57.28 | |Vulture| | Martohtar: I haven't looked at CVS current |
05:57.29 | Math` | |Vulture|: there's an issue for PARKING multiple times |
05:57.46 | infinity1 | opus__: tell me about it. i haven't been downloading tv shows cuz of that |
05:57.52 | |Vulture| | I think transfers as well with a PRI |
05:57.58 | Math` | ah was SIP to SIP |
05:58.04 | infinity1 | opus__: notice my traffic drop over the past few months :) |
05:58.28 | opus__ | \haha |
05:59.02 | Math` | |Vulture|: note that codec transcoding might have been required on the new line (the one that received the x-fer), x-fer done via astman |
05:59.09 | Math` | I'll try to replicate it |
05:59.14 | infinity1 | i really need to switch windows and start making head packages for debian. |
05:59.24 | infinity1 | but its funner to just chil here. lol |
05:59.34 | *** join/#asterisk clive- (~pirch@rndf-146-9-138.telkomadsl.co.za) |
05:59.51 | opus__ | math dude thanks!!! the.daily.show.07.11.05.marci.hamilton.dsr.xvid-crimson.[VTV].torrent |
05:59.59 | Math` | np |
06:00.28 | opus__ | i freaking pay for cable but have no tv |
06:00.37 | Math` | thats bad |
06:00.43 | Math` | I freaking pay for cable tv but rarely watch it |
06:00.46 | infinity1 | all the cable is for inet? |
06:01.03 | MustDie | opus: no biggie, it's nothing on tv anyway |
06:01.14 | justink | any suggestions for unlimted SIP VoIP provider that offers GSM or equavilent economy codec? |
06:01.23 | MustDie | there are about 6000 channels and only 300 has something to watch on ;) |
06:01.36 | opus__ | justink how much do you want to spend |
06:01.51 | Qwell | opus__: OT - piratebay too |
06:01.52 | justink | as little as possible-its residential service. |
06:02.19 | Qwell | justink: How many minutes per month would you say you use? |
06:02.22 | DarthClue | justink: you want PSTN terminatin with that, or just VOIP? |
06:02.42 | opus__ | justink - well you can get very cheap service but there is no guarntee that the service will be up the next day. your better off paying $24.99 or buying your own PRI and reselling it yourself |
06:03.06 | Math` | voipjet seems reliable |
06:03.13 | justink | Def want PSTN termination, availity to have virtual numbers, full feature set...and prefer not to use G711 as Im a bandwidth hog. |
06:03.45 | opus__ | well i work for a few providers so if i recommended them it'd be a conflict of interest:) |
06:03.58 | Math` | lol |
06:04.00 | opus__ | what i want is g711 wideband |
06:04.03 | justink | read that broadvoice uses 711, teliax offers gsm. |
06:04.13 | DarthClue | opus__: it is always a conflict of interest, that doesn't stop most of us. |
06:04.16 | Qwell | opus__: no, it'd be a shameless plug |
06:04.20 | justink | you and I are after diff things then opus |
06:04.22 | Qwell | nothing wrong with that... |
06:04.54 | Qwell | opus__: If somebody asks you what a good provider is... If you believe in your company, hell, promote it |
06:05.03 | opus__ | ahaah! |
06:05.04 | opus__ | haha |
06:05.17 | Qwell | I said if. :P |
06:05.18 | justink | No need for a CD quality phone call...at least for me. |
06:05.22 | darylp | is anyone having problems with free world dialup? |
06:05.33 | *** part/#asterisk jaike (~a@203.131.137.76) |
06:05.34 | justink | I'll check out voiphet |
06:05.43 | Math` | darylp: Im talking on it right now |
06:05.43 | justink | voipjet |
06:05.44 | Qwell | isn't voipjet outgoing only? |
06:05.47 | Math` | it is |
06:05.48 | DarthClue | justink: virtual numbers could be an issue with some providers. what countries do you intend to call? asterlink offers 2c per minute voip termination to the US48 with a toll free number. They support gsm and sip as well. |
06:06.13 | opus__ | and if you have a credit card right now you can dcc chat me we can start an account right now |
06:06.14 | justink | continental US only. |
06:06.22 | darylp | hmm, then I must have a problem |
06:06.34 | Math` | opus__: who are u working for? |
06:06.42 | Qwell | justink: If you need incoming calls, you might want to reconsider voipjet |
06:07.00 | opus__ | math a few resellers, one call center.. just the dumb shit. i got my own dream so fuck em |
06:07.00 | justink | if its outgoing only youre right. |
06:07.10 | Math` | ok |
06:07.18 | DarthClue | justink: check out asterlink, 2c per minute with a toll free, incoming and outgoing, they can set you up for sip and gsm. |
06:07.40 | darylp | with iax debug on should I see something when I try to call fwd from outside my system? |
06:07.46 | justink | ok-will do some digging and googling. thanks. |
06:08.13 | darylp | of course the fwd number is registered |
06:08.13 | *** join/#asterisk marshall (~test@24.77.245.158) |
06:08.25 | DarthClue | darylp: if it is using iax, yes. |
06:08.52 | marshall | Can someone please tell me how to move to the next line of the dialplan when a call exits non-zero |
06:09.13 | darylp | I'm seeing nothing when I call a local access number to fwd extension using pots |
06:09.25 | infinity1 | awesome. i think i found a solution to my debian packages! checkinstall! |
06:09.27 | DarthClue | marshall: pastebin the extensions.conf file |
06:09.28 | Qwell | marshall: show application dial |
06:09.44 | ender | infinity1: doesn't work nearly as well as you might wish. |
06:09.47 | DarthClue | Qwell: but is he using dial? |
06:09.55 | infinity1 | ender: don't tell me that!!!!!!! |
06:09.56 | Qwell | DarthClue: oh, dunno |
06:10.00 | marshall | thanks Qwell, I'll look through that |
06:10.15 | ender | infinity1: it's the equal to maybe making a tarball of compiled binaries into a package, w/out any logic to where they go and whatnot that makes a package desireable. |
06:10.17 | DarthClue | infinity1: nothing ever works the way you want it to. |
06:10.40 | marshall | Darth I am using Dial |
06:10.44 | infinity1 | ender: thats good engouh. i just want something to keep track of where * puts shit |
06:10.51 | infinity1 | ender: cuz the * makefix sux arse. |
06:10.53 | marshall | want a failover onto a zap channel if my IAX trunk is down for whatever reason |
06:10.54 | infinity1 | er makefile |
06:11.08 | ender | infinity1: ah, well that works then I suppose |
06:11.10 | Math` | marshall: use a dial macro |
06:11.25 | Math` | it just to exten+101 in case of failure |
06:11.27 | infinity1 | ender: sweet. i'm going to try it. |
06:11.45 | marshall | Math n+101 is only on a zero exit is it not, this is a non-zero exit |
06:11.53 | opus__ | darylp - try this, edit /etc/asterisk/logger.conf and make console => or what ever have everything. start asterisk with asterisk -vvvvvvvvvvvvvvvvvvgcddddddddddddddd |
06:11.54 | marshall | if the trunk is dead |
06:12.08 | Math` | marshall: if the trunk is dead the call will fail and go to n+101 |
06:12.32 | opus__ | darylp - is that fails i am sure there is a command 'iax2 debug' lemme check real quick |
06:12.55 | litage | which ATAs also have a regular RJ-11/POTS connection that the ATA fails over to in case an internet connection isn't available? |
06:13.05 | Math` | marshall: hmm it just jumps to pri+1 sorry |
06:13.15 | Qwell | litage: I think one of the SPA's did |
06:13.24 | marshall | doesnt seem to work like I need it to |
06:13.29 | litage | thanks Qwell |
06:13.30 | *** join/#asterisk kd5uzz (~nntpsmurf@139.78.167.69) |
06:13.47 | darylp | yeah, I have iax2 debug turned on |
06:13.57 | opus__ | modify logger.conf |
06:14.49 | opus__ | chances are, you have a context/dialplan problem or don't use fwd |
06:15.16 | darylp | it was working until this afternoon |
06:15.28 | opus__ | then, they are down. |
06:15.35 | darylp | when I went to demo the access numbers to someone and then it stopped |
06:15.50 | marshall | Math, any other ideas? Im stumped |
06:15.55 | opus__ | i think thats a moore law |
06:16.11 | darylp | well, it's murpy's law at any rate |
06:16.15 | Math` | marshall: I got more than 1 PSTN termination providers and if 1 fails, it jumps to the second one |
06:16.25 | Math` | marshall: the 1st one uses prio1, the 2nd uses prio2 |
06:16.46 | litage | Qwell: any idea what sipura might call that feature? |
06:16.52 | opus__ | darylp haha right on. |
06:17.00 | marshall | Im trying to prevent dead air on an inbound DID call to my PRI, if the trunk is down to the remote site |
06:17.08 | Qwell | litage: dunno |
06:17.27 | marshall | I was hoping to bounce it to a backup cell phone or basic voicemail box |
06:17.28 | opus__ | darylp - i'd help but i opened a FWD account in 2002 and forgot my password :( |
06:17.33 | marshall | on the main server |
06:17.41 | Math` | marshall: just put a dial as prio+1 |
06:17.42 | opus__ | hehe they wern't even around back then |
06:17.55 | darylp | when I use the call me app I see nothing |
06:18.16 | marshall | Math: where would I find the syntax for that? |
06:18.32 | Math` | show application dial |
06:18.34 | Math` | in the CLI |
06:22.47 | opus__ | the cli is pretty funny. what is it suppose to be sql |
06:24.16 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
06:24.17 | opus__ | darylp - do you have any adavnce IAX2 security features turned on? |
06:24.58 | infinity1 | wow. checkinstall worked nicely for libpri |
06:25.33 | Math` | checkinstall always works nice :P |
06:25.33 | Math` | <PROTECTED> |
06:25.34 | DarthClue | darylp: what is your fwd number? i can give ya a call and see what it does. |
06:25.38 | Math` | uhu now I got that with fwd |
06:26.17 | darylp | 671901 |
06:26.20 | darylp | thanks |
06:26.24 | wunderkin | math thats fine |
06:27.07 | Math` | wunderkin: yeah.... 10000 times |
06:27.25 | DA-MAN | hehe never tried checkinstall |
06:27.25 | DarthClue | darylp: what do you get when you type 'iax2 show registry' on the CLI? |
06:27.26 | Math` | ok I hung up before the 10 000'th time |
06:27.59 | infinity1 | forget freeworld, just peer with everyone :) |
06:28.05 | darylp | all of my iax accounts, including that one, are registered |
06:28.12 | Math` | infinity1: lol |
06:28.40 | darylp | do you get that my number is busy? |
06:28.53 | DarthClue | darylp: yes. |
06:29.01 | darylp | yep, that's what I get too |
06:29.16 | Math` | I love it when I google something, and the first link google gives me is a mailing thread of [asterisk-users] being told to google for it |
06:29.28 | infinity1 | lol |
06:29.28 | DarthClue | darylp:what codecs do you allow? |
06:29.37 | kd5uzz | lol |
06:29.38 | Math` | what's rtp codec id 72? |
06:29.51 | darylp | hmm, I don't know off the top of my head |
06:30.02 | emp | what's the best way to allow >1 incoming calls? more physical lines? use a hosted pbx solution? something else? |
06:30.06 | darylp | I get this: |
06:30.10 | darylp | <PROTECTED> |
06:30.10 | darylp | <PROTECTED> |
06:30.10 | darylp | <PROTECTED> |
06:30.10 | darylp | <PROTECTED> |
06:30.10 | darylp | <PROTECTED> |
06:30.19 | opus__ | emp - pri or provider. provider is cheaper |
06:30.21 | Math` | emp: a PRI line or a voip DID provider |
06:30.25 | Newbie___ | hi guys, i am using h323, is there any way to know i am connected to a gatekeeper? |
06:30.28 | opus__ | math i beat you |
06:30.35 | Math` | u did |
06:30.40 | emp | :) thx |
06:30.41 | DarthClue | darylp: pastebin your iax.conf with user/pass XXed out |
06:30.53 | Math` | rtp.c:508 ast_rtp_read: Unknown RTP codec 72 received |
06:31.08 | darylp | that's a lot of editing, hold on |
06:31.37 | darylp | just ulaw |
06:31.38 | kd5uzz | are there any useful codes, etc you can get from the phone company using normal phone lines using Asterisk? CID info that sort of thing? |
06:31.43 | darylp | I have |
06:31.54 | darylp | disallow=all |
06:31.56 | darylp | allow=ulaw |
06:32.32 | opus__ | kd5uzz electromagnetism |
06:32.49 | DarthClue | darylp: pastebin the complete cli output |
06:33.07 | *** join/#asterisk Assid (~assid@203.115.64.62) |
06:33.15 | darylp | ok, I apologize, but I don't know what "pastebin" is |
06:33.20 | DarthClue | ~pastebin |
06:33.21 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
06:33.39 | kd5uzz | opus__ cool, just what I was hoping for :-) |
06:33.42 | darylp | thanks |
06:33.53 | darylp | learn somthing new every day |
06:34.06 | darylp | darth, you mean when I try to call myself? |
06:34.10 | darylp | that output? |
06:34.23 | Math` | opus__: already got that unknown rtp codec 72 thingy? |
06:34.27 | opus__ | kd5uzz did you know "In classical electromagnetism, the electromagnetic field obeys a set of equations known as Maxwell's equations, and the electromagnetic force is given by the Lorentz force law" |
06:34.28 | DarthClue | darylp: yes. |
06:34.43 | opus__ | Math - no, did you try cscope |
06:35.01 | infinity1 | :) |
06:35.27 | opus__ | math - cd /usr/src/asterisk && cscope -R |
06:35.48 | infinity1 | yay. leno is on |
06:36.26 | opus__ | kd5uzz - use the force |
06:37.11 | Math` | opus__: great, The payload type range 72-76 is marked "reserved" so that RTCP and |
06:37.11 | Math` | <PROTECTED> |
06:37.27 | darylp | http://pastebin.ca/17523 |
06:37.30 | opus__ | well, now you know. |
06:38.33 | Math` | wiki entry for RTCP: See RTP |
06:38.41 | opus__ | math - my wild guess, with no experience what so ever, is that you're getting some kind of control packet ? rtcp? that is messing shit up, perhaps a reinvite |
06:38.48 | opus__ | i'm guess rtcp stands for real time control packet |
06:39.00 | DarthClue | darylp: what operating system? |
06:39.08 | opus__ | no, wait, it wouldn't make sense to to revinites in RTP.. hmmm |
06:39.10 | Math` | RTP COntrol Protocol |
06:39.28 | darylp | fedora core 1 |
06:39.45 | darylp | I'm running asterisk at home, which is why pasting confs can be a bit of a mess |
06:40.02 | Math` | how do u exit cscope lol |
06:40.15 | DarthClue | darylp: i would recommend using either 1.0.9 or CVS-HEAD. you may be running into a bug in 1.0.7 |
06:40.48 | opus__ | math ok my second guess is that it has something to do with comfort noise |
06:40.50 | Math` | ah ok it wants ^d |
06:40.53 | opus__ | math yeah d |
06:40.55 | DarthClue | darylp: do you have to run *@h or can you run just pure *? |
06:41.27 | darylp | well, I really like AMP, the rest of at home I don't care much about |
06:41.32 | litage | what's the difference between the SPA-1000 and SPA-2000 lines? |
06:42.10 | DarthClue | darylp: i think you can install * and then add AMP on top of it without the whole *@h setup. most of us don't like *@h at all. |
06:42.14 | DA-MAN | SPA-2000 has two physical ports |
06:42.20 | DA-MAN | physical phone portz |
06:42.29 | darylp | yeah, I've noticed ;) |
06:42.34 | Math` | opus__: any cng support in x-lite |
06:42.38 | opus__ | math -- hey, its for feedback error repair. http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-feedback-11.txt |
06:42.59 | opus__ | 'of course' |
06:43.03 | litage | DA-MAN: the SPA-1000 also has 2 POTS ports |
06:43.11 | newmember | DarthClue: what are some of the issues with *@h? |
06:43.22 | opus__ | math whats cng? |
06:43.23 | darylp | there's a lot I don't like about it, but, it does a lot of configuration for me |
06:43.27 | Math` | opus__: comfort noise gen |
06:44.08 | opus__ | oh we don't need that here i leave on the beach |
06:44.12 | opus__ | live |
06:44.39 | Math` | opus__: yeah but it was your "best bet" of the problem source so I wanted to check if it wasnt enabled somewhere |
06:44.43 | DarthClue | newmember: if you use *@h then you won't ever really know how * works, and if you don't know how it works, you won't ever know where to start when you need to fix it. * is not difficult enough to need the gui on a fresh install. if you can follow written directions it takes about 5 minutes to get a base install running excluding compile time. |
06:45.11 | opus__ | darthclue - exactly. |
06:45.42 | darylp | yeah, I've installed pure asterisk, in fact, it's working on one of my macs, but the configuration is overwhelming at first |
06:46.08 | darylp | I'm learning quite a bit from at home, and for the most part, it's been working |
06:46.17 | darylp | I consider it a transition tool |
06:46.29 | DarthClue | just run make samples and modify what you need from there. it is quite simple as long as you keep a paper bag handy to keep from hyperventilating. |
06:46.37 | darylp | haha |
06:46.38 | Math` | whats so different with *@h |
06:46.43 | opus__ | big mistake a@h delieverd 1.0.8 :( |
06:46.46 | tzafrir | DarthClue, excluding compile time, and the time to get the rlevant libraries, and figuring out what the heck they are, etc. |
06:46.54 | darylp | I didn't try make samples |
06:47.18 | DarthClue | tzafrir: no issues with libraries on this end, i guess that's just a debian problem. |
06:48.25 | darylp | tnx darth, I have done that, in fact, I have head compiled on my mac |
06:48.35 | opus__ | nice |
06:48.51 | opus__ | darthclue why does sse2 break ilbc |
06:48.52 | opus__ | :) |
06:48.53 | tzafrir | let's compare that to the build-deps on debian... |
06:49.22 | DarthClue | tzafrir: that's why i don't recommend debian. Debian is for people that want to deal with that kind of crap, not for people who need to get it done now. |
06:50.22 | Math` | DarthClue: debian is good when u know what ur doing |
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06:50.42 | Math` | Gentoo is good when u want to waste time recompiling |
06:50.55 | opus__ | yeah |
06:50.59 | tzafrir | Build-Depends: debhelper (>= 4.0.4), sed (>= 3.95), zlib1g-dev, libgtk1.2-dev | libgtk-dev, libreadline4-dev | libreadline-dev, libgsm1-dev, libssl-dev, libzap-dev, libtonezone-dev (>= 1:1.0.6-1), doxygen, bison, libasound2-dev, postgresql-dev, unixodbc-dev, libpri-dev (>= 1.0.6-1),, dpatch (>= 2.0.10), zaptel-source (>= 1:1.0.6-1), autotools-dev, libnewt-dev, libsqlite-dev, libspeex-dev |
06:51.03 | opus__ | gentoo is for kids |
06:51.19 | darylp | I tried gentoo once...once is enough |
06:51.23 | infinity1 | opus__: agreed |
06:51.36 | tzafrir | Let's remove dependecies that were added because of debian scripts, and inter-asterisk-packages |
06:51.38 | infinity1 | tzafrir: hey. whats with your debian packages? |
06:51.41 | DarthClue | Math`: that's my point. most of the people who want help with * don't know what they are doing. most people don't want to waste time. most people want instant gratification, therefore, i recommend a rh variant, either FC or CentOS which are known to work and work quickly with few if any complications. |
06:52.09 | tzafrir | infinity1, what do you mean? |
06:52.13 | darylp | fwiw I didn't have any real trouble compiling head on os x |
06:52.17 | infinity1 | tzafrir: don't you make deb packages? |
06:52.26 | tzafrir | infinity1, I do |
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06:52.33 | infinity1 | tzafrir: head? |
06:52.36 | darylp | a couple of libraries were necessary, same with ubuntu (debian) |
06:52.39 | tzafrir | not yet |
06:52.42 | darylp | but synaptic makes that childs play |
06:52.51 | infinity1 | tzafrir: why not?? |
06:52.53 | infinity1 | :) |
06:53.04 | infinity1 | tzafrir: i'm one of those instant gradification bitches :) |
06:53.05 | tzafrir | head is still a moving target |
06:53.16 | opus__ | darylp - i will beat you with my my wget && tar xvfz && make , skillz |
06:53.16 | tzafrir | the relevant build dependencies: |
06:53.18 | DarthClue | darylp: so i've heard. i'm being encouraged by the storm troopers to switch to mac but i can't afford it at the moment. |
06:53.20 | infinity1 | tzafrir: any plans to? like pakcage it once a week or something |
06:53.50 | infinity1 | tzafrir: i don't follow it closely. it might move to fast for packaging once a week for all i know. |
06:54.02 | tzafrir | infinity1, no, I don't have the time for that. |
06:54.13 | darylp | macs are expensive, I don't have a fast enough one which is why I'm not running * on it |
06:54.13 | infinity1 | tzafrir: k. just checking. |
06:54.16 | tzafrir | I mean: I can automate the build process, but not the testing process |
06:54.37 | darylp | well, the better half is hungry, and my problem can wait a little while |
06:54.57 | hellop | So, whats the word on Outlook shared Calander replacement these days? |
06:54.58 | darylp | thanks for the help guys, I'm off to dennys and I'll attack it on a full stomach in a little bit |
06:54.59 | DarthClue | darylp: just don't feed them after midnight...or get them wet. |
06:55.04 | tzafrir | infinity1, there debs of head, but I haven't checked them for a long time |
06:55.04 | infinity1 | tzafrir: i think it would be nice to have like nightly packages if it was automated. just a thought. maybe i'm on crack. |
06:55.24 | infinity1 | tzafrir: the only debs of head out there are from like dec or jan 04/05 |
06:55.28 | darylp | good advice darth, you wanna keep em on your good side |
06:55.30 | tzafrir | anyway, soon we will have head packages in experimental |
06:55.41 | infinity1 | debian experimental? |
06:55.43 | tzafrir | Unstable will still have 1.0 for a while |
06:55.56 | infinity1 | tzafrir: that would be sweet. |
06:58.19 | tzafrir | infinity1, so do it yourself |
06:58.27 | tzafrir | the scripts are mostly in place |
06:58.41 | infinity1 | i made a few crude packages. |
06:58.51 | infinity1 | very ugly. |
06:59.23 | tzafrir | infinity1, the pkg-voip svn repository has a script for building all of their packages from svn |
06:59.33 | tzafrir | Haven't tried setting it up yet |
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07:00.17 | tzafrir | But it would be interesting. Contact me if you have any problems with it. http://svn.debian.org/wsvn/pkg-voip/ |
07:01.15 | infinity1 | my main goal right now is getting asterisk working in production. |
07:01.32 | Math` | infinity1: whats stopping you? |
07:02.06 | infinity1 | Math`: well, i just got head install. i was runing 109 |
07:03.38 | infinity1 | where is the new syntax for head documented? |
07:03.52 | infinity1 | thats the main reason i wanted to upgrade. i still haven't seen docs for it |
07:04.07 | Math` | what are u looking for |
07:04.11 | Math` | docs of what |
07:04.43 | infinity1 | you know the replacement for using s,1 - s,2, etc. and use s,1 - s,n - s,n - etc. |
07:05.14 | Math` | uhu |
07:05.18 | infinity1 | i've seen it used, haven't ran across docs |
07:05.26 | Math` | never saw that heh |
07:05.47 | DarthClue | infinity1: did you check the wiki? |
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07:06.01 | infinity1 | DarthClue: yea. i searched around. didn't see it. |
07:06.19 | emp | anyone have anything good/bad to say about teliax? |
07:07.29 | infinity1 | DarthClue: you use it? did you learn it by reading other exmaples? |
07:08.49 | infinity1 | oh shit. head has a color console. |
07:09.00 | infinity1 | p.i.m.p. :) |
07:09.02 | DarthClue | infinity1: no, i don't personally, but it should be pretty straightforward. |
07:09.18 | DarthClue | infinity1: all the cool toys are in HEAD. That's why most of us recommend it. |
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07:09.34 | infinity1 | what else do i get? |
07:09.36 | infinity1 | heh |
07:09.50 | DarthClue | infinity1: i can tell you what you don't get...the kitchen sink. |
07:10.01 | infinity1 | uhh. |
07:10.06 | infinity1 | the disposal? |
07:11.25 | infinity1 | typing "stop now" is too much work. ctrl-d ?? |
07:12.03 | hellop | "up arrow" |
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07:12.16 | hellop | infinity1.. |
07:12.18 | DarthClue | use stop gracefully, it's much kinder. |
07:12.20 | infinity1 | maybe |
07:12.31 | infinity1 | DarthClue: even for devleopment |
07:12.31 | infinity1 | ? |
07:12.51 | DarthClue | yes, if there is nothing happening, then gracefully will stop just as quickly as stop now. |
07:13.09 | hellop | infinity1, I type the up arrow button to go back to last command.. |
07:13.17 | infinity1 | lets see if the cli can interpret shortcuts |
07:13.27 | infinity1 | ah boo |
07:13.56 | DarthClue | stop now is for when the drunken pulls on to the train tracks and you are traveling at 50 mph with a mile of cargo behind you. |
07:14.02 | infinity1 | where did they get this cli? 3com netbuilder series? heh |
07:14.24 | infinity1 | well, i like the color. |
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07:34.19 | Newbie___ | hi , i am using h323, is there any way to know i am connected to a gatekeeper? |
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07:38.29 | RoyK | Newbie___: don't use h.323 with asterisk :P |
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07:39.05 | Newbie___ | RoyK: i am more familiar with SIP, but the other end insist on h323 |
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07:40.12 | hellop | Anyone got a Symbol MC50? they look cool |
07:40.15 | Newbie___ | looking at http://www.gnugk.org/ |
07:41.46 | tzafrir | BTW: should ctrl-c on the cli (asterisk -r, not -c) save the history? any reason for it not to? (apart from "nobody has yet bothred") |
07:41.47 | tzafrir | ? |
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07:42.24 | TheEmperor | hello |
07:42.34 | TheEmperor | can someone tell me how to fix this? : Jul 12 15:33:05 NOTICE[1289744832]: rtp.c:275 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
07:42.34 | TheEmperor | Jul 12 15:33:05 NOTICE[1289744832]: rtp.c:304 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 |
07:45.54 | Math` | change the dtmfmode |
07:46.04 | Math` | because your codec doesnt support rfc3389 |
07:46.07 | Zeeek | TheEmperor IIRC the first has to do with transmitting silence which youneed to disable in X-Lite. Or VAD |
07:46.29 | Zeeek | what is the client? |
07:46.36 | Math` | ah 2833 is dtmf, nevermind my comment |
07:46.50 | Zeeek | interesting while it lasted |
07:47.25 | RoyK | Math`: rfc3389 has nothing to do with dtmf |
07:47.39 | RoyK | TheEmperor: turn off silence suppression on the client |
07:48.12 | brenda | if a one minute conversation takes up 60k on my hard drive... how many minutes will fit on a theorectically empty 120gig hard drrive? |
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07:49.35 | DarthClue | brenda...rough number here...but about 1.5 million minutes. |
07:50.03 | brenda | wow! |
07:50.16 | TheEmperor | i'm using x lite as well as a hard phone |
07:50.39 | TheEmperor | disable vad? |
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07:51.24 | TheEmperor | royk: how to do i turn off silence supression on the client? |
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07:51.39 | Zeeek | what client? |
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07:51.53 | TheEmperor | Zeeek: client means mine or the provider? |
07:52.10 | brenda | so about 25,000 hours |
07:52.10 | Zeeek | client is the phone |
07:52.10 | Zeeek | what is it? |
07:52.10 | TheEmperor | it's an atcom 320 |
07:52.28 | Zeeek | uncheck vad |
07:52.59 | TheEmperor | should i leave agc and aeg checked? |
07:53.21 | Zeeek | aec is checked on mine so I authorize to leave it checked |
07:53.28 | TheEmperor | ok |
07:54.35 | TheEmperor | i put dtmf as rfc 2883, is that ok? |
07:55.04 | Zeeek | why change 50 things at once? It'll only make it harder to find the problem |
07:55.16 | TheEmperor | ok |
07:55.39 | TheEmperor | i was just wondering what the optimal settings were |
07:56.01 | Zeeek | depends on what you're connecting to |
07:56.06 | Zeeek | that's why there are choices |
07:57.32 | Zeeek | I wonder if I should have a coffee or install HEAD? |
07:58.15 | DarthClue | Zeeek: install HEAD, you can get coffee while asterisk is building. |
07:58.33 | Zeeek | already built - just a make install needed |
07:58.42 | Zeeek | it's an old head |
07:58.53 | DarthClue | Zeeek: get new HEAD, then get coffee. |
07:58.54 | Zeeek | I'm scared to run head in production |
07:59.02 | Zeeek | is new HEAD good? |
07:59.19 | Zeeek | this is June18th |
07:59.25 | DarthClue | Zeeek: I'm pretty sure i saw bkw get HEAD from earlier today put it in production. |
07:59.26 | Zeeek | shit that's older than I thought |
07:59.41 | DarthClue | Zeeek: you might be surprised how well head runs in production. |
07:59.58 | Zeeek | I know but I don't wanna be surprised by horrible problems |
08:00.10 | Zeeek | "better safe than sorry" |
08:00.18 | DarthClue | Zeeek: then don't use asterisk. asterisk is a horrible problem. |
08:00.23 | TheEmperor | still get the same errors |
08:00.29 | TheEmperor | Jul 12 15:51:27 NOTICE[1289744832]: rtp.c:275 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
08:00.29 | TheEmperor | Jul 12 15:51:27 NOTICE[1289744832]: rtp.c:304 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 |
08:00.50 | Zeeek | no on the contrary, when you have it working and don't need any new features it is a mystery why I'm even considering updating! :) |
08:01.11 | DarthClue | Zeeek: because you need to, everyone should run head. |
08:01.15 | Zeeek | haha |
08:01.22 | Zeeek | everyone should GET HEAD |
08:01.24 | TheEmperor | Zeeek: any ideas? |
08:01.29 | Zeeek | beyond that, I dunno |
08:01.30 | DarthClue | Zeeek: that too. |
08:01.48 | Zeeek | TheEmperor what are yopu calling when you get that error? |
08:02.01 | Zeeek | just you and your phone or a provider? |
08:02.18 | TheEmperor | Zeeek: i am putting the call through to a provider |
08:02.20 | Zeeek | also look that up on the mailing list it's a common message |
08:02.21 | TheEmperor | using h323 |
08:02.25 | TheEmperor | ok |
08:02.38 | Zeeek | H323!!!! fuck. If I'da known that... |
08:03.05 | TheEmperor | ? |
08:03.15 | Zeeek | I don't know anything about 323 at all |
08:03.24 | TheEmperor | o |
08:03.32 | Zeeek | my advice is even more useless than usual |
08:03.44 | TheEmperor | haha |
08:03.46 | Zeeek | "try reinstalling windows 95" |
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08:03.49 | TheEmperor | ok thanks anyway :) |
08:03.51 | Delvar | hehe |
08:03.58 | Zeeek | "over XPPro" |
08:03.59 | DarthClue | Zeeek: but he's running windows 1.0? |
08:04.11 | RestLessGemini | Hi All |
08:04.15 | Zeeek | I'm old enuf to have used WIndows 3 |
08:04.29 | Zeeek | God did windows SUCK BAD in those days |
08:05.07 | DarthClue | Zeeek: I remember windows 1.0...And windows didn't suck, it created and vortex that destroyed everything that got near it. |
08:05.09 | Zeeek | my TRS-80 was more stable (no gui tho) |
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08:05.51 | Zeeek | adding the 32k RAM was a rush :) |
08:06.05 | Zeeek | quite right |
08:07.01 | Zeeek | why do I have a module called "isdn" in lsmod ? |
08:07.20 | djin | Hope someone can point me in the right direction for the following. We use Cisco 7940/7960's to connecto to *. Sometimes while calling another person the caller the call gets 'muted'. It seems to pickup again if when the called talks. In the mentime there is a very clear silence. |
08:07.26 | RoyK | Zeeek: modinfo isdn? |
08:07.47 | RoyK | Zeeek: using your favourite 'add everything there is' kernel distro? |
08:07.56 | Zeeek | but I don't have anything related to ISDN on this box |
08:08.09 | tuxinator_linuxM | Zeeek: It was crazy when I upgraded my ram from 512k to 2 Megs on my 286 |
08:08.33 | Zeeek | they wanted $12,000 to add 2 Megs to our miniVAX |
08:08.57 | tuxinator_linuxM | Cost me several hundred |
08:09.08 | tuxinator_linuxM | Don't really miss those days |
08:09.16 | Zeeek | I think I remember 1 meg at $20 at one point |
08:09.47 | tuxinator_linuxM | these were chips you had to stick on a daughter board |
08:09.53 | Zeeek | but then my first printer cost $600 (no cable incl) |
08:10.17 | tuxinator_linuxM | I had a nice 24 pin |
08:10.24 | justink | i assume that wasnt a dot matrix for $600 |
08:10.41 | Zeeek | so is there any point in removing drivers that are "used by 0" ? |
08:10.50 | tuxinator_linuxM | justink: Evening |
08:10.56 | Zeeek | Epson MX80 dot matrix, slow as molasses |
08:11.19 | justink | tuxinator: morning to ya. |
08:11.26 | tuxinator_linuxM | Zeeek: But you had some much fun with Print Shop |
08:11.28 | Zeeek | pardon my flood |
08:11.29 | Zeeek | i810_rng 2656 0 (unused) |
08:11.29 | Zeeek | hisax 449444 0 (unused) |
08:11.29 | Zeeek | isdn 117184 0 [hisax] |
08:11.29 | Zeeek | slhc 5040 0 [isdn] |
08:11.29 | Zeeek | isa-pnp 30724 0 [hisax] |
08:11.41 | Zeeek | i810_rng 2656 0 (unused)any reason to care? |
08:11.54 | Zeeek | oops I went over the line there |
08:12.02 | Zeeek | hahah POrint SHop |
08:12.30 | Zeeek | I don't think there are any ISA slots |
08:12.49 | Zeeek | still haven't chosen between HEAD and coffee |
08:13.01 | tuxinator_linuxM | Zeeek: Sorry dude, a little late for that stuff |
08:13.24 | tuxinator_linuxM | Zeeek: Eat a grapefruit |
08:13.28 | Zeeek | moot point. I forgot to turn on the espresso machine :( |
08:13.53 | Zeeek | Voluto, anyone? |
08:15.06 | tuxinator_linuxM | Volu...who? |
08:17.11 | RoyK | Zeeek: what is i810_rng? |
08:17.12 | Zeeek | It's my favorite in the morning |
08:17.30 | RoyK | anyway - hisax is just the base driver. no card driver is loaded |
08:17.36 | Zeeek | random number generator |
08:17.45 | RoyK | ah |
08:17.57 | Zeeek | still hisaw is half a meg |
08:18.12 | Zeeek | no audio card, no usb, no scsii |
08:18.15 | RoyK | Ăºsing 2.4 or 2.6? |
08:18.23 | Zeeek | 2.4 |
08:18.33 | RoyK | configure modutils to stop loading it, then |
08:18.52 | RoyK | isa-pnp might as well be unneeded :) |
08:18.58 | Zeeek | I wonder why ideèscsi says no? there is an ide |
08:19.36 | Zeeek | well, if it works... |
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08:33.50 | michael1234 | anyone in here know how to send a sip call to a billion |
08:34.36 | djin | michael1234, not many here would be willing to support that. |
08:34.38 | afv-13 | i seem to have a delay when the caller hits a key during a background, and when it moves to the next extension |
08:34.52 | djin | Hope someone can point me in the right direction for the following. We use Cisco 7940/7960's to connecto to *. Sometimes while calling another person the caller the call gets 'muted'. It seems to pickup again if when the called talks. In the mentime there is a very clear silence. |
08:36.03 | michael1234 | djin: I have my own asterisk box just trying to work out how to send a call to my registered billion box |
08:36.51 | djin | "billion box' ? |
08:36.53 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
08:36.58 | fenlander | djin: sounds like a problem with VAD. Asterisk needs a continuous rtp stream to clock the outgoing data |
08:37.26 | fenlander | djin: turn off any silence supression |
08:37.50 | michael1234 | djin: http://www.billion.com.au/product/voip/bipac7402vl.php |
08:37.57 | djin | fenlander, I was looking into that, but does is require VAD=0 or 1? |
08:38.17 | djin | michael1234, ah /me was thinging of voice spam ;) |
08:38.28 | fenlander | djin: I have vad: 0 in my SIPDefault.cnf |
08:39.19 | fenlander | djin, sorry, enable_vad: 0 :) |
08:39.58 | djin | fenlander, vad was disabled by default, so wasn't sure this would help. |
08:41.00 | fenlander | djin: might be worth checking that it is working by running ethereal on the asterisk box |
08:41.13 | djin | there is no 'comfort noice' setting, is there? |
08:41.27 | djin | ok, will try that. |
08:42.53 | *** join/#asterisk florinm (~florin@81-178-45-7.dsl.pipex.com) |
08:43.01 | florinm | morning boys |
08:43.12 | florinm | anyone heard about goldpbx? |
08:44.03 | florinm | no , the software :P |
08:44.35 | florinm | i found the goldbpx somewere and is fully asterisk, but all name changed |
08:44.44 | florinm | and changed the copyright info |
08:44.51 | RoyK | strange |
08:44.58 | RoyK | google didn't give any hints |
08:45.00 | djin | RoyK was a fan of Beavis & Butthead: "Fire Fire Fire Fire Fire !!!" :) |
08:45.02 | florinm | and i was wandering if this doesn't break the licence of the asterisk |
08:45.14 | fenlander | Maybe they have a commercial license |
08:45.40 | RoyK | fenlander: i somehow doubt digium would allow people to re0wrap it |
08:45.49 | RoyK | florinm: have you got the software somewhere? |
08:45.53 | RoyK | can i have a look? |
08:45.57 | florinm | i didn't saw anywere saying about digium and asterisk |
08:46.09 | Newbie___ | RoyK: what do u recommend if i were to to h323 |
08:46.16 | Delvar | shot to the head |
08:46.25 | RoyK | Newbie___: dunno. yate might help |
08:46.27 | florinm | i need somewre to upload it |
08:46.31 | RoyK | ~lart Delvar |
08:46.32 | florinm | as is 20 meg |
08:46.37 | Delvar | :) |
08:46.40 | florinm | (is allreadu v 4.0.1 |
08:47.25 | fenlander | florinm: what is the license it came with? |
08:48.04 | Newbie___ | RoyK: yate does SIP to h323 signaling |
08:48.04 | florinm | i saw no licence info on it |
08:48.23 | florinm | let me check it :P |
08:48.34 | RoyK | Newbie___: I don't know if it does SIP any good |
08:48.38 | RoyK | Newbie___: perhaps chan_woomera |
08:48.44 | RoyK | Newbie___: that might be best |
08:48.51 | RoyK | that's with asterisk |
08:49.30 | Newbie___ | i am shit out of luck with oh323 and h323 |
08:50.06 | RoyK | Newbie___: try chan_woomera |
08:51.45 | Newbie___ | RoyK: installation seem simple enough |
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08:52.52 | Zeeek | lo PoWeRKiLL |
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08:54.28 | bogdanro | hello |
08:54.57 | RoyK | hi |
08:56.51 | Zeeek | ok now I'm in real trouble |
08:57.01 | *** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com) |
08:57.09 | bogdanro | can anybody here help me to setup a oh323 trunk on my asterisk ? |
08:57.16 | limbique | hi |
08:57.43 | limbique | i have found some wicked things on the asterisk |
08:58.05 | limbique | when i call from external to a voip phone |
08:58.32 | limbique | when i hangup the voip phone, the asterisk reserves an extra zap channel |
08:58.38 | limbique | anyone knows? |
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09:03.16 | limbique | kinda dead here? |
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09:05.23 | djin | fenlander, are you getting these "RFC3389 support incomplete." messages as well? |
09:09.29 | fenlander | djin: don't remember them |
09:09.45 | fenlander | what is 3389? |
09:11.13 | djin | fenlander: |
09:11.14 | djin | What does this RFC3389 error message in the CLI/my log try to tell me? |
09:11.14 | djin | This RFC refers to silence suppression. For example X-Lite has a configuration setting called "Transmit silence" where the behaviour can be controlled by the user. Anyway, Asterisk is complaining about silence supression. It's a harmless message, but if you can turn off silence suppression in your SIP client the message will go away. Related keywords are "VAD" and "comfort noise". |
09:11.34 | *** part/#asterisk pif (ldm@zenon.apartia.fr) |
09:11.35 | Zeeek | I have a major problem: I can't seem to get back to STABLE. |
09:11.58 | Zeeek | how is /etc/moidules.conf created? is it touched during install ? |
09:13.24 | fenlander | Zeek: I thought it just came out of configs/*.sample like the others |
09:13.42 | fenlander | Zeek: what is the error? |
09:14.01 | Zeeek | talking about the /etc/modules.conf file |
09:14.02 | fenlander | djin: must have some form of VAD still turned on somewhere? |
09:14.11 | fenlander | Zeek - oh |
09:14.20 | Zeeek | The change from wcfxs to wctdm has fscked up something on my machine |
09:14.39 | limbique | i have an some like, that asterisk reserves and immidiate release an extra channel when i hangup a external call from a voipphone |
09:14.50 | limbique | extra zap channel |
09:15.09 | Zeeek | astrisk is now dying at a random point during its startup |
09:15.24 | Zeeek | perhaps I need to delete all modules and start again |
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09:16.05 | djin | fenlander, "Enable VAD NO" |
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09:17.15 | indra | Hello all |
09:20.08 | *** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122) |
09:20.11 | TheEmperor | hi guys |
09:20.17 | TheEmperor | anyone know how to fix this? |
09:20.20 | TheEmperor | Unknown RTP codec 19 received |
09:22.15 | fenlander | TheEmperor: sounds like confort noise - turn it off on your phone (what is it with comfort noise today?) |
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09:22.52 | TheEmperor | ok |
09:23.38 | fenlander | djin: all I can say is that it works for me :-( You could try rebooting the phone I guess |
09:23.50 | Delvar | Zeeek: yeah delete /modules dir and do a make install. |
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09:24.02 | kajtzu | hmph |
09:24.11 | Zeeek | I just deleted the "wrong" stuff but screwed up I think on something else |
09:24.23 | Zeeek | I removed a reference to wctdm in /etc/modules.conf |
09:24.45 | Zeeek | asterisk is fine now but I bet if I reboot the drivers don't load right |
09:25.04 | Delvar | oh wait.. your talking about linux kernel modules?.. thought you were talking about asterisk |
09:25.07 | Zeeek | someone tell me about these post-install lines? |
09:25.22 | Zeeek | NO the modules I eliminated WERE asterisk |
09:25.37 | Zeeek | but now for zaptel, I think I screwed up the /etc/modules.conf |
09:25.38 | Delvar | im confused! |
09:25.57 | djin | fenlander, thanks. I'll continue my search :) |
09:25.59 | Zeeek | that's what they get for have two modules.conf files |
09:26.03 | Delvar | eh its still morning and my brain isnt working |
09:26.13 | Zeeek | what do the post-install lines do? |
09:26.28 | Zeeek | and where did they come from? |
09:26.31 | *** join/#asterisk jansaell (~jan@c80-216-185-161.cm-upc.chello.se) |
09:26.56 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
09:26.57 | RoyK | goldenpbx from http://www.xoasisnetworks.com/ seems to be extremely similar to asteriskk |
09:27.02 | puzzled | morning |
09:27.48 | RoyK | puzzled: morning |
09:28.35 | *** join/#asterisk helpmeguys (~helpmeguy@222.253.87.143) |
09:29.27 | *** join/#asterisk lobbin (~lobbin@c83-253-149-45.bredband.comhem.se) |
09:29.59 | Zeeek | what is this: |
09:30.00 | Zeeek | zt_rbs: Tried to set RBS hook state 0 on channel WCTDM/0/0 while span WCTDM/0 lacks rbsbits or hooksig function |
09:30.23 | Delvar | http://www.xoasisnetworks.com/products/prodigy-voip/screenshots.php?img=ss-wop.png thats flash operator pannel :) |
09:30.30 | Delvar | mut be asterisk |
09:30.45 | fenlander | RoyK: I don't see any goldenpbx - where is it on the site? |
09:30.56 | Zeeek | what is zt_rbs? |
09:30.58 | djin | 2005-07-11 - Xoasis Networks offers 20% discount code on all IP PBX products to VoIP Info readers, coupon code: voip. |
09:31.02 | djin | voip-info.org |
09:32.59 | PoWeRKiLL | Salut Zeeek |
09:33.02 | RoyK | fenlander: florinm got through some hole there and found the code |
09:33.24 | Zeeek | jour PoWeRKiLL |
09:33.35 | Zeeek | cat modules.conf |
09:33.48 | Zeeek | that's where I am, typing in the wrong windo |
09:34.19 | Zeeek | someone help me with linux post-install lines ? |
09:35.06 | TheEmperor | RoyK: any idea on this? rtp.c:434 ast_rtp_read: Unknown RTP codec 19 received |
09:37.58 | *** join/#asterisk mithro (~tim@195.177.247.8) |
09:38.56 | RoyK | fenlander: http://pastebin.ca/17529 |
09:39.36 | Zeeek | let me try another angle: I already have modprobes in rc.local for startup. WHat are the post-install lines in /etc/modules.conf ? They seem to have been added by my failed install of HEAD? |
09:40.27 | puzzled | RoyK: if that isn't * I don't know what is |
09:41.52 | RoyK | puzzled: right |
09:43.34 | fenlander | RTP codec 19 was for Comfort Noise in a draft spec. It is now reserved, but you will find some devices that still use it. |
09:45.20 | RoyK | puzzled: seems this company creates 'embedded pbx systems' |
09:45.40 | Zeeek | RoyK what distro are you running? |
09:45.58 | Zeeek | no wait I need someone who has both X100 and TDM400 ? |
09:47.44 | *** join/#asterisk outsidefactor (chrismarti@203-206-254-183.dyn.iinet.net.au) |
09:48.01 | Newbie___ | i have a X100 and TE410 |
09:48.05 | Zeeek | I see that zaptel Makefile writes those lines to /etc/modules.conf |
09:48.21 | Zeeek | HEAD apparently writes a lot more lines |
09:48.39 | Zeeek | pciradio etc |
09:49.36 | puzzled | Zeeek: yup, check the Makefile. it's all in there |
09:50.00 | Zeeek | yeah but in the meantime I'd like to know what has to be in the file so I can reboot |
09:50.02 | puzzled | Zeeek: alternatively you can just load them "manually" in /etc/rc.d/rc.local |
09:50.11 | Zeeek | I do that |
09:50.15 | puzzled | me too |
09:50.24 | Zeeek | so I don't need any of the modules.conf post-install et al? |
09:50.46 | Zeeek | it also looks like it added options |
09:50.55 | Zeeek | and what is torisa? |
09:50.57 | puzzled | I don't use it. just load them from rc.local. iirc cause I couldn't figure out the modprobe.conf stuff :) |
09:51.04 | Zeeek | and tor2 ? |
09:51.23 | puzzled | which card do you use? |
09:51.23 | TheEmperor | fenlander:how do i get rid of that? |
09:51.37 | Zeeek | X100P and TDM400P |
09:51.42 | TheEmperor | fenlander: the rtp 19 codec i mean, is that coming from my side or the otherside? |
09:52.28 | puzzled | Zeeek: then I think you just need the wcfxo module for the X100P. dunno the module's name for the tdm400p |
09:52.38 | Zeeek | wcfxs |
09:52.59 | Zeeek | I have this in my rc.local: modprobe of zaptel, wcfxo, wcfxs ztcfg |
09:53.24 | puzzled | think that should work |
09:53.36 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
09:53.45 | Zeeek | since they changed the tdm driver to wctdm (no more wcfxo) |
09:58.50 | Zeeek | what is torisa ? anyone ? |
09:59.04 | puzzled | i think those are the really old cards |
10:00.00 | Zeeek | this backstage shit has me confused. No wonder I get so scared when I try to update |
10:00.30 | puzzled | just make some good notes how you made it work like you want it to and keep those handy when updating |
10:02.03 | Zeeek | muhahahaha notes! I'd have to find the notebook! |
10:02.13 | Zeeek | before and after |
10:03.03 | *** join/#asterisk TimmiG (~TimmiG@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
10:03.13 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
10:03.42 | lobbin | I'm having some problems with my sip phone, I can dial-in but I can't dial-up |
10:05.14 | Zeeek | plus delete all modules"what are the symptoms? |
10:05.28 | Zeeek | what are the symptoms? |
10:05.40 | Zeeek | asterisk -r |
10:05.44 | TimmiG | Hi, I've a question, can asterisk use Advice of Charge from the ISDN side? And how I have to configure it. I use a zaphfc-card |
10:06.11 | puzzled | TimmiG: search the list archives |
10:06.21 | TimmiG | OK, I try |
10:06.52 | Zeeek | lobbin you have to tell us more about what happens and what you are trying to do |
10:07.17 | RoyK | how are US laws with regard to GPL protection? |
10:07.27 | Zeeek | they lock you up |
10:07.52 | lobbin | Zeeek: it tries to make the call to my sip provider, but asterisk seems to send wrong information |
10:08.41 | Zeeek | how can anyone know what the problem is if you don't describe it? Are you running @home or AMP by any chance? |
10:09.44 | Zeeek | lobbin need to see the dial line and the errors asterisk shows |
10:09.48 | lobbin | http://pastebin.com/311772 |
10:09.50 | Zeeek | ~pastebin |
10:09.50 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
10:09.56 | Zeeek | ok you're ahead of me :) |
10:09.59 | lobbin | :) |
10:10.17 | Zeeek | what part of WRONG PASSWORD do you not get? |
10:11.20 | lobbin | If you look at the end of that line, in the <sip:>, it seems to send asterisk as my username? |
10:11.42 | Zeeek | paste you peer config |
10:11.50 | Zeeek | sans passwords of course |
10:12.25 | Zeeek | digisip is the provider? |
10:12.39 | lobbin | http://pastebin.com/311774 |
10:12.41 | lobbin | Yeah |
10:13.11 | Zeeek | you have any other SIP providers working? |
10:13.34 | lobbin | no, but it did work before I tried to put it trough asterisk |
10:14.05 | Zeeek | you mean it works with just a phone? |
10:14.35 | lobbin | No, I got a dlink 1120 that I'm using, I need to use it through asterisk as this pos doesn't work behind a nat |
10:19.18 | Zeeek | ok I was interrupted for a moment |
10:19.32 | Zeeek | something is wrong with your username or password |
10:20.19 | lobbin | Yeah, but what, it seems that asterisk is sending asterisk as my username? |
10:20.48 | *** join/#asterisk ady (~root@202.5.145.13) |
10:21.03 | lobbin | You can see the header here: http://pastebin.com/311776 |
10:21.09 | Zeeek | no that's normal |
10:21.31 | Zeeek | <PROTECTED> |
10:22.01 | lobbin | ok |
10:22.38 | lobbin | Hmm, it does seem to send Proxy-Authorization further down |
10:22.40 | PoWeRKiLL | Is there a possibility to generate a random number without using agi in extension.conf ? (If i remember there vas a variable rand ?) |
10:22.55 | Zeeek | I think there may be in HEAD |
10:23.44 | Zeeek | PoWeRKiLL quel est ton prénom, je les confond avec les pseudos |
10:24.00 | tzafrir | PoWeRKiLL, I was looking for something similar recently |
10:24.15 | *** join/#asterisk eldu (~damajor@tuxmania.org) |
10:24.19 | Zeeek | or it would be easy to write the app in C |
10:24.24 | eldu | hello |
10:24.38 | tzafrir | I want to make a nice ivr-like context with all the silly sounds in asterisk-sounds |
10:24.58 | Zeeek | remind me not to call you for mission-critical info |
10:25.38 | *** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122) |
10:26.04 | tzafrir | Zeeek, asterisk is not exactly on-par with some "proffesional" PBX-s. But which PBX has a prompt "will you marry me?" |
10:26.48 | lobbin | Zeeek: looking at the headers it seems like asterisk is sending correct headers, so why is the password "wrong"? Atleast the username is sent correctly |
10:27.16 | Zeeek | lobbin that "no record" message may mean you are trying to register with a wrong proxy |
10:27.31 | eldu | is there a frontend to easily manage dialplans ? |
10:27.58 | tzafrir | eldu, not exactly |
10:28.00 | Zeeek | there are a few, but you won't hear a lot of people recommending them here |
10:28.13 | eldu | hmmm |
10:28.27 | tzafrir | there are some front ends for that, which will limit you with the choice of actions |
10:28.43 | eldu | i saw that there r many web interfaces to manage sip/iax client but not for dialplans |
10:28.57 | tzafrir | Others are basically glorified editors, and to use them you have to understand how the dialplan works |
10:29.02 | eldu | tzafrir: that's what i saw |
10:29.23 | eldu | ok ok |
10:29.31 | Zeeek | the manager allows adding extensons dynamically |
10:29.36 | tzafrir | take your pick. If you want the full power, you'll have to understand how it works |
10:30.15 | Zeeek | that's my problem with linux. I don,'t know anything about it |
10:30.16 | eldu | Zeeek: its a good idea but a global view of the dialplan is also a feature i need |
10:30.22 | *** join/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
10:30.33 | MuppetMaster | Hello everyone. |
10:30.40 | Zeeek | you could write something thatwould be like a FOP for dialplan - that's be cute |
10:30.50 | tuxinator_linuxM | Night guys |
10:30.51 | eldu | tzafrir: no prob understanding what i did :) |
10:31.09 | MuppetMaster | I am having a real nerve wracking issue with a Sipura 3K and the latest CVS HEAD of Asterisk (as of yesterday). |
10:31.12 | Zeeek | bye tuxinator_linuxM |
10:31.13 | eldu | Zeeek: i'll plan to do something if i really need it ;) |
10:31.38 | MuppetMaster | I have logged the issue in the Voxilla forums for the Sipura users group. http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&p=21329#21329 |
10:31.59 | *** join/#asterisk pelo (~tcs@adsl203-149-051.mclink.it) |
10:32.26 | *** part/#asterisk florinm (~florin@81-178-45-7.dsl.pipex.com) |
10:32.55 | tzafrir | Zeeek, things like AMP aim to do that. But the price is that they have to make some assumptions about the dialplan |
10:33.06 | MuppetMaster | In essence, the Line 1 on my SPA3K registers, but not completely (somehow). When I try to dial I get: NOTICE[3209]: chan_sip.c:6208 check_user_full: From address missing 'sip:', using it anyway |
10:33.14 | MuppetMaster | And WARNING[3209]: chan_sip.c:5801 get_destination: Huh? Not a SIP header (3006)? |
10:33.14 | Zeeek | I've never seen AMP or @home |
10:33.28 | MuppetMaster | Thing is that I have a Sipura 2K with the same configuration working without a problem. |
10:33.38 | MuppetMaster | And the PSTN line of the device registers without a problem. |
10:33.42 | lobbin | Zeeek: I guess that something is not correct in my settings then? |
10:34.02 | Zeeek | lobbin sometimes providers have a proxy system with two different servers |
10:34.13 | Zeeek | I have one like that, it's a pain to set up |
10:34.47 | Zeeek | are you registering tothis provider? |
10:35.05 | lobbin | Zeeek: Yes, register to the provider works including incoming calls |
10:35.25 | Zeeek | you see, outgoing may require a different ip |
10:35.38 | Zeeek | were there no instructions for configuring? |
10:35.46 | Zeeek | heh maybe not with asterisk |
10:36.10 | lobbin | I have instructions for asterisk and I followed them and ended up with this :) |
10:36.33 | Zeeek | is there always the same ip or domain name inthe instructions or more than one? |
10:36.42 | lobbin | it's always the same |
10:36.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
10:37.18 | MuppetMaster | I am also getting this: Jul 12 12:32:43 NOTICE[23730]: chan_sip.c:5675 register_verify: Invalid to address: '3006' from my_ip (missing sip:) trying to use anyway... |
10:38.18 | Zeeek | lobbin it appears right - you sure there isn't a typo in user or pass? Perhaps you want to limit codecs? disallow=all allow=ulaw or sth |
10:40.04 | Zeeek | no it's clear that it doesn't like the pasword |
10:40.32 | Zeeek | you have a username AND a number ? Maybe you're supposed to log in with the number? |
10:42.46 | lobbin | yea, I have a username and a number |
10:43.02 | lobbin | I can try, but the the userguide says it should be the username |
10:47.17 | PoWeRKiLL | Zeeek pv |
10:47.31 | PoWeRKiLL | tzafrir and you find something ? |
10:48.37 | Zeeek | PoWeRKiLL I'm going to lunch - back in 30 or so min |
10:48.55 | PoWeRKiLL | ok |
10:48.59 | PoWeRKiLL | Bon appetit |
10:49.11 | *** join/#asterisk chezgi_ (~user@217.219.15.242) |
10:49.20 | lobbin | Zeeek: no luck changing the criterias |
10:49.26 | Zeeek | c olivier? je suis perdu entre les pseudos |
10:49.47 | Zeeek | lobbin, if it makes you feel any better, I'm having major linux stupidity (mine) here |
10:50.01 | lobbin | it does :) |
10:50.28 | Zeeek | lobbin try logging in with a phone? |
10:52.27 | Zeeek | what is zt_rbs ? |
10:53.27 | Zeeek | aha |
10:53.43 | Zeeek | I can't get rid of the damn calls to wctdm - well lunch for now |
10:54.50 | MuppetMaster | Also, I can see the address is missing in the Asterisk internal Database as seen from the CLI: |
10:55.04 | MuppetMaster | What would cause the SIP URI to be missing in the Asterisk database as per 3006? |
10:55.13 | MuppetMaster | While 3000 from the same Sipura 3000 contains the SIP URI. |
10:55.25 | lobbin | I'm out for lunch now as well |
10:56.28 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
10:56.43 | onkeltimm | helay |
10:57.18 | onkeltimm | if i do an attended transfer, and the person i want to transfer to does not pick up, how do i get the call back? |
10:57.41 | RoyK | hm... now if digium was to license asterisk to someone |
10:58.08 | RoyK | would they also allow that company to replace all references to digium/asterisk/developers by their own people? |
10:58.32 | RoyK | and rename the suite |
10:58.37 | RoyK | and set a new version on it |
11:00.35 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
11:02.04 | *** join/#asterisk goldenear (~goldenear@m209.net81-64-245.noos.fr) |
11:02.46 | MuppetMaster | RoyK You will most likely need to sign an NDA with Digium and ask them what their licensing terms are. |
11:02.58 | RoyK | http://www.xoasisnetworks.com/ |
11:03.02 | RoyK | see the demo |
11:03.08 | MuppetMaster | RoyK If they are willing to do a true OEM with the license than yes. But maybe they are not. No one may answer that but Digium themselves. |
11:04.01 | *** join/#asterisk alphaque (~alphaque@60.48.197.184) |
11:04.40 | kd5uzz | is there any software out that would allow a pocketPC to connect to an asterisk server as a voip phone? |
11:04.57 | goldenear | I wonder, is adding "exten => h,1,Hangup" in a dial plan useful ? |
11:04.58 | tzafrir | RoyK, did you try contacting the company ans asking them before posting on a public mailing list? |
11:05.05 | goldenear | kd5uzz, try ziax |
11:05.13 | kd5uzz | thanks |
11:05.30 | goldenear | if you run linux on your PDA |
11:05.39 | RoyK | tzafrir: nope... didn't think about it. i was just given the info i posted. should have done so, sorry |
11:05.53 | tzafrir | BTW: where can I get this libpri 4.0.1? these guys seem to live well ahead of us |
11:05.58 | goldenear | else you can try something comercial from xten |
11:06.22 | kd5uzz | doh..k |
11:06.31 | goldenear | kd5uzz, make a search at voip-info.org |
11:06.31 | *** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com) |
11:06.54 | Zeeek | somebody gotta help me wit this zt_rbs problem |
11:06.58 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
11:09.11 | MuppetMaster | kd5uzz: SJPhone runs on PocketPC: http://www.sjlabs.com/products.html |
11:09.18 | MuppetMaster | Xten might too, just not sure. |
11:11.04 | goldenear | http://www.xten.com/index.php?menu=products&smenu=xproppc |
11:11.19 | *** join/#asterisk Morex (~blah@host81-157-165-204.range81-157.btcentralplus.com) |
11:11.23 | Morex | Hi there |
11:11.34 | goldenear | hi Morex |
11:11.35 | Morex | Anybody else having trouble logging on to cvs.digium.org? |
11:11.43 | Morex | 'Sup Goldenear |
11:12.12 | fenlander | cvs is good for me |
11:12.16 | Morex | Huh... |
11:12.18 | Morex | OK thanks |
11:12.32 | *** part/#asterisk vuvie (~vuvie@bb219-74-45-31.singnet.com.sg) |
11:13.50 | goldenear | Could you help me please I have 2 (existential ?) questions about dial plan / extensions.conf : |
11:14.19 | goldenear | 1) is exten=>h,1,Hangup necessary or is it implicit ? |
11:14.53 | ManxPower | goldenear: You NEVER need exten => h,1,Hangup. |
11:15.01 | goldenear | 2) when exactly it's necessary or not to use the "r" option of Dial ? |
11:15.07 | ManxPower | Extension h is only called when the call has ALREADY hungup. |
11:15.14 | Zeeek | ManxPower when you make zaptel, where does it install modules? |
11:15.16 | ManxPower | goldenear: you almost never need "r" option to dial. |
11:16.16 | goldenear | does "r" really kills call progress information |
11:16.20 | goldenear | ? |
11:16.44 | ManxPower | goldenear: Yes. |
11:16.49 | goldenear | ok |
11:16.59 | ManxPower | which is why you almost never need it. |
11:17.29 | goldenear | I was confused because I found a lot of extensions.conf examples using "r" all the time ;) |
11:17.42 | ManxPower | goldenear: they are written by newbies |
11:18.18 | goldenear | I guess... |
11:19.52 | goldenear | also, I guess Hangup is needed at the last priority for each extensions, isn't it ? |
11:20.10 | goldenear | exten => 10,1,Dial(IAX2/user) |
11:20.21 | goldenear | exten => 10,2,Hangup |
11:21.45 | ManxPower | goldenear: Most of the time Asterisk will do the right thing, but it's still a good idea to have Hangup in that case. |
11:22.05 | goldenear | I notice that, when Hangup is not used and the called part is not registred, Calling part needs more time to hangup |
11:22.54 | goldenear | and with iaxclient based client, it will just ring ring ring infinitly :) |
11:23.01 | *** join/#asterisk albators (~albaneagr@81-178-45-7.dsl.pipex.com) |
11:23.26 | goldenear | That's a good idea indeed :) |
11:23.49 | goldenear | and what about a Hangup at priority n+101 ? |
11:23.53 | Zeeek | this must be a job for bkw_ |
11:24.14 | *** join/#asterisk zeitgeist_y2k (~ceicke@mail.inmedias.it) |
11:24.27 | goldenear | exten => 10,102,Hangup |
11:25.07 | goldenear | or exten => 10,102,Busy (but I guess this is not necessary as it is implicit, right ?) |
11:25.40 | Zeeek | ok, one last attempt at linux question: the files in /lib/modules/2.4.22/misc |
11:25.52 | Zeeek | are they autoloaded (like asterisk modules)? |
11:26.10 | *** part/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
11:27.01 | goldenear | ManxPower, so what do you think about Hangup or Busy at n+101 ? |
11:27.48 | ManxPower | goldenear: If you use Busy, you don't need hangup. |
11:29.05 | goldenear | so is exten => 10,2,Hangup a good idea ? or is it just not necessary ? |
11:29.43 | Morex | Does anybody hear use FastAGI? |
11:29.47 | Morex | here even? |
11:30.07 | *** join/#asterisk Twister (~jason@pool-151-205-68-214.char.east.verizon.net) |
11:30.18 | goldenear | ManxPower, I guess it's killing Busy indication, so it's should be avoided... |
11:35.35 | goldenear | Thank you ManxPower for all this information. Now my dial plan should be cleaner :) |
11:35.58 | Zeeek | ManxPower you run HEAD or STABLE? (i seemto remember STABLE) |
11:35.58 | Twister | YAY!!! *jumps for joy* |
11:37.47 | zeitgeist_y2k | can anyone help me with compilation problems of res_mysql.so in combination with the bristuff package? |
11:38.05 | ManxPower | goldenear: no, Hangup as the priority after Busy or Congestion will never be called, since Busy does not exit until the caller hangs up. |
11:38.13 | zeitgeist_y2k | sorry... it's res_config_mysql.so |
11:38.20 | ManxPower | Zeeek: STABLE only. |
11:38.45 | Zeeek | see my speech in #asterisk-stable |
11:39.57 | *** join/#asterisk pa (~Paolo@pa.user) |
11:48.38 | *** part/#asterisk pelo (~tcs@adsl203-149-051.mclink.it) |
11:53.47 | alphaque | goldenear: yes, because the Dial cmd has no timeouts |
11:54.30 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
11:56.30 | limbique | hi |
11:56.42 | limbique | does anyone knows what i can do with AGI? |
11:57.35 | limbique | And what is "zap show cadences"? |
11:58.42 | limbique | <PROTECTED> |
11:58.42 | limbique | Jul 12 13:58:34 WARNING[229390]: chan_zap.c:2022 pri_find_dchan: No D-channels available! Using Primary on channel anyway 3! |
11:58.42 | limbique | <PROTECTED> |
11:58.43 | *** part/#asterisk RestLessGemini (~root@202.142.182.154) |
11:58.45 | limbique | ? |
11:58.48 | *** join/#asterisk krazykrab (~Krab@203.81.238.65) |
11:59.01 | krazykrab | hi |
11:59.17 | krazykrab | any one can help me out with Asterisk AAH |
11:59.51 | zeitgeist_y2k | krazykrab, shoot... maybe i can help |
11:59.56 | krazykrab | thanks |
12:00.14 | krazykrab | i just installed the AAH . and have the broadvoice connection to test |
12:00.16 | ManxPower | limbique: Did you have a specific question? |
12:00.29 | limbique | hi |
12:00.31 | krazykrab | but when ever i call .. in my panel it shows incoming |
12:00.54 | limbique | um, yes i have many questions.. :) |
12:01.05 | krazykrab | i have followed the instruction but didnt help it |
12:01.05 | limbique | i'm a developer |
12:01.18 | zeitgeist_y2k | krazykrab, i have no idea about asterisk@home |
12:01.25 | krazykrab | oh ok |
12:01.29 | ManxPower | I can't really help with Flash Operator Panel, limbique |
12:02.30 | krazykrab | does any one knows any site addy for asterisk@home manual |
12:02.56 | ManxPower | limbique: Do NOT message me unless you have confidential information that cannot be put on the channel. |
12:03.09 | limbique | ok, sorry |
12:03.56 | krazykrab | ManxPower can you help me on Asterish@home |
12:04.39 | limbique | i have something like this. when ik call from external to a voipphone, when i disconnect the call, asterisk reserves an new zap channel and releases it immidiately.. |
12:04.44 | limbique | do you know why? |
12:05.16 | Zeeek | fucking wengo changed their server domain again without warning! </rant> |
12:06.20 | limbique | Is there a way to get information from a manager connection, to get all active links? (not the connections but who's connected to who) |
12:06.38 | ManxPower | krazykrab: no. |
12:06.51 | ManxPower | limbique: I cannot help you with developement questions |
12:07.13 | ManxPower | limbique: look at how Flash Operator Panel does things. |
12:07.16 | limbique | ManxPower: tnx.. what do you do? installing etc? |
12:07.35 | ManxPower | limbique: I do network and PBX management, installation, and service for my clients |
12:07.47 | limbique | i'll give a watch on Flash Operator Panel |
12:07.53 | limbique | is that a windows application? |
12:07.59 | ManxPower | No. |
12:09.52 | Zeeek | part of it is :) |
12:10.02 | limbique | a, perl |
12:10.18 | ManxPower | Zeeek: No, the GUI part is Flash. |
12:10.32 | Zeeek | which does run on WIndoze |
12:10.38 | ManxPower | The GUI works under Linux |
12:10.39 | Zeeek | FOP is a great tool |
12:10.57 | Zeeek | because it does run on anything Flash runs on |
12:12.42 | goldenear | Ok, thank you ManxPower :) |
12:13.03 | goldenear | alphaque, what do you mean ? |
12:16.22 | tzanger | well |
12:16.24 | tzanger | there goes the cat |
12:16.43 | tzanger | came home at about 6:45am, went out at 7:45am, cat was on the road between that time |
12:17.14 | Prion | Anyone having any issues with Voicepulse connect? |
12:17.34 | lobbin | back.. |
12:18.40 | limbique | ManxPower: ok, tnx.. i'm a noob with linux etc... |
12:19.28 | Zeeek | lobbin any progress? |
12:21.51 | alphaque | goldenear: sorry. meant that without a timeout, no call progress is being handled by Dial, so it continues forever |
12:22.39 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
12:22.52 | limbique | but is generates flash... thats cool |
12:24.43 | *** join/#asterisk FreezeS (~gido_b@83.103.170.130) |
12:24.50 | FreezeS | hi guys |
12:25.02 | FreezeS | do you know how can I route a call based on the destination number ? |
12:25.46 | FreezeS | I know I can route based on the calling number ( exten => number/s,1,Dial...) |
12:25.59 | FreezeS | but how can I route based on the number that is called ? |
12:26.01 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
12:27.45 | FreezeS | anyone ? |
12:30.10 | albators | yes |
12:30.29 | albators | extren => numberdialed,1, ..... |
12:33.54 | lobbin | Zeeek: yea, right now, I added "fromuser=" and then my username, after that it send that instead of asterisk and now it works |
12:34.02 | Zeeek | great! |
12:34.07 | lobbin | Yeah :) |
12:34.24 | lobbin | Caller ID seems fubar, but I guess that isn't Asterisk's fault? |
12:38.42 | *** join/#asterisk lodeon (~not4u@as1-6-4.ld.bonet.se) |
12:39.28 | goldenear | alphaque, of course I know it ... I just forgot the time out in my example ( exten => 10,1,Dial(IAX2/user,20) ) |
12:40.35 | goldenear | What I did not know is if 10,102,Busy was implicit or not ... |
12:41.16 | FreezeS | so, anyone can help me with my problem ? |
12:42.41 | ManxPower | FreezeS: exten => _31XX,1,Whatever will route all calles to any 4 digit extension beginning with "31" to the application Whatever |
12:42.42 | *** join/#asterisk razu (~razu@80-235-89-85-dsl.prn.estpak.ee) |
12:44.23 | goldenear | FreezeS, indeed it's just as simple as ManxPower says ... |
12:44.32 | tzanger | ManxPower: I heard that Whatever() app was kind of dangerous |
12:44.46 | tzanger | it turned the temperature on my freezer way up and made my ice cream go all melty |
12:45.25 | *** join/#asterisk toot (chris@212.20.250.186) |
12:45.31 | darwin35 | anyone having issues dialing interal sip extensions |
12:45.55 | goldenear | what kind of issues ? |
12:45.59 | darwin35 | my mapped exten work fine but when I dial sip/1001 or 1002 I get a 404 on my phone |
12:46.02 | ManxPower | tzafrir: It sometimes does that. The developers have not personally experienced this problem so they always close bug reports about it. |
12:46.15 | tzanger | ManxPower: :-) |
12:46.42 | tzanger | hee hee hee |
12:46.46 | tzanger | that word always makes me giggle |
12:46.50 | tzanger | toot |
12:46.52 | tzanger | hee hee hee |
12:46.55 | *** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net) |
12:47.19 | lobbin | Zeeek: never mind, I solved the caller id bit as well |
12:47.52 | Zeeek | $*%%*$$%%£££!!! |
12:48.27 | Zeeek | ^^^^^ read with Donald Duck voice |
12:48.33 | goldenear | darwin35, are you sure 1001 and 1002 are well defined in sip.conf ? |
12:48.41 | darwin35 | yes |
12:49.03 | goldenear | can you see them with sip show peers ? |
12:49.08 | ManxPower | A 404 indicates the phone is getting the request. |
12:49.27 | ManxPower | darwin35: Try Dial(SIP/1001@1001) |
12:49.40 | ManxPower | You should not need that, but it can't hurt to try. |
12:49.59 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
12:50.03 | darwin35 | 1001/1001 10.0.0.2 D 255.255.255.255 5060 Unmonitored |
12:50.10 | FreezeS | my problem is this: I have an ISDN line with multiple numbers. I want to allocate each number to a SIP phone. This way, there can only be 2 simultaneous calls, but each user can be called with a different nummber, from the exterior |
12:50.22 | darwin35 | 1002 is offline at the min |
12:50.40 | *** part/#asterisk alphaque (~alphaque@60.48.197.184) |
12:51.03 | *** join/#asterisk alphaque (~alphaque@60.48.197.184) |
12:51.28 | FreezeS | I've discovered in the documentation ${DIALEDPEERNUMBER} .. how can I route based on this variable ? |
12:52.01 | ManxPower | FreezeS: Incoming calls that haved the dialed number associated with them will match an extension line. |
12:52.28 | tzanger | FreezeS: Goto(somecontext,${DIALEDPEERNUMBER},1) ?? |
12:52.33 | ManxPower | For example, my provider sends me the entire number dialed when a call comes in. So I have exten => 5045551212,1,Goto(1212,1) |
12:52.50 | FreezeS | ManxPower: super, thanx |
12:52.54 | FreezeS | that's what I needed |
12:53.05 | *** part/#asterisk alphaque (~alphaque@60.48.197.184) |
12:53.08 | *** join/#asterisk alphaque (~alphaque@60.48.197.184) |
12:53.13 | *** part/#asterisk alphaque (~alphaque@60.48.197.184) |
12:53.38 | *** join/#asterisk alphaque (~alphaque@60.48.197.184) |
12:53.43 | goldenear | darwin35, I did the test, I only have 404 when the extension dialed doesn't exist ... |
12:53.44 | ManxPower | FreezeS: The telco may send 1 digit, 4 digits, or some other number of digits, depending how your phone company has the line set up. |
12:54.39 | darwin35 | hmm |
12:54.53 | darwin35 | all the mapped exten work fine |
12:55.20 | goldenear | so you sould not have a 404 :/ |
12:56.01 | goldenear | also are you in the right context ? |
12:57.44 | Katty | mew |
12:57.53 | darwin35 | re reading hold a min |
12:58.17 | Katty | is it morning? |
12:58.45 | kajtzu | sup |
12:58.50 | lobbin | Zeeek: thanks for the help |
12:59.12 | Zeeek | np |
13:00.50 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
13:01.09 | Katty | Zeeek: my new card came in today :> |
13:01.35 | Katty | oh. |
13:01.37 | Katty | no |
13:01.38 | Katty | i take it back |
13:01.45 | Katty | k, all better |
13:02.32 | *** join/#asterisk albators (~albaneagr@81-178-45-7.dsl.pipex.com) |
13:02.41 | Zeeek | new credit card? |
13:02.53 | Katty | no |
13:02.58 | Katty | the 4 port card for another 4 phone lines (= |
13:05.35 | ManxPower | It was for your old boss. |
13:06.30 | DarthClue | Yesterday was the last day at my other job. Problem being, the head man didn't show up so I couldn't tell him why and what my plans are. He is freaking out because they are about replace the entire phone system / dialer and I was heading up the project. I guess he doesn't realize that if ya don't pay me I won't stick around. |
13:06.43 | iCEBrkr | Ok, so I spent about 5hrs hacking up the EAGI example and res_agi last night and I still can't figure out how to 'listen' to the audio. |
13:06.44 | *** join/#asterisk toot (toot@212.20.250.187) |
13:07.09 | toot | sob - i can't reach the power for my new TDM400P |
13:07.14 | iCEBrkr | DarthClue: Yeah! Bend 'the man' over.. :D |
13:07.25 | Katty | Zeeek: hmm..when i put that card in |
13:07.31 | Katty | Zeeek: and define my channels |
13:07.39 | Katty | Zeeek: do i have two sets of 1-4 or does it just become 1-8? |
13:07.45 | DarthClue | Katty: 1-8 |
13:07.59 | Katty | DarthClue: and the second card? how does it know which card is number 2? |
13:08.31 | Katty | ports 7 and 8 i want in a different group |
13:08.47 | DarthClue | Katty: it depends on how they are detected on boot. Worst case scenario, you have to move the lines that are plugged into the cards or move the cards. |
13:08.52 | Katty | hopefully it won't think 7 and 8 are somehow card one, and not card two |
13:09.22 | Katty | DarthClue: k |
13:09.32 | Katty | DarthClue: anything special about putting that card in? |
13:09.43 | DarthClue | Katty: configure them and see what it thinks, if it thinks card 1 is 5-8 then just move the lines. |
13:09.45 | Katty | shouldn't have to load drivers twice, surely |
13:10.07 | DarthClue | Katty: should use the same drivers as the other card. |
13:10.11 | Katty | k |
13:10.14 | Katty | just...put the card in? |
13:10.51 | DarthClue | Katty: turn the box off first...and ya know, maybe a few other things to ensure that you don't fry it. |
13:10.57 | Katty | ;> |
13:10.59 | Katty | butbutbut |
13:11.07 | Katty | i wanna work with the computer /on/! |
13:11.30 | DarthClue | Katty: That procedure isn't supported and will void your warranty and terminate your support contract. |
13:11.48 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
13:11.49 | Hmmhesays | heh |
13:12.16 | ManxPower | If you plug a phone line into a Digium FXS port, the FXS module will be destroyed when the first call comes in |
13:12.44 | DarthClue | ManxPower: you serious? |
13:12.53 | kajtzu | ManxPower: never happened on a cisco fxo/fxs interface |
13:13.00 | Katty | DarthClue: i /am/ the support contract ;) |
13:13.04 | JerJer | DarthClue: as a heart attack |
13:13.07 | kajtzu | (and people plug them wrong all the time) |
13:13.18 | JerJer | kajtzu: then it never rang |
13:13.28 | ManxPower | DarthClue: Yes, I am serious |
13:13.33 | zoa | any bri experts here ? |
13:13.48 | Hmmhesays | morning Katty :) |
13:13.50 | ManxPower | The Cisco may be designed to not have that problem. |
13:13.55 | Katty | DarthClue: i'm going to put the card in over lunch (= |
13:13.59 | DarthClue | ManxPower: k. Seems like a design flaw. |
13:14.08 | JerJer | no i've seen cisco FXS ports smoke on ring of an FXO line |
13:14.11 | darwin35 | ok I did a asterisk -vvvvvvvvvgc and get no errors or warnings but still cant dial local sip exten |
13:14.14 | Katty | DarthClue: someone mentioned something about going into bios and making sure they were on different interrupts or something |
13:14.22 | darwin35 | this pisses me off |
13:14.37 | Katty | DarthClue: should i disregard that? |
13:14.50 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
13:15.09 | Katty | hi ariel_ (= |
13:15.12 | darwin35 | I would first move the cards around before messing with the bios |
13:15.16 | yxa | darwin35 sometimes you need to reboot :) |
13:15.25 | darwin35 | I have 100 times |
13:15.26 | ariel_ | Katty, morning hope your dong well today. |
13:15.28 | DarthClue | Katty: yes, irq settings may cause a conflict. as long as the devices are on different irqs it shouldn't be an issue. what is the motherboard on the system? maybe the mobo docs will say which slots share irqs. |
13:15.31 | Katty | ariel_: yes thanks (= |
13:15.35 | goldenear | darwin35, what does asterisk CLI tell you when you call a local sip phone ? |
13:16.10 | goldenear | what the log says ? |
13:16.15 | Katty | DarthClue: k, i'll go look a little later |
13:16.42 | *** join/#asterisk Mike9 (~sturdee@ireland.pathwaynet.com) |
13:17.28 | iCEBrkr | darwin35: Dude, haven't you gone to bed yet? |
13:18.01 | DarthClue | iCEBrkr: it's 815am, why would anyone want to be in bed now? |
13:18.52 | iCEBrkr | DarthClue: I just recall him being fairly active in channel last night. |
13:19.00 | iCEBrkr | Just seemed like he hasn't stopped. :D |
13:19.29 | DarthClue | iCEBrkr: yeah well, some of just don't sleep normal. and that includes me. |
13:19.40 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
13:19.47 | iCEBrkr | Same here, but I had to call it a night early last night. 12:30 and I was passed out. |
13:19.48 | darwin35 | http://pastebin.ca/17536 there is a sip debug |
13:20.21 | Katty | yawn. |
13:20.22 | RoyK | anyone here that knows how i can change the audio file for a particular call to Authenticate? |
13:20.40 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
13:20.55 | DarthClue | RoyK: audio file Authentication? |
13:21.15 | RoyK | DarthClue: huh? |
13:21.15 | iCEBrkr | DarthClue: Sun and Mon were 3am nights for me. I couldn't walk away from the keybaord. |
13:21.41 | iCEBrkr | RoyK: Yea, what do you mean by audio file authentication?! |
13:21.53 | DarthClue | iCEBrkr: Last week was hell. A couple of days in row where I was up for 24 hours straight, crashed for 6, rinse and repeat. |
13:22.07 | iCEBrkr | DarthClue: ouch |
13:22.08 | RoyK | i mean i call Authenticate in asterisk and that plays off some file |
13:22.16 | RoyK | i want it to play my own voice in my own language |
13:22.27 | goldenear | darwin35, and could you show the general asterisk log, not the sip debug ? |
13:22.28 | RoyK | one per purpose |
13:22.42 | darwin35 | the log is empty |
13:22.46 | Katty | iCEBrkr: thx, all better |
13:22.53 | iCEBrkr | Katty: :D |
13:22.54 | darwin35 | no errors or warnings |
13:23.17 | Katty | will ssh tunneling work for softphones? |
13:23.40 | goldenear | darwin35, but does it execute the correct commands ? |
13:23.46 | Katty | connect to an ssh through a firewall while specificying localport:destination:anotherport |
13:23.48 | iCEBrkr | DarthClue: I've been hacking away at this EAGI audio crap for the past 3 days. I still don't understand what it's (not) doing. I'd like to eventually get sphinx integrated. I have sphinx working without a hitch. |
13:23.56 | DarthClue | RoyK, how about using Read / Playback instead? |
13:24.01 | Katty | and then connect to localhost on port like 1234 |
13:24.09 | Katty | and have if forward to destination:whateverport |
13:24.19 | iCEBrkr | Katty: I've heard it's been done, but I'm not so sure it's 'efficient'. |
13:24.20 | darwin35 | wich command ? |
13:24.34 | Katty | iCEBrkr: why? |
13:24.40 | darwin35 | its registering all the extensions and every thing else works |
13:24.47 | zeitgeist_y2k | Katty, you need more than one TCP channel for SIP connections |
13:24.51 | darwin35 | inbound from outdide works |
13:24.54 | zeitgeist_y2k | some UDP too |
13:24.56 | Katty | zeitgeist_y2k: yes, but not iax (= |
13:25.00 | iCEBrkr | Katty: I didn't the follow the conversation that closely. :) |
13:25.04 | darwin35 | all the mapped exten work |
13:25.07 | Katty | zeitgeist_y2k: iax only uses one port :) |
13:25.11 | JerJer | asterisk doesn't deal with TCP based SIP |
13:25.14 | zeitgeist_y2k | Katty, I should read before writing :-) |
13:25.25 | Katty | yeah, it's udp |
13:25.31 | JerJer | you need a GRE tunnel |
13:25.42 | Katty | is that something like a green tunnel? |
13:25.45 | Katty | or a subway tunnel? |
13:25.57 | iCEBrkr | Katty: There was someone in here all concerned about someone 'listening in' on their VoIP calls. I kinda laughed. |
13:25.57 | JerJer | or other udp encapsulated tunneling method |
13:26.02 | DarthClue | Katty: VPN |
13:26.06 | Katty | DarthClue: oh |
13:26.12 | JerJer | VPN is a generic term |
13:26.20 | Katty | JerJer: yes, but i understand vpn |
13:26.34 | DarthClue | jerjer: it may be generic, but it is understood by many. |
13:26.36 | JerJer | but you cannot use just any VPN and expect VoIP to work |
13:26.37 | darwin35 | this makes no sense |
13:26.44 | Katty | JerJer: hahahahhahahaha |
13:26.49 | Katty | JerJer: HAHAHAHA, kthx it works |
13:26.54 | Katty | JerJer: on DIALUP even |
13:27.05 | goldenear | darwin35, what is the context for inbound calls ? |
13:27.13 | zoa | does somebody know if a bri card can be used to connect asterisk to a pbx ? |
13:27.20 | DarthClue | ok...PPTPD or IPSEC VPN which uses GRE, etc, etc, etc |
13:27.26 | zoa | can asterisk play the net side on a bri ? |
13:27.34 | kajtzu | you dont need gre with ipsec |
13:27.41 | Katty | the Dark side of a bri |
13:27.57 | kajtzu | in fact, gre is completely unnecessary unless you intend to run a routing protocol on top of ipsec :) |
13:28.06 | RoyK | is the asterisk db persistent between restarts? |
13:28.22 | DarthClue | kajtzu: yeah, i know, but it's too early to sift thru the mess in my head. |
13:28.36 | goldenear | darwin35, the asterisk log (with -vvvvvv) should show you why you have a 404 |
13:28.42 | Katty | DarthClue: it needs a shower first |
13:28.46 | JerJer | you don't want IPSEC - adds too much latency |
13:28.52 | *** part/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com) |
13:28.55 | darwin35 | http://pastebin.ca/17538 there is the sip 101 entry |
13:28.55 | iCEBrkr | Oh? Sweet. NuFone offers 800 DID's? |
13:28.57 | zoa | ipsec could do udp |
13:29.01 | zoa | so that doesnt have to be a problem |
13:29.23 | DarthClue | Katty: true, but I don't have to leave the house till about noon so I'm in no hurry. |
13:29.41 | kajtzu | JerJer: uh? |
13:29.51 | Katty | DarthClue: k |
13:30.00 | Katty | my hair gets cut today :< |
13:30.02 | Katty | i'm skeered bad |
13:30.07 | iCEBrkr | DarthClue: must be nice. |
13:30.20 | *** join/#asterisk wiseguy_ (chivilis@85.206.10.242) |
13:30.21 | toot | can i do anything with the TDM400P without the power supply hooked in? |
13:30.24 | DarthClue | iCEBrkr: it is. one of the perks of working for the dark side. |
13:30.24 | wiseguy_ | hellow people |
13:30.31 | toot | its a public holiday over here and i won't be able to get one for a few days |
13:30.33 | Katty | noooooooooo |
13:30.37 | iCEBrkr | DarthClue: hehe |
13:30.38 | wiseguy_ | anyone have done a calling center with asterisk? |
13:30.40 | JerJer | everytime i've gone into a network that used an IPSEC tunnel and tried to use VoIP the latency was very noticeable |
13:30.57 | JerJer | toot: no |
13:31.02 | kajtzu | ipsec itself doesn't add much to latency |
13:31.04 | JerJer | you must connect it to power |
13:31.11 | DarthClue | wiseguy_: yes, why? |
13:31.13 | toot | sob |
13:31.14 | darwin35 | Jul 12 08:10:07 WARNING[54145] res_indications.c: Invalid ringcadance given '' at line 62. |
13:31.14 | darwin35 | Jul 12 08:10:08 NOTICE[54145] res_indications.c: Duplicate entry 'stutter', skipped. |
13:31.15 | iCEBrkr | I need to get a haircut myself. I'm starting to get that 70's bed-head shag.. |
13:31.17 | JerJer | kajtzu: 30-40 ms |
13:31.17 | kajtzu | we're talking sub millisecond range |
13:31.20 | kajtzu | JerJer: nooo |
13:31.20 | darwin35 | thasts new |
13:31.25 | wiseguy_ | DarthClue: maybe pm |
13:31.44 | darwin35 | thats the only error I have |
13:31.57 | JerJer | kajtzu: sub ms - only if the network has hardware assisted ipsec |
13:32.03 | goldenear | darwin35, and what's the log says when you call the sip phone ? |
13:32.14 | goldenear | (errors or not) |
13:32.29 | kajtzu | JerJer: of course it would have. what's the point of designing a ipsec network and then utilizing underpowered cpus? :) |
13:32.53 | JerJer | small businesses are not always going to invest in the latest and greatest technology |
13:33.00 | JerJer | or home users |
13:33.03 | clive- | zoa yes klaus's cards can do NT mode |
13:33.08 | kajtzu | HiFn chips came on the market in 1998 |
13:33.17 | zoa | so NT mode is what i need ? |
13:33.22 | zoa | i dont know a thing about NT mode |
13:33.25 | kajtzu | and the cost of the add on module or chip isnt much |
13:33.26 | zoa | or BRI in general |
13:33.26 | clive- | zoa yup |
13:33.33 | zoa | k do you have any docs on this ? |
13:33.37 | zoa | or links ? |
13:33.40 | darwin35 | the log says nothing |
13:33.44 | JerJer | i use a very trivial GRE tunnel and VoIP works flawlessly |
13:33.52 | *** join/#asterisk zpn (~xpn@207.111.174.1) |
13:33.55 | darwin35 | and I pastebin the sipdebug |
13:33.56 | toot | anyone know where i can buy a Molex Power extension cable in the uk? i'm struggling to find one |
13:34.10 | toot | i imagine other people have had the same issues |
13:34.12 | JerJer | Radio Shack UK |
13:34.13 | JerJer | :) |
13:34.16 | iCEBrkr | JerJer: hehe |
13:34.19 | kajtzu | JerJer: it doesn't protect the contents either. wasnt this thread about protecting |
13:34.22 | iCEBrkr | Rad-Shack |
13:34.27 | toot | ah - cheers |
13:34.33 | JerJer | kajtzu: who cares? |
13:34.39 | JerJer | encrypt at the application layer then |
13:35.39 | kajtzu | yeah, encrypted signalling and rtp anyone? ;-) |
13:35.42 | DarthClue | Just make sure all your VOIP calls are in a language that noone else can understand, like Navaho. |
13:36.35 | goldenear | darwin35, that's the problem so ... the log should show you what happens during the call ... |
13:36.40 | *** join/#asterisk coppice (~chatzilla@244.203.17.210.dyn.pacific.net.hk) |
13:36.48 | JerJer | encrypt=yes |
13:36.50 | JerJer | problem solved |
13:37.05 | ManxPower | BTW, Asterisk Business Edition is based on CVS-HEAD and ONLY gets bug fixes. It might be the solution for people that need some of the features of CVS-HEAD, but don't want behavoral changes. |
13:37.28 | DarthClue | bureaucracy=US, problem solved. |
13:37.40 | goldenear | darwin35, your phone may not use asterisk as the default proxy ... |
13:37.49 | ManxPower | Of course, Asterisk Business Edition does not have Realtime documented. Apparentlys TPTB didn't think it was stable enough. |
13:38.05 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
13:38.10 | MikeJ[Laptop] | Calling all asterisk head users... the bug tracker needs you... go adopt a bug. |
13:38.13 | JerJer | one should only ever use realtime if one has a very good reason to use it |
13:38.25 | darwin35 | ok fixed |
13:38.26 | JerJer | a 'business' does not need realtime |
13:38.35 | MikeJ[Laptop] | my house does ;) |
13:38.37 | MikeJ[Laptop] | hehe |
13:38.43 | ManxPower | JerJer: *nod* That's why ABE would work well for us. |
13:38.44 | darwin35 | even thou the phone is not behind nat I had to set nat=yes |
13:38.50 | MikeJ[Laptop] | there's a smart way to watch tv |
13:38.55 | darwin35 | but that did not work earlier |
13:39.01 | fenlander | Can we expect "Asterisk Service Provider Edition" soon? |
13:39.02 | darwin35 | now it just did |
13:39.05 | DarthClue | MikeJ[Laptop]: I don't want to adopt any bugs, that requires time that I don't have. |
13:39.06 | ManxPower | And apparently it actually comes with a real manual. |
13:39.12 | ManxPower | fenlander: that would be called "CVS-HEAD" |
13:39.28 | goldenear | darwin35, ah sip mysteries ... |
13:39.46 | darwin35 | yeah |
13:39.48 | ManxPower | darwin35: All the COOL service providers are using it! |
13:40.06 | darwin35 | what nat=yes |
13:40.19 | darwin35 | or the manual |
13:41.18 | darwin35 | well its all working again |
13:41.24 | [TK]D-Fender | Neat info I just got on * : A company called ScopServ (on the Wiki) is selling a turn-key * solution (all hardware & software setup up and supported). Their GUI is actually DAMN solid looking and in gearing up for MASS production and distribution. They are aligning themselves with some major OEM people and they seem to be taking serious aim at the market, |
13:41.27 | ManxPower | darwin35: and what does "sip show peers" show. |
13:41.37 | darwin35 | its all working now |
13:41.53 | darwin35 | but earlier when I had nat=yes it was not going threw |
13:42.06 | darwin35 | then I set nat=no and still did not work |
13:42.20 | darwin35 | then just now set nat=yes and it all works |
13:42.33 | bkw_ | what |
13:42.44 | darwin35 | nat is a thorn is the butt of asterisk |
13:42.50 | [TK]D-Fender | Website : http://www.scopserv.com/ and an on-line demo of the GUI - http://demo.scopserv.com/ |
13:43.20 | Nivex | darwin35: maybe only in the butt of SIP |
13:43.28 | Nivex | IAX and NAT do quite well |
13:43.28 | darwin35 | 1001/1001 10.0.0.2 D N 255.255.255.255 5060 Unmonitored |
13:43.30 | *** join/#asterisk rjreb (~rjreb@greatwall.amer.net) |
13:43.42 | ManxPower | DarthClue: I have almost no SIP problems with my NAT SIP clients. |
13:43.54 | ManxPower | Nivex: IAX and NAT has some issues too. |
13:44.33 | bkw_ | iax has no nat issues.. unless you're using PAT which isn't really NAT |
13:44.37 | ManxPower | Two IAX2 phones behind different NAT routers, but talking to the same Asterisk server can't use IAX2 transfers. |
13:44.39 | bkw_ | IAX has sound quality issues |
13:44.39 | JamesDotCom | [TK]D-Fender: WOW, now you're gonna tell us you have no affilliation with the company except a satisfied customer, right?!?! |
13:45.03 | Nivex | why would IAX have sound quality issues? that's dependant on the codec |
13:45.10 | bkw_ | no its not |
13:45.14 | bkw_ | IAX is pure ass |
13:45.16 | coppice | Nivex: wrong |
13:45.16 | ManxPower | [TK]D-Fender: I'll wait for user reports of using it. |
13:45.26 | bkw_ | its got issues |
13:45.38 | ManxPower | Apparently bkw_ mind control programming has worn off and he's seeing the light. |
13:45.38 | Nivex | I'm being goaded here... I can feel it. |
13:45.41 | darwin35 | bkw how much to get your help with the call back busy |
13:45.46 | DarthClue | Nivex: IAX has issues. I can confirm it. |
13:45.48 | bkw_ | ManxPower, no I have always known this |
13:45.52 | bkw_ | asterisk in general Sucks |
13:45.54 | bkw_ | majorly |
13:46.01 | bkw_ | it has a few fatal flaws |
13:46.06 | ManxPower | bkw_: The FORMER Asterisk cheerleader. |
13:46.07 | bkw_ | but nobody will listen to us |
13:46.08 | coppice | Nivex: no. you're just wrong |
13:46.16 | ManxPower | bkw_: so fork the code and fix it. |
13:46.18 | Nivex | bkw_: I hear there's a nice Windows offering you can use :) |
13:46.38 | bkw_ | forking asterisk to make a better PBX is like starting with a gun with the intention of making a small childs toy. |
13:46.39 | DarthClue | Welcome to the dark side. Once you go dark, you never see the light again. |
13:46.43 | ManxPower | bkw_: and you KNOW the only way you'll get your issues fixed is by forking Asterisk. |
13:46.44 | Nivex | coppice: I was referring to bkw_'s comment. If you would like to inform me in what manner I am wrong I'd be glad to listen. |
13:46.59 | bkw_ | ManxPower, not really.. just write something else |
13:47.00 | ManxPower | bkw_: So you are moving over to YATE then? |
13:47.03 | bkw_ | no |
13:47.13 | ManxPower | bkw_: best of luck with that. |
13:47.20 | [TK]D-Fender | JamesDotCom : No affiliation. I am trying to get * into my company and the bosses here don't like the "programmer only" maintenance needs * demands. ScopServ is just a way of validating getting * in the door and kicking out Nortel's BCM solution we are being proposed. |
13:47.39 | coppice | Nivex: codec is a biggy in sound quality. several things affect the smoothness of the stream and they have just as big an effect on sound quality |
13:47.51 | bkw_ | what coppice said |
13:48.05 | DarthClue | [TK]D-Fender: tell your bosses that you can get someone to support it no questions asked for less than the cost of your nortel system. |
13:48.24 | puzzled | bkw_: mind elaborating a bit on what you think is wrong with *? |
13:48.29 | [TK]D-Fender | JamesDotCom : Personally I'd like the just white-box it myself, but they don't like the word "home-grown" even if I DID find 1/2 dozen consultants to set it up. These people are friggen GUI centric where everything has to be configurable bya MORON. |
13:48.55 | bkw_ | puzzled, every call uses the same port.. same queue.. as the box gets loaded with traffic.. the more and more calls you have the worse it gets. |
13:49.00 | DarthClue | [TK]D-Fender: i can make that happen for less than your nortel system. |
13:49.01 | [TK]D-Fender | DarthClue : Didn't work, they want turn-key...... ScopServ made the best offering to get it "in the door" |
13:49.21 | ManxPower | [TK]D-Fender: Let us know how well it works. |
13:49.22 | [TK]D-Fender | DarthClue : So can I! The point is they want something that makes them feel "warm & fuzzy" :( |
13:49.40 | bkw_ | puzzled, can you see what i'm talking about? |
13:49.45 | DarthClue | [TK]D-Fender: pm me the specs that you need and what I need to compete with and I'll see if i can do better. |
13:49.47 | coppice | they want to resemble pubic hair? :-\ |
13:49.59 | ManxPower | We're in a SAVE MONEY mode. Apparently the company has not made any money so far this year. |
13:50.01 | puzzled | bkw_: yup (as far as my understaning of C goes) |
13:50.17 | [TK]D-Fender | Franly my test server is VERY nice right now and it'd take about 10 hours to finalize it & SpanDSP I'm sure, but they want "idiot-proof" and "firiendly" and "Supported". They are too used to "stock" systems |
13:50.26 | bkw_ | now if IAX could punch holes in NAT and have a monitor thread per call.. now that might change things |
13:50.37 | bkw_ | oh and a double legged IAX connection is pure evil |
13:50.52 | *** join/#asterisk Ahrimanes (~aron@hobbes.bsd-dk.dk) |
13:50.56 | [TK]D-Fender | DarthClue : Anyone can offer me * at a better price including my doing it ALL from scratch (which I have actually just done (mostly). But Cost isn't their point :/ |
13:50.59 | sivana | so is IAX better than SIP overall? |
13:51.10 | bkw_ | Zap => IAX => BOX => IAX => ZAP <--EVIL |
13:51.11 | ManxPower | bkw_: I already talked to someone about better IAX2 NAT traversal. |
13:51.17 | puzzled | bkw_: a little patch here, a little patch there, rinse and repeat :) |
13:51.20 | bkw_ | sivana, no |
13:51.21 | DarthClue | [TK]D-Fender: i can give them turnkey, i just need to know what I have to work with on your end. |
13:51.24 | coppice | overall the PSTN is best "-) |
13:51.28 | [TK]D-Fender | Its "user maintainable" and "friendly" that counts. |
13:51.35 | ManxPower | bkw_: You've been spending too much time with timecop |
13:51.44 | bkw_ | SIP will whip IAX in quality |
13:51.47 | DarthClue | [TK]D-Fender: i can do that. |
13:51.54 | [TK]D-Fender | DarthClue : You'd also HAVE to have a LOCAL physical presence. AKA Montreal, QC. |
13:52.00 | bkw_ | ManxPower, nope just telling it like it is |
13:52.04 | sivana | hrm.. and here I thought IAX was better |
13:52.13 | [TK]D-Fender | Thats the curse * has with corporate approval. |
13:52.22 | bkw_ | sivana, you must be drinking out of the IAX punch bowl |
13:52.26 | sivana | ya :) |
13:52.27 | DarthClue | [TK]D-Fender: is ScopServ local? |
13:52.44 | puzzled | bkw_: is there any agreement amongst the developers that these are indeed issues that need to be addressed? |
13:53.04 | bkw_ | puzzled, yes but asterisk development is a one way street as of late |
13:53.12 | zoa | mark seems to agree with changes needed for iax2 |
13:53.14 | Nivex | and here I've been espousing the virtues of IAX to my friends of late. |
13:53.18 | [TK]D-Fender | DarthClue : Yup, Offices in Montreal and 2 surrounding cities |
13:53.22 | zoa | i had a talk with him on astricon about it |
13:53.26 | bkw_ | zoa knows of the flaws in IAX also |
13:53.39 | zoa | i think we need an iax3 |
13:53.42 | bkw_ | yes |
13:53.47 | iCEBrkr | Dive! Dive! Dive! |
13:53.49 | Ahrimanes | ah i was there too :) |
13:53.49 | puzzled | bkw_: ok, well, realizing it is a start. babysteps I guess |
13:53.50 | ManxPower | Maybe if mark did more coding and less traveling.... |
13:53.54 | zoa | oh brian, i have a patch for sip nat for sale |
13:53.57 | zoa | not my patch though |
13:53.58 | DarthClue | [TK]D-Fender: too bad, i bet i could easily compete and give you and your bosses a system that would make all of you happy. |
13:54.02 | Nugget | A lot of IAX advocacy is just a groupthink feedback loop. |
13:54.24 | iCEBrkr | ManxPower: Naa, bkw just needs to code faster. |
13:54.25 | Nugget | And a lot of the anti-SIP sentiment is just misplaced anger at NAT. |
13:54.25 | zoa | well i like iax2, but you need to know when to use it and how to use it |
13:54.25 | darwin35 | bkw you have not answerd me |
13:54.30 | Twister | when i dial *77 and make a recording it will play through playback when when i do echo "text" >> text2wave hello.wav it wont play, what options am i missing for text2wave? |
13:54.52 | puzzled | Nugget: my anger is totally reserved for NAT and stupid alcatel adsl modems :) |
13:54.54 | ManxPower | Yay! My callerid-fixup.agi is working |
13:55.06 | darwin35 | cool |
13:55.08 | iCEBrkr | F NAT. I wish NAT was never invented. :D |
13:55.42 | coppice | Nay wasn't invented |
13:55.43 | [TK]D-Fender | DarthClue : I quoted a system with bottom-dollar hardware costs on all components (best of breed purchasing). I then added on $1500 for 10hrs@150$/hr of support (IF NEEDED) just to set up the support relationship. I was proposing to set it all up myself. Hard to beat that.... but that wasn't what they wanted. They want someone else to set it up make it idiot-proof, and support it physically. |
13:55.49 | coppice | NAT wasn't invented |
13:55.59 | coppice | It sort of festered into life |
13:55.59 | Ahrimanes | it was summoned from below.. |
13:56.08 | bkw_ | haha |
13:56.16 | iCEBrkr | coppice: Yea, I know.. :/ |
13:56.28 | bkw_ | nat spells 628 so its on the same block as 666 |
13:56.35 | ManxPower | I just don't have many problems with NAT |
13:56.36 | iCEBrkr | haha |
13:56.56 | DarthClue | [TK]D-Fender: luckily my old boss is a cheap bastard. we are replacing our entire pbx and dialer for $20,000 and that gives us ton's of growing room. |
13:57.13 | iCEBrkr | ManxPower: With a properly setup router sure. But it's just a headache. My setup has been playing nicely with NAT so far. Kinda scary. |
13:57.13 | [TK]D-Fender | You couldn't beat MY price, but you might beat ScopServ's. Theirs is +/- $12,000 for a 4 port T1 server. (Rack mount SCSI RAID, fully congured and GUI'd up) Frankly for the actual increase from white-boxing I consider it an acceptable trade-off since I will have root access to it. |
13:57.17 | bkw_ | DarthClue, you didn't quite that job yet? |
13:57.36 | [TK]D-Fender | DarthClue : Lucky for you your boss takes your word for it. I know I'm right, but they don't care that much. |
13:57.44 | darwin35 | I just wish asterisk would drop mpg123 and just move to madplayer it works great |
13:57.52 | darwin35 | no issues |
13:57.57 | ManxPower | iCEBrkr: it took me 1 hour to configure my SIPura, my Asterisk, and my Cisco router to handle NAT. The phone can move between the local lan and the general internet transparently. |
13:57.57 | bkw_ | darwin35, please tell us why? |
13:58.02 | iCEBrkr | darwin35: hack the source. |
13:58.11 | bkw_ | ya really |
13:58.20 | DarthClue | bkw_: i found out this morning he is in panic mode. I am going in this afternoon to tell him i am part time only starting yesterday. |
13:58.55 | iCEBrkr | ManxPower: I'm not a network guy, so that kinda stuff is a bit confusing at first. But like I said, all my stuff just seems to work and I was rather surprised. :D |
13:58.59 | DarthClue | bkw_: he never showed up yesterday before i left. |
13:59.15 | bkw_ | sounds like you need to call your big boss to get the boss fired |
13:59.17 | ManxPower | Of course if you have a cheap-ass old Linksys router that randomly reboots when it has to NAT SIP packets, then that's not a NAT problem. |
13:59.20 | toot | i used to have a x100p clone - it stopped working - i wondered might it be anything to do with the fact my phone rings once after i hang up a call? i have now bought a tdm400p and don't want it to go the same way - any thoughts? |
13:59.28 | iCEBrkr | ManxPower: hehe |
13:59.38 | iCEBrkr | Randomly reboots haha |
14:00.28 | ManxPower | iCEBrkr: each of the 3 older (hardware rev a) Linksys routers start rebooting when I put a SIP phone behind them. Running the latest firmware available for that hardware. |
14:00.42 | DarthClue | bkw_: neither my manager, nor the big boss showed up yesterday. |
14:00.44 | ManxPower | I've almost finished switching the 2 locations to m0n0wall |
14:00.44 | Katty | hi |
14:00.47 | Katty | i have query |
14:00.48 | Katty | what does this mean: |
14:00.52 | Katty | Jul 12 08:54:08 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Critical Request) |
14:00.52 | ManxPower | ..er.. 3 locations |
14:00.55 | Katty | Jul 12 08:54:12 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Non-critical Request) |
14:00.57 | iCEBrkr | ManxPower: that's fizzucked. |
14:01.06 | ManxPower | Katty: it means asterisk can nolonger see the phone. |
14:01.20 | ManxPower | Either the phone is off, the NAT translation timed out, or the phone changed IP addresses. |
14:01.23 | ManxPower | iCEBrkr: Yup! |
14:01.30 | Katty | must be NAT timed out |
14:01.36 | bkw_ | ManxPower, those messages happen even on the local lan |
14:01.39 | Katty | cause the phone is still on and they're set to manual ip |
14:01.42 | ManxPower | Katty: qualify-yes will fix that. |
14:01.52 | ManxPower | qualify=yes that is |
14:01.53 | Katty | ManxPower: in sip.conf? |
14:02.00 | ManxPower | Katty: yes, in the entry for the phone |
14:02.18 | [TK]D-Fender | DarthClue : My 6187$ home made boring server woul be replaced by a SCSI redundant supported server costing $12000. My original solution TOTAL cost was $19,000, and Nortel's bid was $45,000. If I go up $6000 and still get to KEEP * Halla-fucking-lugiah! *THEN* I will twist it to my designs :D |
14:03.11 | [TK]D-Fender | For me ScopServ is a means to an end. I'd prefer to bank the $6,000 mind you, but if it gets me what I want, I'll do it. |
14:04.00 | DarthClue | [TK]D-Fender: is that canadian? |
14:04.24 | [TK]D-Fender | But I do have to admit I'm DAMN impressed by their GUI especially because it does Queue statistic reports as well which is only really avaliable by the XC-AST add-on for almost $1000 CDN. Add that to my $6187, and $12,000 looks better again |
14:04.35 | [TK]D-Fender | DarthClue : Yes, All CDN |
14:04.44 | *** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
14:05.01 | zoa | i have queue statistics reports too |
14:05.17 | [TK]D-Fender | zoa : Using what? |
14:05.28 | DarthClue | [TK]D-Fender: take it, i doubt i could beat that. |
14:05.37 | [TK]D-Fender | zoa : And I'm something web-based and pretty. |
14:06.03 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
14:06.03 | bkw_ | you do realize that chan_agent and app_queue are CRAP |
14:06.11 | bkw_ | aka chan_deadlock |
14:06.16 | bkw_ | and app_segfault |
14:06.19 | [TK]D-Fender | DarthClue : ScopServ's advantage is they are heading to mass-distribution to be sold as commoditiy like the Nortel BCM is without the added licensing costs. |
14:06.38 | maik | who killed the cvs server? |
14:07.00 | [TK]D-Fender | bkw_ : So how long before ICD becomes better documented and merged in? |
14:07.14 | DarthClue | [TK]D-Fender: I am working with someone else to head in that direction, but ours aren't quite to that level yet where we can compete with another VOIP system. |
14:07.29 | ManxPower | [TK]D-Fender: I highly doubt that ICD will EVER be merged into Asterisk. |
14:07.30 | [TK]D-Fender | bkw_ : It didn't leave a good taste in my mouth when I first read up on it. Too cryptic for me at the time |
14:08.09 | [TK]D-Fender | DarthClue : They are already targeting the BCM and have channel parters I know well already lined up. This could be BIG for * in the long run. |
14:08.40 | bkw_ | ICD has gotten out of hand |
14:08.54 | [TK]D-Fender | bkw_ : So what does that leave me? |
14:09.09 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
14:09.41 | [TK]D-Fender | ..besides screwed... |
14:09.52 | bkw_ | nowhere |
14:09.54 | *** join/#asterisk mcreedjr (~anon@oh-65-41-206-34.sta.sprint-hsd.net) |
14:10.50 | [TK]D-Fender | bkw_ : And there you have it. So I guess I'll just have to sit back and pray that *'s call center woes get resolved by SOMEBODY in some way... |
14:10.59 | mcreedjr | Can anyone point me to a good explaination of the difference between FXO/FXS? I am wondering if I need one FXO port and one FXS port to be able to send and receive calls from the PSTN. |
14:11.02 | *** join/#asterisk wiseguy_ (chivilis@85.206.10.242) |
14:11.15 | wiseguy_ | i can't login as queue agent with grandstream phone |
14:11.30 | wiseguy_ | how do i need to send dtmf, as sip info maybe? |
14:11.30 | ManxPower | ~fxofxs |
14:11.30 | jbot | from memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
14:11.53 | wiseguy_ | any one using queues? |
14:11.56 | DarthClue | wiseguy_: yes, info. |
14:12.33 | mcreedjr | Jbot: So I need to use FXO ports to access my outside PSTN lines, and then I need FXS ports as extensions to hook my phones to then? |
14:12.48 | [TK]D-Fender | DarthClue : ScopServ is coming on-site with a server & 2 phones for a demo and proposal. |
14:13.00 | wiseguy_ | okey, i will try |
14:13.16 | mosty | mcreedjr, by jove you've got it :) |
14:13.18 | DarthClue | [TK]D-Fender: does that cost include phones or just the box? |
14:13.32 | mcreedjr | Mosty: Thanks for the 'atta-boy :) |
14:14.21 | *** join/#asterisk brookshire (~pfffft@207.111.174.1) |
14:14.22 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
14:14.27 | DarthClue | MikeJ[Laptop]! |
14:14.28 | mcreedjr | So an FXO will answer inbound lines and also allow me to call out on the PSTN then? |
14:14.37 | [TK]D-Fender | DarthClue : just the box & support. I'm looking to get all Polycom IP 600 phones. SCopserv already auto-provisions Cisco & Aastra phones and they're looking to add Polycom to that list |
14:14.42 | MikeJ[Laptop] | hello |
14:14.45 | mosty | mcreedjr, you only need fxs ports if you want to use regular (non voip) phones |
14:15.10 | MikeJ[Laptop] | Calling all asterisk head users, today is adopt a mantis bug day. Adopt a bug and fix it! |
14:15.23 | [TK]D-Fender | DarthClue : For the rest of my solution I can bottom-dollar source it (like my Rhino failover / analog solution) |
14:15.27 | MikeJ[Laptop] | it's your time to give back a little bit. |
14:15.47 | DarthClue | [TK]D-Fender: how soon do you need an answer? and is having someone local a requirement? I might be able to beat that price if it is the box and support (how many hours). |
14:15.48 | brookshire | mikej: you're awesome ;) |
14:15.57 | mcreedjr | Mosty: Okay, great, makes things easier. Thank you for your help! |
14:16.04 | mcreedjr | Jbot: Thanks for the help. |
14:16.04 | jbot | gern geschehen, mcreedjr |
14:16.52 | DarthClue | jbot: are you dead? |
14:17.27 | brookshire | jbot: purr |
14:17.27 | jbot | ACTION *purrs purrs purrs* |
14:17.30 | brookshire | :D |
14:17.36 | DarthClue | jbot: explode |
14:17.36 | jbot | BOOOM |
14:17.40 | DarthClue | jbot: die |
14:17.40 | jbot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
14:17.42 | ManxPower | jbot doesn't seem to do google searches anymore |
14:17.52 | brookshire | :( |
14:17.52 | DarthClue | jbot: google digium |
14:18.00 | brookshire | :((((((((((((((((( |
14:18.13 | DarthClue | jbot: fsck brookshire |
14:18.14 | jbot | e2fsck /dev/brookshire : warning! filesystem contains idiots! |
14:18.24 | DarthClue | jbot: fsck jbot |
14:18.24 | jbot | e2fsck /dev/jbot : warning! filesystem contains dickheads! |
14:18.43 | cpatry | ~google cluecon |
14:18.50 | DarthClue | ~cluecon |
14:18.50 | jbot | i heard cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
14:19.00 | MikeJ[Laptop] | brookshire, no.. I am naggy... |
14:19.04 | [TK]D-Fender | DarthClue : Local = MUST. Instant failure for you. And you don't have to corporate image and backing to convince people here. I've already called in professional consultants here for this. |
14:19.05 | MikeJ[Laptop] | but I enjoy it... |
14:19.40 | mcreedjr | Can anyone recommend a good, cheap IP phone. I am pilot testing an Asterisk solution and I just want to try VoIP out. |
14:19.46 | coppice | cluecon puts the con in conference :-) |
14:20.01 | DarthClue | mcreedjr: Grandstream bt100 |
14:20.16 | DarthClue | coppice: cluecon will have the best of the VOIP world and then some. |
14:20.20 | ManxPower | mcreedjr: There ARE no cheap good phones. |
14:20.36 | ManxPower | mcreedjr: Try the SIPura SPA-841. It's the best phone for under $100. |
14:20.46 | ManxPower | We use Polyco IP 300 and IP500 |
14:20.47 | coppice | no it won't. I'm not going :-) |
14:20.58 | ManxPower | me neither |
14:21.06 | ManxPower | There are two "best" that won't be there. |
14:21.11 | mcreedjr | ManxPower: I was just waiting for the no good, cheap phone thing :) Thank you for your input |
14:21.34 | ManxPower | mcreedjr: Grandstream would qualify as "cheap and no good" |
14:21.38 | DarthClue | coppice and ManxPower: i'm not commenting. |
14:21.40 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
14:21.56 | DarthClue | Grandstream qualifies as a good test phone. Period. |
14:21.58 | MikeJ[Laptop] | coppice, but you should.. |
14:22.00 | ManxPower | And honestly, I don't give a flying fuck about non-asterisk related stuff. |
14:22.13 | MikeJ[Laptop] | cuz I really want to finally meet you. |
14:22.19 | coppice | hold it in asia and i'll consider going |
14:22.22 | lyroy | Does someone could tell me how can I insert VALUES in a MYSQL database in my dialplan? |
14:22.28 | MikeJ[Laptop] | where are you? |
14:22.31 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
14:22.37 | coppice | duh! asia |
14:22.42 | MikeJ[Laptop] | I got that much. |
14:22.46 | RoyK | hk |
14:22.50 | *** join/#asterisk darwin35 (~darwin35@ip70-186-117-198.ma.dl.cox.net) |
14:22.54 | RoyK | if he hasn't moved lately |
14:22.55 | mosty | lyroy, what do you mean by values? |
14:22.55 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
14:23.01 | lehel | hello |
14:23.10 | darwin35 | ok major issues with head as of 8:45 am |
14:23.12 | ManxPower | coppice IRCs from a remote village in China and uses GRPS and SAT links for IRC. |
14:23.17 | MikeJ[Laptop] | hehe |
14:23.19 | RoyK | lyroy: use application MYSQL from asterisk addons |
14:23.23 | darwin35 | audio for voicemail craps out |
14:23.43 | lyroy | mosty exten => 5,3,MYSQL(Query resultid ${connid} INSERT\ INTO\ foward\ VALUES\(\'XXXXXXXXXX\'\,\'XXXXXXXXXX\'\)) |
14:24.22 | lyroy | something like that |
14:24.58 | darwin35 | it crashes asterisk when it cant read the vm password also |
14:25.09 | Katty | aroo! |
14:25.14 | mosty | lyroy, you probably want to lookup the "asterisk gateway interface" (agi): http://www.voip-info.org/wiki-Asterisk+AGI |
14:26.13 | *** join/#asterisk cybyc (~CYB@bi01p1.nc.us.ibm.com) |
14:27.16 | darwin35 | I also get constant speaker beeps |
14:27.18 | Katty | woah |
14:27.22 | Katty | there are still drive in theators |
14:27.28 | Katty | Jul 12 08:54:08 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Critical Request) |
14:27.31 | Katty | erm, wrong paste |
14:27.34 | Katty | Jul 12 08:54:12 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Non-critical Request) |
14:27.43 | Katty | http://www.driveintheater.com/index.htm |
14:27.49 | *** part/#asterisk cybyc (~CYB@bi01p1.nc.us.ibm.com) |
14:27.52 | Katty | that's neat :> |
14:27.55 | DarthClue | Katty: yes...We have one here that shows first run movies for less than the actual theatre. |
14:28.15 | goldenear | ManxPower Two IAX2 phones behind different NAT routers, but talking to the same Asterisk server can't use IAX2 transfers. <-- it works for me :) |
14:28.39 | Katty | DarthClue: hot |
14:28.44 | Katty | i wanna go to a drive in movie |
14:28.46 | darwin35 | Jul 12 09:15:00 WARNING[70034] chan_sip.c: sip_xmit of 0x8158e1c (len 451) to 10.0.0.2 returned -1: Bad file descriptor |
14:28.58 | darwin35 | its filling my log with loads of these |
14:30.00 | goldenear | ManxPower, if the * server has a public ip and if the NAT is nos symmetric, IAX2 transfers will work. I did the test. |
14:30.55 | ManxPower | goldenear: between IAX2 phones behind different NATs. I don't mean "transfer the call". I mean IAX2 native transfers so the phones talk directly to each other once the call is set up, bypassing Asterisk. |
14:31.18 | ManxPower | It really should be called "IAX2 reinvites" or something other than "transfer" |
14:32.10 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
14:32.41 | darwin35 | well thats screwed |
14:33.12 | goldenear | ManxPower, that what I'm talking about |
14:33.27 | goldenear | and it works :) |
14:33.51 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
14:34.04 | ManxPower | goldenear: it will work with port forwarding, of course. |
14:34.05 | *** join/#asterisk JakubS_ (~qbast@pe72.czempien.sdi.tpnet.pl) |
14:34.42 | JakubS_ | hello |
14:34.59 | JakubS_ | do all G.7xx codecs are lossless? |
14:35.19 | coppice | JakubS_ none are |
14:36.10 | ManxPower | JakubS_: not G711 |
14:36.24 | JakubS_ | are they enough to transmit fax ? |
14:36.41 | coppice | even G.711 is not lossless |
14:37.11 | goldenear | ManxPower, it worked for me without any manual seting on the NAT/router |
14:37.47 | ManxPower | goldenear: How do you know that the iax2 native transfer worked? |
14:37.58 | goldenear | I did the test |
14:38.06 | ManxPower | goldenear: WHAT TEST? |
14:38.29 | ManxPower | Did you use tcpdump? Did you look at the console? Did you get telepathetic reports from alians from Alpha Centauri? |
14:38.57 | *** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155) |
14:39.03 | ManxPower | packet sniffing is really the only way to know if the IAX2 native transfer worked or not. |
14:39.27 | coppice | is packet sniffing addictive? |
14:39.35 | goldenear | user1 behind NAT 1, user2 behind NAT 2, * server with a public IP (different than each NAT/router IP) |
14:39.38 | Nugget | heh |
14:39.42 | mariogamboa | i have a question? i want to use the g729 codec but i see i need pay royalities is true this or exist a open source of the codec |
14:39.52 | Nugget | mariogamboa: it is true. |
14:40.01 | Ahrimanes | coppice: quite so |
14:40.07 | MikeJ[Laptop] | coppice. yes... and it gets you high |
14:40.09 | ManxPower | mariogamboa: you need to pay a license fee. Digium sells the license in partnership with the patent holder. |
14:40.13 | mosty | mariogamboa, you need to buy licenses for the g729 patent. it's $10 usd per channel |
14:40.16 | goldenear | user1 call user2 --> asterisk log show native transfer |
14:40.27 | ManxPower | goldenear: that means nothing. |
14:40.34 | ManxPower | goldenear: try using a packetsniff. |
14:40.47 | Ahrimanes | ethereal is your pal |
14:41.15 | mariogamboa | in this channel is when a ip phone call to another ip phone i need to buy 1 license or need 2 licenses |
14:41.35 | ManxPower | mariogamboa: if both phones support G729 then you don't need any license. |
14:41.49 | Katty | how much do you tip a hair dresser? |
14:41.54 | goldenear | (also monitoring with successive iax2 show channels show me that the connection between usr2 and usr2 is released as usr1 and usr2 are effectly still speaking ) |
14:41.57 | Katty | no cow tipping comments please (= |
14:42.01 | bkw_ | Cresl1n, boi whats up |
14:42.05 | Hmmhesays | ha! |
14:42.09 | Katty | bkw_: how much do you tip your hair dresser? |
14:42.09 | mosty | manxpower: you do if you want to do voicemail, or anything else that "speaks" to something that only supports g729 |
14:42.10 | mariogamboa | i have 2 grandstream |
14:42.11 | Cresl1n | howdy all ;-) |
14:42.19 | ManxPower | but if you want the phone to connect to the PSTN, Asterisk voicemail. "t/T" option to Dial, use Zapscan, Zap Barge, or any other application or part of asterisk that has to listen to or play audio -- then you need a licenmse. |
14:42.19 | goldenear | ManxPower, why do it means nothing ? |
14:42.20 | Katty | bkw_: TELL ME |
14:42.22 | Katty | bkw_: I MUST KNOW |
14:42.23 | bkw_ | Katty, I don't.. I give her gossip |
14:42.28 | Katty | bkw_: oh. k |
14:42.34 | bkw_ | we gossip like crazy |
14:42.41 | bkw_ | because her ex-husband married my neighbor |
14:42.45 | Ahrimanes | lol |
14:42.48 | Katty | bkw_: do you know what you're supposed to tip? |
14:42.49 | ManxPower | goldenear: Asterisk is pretty bad this sort of stuff when it comes to the console messages. |
14:42.50 | mariogamboa | but if i don't add the g729 codec to asterisk the grandstream phones only conect in g711a |
14:42.59 | Hmmhesays | nothing if they suck at hair dressing |
14:43.00 | mariogamboa | can't connect in g729 |
14:43.05 | Katty | Hmmhesays: and if they don't? |
14:43.12 | ManxPower | mariogamboa: that is correct. |
14:43.13 | bkw_ | Katty, I have simple needs |
14:43.15 | bkw_ | I shouldn't have to tip |
14:43.20 | Katty | argh |
14:43.23 | bkw_ | I think 13 bucks for a haircut is plenty |
14:43.25 | bkw_ | for 10 min of work |
14:43.25 | Cresl1n | mariogamboa: yep |
14:43.26 | Katty | you're going to make me ask google :< |
14:43.32 | mariogamboa | in this case manx i need to buy the license is correct? |
14:43.34 | Cresl1n | mariogamboa: that's the way it is |
14:43.39 | ManxPower | but if you want the phone to connect to the PSTN, Asterisk voicemail. "t/T" option to Dial, use Zapscan, Zap Barge, or any other application or part of asterisk that has to listen to or play audio -- then you need a licenmse. |
14:43.43 | Cresl1n | mariogamboa: right again |
14:43.45 | Hmmhesays | yeah I normally will give them a buck |
14:43.45 | goldenear | ManxPower, I also verified it with netstat :) |
14:43.46 | mosty | mariogamboa, do the grandstream phones support gsm? |
14:43.47 | Katty | bkw_: well my hair is going to take a wee bit more than 10 minutes |
14:43.55 | goldenear | it works I tell you ! |
14:43.56 | Katty | bkw_: try about 50 minutes :< |
14:43.57 | Hmmhesays | 15% |
14:44.04 | ManxPower | goldenear: What universe do you live in? I want to move there. |
14:44.05 | Nugget | The difference between a $12 haircut and a $120 haircut is about two days. |
14:44.06 | Katty | Hmmhesays: google said 20% |
14:44.13 | Cresl1n | Katty: that's a long time |
14:44.15 | ManxPower | My haircuts are free. |
14:44.18 | Katty | that's like a dollar for every 5 |
14:44.20 | Hmmhesays | google wants you to put more money into the economy |
14:44.25 | Cresl1n | Katty: how much does it cost? |
14:44.30 | Katty | Cresl1n: don't know yet |
14:44.31 | mariogamboa | ok |
14:44.35 | Cresl1n | hrm... |
14:44.39 | Katty | Cresl1n: just want to know how much to tip (= |
14:44.43 | Cresl1n | Katty: Maybe you could check froogle |
14:44.46 | Cresl1n | :-) |
14:44.46 | Katty | Cresl1n: trust me, it will take awhile...my hair is down to my hips. |
14:44.58 | Katty | Hmmhesays: you missed webcam last night! |
14:45.02 | goldenear | ManxPower, I live on planet IAX, constellation of asterisk :) |
14:45.03 | mariogamboa | anyhere have the pinout of the wildcard te110p |
14:45.08 | Katty | Hmmhesays: last picture is still up if you wanna go see (= |
14:45.09 | ManxPower | Katty: a clippers will fix that very fast. |
14:45.14 | Katty | ManxPower: noooo |
14:45.21 | Hmmhesays | I did huh, I was sleeping all night long. sure |
14:45.41 | Katty | Hmmhesays: it's in usual spot (= |
14:45.45 | DarthClue | Katty: warn us next time web cam is gonna be on. |
14:45.59 | Katty | DarthClue: it'll be like 2 motnhs |
14:46.02 | Katty | i mean months |
14:46.12 | Hmmhesays | you'll have to remind me, It's been like 9 months since i've been there |
14:46.17 | Katty | Hmmhesays: oh, and see new.jpg, new2.jpg, and new3.jpg for the new heels i bought (and am wearing today!) |
14:46.28 | DarthClue | Katty: that's ok, we'll see ya in less than a month at cluecon. |
14:46.31 | goldenear | ManxPower, and on my planet SIP means local net or ipv6 only :) |
14:46.41 | Katty | DarthClue: ya (= |
14:47.04 | *** join/#asterisk jhava (~icechat5@200.58.26.21) |
14:47.16 | Katty | i'll be the Voice From Behind |
14:47.23 | Katty | to quote a famous book |
14:47.29 | bkw_ | ok that sounds just wrong |
14:47.33 | Katty | k |
14:48.02 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
14:48.03 | sneak | hi |
14:48.07 | Katty | bkw_: s'ok, i don't want people staring at me anyway (= |
14:49.06 | goldenear | ManxPower, why IAX2 native transfers shouldn't work with [non symmetric] NAT ? |
14:49.10 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
14:49.15 | Katty | DarthClue: until i steal it |
14:49.29 | Katty | actually, i think i'll nap |
14:49.49 | *** join/#asterisk Madkiss (madkiss@madkiss.staff.freenode) |
14:49.52 | Madkiss | Hi folks |
14:50.27 | bkw_ | i'm about to murder greg |
14:50.33 | Katty | bkw_: :<<< |
14:50.36 | bkw_ | he thinks he's about to die.. he's not feeling well |
14:50.41 | Madkiss | What does "insecure=very |
14:50.42 | Katty | rut roh |
14:50.43 | Madkiss | " cause in sip.conf? |
14:50.51 | Katty | bkw_: that calls for hugging, sometimes :< |
14:50.52 | bkw_ | he does this every single time he feels bad |
14:51.04 | bkw_ | then he goes to the dr. which costs out the ass |
14:51.07 | DarthClue | bkw_: maybe he knows that you are about to kill him? |
14:51.23 | Katty | doctor visit costs me 30 |
14:51.29 | Katty | which includes whatever drugs they decide to put me on |
14:51.46 | ManxPower | goldenear: because the two phone have never communicated directly before, they have all been talking to Asterisk, so there is no dynamic NAT translations for the phones going |
14:51.51 | Katty | but i've got insurance, too |
14:51.54 | DarthClue | i need an expiration date. what would be a good expiration date for a jug o milk. |
14:52.19 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:52.19 | *** mode/#asterisk [+o anthm] by ChanServ |
14:52.25 | Katty | how about a jug o DarthClue |
14:52.39 | ManxPower | i.e. Phone at 172.16.7.12 is communicating with IP address 209.45.3.16. If another device then tries to communicate with the phone, well the NAT router has never seen an outbound packet to the 2nd phone's IP address and so doesn't have a NAT translation set up for it. |
14:52.52 | Katty | :< |
14:53.03 | Katty | you can't leave me alone with bkw_! |
14:53.20 | DarthClue | Katty: bkw_ is harmless, mostly. |
14:53.23 | *** join/#asterisk wasim (~wasim@wasim.active.supporter.pdpc) |
14:53.26 | Katty | :<<< |
14:53.32 | goldenear | it does, as the * server know on wich port the Phone is connecting from |
14:53.43 | ManxPower | goldenear: Correct, but the NAT router does NOT. |
14:54.20 | goldenear | but if the NAT is not symmetric, the used port is kept for any connection |
14:54.24 | goldenear | so it will work |
14:55.08 | ManxPower | goldenear: NAT is done on PORT+IPADDRESS |
14:55.18 | *** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net) |
14:55.25 | goldenear | and so ? |
14:55.32 | ManxPower | Even if the packet is coming from the correct source port, the address will be different from that of the Asterisk server. |
14:56.07 | ManxPower | it would be like doing a DNS query thru a NAT router, and the response comes from a different port and IP adddress. It won't get thru the router. |
14:56.12 | goldenear | but it will be accepted, as the phone also try to contact the other one |
14:56.40 | ManxPower | goldenear: and the NAT router then picks a random source port for NAT and the packet fails. |
14:57.04 | goldenear | you think symmetric NAT ! |
14:57.22 | goldenear | I said, if the NAT is symmetric is won't work |
14:57.31 | Madkiss | Oh, does somebody know whether the voicemail-mesages are available in german language? |
14:57.34 | goldenear | but if the nat is non symmetric, it will work |
14:57.52 | ManxPower | goldenear: so you admit there ARE nat issues with IAX2 native transfers. |
14:58.10 | mosty | madkiss: i see a package in debian (sarge) called asterisk-prompt-de |
14:58.50 | *** join/#asterisk mrtwister (~user@cable-1-32.cgates.lt) |
14:59.00 | goldenear | but less than with SIP |
14:59.22 | *** part/#asterisk mrtwister (~user@cable-1-32.cgates.lt) |
15:00.00 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net) |
15:00.06 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
15:00.25 | goldenear | and if native transfer is impossible because of symmetric NAT, the call will just smoothly keep to go through the * server |
15:00.29 | ManxPower | goldenear: SIP reinvites simply don't work with NAT unless, maybe, you have an external "thing", SER, STUN, etc. |
15:00.31 | Madkiss | mosty: Cool, thank |
15:00.44 | *** join/#asterisk montag___ (~montag@host187-252.pool8175.interbusiness.it) |
15:01.26 | montag___ | what's the best way to interconnect a PABX with asterisk for a ibrid system (asterisk with some VoIP clients <---> pabx with phones) |
15:01.28 | montag___ | ? |
15:01.42 | goldenear | symmetric NAT are so evil ! |
15:01.53 | alphaque | montag___: depends how many lines. if its large, consider E1 otherwise a few FXOs |
15:01.54 | goldenear | that's the issue, not IAX |
15:02.16 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
15:02.44 | goldenear | yep, SIP needs help to work with NAT |
15:03.01 | lehel | why i can't run asterisk as "asterisk" while i'm loading chan_capi ?? |
15:03.02 | montag___ | 8 lines...i think a couple of tdm400 connected to analog extensions on pabx, but i think it's better E1. But TE410P E1 card work well with all pabx ???? |
15:03.21 | ManxPower | I recommend you stay away from the TDM400Ps |
15:03.30 | alphaque | montag___: u'd need an E1 conn from the pbx too |
15:03.44 | coppice | i recommend staying away from anything electronic |
15:03.57 | ManxPower | lehel: use a channel bank if your PBX has no E-1/T-1 ports |
15:04.27 | montag___ | with FXO lines two problem: call delay (waiting for the first ring...), and when i receive a call from the analog pabx and i trasnfer the call to another analog pabx extension i use 2 analog lines :-( |
15:04.45 | montag___ | i've tried only tdm400, there's better on the market ??? |
15:04.52 | alphaque | montag___: cant u use flash() to do that ? |
15:05.00 | ManxPower | montag___: Channel bank |
15:05.06 | lehel | what do you mean channel bank ManxPower ? |
15:05.19 | montag___ | alphaque: flash ??? |
15:05.42 | alphaque | yes, flash hook transfer. should free up the line connected to pbx and thus let call pass within pbx |
15:06.03 | montag___ | alphaque: have u some samples for this ??? |
15:07.30 | razu | can anyone help me loading ztdummy module under slackware 10.1 ... i'm getting errors :S |
15:08.13 | *** part/#asterisk santiago (~santiago@63.245.86.188) |
15:08.23 | alphaque | montag___: www.voip-info.org |
15:08.30 | goldenear | who said that iax2 has some audio issues ? |
15:08.58 | goldenear | bkw_, ? |
15:09.09 | ManxPower | bkw_ says a lot of stuff. |
15:09.39 | goldenear | does he take drug or something ? ;) |
15:09.53 | ManxPower | goldenear: I hope so! |
15:10.14 | goldenear | me too! |
15:10.37 | goldenear | he's scary ;) |
15:10.54 | ManxPower | goldenear: he's also right much of the time. |
15:10.56 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
15:11.01 | ManxPower | and he's smart as well. |
15:11.02 | lehel | ManxPower: i read about Asterisk+Channel+Bank on voip. but still 'don't get.. |
15:11.17 | goldenear | so he is even more scary !!! |
15:11.46 | ManxPower | lehel: your analog lines into FXO ports on the channel bank, the FXS ports to your PBX, then a T-1 card to Asterisk from the channel bank. A channel bank converts t-1/e-1 channels into analog ports |
15:13.37 | *** part/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
15:13.45 | *** join/#asterisk Malthus (~herman@port0129-abn-s-adsl.cwjamaica.com) |
15:13.56 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
15:14.27 | lehel | ManxPower: i've no doubt that this is a useful information, but not for me yet:-/ |
15:14.46 | lehel | and i don;t know what is the connection between my Fritz card |
15:16.05 | lehel | AVM Fritz! CAPI .. |
15:16.26 | Katty | lehel: you've insaned. |
15:17.20 | lehel | ? |
15:18.36 | lehel | specify Katty |
15:19.50 | Katty | lehel: you. have. insaned. |
15:25.02 | lehel | explain |
15:25.09 | Katty | ... |
15:25.31 | lehel | then shut up |
15:25.40 | Katty | gosh |
15:25.47 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
15:25.54 | Katty | lehel: don't be hateful )= |
15:26.19 | lehel | k i'll try no to.. :P |
15:26.24 | MikeJ[Laptop] | whos a hater? |
15:26.34 | MikeJ[Laptop] | yous a hater! |
15:26.45 | Katty | hater? i hardly know 'er! |
15:27.34 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM] |
15:27.34 | *** join/#asterisk johnh51 (~john@adsl-66-218-62-115.dslextreme.com) [NETSPLIT VICTIM] |
15:27.34 | *** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM] |
15:27.34 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM] |
15:28.20 | lehel | zbye |
15:28.23 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
15:28.39 | *** join/#asterisk kkq (~christ@234-200-29-134.hcc.mnscu.edu) |
15:28.50 | kkq | hello |
15:33.12 | *** join/#asterisk tomtom_ (~root@83-217-70-161.reverse.realroot.be) |
15:33.14 | tomtom_ | hi |
15:35.21 | kkq | hi |
15:35.41 | RoyK | is robert webb here? |
15:35.41 | *** join/#asterisk pdugas (~pdugas@h79.95.40.69.ip.alltel.net) |
15:36.13 | kkq | jack webb is on my television |
15:36.41 | *** join/#asterisk LOT (~Methos@S0106000f6694b86f.ed.shawcable.net) |
15:37.07 | *** join/#asterisk rjreb (~rjreb@greatwall.amer.net) |
15:37.51 | ManxPower | ~docs |
15:37.51 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:37.52 | ManxPower | ~mailinglist |
15:37.52 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
15:38.49 | albators | ~seen royk |
15:38.49 | jbot | royk is currently on #asterisk (8h 20m 42s). Has said a total of 48 messages. Is idling for 3m 8s |
15:38.58 | albators | ~history |
15:38.58 | jbot | history is, like, at http://ibot.rikers.org/oe/ |
15:39.11 | albators | ~asterisk |
15:39.11 | jbot | methinks asterisk is the symbol that looks like a star (shift-8 on north american English keyboards) |
15:39.32 | albators | ~;ls |
15:39.46 | albators | :P |
15:40.26 | albators | ~did |
15:40.26 | jbot | well, did is Direct Inward Dialing |
15:40.43 | albators | ~CLIP |
15:40.50 | albators | ~CLI |
15:40.50 | jbot | methinks cli is a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
15:41.06 | albators | ~crc |
15:41.06 | jbot | crc is, like, (Cyclic Redundancy Check) A mathematical calculation that produces a number that can be used to check the integrity of a file by regenerating the number and comparing the results. |
15:41.47 | albators | ~ping me |
15:41.48 | jbot | pong albators |
15:41.56 | albators | ~time |
15:41.56 | jbot | You are educated stupid and therefore too dumb to understand nature's perfect time cube! |
15:42.12 | albators | :D |
15:42.39 | albators | can i crash it ;) ? |
15:43.01 | RoyK | albators: ? |
15:43.12 | RoyK | ~lart albators |
15:43.33 | ManxPower | albators: better people than you have tried and failed |
15:43.45 | albators | never say never :P |
15:44.11 | albators | everyone has a bug :P |
15:44.18 | albators | is just matter of time to be found :P |
15:45.16 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
15:45.28 | jeremywhiting | hi all |
15:46.07 | jeremywhiting | just checking to see if I've got this right, if I want to dial from one asterisk to another asterisk to a zap channel that may or may not be in use |
15:46.28 | jeremywhiting | should I just dial the second box and have alternate plans if dial doesn't work e.g. n+101, etc |
15:46.59 | jeremywhiting | or do you have to do chanisavail in there somewhere? |
15:47.44 | MikeJ[Laptop] | chanisavail is not necessary, just let it roll baby... |
15:48.13 | jeremywhiting | and if there's no channel available on the second asterisk box for dialing out, would the first dial come back to n+1 or n+101? |
15:48.27 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
15:48.35 | jeremywhiting | or do I tell it which when the second asterisk box dial doesn't work with a busy or something? |
15:48.53 | *** join/#asterisk logicalonline (~Ken@border.logicalonline.com) |
15:49.00 | kaldemar | it comes to n+101. |
15:49.12 | jeremywhiting | ok, thanks |
15:49.27 | kaldemar | are you familiar with Noop? |
15:49.41 | kaldemar | quite useful for printing debug information. |
15:49.48 | alphaque | jeremywhiting: it goes to n+101 on the /second/ asterisk box |
15:50.03 | jeremywhiting | so Box1 Dial(IAX2/box2) -> Box2 Dial(Zap/g1) fails then how do I kick it back so box1 knows it failed |
15:50.10 | jeremywhiting | or was busy or whatever |
15:50.19 | alphaque | jeremywhiting: the first dial succeeded to the second box |
15:50.31 | kaldemar | oh yes, but if you have Busy in the second box, it comes to n+101 on the first one. |
15:50.58 | jeremywhiting | kaldemar: really, I sure hope so, that would be nice |
15:50.58 | alphaque | jeremywhiting: like kaldemar says, use Busy() or Congested() in the second box which would cause the first to go to n+101 |
15:51.07 | jeremywhiting | ok, got it |
15:51.08 | jeremywhiting | thanks |
15:51.21 | *** part/#asterisk logicalonline (~Ken@border.logicalonline.com) |
15:51.22 | jeremywhiting | you guys really are the best |
15:51.31 | alphaque | jeremywhiting: aww shucks. :) |
15:51.32 | kaldemar | i tried that chain with several boxes some time ago. |
15:52.07 | jeremywhiting | yeah, the boss wants to use the cheaper lines for long distance from the home office while at the remote office if he can |
15:52.11 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
15:52.15 | jeremywhiting | trying to save some $$ to pay me with I guess |
15:52.25 | *** join/#asterisk jansen (~Jan@p5487894B.dip0.t-ipconnect.de) |
15:55.29 | DarthClue | ~cluecon |
15:55.29 | jbot | rumour has it, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
15:55.44 | file | ^^^ Everyone should go that ^^^ |
15:56.42 | ManxPower | Everyone should stop spamming the channel about Cluecon. |
15:56.51 | mutilator | NOOOOOOOOO |
15:56.58 | mutilator | NOT CLUECON! |
15:57.18 | ManxPower | In fact everyone should boycott Cluecon if they continue to spam the mailing lists and the channel. |
15:57.29 | mutilator | heh |
15:57.32 | *** part/#asterisk Myshenka (~gunde@217.9.101.85) |
15:57.33 | DarthClue | ManxPower: it would appear that your definition of spam is anything that mentions anything you are interested in, am i right? |
15:57.41 | cpatry | ManxPower: i think u really have something against cluecon. |
15:57.45 | ManxPower | DarthClue: only partially. |
15:57.47 | anthm | ManxPower, perhaps everyon involved in cluecon should boycott asterisk |
15:57.53 | Malthus | lol |
15:57.57 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
15:58.15 | ManxPower | DarthClue: Three messages to asterisk-users from bkw_ about the SAME speaker at cluecon exceeded my annoyance threashold. |
15:58.26 | mutilator | perhaps i should go eat some hotdogs |
15:58.34 | ManxPower | That and the fact that he seems to send 2 or so messages (seperate messages) about EACH AND EVERY speaker. |
15:58.36 | DarthClue | it would also appear that some people drink too much of the asterisk voodoo and fail to realize just how much the cluecon sponsors actually do for asterisk. |
15:58.57 | ManxPower | DarthClue: Sangoms? |
15:59.04 | ManxPower | ..er..Sangoma does a lot for Asterisk? |
15:59.11 | DarthClue | ManxPower: anthm, bkw_, file, etc. |
15:59.15 | anthm | perhaps that's because we get new speakers every day and cannot predict the future |
15:59.33 | anthm | how can we send an email today telling you who decided to speak tommorow |
15:59.40 | ManxPower | DarthClue: This must be some previously unknown definition of "sponsor". Perhaps you meant "organizers"? |
16:00.11 | ManxPower | anthm: perhaps a simple web page listing all the speakers and/or one e-mail with all the speakers. |
16:00.23 | ManxPower | I certinally don't get 2 e-mails about every speaker at Astricon |
16:00.44 | bkw_ | ManxPower, you do realize thats a bug in the mailing list software |
16:00.49 | zoa | i get 9 for every speaker :) |
16:00.53 | anthm | well I've been told the mailing list has a bug in it and sends the same email more than once |
16:00.55 | bkw_ | I sent it once to all three lists |
16:01.01 | DarthClue | ManxPower: 3? why is it that I only see one? |
16:01.03 | bkw_ | when you do that it triggers the bug |
16:01.14 | Malthus | I see two |
16:01.16 | ManxPower | bkw_: no, I did not know that. |
16:01.17 | bkw_ | it causes it to dupe all messages 3 times |
16:01.28 | bkw_ | three per list or something stupid like that |
16:01.30 | bkw_ | look at the title |
16:01.31 | zoa | look at the subjects, they are slightly different |
16:01.35 | bkw_ | yep |
16:01.37 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:01.37 | ManxPower | It's still not needed to send a seperate message about each and every speaker. |
16:01.55 | bkw_ | ManxPower, I would have announce them together but at the time I didn't know that Nix was going to show up |
16:01.57 | bkw_ | or I would have |
16:02.03 | ManxPower | If Astricon did that there would be riots in the street. |
16:02.18 | bkw_ | Lets count the 500+ emails I got from astricon about the hot tub |
16:02.23 | bkw_ | that went on for a week |
16:02.27 | bkw_ | now that was anoying |
16:02.35 | mutilator | send me a hottub! |
16:02.39 | Juggie | nothing wrong with a hot tub |
16:02.49 | mutilator | heh |
16:02.49 | *** join/#asterisk zimba02 (~zimba02@200.39.213.62) |
16:02.54 | bkw_ | not at all but when I got 500 emails about it over a week it was pretty anoying :P |
16:02.56 | Juggie | i got drunk in a hottub sat night, good times. |
16:03.13 | anthm | we won't spam the list anymore |
16:03.20 | anthm | after aug5 |
16:03.21 | ManxPower | I still have 24 messages in my trash folder about cluecon since July 7 |
16:03.25 | zoa | :) |
16:03.35 | anthm | then we will resume in sept for cluecon2006 |
16:03.40 | ManxPower | Fortunatly the mailing list issue is nolonger an issue since updated my .procmailrc |
16:03.42 | bkw_ | ManxPower, it wasn't intentional to annoy anyone.. so i'm sorry if it did... I'll try not to annoy! |
16:03.52 | *** join/#asterisk Tili (~Tili@202-133-65-107-dialup.sat.net.pk) |
16:04.04 | ManxPower | bkw_: my level of annoyance is less since you informed me about the mailinglist bug. |
16:04.13 | anthm | maybe you should give up on hating it since it's a collection of all the ppl who give you the code you use to run your business ! |
16:04.13 | zimba02 | hi all, i have a question to make asterisk send tcp information, is this the right place to ask? |
16:04.16 | ManxPower | But still, isn't your cluecon advertizing a little excessive? |
16:04.17 | bkw_ | i'll see if we can get them to update that software to fix that |
16:04.48 | bkw_ | ManxPower, not any more than Astricon really |
16:04.54 | anthm | nope it's a small price to pay for all the stuff we do for free all year |
16:04.58 | anthm | free |
16:05.13 | DarthClue | ManxPower: not really. If you really are tired of it, pay anthm and he'll make it stop. |
16:05.21 | bkw_ | hahahaha |
16:05.23 | ManxPower | DarthClue: How much????????????? |
16:05.41 | Juggie | anthm, did you ever hear of an indian giver? |
16:05.51 | Juggie | thats someone who gives away something for free, and expects something in return |
16:06.03 | ManxPower | if I can't afford it, maybe I can create a non-profit to find the "no clie-con" messages. |
16:06.07 | tomtom_ | anyone has experience with channel grouping in zapata.conf? when i register to two groups (eg: group = 1,0) it doesn't seem to work ... |
16:06.07 | Juggie | you work in open source, be happy if you get anything. |
16:06.09 | bkw_ | Juggie, you're just mad because we won't give away confcall :P |
16:06.32 | Juggie | bkw, my meetme+agi solution is working just fine |
16:06.37 | bkw_ | hehe |
16:06.38 | Juggie | i'll use that for years before i pay for open source |
16:06.42 | file | confcall is sexy |
16:06.44 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
16:06.44 | anthm | yah so what does that make you an "indian taker" |
16:06.50 | bkw_ | hahahahha |
16:07.04 | bkw_ | well to be honest people that take and take and take are worse than we are |
16:07.08 | bkw_ | we give and give and give.. |
16:07.16 | bkw_ | atleast people could say thanks |
16:07.17 | bkw_ | but they don't |
16:07.24 | eKo1 | thanks |
16:07.27 | Juggie | my thanks is helping people in here on irc |
16:07.30 | Juggie | and answering questions |
16:07.31 | ManxPower | A Cluecon announcement is in the /topic. If someone needs jbot to vomit up the announcement to the channel for people....well you don't want those people to come to the con anyway. |
16:07.32 | Juggie | thats my thanks |
16:07.45 | Corydon-w | Anything further on the copyright infringement alleged by xoasis on the dev list? |
16:08.01 | bkw_ | Corydon I was wondering about that also |
16:08.08 | Juggie | and thats more then 99% of the people do. |
16:08.19 | bkw_ | no |
16:08.22 | ManxPower | Corydon-w: I suspect it's the same crap. No proof that they have not purchased a commercial license. No comment from Digium, not even a mention of trying to report the issue to Digium before posting to the mailing list. |
16:08.23 | bkw_ | 1% of the people do that |
16:08.26 | bkw_ | 99% take and make |
16:08.34 | bkw_ | while the 1% gives back |
16:08.36 | Juggie | bkw, we just said the same thing. |
16:08.41 | anthm | Manx, c'mon |
16:08.47 | DarthClue | tomtom_: HEAD or STABLE? |
16:09.04 | Juggie | i said, i come in here, i help people, i give back where i can, thats more then 99% of people do. |
16:09.08 | drumkilla[work] | I have forwarded that email from the -dev list to the Digium licensing director so that it can be addressed through our lawyers |
16:09.10 | bkw_ | ManxPower, I took the sysmaster thing to digium months before I let the beans spill |
16:09.11 | Corydon-w | ManxPower: yes, but there's no license that Digium can make to remove copyright notice |
16:09.14 | Juggie | and you just said the same thing. |
16:09.31 | ManxPower | bkw_: I wasn't referring to you in this case *tease* |
16:09.33 | Juggie | bkw, so we agree there then :) |
16:09.35 | bkw_ | hehe |
16:09.51 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.active.supporter.pdpc) |
16:09.54 | bkw_ | Juggie, agree that the world is full of selfish pricks that could care less? |
16:09.54 | bkw_ | sure |
16:09.55 | ManxPower | Corydon-w: have you seen their commercial license? |
16:10.09 | ManxPower | Corydon-w: I haven't. |
16:10.11 | Zeeek | have I come at a bad time? I have a question for bkw_ |
16:10.11 | Corydon-w | drumkilla[work]: sure would be nice to know that the lawyers are doing SOMETHING about the infringement, even if we have no details |
16:10.24 | bkw_ | if digium doesn't do anything thats just bad |
16:10.27 | Silik0n | is it ever a bad time to come |
16:10.28 | bkw_ | I suspect they will |
16:10.31 | anthm | hmm so Juggie we donate like $100k in services and code per year and we want people to come and learn more about voip and that makes me an indian giver? |
16:10.35 | Corydon-w | ManxPower: you cannot remove a copyright notice, period, except if you have permission from the copyright owner |
16:10.48 | ManxPower | bkw_: Then the people that DO have copyright claims to pieces of Asterisk can start sueing 8-) |
16:10.51 | Zeeek | this is about premiumgoldplatinumpbx? |
16:11.13 | anthm | I'd love to hear some more about that |
16:11.15 | Zeeek | file you might know |
16:11.17 | Juggie | anthm, the part i was referencing was the bombardement of advertising. |
16:11.34 | file | I know lots of things. |
16:11.40 | bkw_ | nobody says anything when we bombard cvs with patches and such |
16:11.48 | bkw_ | double standard |
16:11.50 | bkw_ | :P |
16:11.53 | Zeeek | I have a friend who wanted to try HEAD but wanted to be able to revert to STABLE 1.0.6 if it didn't work out |
16:12.02 | ManxPower | bkw_: we would if they were posted to -users or -dev and posted to the channel. |
16:12.02 | file | bkw_: let's get the anthm contribution list! |
16:12.05 | bkw_ | Zeeek, it will wokr |
16:12.17 | anthm | oh like the advertising that comes with free help and megabytes of free code? |
16:12.27 | Zeeek | so he tried HEAD had some problems and went back, removed the modules redid make install on the old STABLE. But... |
16:12.41 | anthm | as i said it's takers we are talking about not givers |
16:12.47 | bkw_ | cd /usr/include |
16:12.49 | bkw_ | rm -rf asterisk |
16:12.52 | bkw_ | before you go back to stable |
16:12.59 | ManxPower | jbot!*@* added to ignore list. <-- that should help |
16:13.09 | anthm | manx you are silly |
16:13.16 | ManxPower | between /ignore and .procmailrc perhaps I'll see less cluecon stuff. |
16:13.18 | twisted[work] | hussha |
16:13.18 | Katty | ManxPower: silly rabbit |
16:13.20 | Zeeek | My "friend" had a shitload of wacky linux issues suddenly because apparently HEAD Makefile writes to /etc/modules.conf alnd all that and it kept wanting to load wctdm and all that |
16:13.22 | DarthClue | ~cluecon |
16:13.22 | jbot | it has been said that cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
16:13.32 | Nugget | Linux is poo. :) |
16:13.37 | ManxPower | Zeeek: 1.0.x also writes to that |
16:13.50 | *** join/#asterisk jhava (~icechat5@200.58.26.21) |
16:13.50 | Zeeek | bkw_ ah ok if that was it it's good to know. I'm sure my "friend" will be happy |
16:14.01 | zimba02 | need some help making a small tcp application work with asterisk, any help appreciated |
16:14.09 | Zeeek | but in the meantime there were no phones inthe office for two hours |
16:14.51 | ManxPower | zimba02: The only TCP stuff in Asterisk is for manager interface and remote console. All VoIP is UDP. |
16:15.05 | DarthClue | Cluecon is the only conference where you won't have to hear ManxPower give a presentation. |
16:15.07 | Katty | hmm, i've noticed my polycoms acting funny |
16:15.10 | Zeeek | ManxPower "my friend" was confused because he doesn't use /etc/modules but has all the modprobe stuff in rc.d/rc.local |
16:15.18 | eKo1 | DarthClue: hahaha |
16:15.20 | DarthClue | Katty: you shouldn't have let bkw_ touch them. |
16:15.24 | Katty | funny strange, not funny haha..................after awhile, you can pick up, but not hear anything.....but the other person can hear you |
16:15.30 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
16:15.31 | Katty | rebooting the phone seems to fix it |
16:15.32 | anthm | unless he want's to he's welcome of course |
16:15.40 | bkw_ | yep |
16:15.44 | bkw_ | I think I offered |
16:16.10 | anthm | but he's gonna go ignore us and not give us the same respect we give him I think |
16:16.21 | Zeeek | then one of my firned's providers suddenly switched domains with no notice excapt all calls were answered by "your are calling the wrong server... please change to ..." |
16:16.22 | anthm | that's too bad |
16:16.31 | DarthClue | No respect I tell ya, no respect! |
16:16.36 | Katty | RESPECT |
16:16.41 | Katty | show you what it means to me! |
16:16.42 | Silik0n | god thats a blantant rip off of FOP |
16:16.43 | Katty | do do do do do do |
16:16.44 | mutilator | sing it! |
16:16.50 | Zeeek | R E S P E C T Fine Out what it mean to me |
16:16.54 | twisted[work] | Katty, keep your do do to yourself ;P |
16:16.54 | zoa | id go, but it cant make it |
16:16.55 | zimba02 | i'm looking to acomplish the follow, I have written a small c program to send information to an external application via tcp, i need asterisk to send cdr informatin over tcp |
16:17.05 | Zeeek | socket.2me socket.2me socket.2me socket.2me socket.2me |
16:17.12 | *** join/#asterisk brimstone (~brimstone@brimstone.digium.sponsor.pdpc) |
16:17.15 | Katty | twisted[work]: pffft |
16:17.20 | Zeeek | oh shiw I won't be able to sleep now |
16:17.27 | file | zimba02: you can write custom CDR handlers ya know |
16:17.33 | twisted[work] | Katty, heheh ;) |
16:17.41 | cpatry | cdr_custom.c |
16:17.48 | ManxPower | No, sweetie, I'm ignoreing the conference. |
16:17.51 | ManxPower | Not YOU. |
16:17.58 | Katty | i'll pun you in a minute. |
16:18.03 | twisted[work] | promise? :P |
16:18.06 | Katty | no |
16:18.06 | anthm | the conference is all me! |
16:18.08 | JerJer | cpatry: blah |
16:18.09 | twisted[work] | lol |
16:18.14 | zoa | :) |
16:18.24 | anthm | anyone doesnt like it it's all my fault |
16:18.31 | JerJer | cpatry: anyone can register their own cdr handler by simply calling one asterisk API function |
16:18.31 | ManxPower | I don't give a flying fuck about the conference, but anthm and the rest have made significant contributions to Asterisk. |
16:18.35 | Katty | it's always your fault, anthm |
16:18.40 | Katty | YOUR FAULT |
16:18.49 | bkw_ | yeppers |
16:18.51 | DarthClue | Come to Cluecon. The only conference with a high female / geek ratio! |
16:18.53 | bkw_ | JerJer, is right |
16:19.00 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
16:19.02 | Zeeek | 3 to 2000 ? |
16:19.04 | ManxPower | The only time I ever put bkw_ on /ignore is when he's being an asshole. |
16:19.11 | ManxPower | 8-) |
16:19.13 | DarthClue | Zeeek: even better. |
16:19.17 | bkw_ | I am never an asshole.. i'm always a bitch |
16:19.20 | Zeeek | 4 : 2000 ? |
16:19.20 | *** join/#asterisk Shaneful (~sharper@d154-20-37-11.bchsia.telus.net) |
16:19.22 | bkw_ | :P |
16:19.25 | Silik0n | damnit DarthClue enuff spamming about cluecon already |
16:19.27 | file | bkw_: very much so |
16:19.31 | cpatry | anthm: we have to go in city, ive a large room for myself alone! :P |
16:19.32 | Silik0n | we get it |
16:19.34 | Nugget | I can stay home and enjoy the perfect equilibrium of a 1/2 female/geek ratio. |
16:19.41 | ManxPower | Silik0n: Just put jbot on /ignore |
16:19.58 | DarthClue | Zeeek: at least 3:40 |
16:20.05 | Katty | DarthClue: how many now? |
16:20.13 | Katty | oh |
16:20.15 | Katty | k |
16:20.19 | JerJer | we need some conf whores to show up |
16:20.21 | zimba02 | i know, i need to solve a billing solution for a hotel, they have their own billing system reading cdr from serial port or old pbx |
16:20.22 | DarthClue | at least, but i don't have the list. |
16:20.24 | Zeeek | jeeze the estrogen ration will be really heavy! not. |
16:20.42 | DarthClue | JerJer: if enough people come, we can always get some high quality girls. |
16:20.46 | file | zimba02: so write an asterisk module to do it |
16:20.57 | Silik0n | its not that everyother word out of that boy is cluecon... I definatly understand the need to promote the conference and I sincerly hope it does well, but damn there becomes a point where its bocming annoying and counter productive |
16:21.04 | JerJer | plan a rave for the last night :) |
16:21.11 | ManxPower | Hey! Maybe we can have conference advert free-for-all! In this corner is Astericon, in this corner is Cliecon, in this corner is VON! They can duke it out on the channel, damn the annoyance! |
16:21.15 | *** part/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
16:21.16 | zimba02 | that's where I need some help or good links to get started |
16:21.16 | zoa | <zimba02> i know, i need to solve a billing solution for a hotel, they have their own billing system reading cdr from serial port or old pbx -> we made such a tool some months ago |
16:21.32 | Shaneful | Is there a calling card application that allows a caller to make as many calls as they need . something like when done with your current call hit pound for 3 seconds to dial the next? |
16:21.39 | Silik0n | now if he was staying STRIPPERS AT CLUECON |
16:21.45 | Silik0n | that would be a different storry |
16:21.51 | Silik0n | but I havent heard shit about the strippers yet!@ |
16:21.52 | twisted[work] | Katty, after the help i've given you you be mean? |
16:21.53 | Cresl1n | Come to RussellCon2005! |
16:22.02 | zimba02 | i have written a c program and work fined, i tried to "incorporated" to asterisk and got lost.. lol |
16:22.04 | cpatry | mouhjahahha |
16:22.05 | DarthClue | Silik0n: if enough people come to cluecon...there will be STRIPPERS AT CLUECON! |
16:22.07 | Cresl1n | it's the only conference that's all about russell |
16:22.09 | Cresl1n | :-) |
16:22.12 | Katty | twisted[work]: i'm always mean |
16:22.12 | twisted[work] | DarthClue, hmm |
16:22.14 | Katty | twisted[work]: VERY mean |
16:22.20 | bkw_ | but keep the divorce lawyers handy too |
16:22.24 | twisted[work] | AHH |
16:22.24 | Silik0n | Cresl1n i thought that was geekcon |
16:22.26 | brimstone | Cresl1n: ? |
16:22.28 | eKo1 | STRIPPERS!? where?! |
16:22.33 | brimstone | sorry, wrong window again |
16:22.36 | zoa | hey creslin |
16:22.38 | DarthClue | and for an additional fee, we cover the tips and lawyers fees. |
16:22.38 | Silik0n | bkw_: what happens in chicago stays in chicago |
16:22.42 | Cresl1n | zoa!!! :-) |
16:22.45 | zoa | they promised to do it on the 22nd |
16:22.47 | bkw_ | Silik0n, ok |
16:23.16 | Silik0n | bkw_ be sure to let me know if you are really going to do strippers so I can break out the stripclub MP3s and bring the right stuff |
16:23.23 | zimba02 | do you have any links or info where to go and searcho for the tool made months ago? |
16:23.34 | file | nights are open to go to a stripclub :P |
16:23.34 | ManxPower | "Candy" and her twin sister "Boffy" will be the "hosts" for the conference party! |
16:24.06 | mogorman | russelcon would be cool |
16:24.10 | mogorman | all stable all the time |
16:24.16 | ManxPower | Yeah! |
16:24.22 | anthm | oh sure, fuck telephony, if it's about strippers we can go on all day talking about it! |
16:24.24 | Cresl1n | russellcon will be a big room that we rent out for 3 days |
16:24.31 | drumkilla[work] | that's hot |
16:24.49 | mogorman | and all stable bugs get kicked around |
16:24.52 | Cresl1n | with a steady supply of red bull and food sent to it |
16:24.58 | mogorman | until 1.2 rumbles out of the ashes |
16:25.03 | Katty | i wonder if redbull is vegan |
16:25.11 | Cresl1n | I'm sure it can be |
16:25.11 | ManxPower | drumkilla: my Dutch Red Bull can was squished in my luggage. |
16:25.12 | Cresl1n | :-0 |
16:25.18 | drumkilla[work] | Katty: I don't think it is |
16:25.19 | Cresl1n | sucks |
16:25.21 | twisted[work] | dude |
16:25.24 | Katty | drumkilla[work]: :< |
16:25.28 | twisted[work] | Taurine isn't vegetarian |
16:25.37 | mogorman | yeah |
16:25.39 | mogorman | exactly |
16:25.39 | Cresl1n | it's just a chemical, no? |
16:25.47 | zimba02 | chat zoa |
16:25.47 | drumkilla[work] | Cresl1n: that's not what I heard ... |
16:25.47 | anthm | we can guarentee strippers.... |
16:25.50 | twisted[work] | Cresl1n, yes, but it's created by bull testicles |
16:25.54 | anthm | the develpers cam |
16:25.55 | Cresl1n | not an animal product |
16:25.55 | Cresl1n | ooh... |
16:25.56 | Cresl1n | wow |
16:26.01 | anthm | "strip the code" |
16:26.04 | twisted[work] | go look up taurine |
16:26.04 | anthm | strip asterisk |
16:26.07 | Nugget | ]:8) No Bull |
16:26.12 | Katty | hi Nugget! |
16:26.13 | anthm | yay more efficient |
16:26.15 | twisted[work] | anthm, talk to drumkilla about releasing astwipe |
16:26.16 | mogorman | my virgin eyes |
16:26.24 | drumkilla[work] | you'd think they were using a synthetic version now though |
16:26.29 | Cresl1n | I heard that asterisk is made of moose testicles |
16:26.29 | bkw_ | virgin eyes my ass! |
16:26.44 | mogorman | lol |
16:26.45 | Katty | gosh, i'm not looking |
16:26.47 | bkw_ | Cresl1n, I have a good picture of you |
16:26.48 | mogorman | bkw_ you know nothing |
16:26.48 | ManxPower | Cresl1n: Only CVS-HEAD. |
16:26.58 | twisted[work] | drumkilla, they may be... but i dono |
16:27.04 | bkw_ | http://www.bkw.org/photos/spring_von2005/100_0328.html |
16:27.06 | Katty | in fact, i'm going to lunch |
16:27.13 | twisted[work] | enjoy, Katty |
16:27.17 | zimba02 | irc question, zoa replied in red to my question, how do I reply the same way? |
16:27.34 | bkw_ | here is Cresl1n under the table http://www.bkw.org/photos/spring_von2005/100_0229.html |
16:28.03 | wasim | zimba02: you need to be in debt |
16:28.05 | *** part/#asterisk zpn (~xpn@207.111.174.1) |
16:28.10 | ManxPower | I dunno what it is about Cresl1n, but he always seems to look stoned. |
16:28.18 | *** join/#asterisk zpn (~xpn@207.111.174.1) |
16:28.18 | file | ManxPower: pfft |
16:28.19 | cpatry | i prefer twisted on fire: http://www.bkw.org/photos/spring_von2005/100_0294.html :P |
16:28.24 | bkw_ | here is zoa and Cresl1n http://www.bkw.org/photos/astricon_2004/100_0921.html |
16:28.38 | zimba02 | sorry, don't know what you mean by debt (that I am with my bank.. lol) |
16:28.43 | Cresl1n | oh yeah, my glasses broke |
16:28.51 | Cresl1n | those ones I bought in san francisco |
16:28.58 | file | Cresl1n: that's sad |
16:29.02 | bkw_ | you had no need for those gay ass glasses |
16:29.03 | bkw_ | really |
16:29.06 | bkw_ | its better they broke |
16:29.19 | bkw_ | here is kram http://www.bkw.org/photos/astricon_2004/100_0920.html |
16:29.23 | bkw_ | look at that look |
16:29.27 | bkw_ | haha |
16:29.28 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
16:29.50 | cpatry | mouhaha |
16:29.54 | bkw_ | here is blitzrage http://www.bkw.org/photos/astricon_2004/100_0807.html |
16:29.57 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com) |
16:30.12 | ManxPower | I like the zoa/Cresl1n tongue pic better. |
16:30.19 | bkw_ | <PROTECTED> |
16:30.21 | bkw_ | it was funny |
16:30.23 | ManxPower | Asterisk Porn! |
16:30.32 | bkw_ | the security guard was getting on me for taking pictures of the Marta |
16:30.49 | bkw_ | "its a security issue" |
16:30.55 | bkw_ | bitch please |
16:31.21 | mogorman | someone hasnt google asterisk on google images |
16:31.35 | zoa | i made a new tongue version with creslin on the last astricon |
16:31.36 | zoa | :) |
16:31.52 | *** join/#asterisk cfrank (~cfrank@bi01p1.co.us.ibm.com) |
16:31.55 | bkw_ | haha |
16:35.31 | ManxPower | Maybe I should do a talk on Asterisk Urban Myths at Astricon/Calif. Things like "reinvite=", "the TDM card is stable", and "SIP and NAT are hard". Total lies like that need to be dispelled. |
16:36.10 | *** join/#asterisk [Jedi] (~fdsafasdf@213.162.200.226) |
16:36.11 | cfrank | bkw_: Is dundi dying? |
16:36.16 | [Jedi] | Hello |
16:36.18 | Zeeek | ManxPower you used to say SIP and double NAT were hard |
16:36.24 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
16:36.25 | Malthus | what's the reinvite lie? |
16:36.26 | [Jedi] | *CLI> pri show span 2 |
16:36.27 | [Jedi] | Primary D-channel: 47 |
16:36.27 | [Jedi] | Status: Provisioned, Down, Active |
16:36.35 | [Jedi] | how can i put it 'Up'? |
16:36.54 | DarthClue | [Jedi]: call your provider? |
16:36.56 | [Jedi] | the remote end is supposed to be enabled |
16:37.38 | ManxPower | Zeeek: until I tried them 8-) |
16:37.42 | cpatry | u sure the cable is right connected? (just in case) :) |
16:37.47 | [Jedi] | yes |
16:37.49 | mkrufky | [Jedi]: what is on the OTHER end of the cable? |
16:37.56 | [Jedi] | mkrufky: a #5ess |
16:38.01 | ManxPower | It took me an hour and that was mostly reading up on DNS SRV records. 8-) |
16:38.02 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
16:38.02 | *** mode/#asterisk [+o anthm] by ChanServ |
16:38.21 | ManxPower | Malthus: The sip.conf option reinvite= does NOT exist. Check the source code. |
16:38.47 | Zeeek | what about makeitworkplease=yes ? |
16:39.01 | Zeeek | I've used that to advantage |
16:39.09 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
16:39.22 | Zeeek | havereadwiki=yes |
16:39.23 | DarthClue | youtookmystapler=yes? burndownthebuilding=yes? |
16:39.35 | mkrufky | [Jedi]: i had a problem like that when i was connecting to another pbx.... Make sure that your asterisk box is set to CPE if the PBX is set to NET ... (or vice versa -- MY pbx is set to CPE, so I had to set the asterisk box as NET) |
16:39.40 | DarthClue | iamyourfather=nooooooooooooooooooooooooooooooooooo? |
16:39.46 | Zeeek | heh |
16:40.00 | Zeeek | ok I submit to the jury another situation |
16:40.11 | Katty | my phone won't work |
16:40.12 | Katty | IT WON"T WORK |
16:40.14 | outtolunc | whatshouldthechanneldo=deadlock <G> |
16:40.16 | Katty | fix it |
16:40.30 | [Jedi] | mkrufky: my "PBX" is a big 128 E1 switch... and it's suposed to be NET |
16:40.42 | mkrufky | Zeeek: I like your fix better.... now let's implement it :-D |
16:40.44 | Malthus | oh |
16:40.45 | Zeeek | I call the provider and the audio goes both ways between my SIP phone, asterisk and their SIP server, hence codecs and the rest are fine. Two-way audio and DTMF are fine |
16:40.47 | mkrufky | j/k |
16:40.55 | *** join/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:41.03 | zimba02 | zoa: do you have any info or link where to find the tool to send tcp information from asterisk? |
16:41.15 | Zeeek | But a call to a phone number from this same provider give a call that rings wand when answered, no audi either way |
16:41.21 | mkrufky | [Jedi]: so your asterisk box should be set as CPE ... just make sure that it already is set that way -- just a suggestion |
16:41.29 | Zeeek | This HAS to be their problem, right? (many users complaining as well) |
16:41.51 | Zeeek | they keep saying, yeah, well, asterisk blah blah, port forwarding |
16:41.56 | [Jedi] | mkrufky: that's in zapata.conf, right? |
16:42.00 | twisted[work] | cpatry, i don't even remember that picture |
16:42.09 | [Jedi] | signalling=pri_cpe |
16:42.13 | cpatry | twisted: so that's really bad then :) |
16:42.25 | twisted[work] | cpatry, lol |
16:42.42 | twisted[work] | cpatry, i don't remember what was going on there... but apparently it was on the exhibit floor while it was open |
16:42.47 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
16:42.50 | twisted[work] | so i'm kinda scared to think what I might have been doing |
16:42.53 | *** part/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:43.08 | mkrufky | [Jedi]: yes... zapata.conf should have -- signalling=pri_cpe |
16:43.56 | cpatry | maybe you're already on playgirl then :P |
16:44.17 | cpatry | apparently, i need to taste to shots too. |
16:44.24 | *** join/#asterisk grolloj (~grolloj@slim-eth0.horizonlive.net) |
16:44.54 | *** join/#asterisk MustDie (voip@205.247.13.252) |
16:45.23 | *** join/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:46.11 | mkrufky | [Jedi]: you also need switchtype=5ess ... but i bet you already know that ;-) |
16:47.05 | puzzled_ | anyone happen to remember that nice webpage about tweaking pci interrupt latencies to help cards work better with asterisk? |
16:48.01 | Katty | should i update my 2.4 kernel? |
16:48.05 | zoa | my asteriskguru page ? |
16:48.17 | Silik0n | Katty: update to 2.0.36 |
16:48.18 | *** part/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:48.21 | puzzled_ | zoa: could be. lemme check |
16:48.25 | zoa | http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
16:48.27 | zoa | is this the one ? |
16:48.28 | Katty | :< |
16:48.29 | Hmmhesays | 2.4 to what? |
16:48.38 | Katty | isn't 2.6 out? |
16:48.40 | Silik0n | ;) |
16:48.44 | puzzled_ | zoa: yup, thanks |
16:48.48 | Hmmhesays | are you having issue's caused by the 2.4 kernel? |
16:48.54 | Katty | Hmmhesays: no (= |
16:48.58 | Silik0n | kajtzu yeah |
16:49.00 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
16:49.04 | Silik0n | err katty yeah 2.6 is out |
16:49.10 | Silik0n | 2.6.11 even |
16:49.15 | Hmmhesays | I personally would not if you have no reason other than just to upgrade |
16:49.18 | Katty | k |
16:49.31 | Hmmhesays | especially on a production box.... if it's a testing box, go for it |
16:49.32 | mkrufky | Silik0n: 2.6.12 is the current mainline kernel |
16:49.56 | mkrufky | ...and 2.6.13-rc2 is current stable prepatch |
16:50.01 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172) |
16:50.26 | mkrufky | ...and 2.6.13-rc2-mm2 is current dev tree (released today) |
16:50.28 | Hmmhesays | hrm, the more I play with perl, the more I like it |
16:50.37 | Katty | Hmmhesays: how do i know when to update asterisk? |
16:50.45 | Silik0n | hah |
16:50.50 | mkrufky | if it's not broken, dont fix it |
16:50.55 | Hmmhesays | production or testing box? |
16:50.56 | Silik0n | Katty when you find something b0rked :) |
16:50.56 | Katty | mkrufky: that's a male attitude |
16:50.59 | mkrufky | lol |
16:51.12 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
16:51.15 | Katty | Hmmhesays: it's not a test machine anymore |
16:51.19 | mkrufky | i agree with you... i only feel that way on my PRODUCTION asterisk box |
16:51.20 | Katty | Hmmhesays: it omgworks |
16:51.21 | DarthClue | katty: just bring it to cluecon, we'll do it for you. |
16:51.27 | Silik0n | it might be a male attitude, but for a production box, thats always a good course of action |
16:51.27 | Katty | DarthClue: shan't |
16:51.31 | [Jedi] | mkrufky: no, my switchtype is a EuroISDN |
16:51.36 | Hmmhesays | my rule of thumb for production boxes is, don't update unless you find something that needs fixing that an update will fix |
16:51.37 | Silik0n | unless a newer version happens to have a feature you need |
16:51.39 | Katty | DarthClue: keep your dirty paws off my silky......err, asterisky ...uhh, box |
16:51.39 | [Jedi] | mkrufky: althought it is a lucent 5ess, it's using euroisdn signaling |
16:51.56 | Katty | Hmmhesays: k |
16:52.05 | Katty | Hmmhesays: so...something equivilent of doing windows updates....... |
16:52.12 | Katty | Hmmhesays: is......non existant? |
16:52.18 | mkrufky | [Jedi]: OOoooh ... ok I'm not familiar with that stuff... hopefully someone else in here knows |
16:52.22 | Katty | Hmmhesays: whether it's the kernel, or asterisk? |
16:52.26 | Katty | or.......debian |
16:52.42 | Katty | Hmmhesays: is there /anything/ i should be updating on a fairly regular basis? :P |
16:52.50 | Hmmhesays | security patches for debian |
16:53.19 | SpaceBass | does debian have anything like RH's up2date or what ever? |
16:53.26 | mkrufky | apg-get |
16:53.28 | Hmmhesays | apt |
16:53.29 | mkrufky | oops |
16:53.31 | mkrufky | apt-get |
16:53.33 | Hmmhesays | it has super cow powers |
16:53.44 | Katty | Hmmhesays: how.....do i do that? |
16:53.55 | mkrufky | Katty: type man apt-get |
16:53.56 | SpaceBass | but will apt-get run as a damean or something that notifis you when security updates are available? |
16:53.58 | Katty | Hmmhesays: apt-get update alapeanutbuttersammiches? |
16:54.23 | mkrufky | SpaceBass: apt-get CAN be set to run regularly by cron |
16:54.30 | SpaceBass | gotcha |
16:54.34 | Hmmhesays | brb |
16:54.51 | SpaceBass | similar to yum? |
16:55.22 | mkrufky | VERY similar... but i like apt better |
16:55.37 | mkrufky | i like apt SO much better that i even use it on my fedora boxes |
16:55.46 | SpaceBass | my only problem with yum- and is apt-get the same? - is that i never know exactly what the package is called |
16:55.49 | JerJer | tar zxf filename.tgz ; cd filename ; make install <--- the only way |
16:55.56 | anthm | "if it ain't broke, don't fix it" is a good policy in the computer industry and is not a "male" thing |
16:56.02 | SpaceBass | for instance... I needed gnome-kerberos and thats what its call everywhere i've ever seen it |
16:56.06 | Malthus | SpaceBass: apt-cache search |
16:56.08 | SpaceBass | but yum couldt find it |
16:56.10 | SpaceBass | ahhh |
16:56.19 | anthm | because computers tend to break the most whilst being "fixed" |
16:56.20 | mkrufky | SpaceBass: apt-cache search gnome kerberos |
16:56.32 | zimba02 | looking to replace a panasonic pbx with asterisk in a hotel that reads cdrs via serial cable with the following solution |
16:56.33 | SpaceBass | good to know |
16:56.34 | SpaceBass | thanks |
16:56.34 | mkrufky | Malthus: :-) |
16:56.44 | Katty | anthm: boy, did i get into the wrong field ;) |
16:56.55 | alphaque | anthm: aint that the truth ! |
16:57.07 | Malthus | mkrufky: :) |
16:57.14 | JerJer | zimba02: you could make asterisk crap out cdr's on a serial interface |
16:57.36 | anthm | Katty, nah it's just a fact you come to terms with after doing it long enuf |
16:57.37 | jeremywhiting | hi all, anyone know how to tell if a line is connected to tdm04b card from asterisk cli? |
16:57.46 | ManxPower | jeremywhiting: no. |
16:57.49 | JerJer | jeremywhiting: zttool |
16:57.52 | Katty | anthm: yeah i know, that's what i do to the windows server :/ |
16:57.53 | ManxPower | You should be able to tell via zttool. |
16:57.55 | *** join/#asterisk InfraRed (~bigboss@master.subhi.com) |
16:58.03 | ManxPower | You can also post a bounty to PAY someone to add that basic feature. |
16:58.04 | jeremywhiting | is zttool the only way? |
16:58.08 | Katty | anthm: which is why it unnerved me when that guy from out here starting poking about on it |
16:58.10 | ManxPower | jeremywhiting: yes. |
16:58.11 | cpatry | jeremywhiting: zap show status? |
16:58.20 | wasim | cat /proc/zap/* |
16:58.21 | InfraRed | hi all |
16:58.25 | zimba02 | jerjer: thanks, the billing system is related to room checkout, etc.. and the software developer has opened a tcp port to recieve text |
16:58.26 | jeremywhiting | cpatry:? no such thing |
16:58.40 | cpatry | jeremywhiting: its only in cvs-head |
16:58.43 | [Jedi] | i I do intense debug in my PRI, I get "Sending Set Asynchronous Balanced Mode Extended" |
16:58.50 | ManxPower | jeremywhiting: what is your specific issue? |
16:58.54 | zimba02 | all i need to send is the sql statment to the tcp port, "inster into bla blah" |
16:59.06 | ManxPower | zimba02: you can't do that. |
16:59.06 | [Jedi] | what may be happening in my E1? |
16:59.08 | jeremywhiting | just trying to tell if phone line is connected right at a remote location |
16:59.17 | jeremywhiting | because asterisk isn't picking up when I call the line |
16:59.19 | *** join/#asterisk jfonsecausa (~jfonseca@c-66-176-57-28.hsd1.fl.comcast.net) |
16:59.24 | anthm | ;delete from blah ;grant me a shell ? |
16:59.27 | InfraRed | question: I am trying to change my voip termination company, i have asterisk server and 6 extensions, do i need one login per phone on the remote server? or do i just need one account for the asterisk server? |
16:59.28 | ManxPower | jeremywhiting: Asterisk does not support line or dialtone detection on analog cards. |
16:59.47 | jeremywhiting | ok |
16:59.57 | ManxPower | jeremywhiting: I have that fairly often with one or two of my TDM400P. Reboot the server (and hope it doesn't crash when unloading zaptel) and prey. |
17:00.01 | ManxPower | and pray too |
17:00.05 | zimba02 | is not an sql connection, it's just a tcp server reading "instert into...." |
17:00.10 | Malthus | InfraRed: just one |
17:00.12 | jeremywhiting | is there a way to see debug messages if I get a call in on the zaptel interface |
17:00.12 | wasim | InfraRed: one login per server would work, as well as one per line, its flexible |
17:00.21 | ManxPower | zimba02: How, exactly, does this relate to Asterisk? |
17:00.25 | jeremywhiting | so after the line is connected I have to restart to get asterisk to use the lines? |
17:00.30 | [Jedi] | jeremywhiting: pri debug |
17:00.30 | bkw_ | ManxPower, yes it does.. if the line is unpluged it goes into red alarm on analog lines |
17:00.40 | ManxPower | jeremywhiting: only if the TDM card blows up. |
17:00.44 | jeremywhiting | [Jedi] not using pri I don't think |
17:00.49 | ManxPower | bkw_: That's Zaptel, not asterisk. |
17:00.52 | zimba02 | manxpower: i will be replacing their existing panasonic pbx with an asterisk box |
17:00.58 | [Jedi] | "Status: Provisioned, In Alarm, Down, Active" |
17:00.59 | InfraRed | ok |
17:01.00 | InfraRed | cool :) |
17:01.05 | bkw_ | you'll get an alarm notice if you have debug on |
17:01.06 | [Jedi] | how can I see the kind of alarm on the PRI? |
17:01.11 | jfonsecausa | Hi, Someone knows haw to dial to a provider of DID's who does not requires registry neither passwoed or userid? |
17:01.15 | ManxPower | [Jedi]: red alarm, yellow alarm, or blue alarm? |
17:01.15 | *** join/#asterisk iswm (iswm@iswm.user) |
17:01.32 | ManxPower | bkw_: Sorry, you are correct about that. |
17:01.36 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
17:01.38 | [Jedi] | ManxPower: I don't know how to check the alarm type |
17:01.43 | [Jedi] | :)))) |
17:01.48 | cpatry | jedi: "zap show status" give u all that info. (only in cvs-head) |
17:02.00 | ManxPower | Actually, I've only see the red alarm on X100P, never on the TDM400P. |
17:02.08 | [Jedi] | No such command 'zap show status' (type 'help' for help) |
17:02.10 | ManxPower | [Jedi]: zttool |
17:02.13 | *** join/#asterisk zpn (~xpn@207.111.174.1) |
17:02.20 | cpatry | u need zttool then. |
17:02.22 | [Jedi] | I don't have zttool |
17:02.27 | [Jedi] | where's that? |
17:02.29 | ManxPower | [Jedi]: What part of "only in CVS-HEAD" are you not understanding. |
17:02.30 | cpatry | then run cvs-head |
17:02.31 | cpatry | :) |
17:02.40 | [Jedi] | I can't run HEAD |
17:02.41 | ManxPower | [Jedi]: zttool will only build if you have newt-devel installed. |
17:02.47 | [Jedi] | ok |
17:02.49 | cpatry | why? |
17:02.52 | [Jedi] | then I'll rebuild |
17:02.56 | [Jedi] | cpatry: because I need BRIstuff |
17:03.04 | cpatry | k |
17:03.05 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
17:03.09 | ManxPower | cpatry: If you have to ask "why can't you run CVS-HEAD" then ANYTHING he tells you will be moot. |
17:03.20 | [Jedi] | why? =) |
17:03.34 | cpatry | nmouhaha |
17:03.42 | jeremywhiting | maybe since I had to patch my wcfxs.c to detect my card it's not working quite right? |
17:03.48 | ManxPower | The 1.0.x people think that people that run CVS-HEAD are crazy lunatics, and the CVS-HEAD people think the same about people running 1.0.x. |
17:03.48 | jeremywhiting | asterisk still doesn't answer the calls |
17:03.56 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
17:04.00 | *** join/#asterisk _ioscanner (~ioscanner@c-67-162-251-133.hsd1.tx.comcast.net) |
17:04.05 | ManxPower | jeremywhiting: perhaps you could try using zttool? |
17:04.12 | gtigene | Does call parking work with SIP phones? I pressed # and nothing happened. |
17:04.15 | jeremywhiting | where's that, just in cvs |
17:04.16 | _ioscanner | Hello all |
17:04.37 | ManxPower | jeremywhiting: If you plug a phone line into a Digium FXS module, the module will blow up at the first call. The ports may not be numbered as you expect. |
17:04.44 | DarthClue | those of us who choose to run CVS-HEAD know who runs the asylum, we just prefer to have more padding on our walls. |
17:04.44 | ManxPower | jeremywhiting: it's in zaptel. |
17:04.45 | [Jedi] | ok, installing newt-devel |
17:04.50 | alphaque | gtigene: did u dial() with the t or T option ? |
17:04.54 | *** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com) |
17:05.06 | ManxPower | jeremywhiting: make sure newt-devel is installed, rebuild zaptel, you will have zttool |
17:05.06 | [TK]D-Fender | Anyone have experience with Channel banks on lines with DSL on them? I'm lloking to use a channel bank for and analog failover to my PRI and want to know if have dsl (behind a filter) should pose a problem with it. |
17:05.31 | ManxPower | gtigene: Is your SIP phone too brain dead to support a transfer button? |
17:05.36 | _ioscanner | I have 1-0-07/05/05 cvs stable code on a box. Asterisk just restarts at at times. Seem mostly when I get SIP calls |
17:05.53 | alphaque | ManxPower: some cheap sip hard phones dont |
17:06.02 | ManxPower | alphaque: those are brain dead phones. |
17:06.12 | alphaque | ManxPower: note i said /cheap/ |
17:06.22 | gtigene | alphaque: no t or T in my Dial commands |
17:06.27 | wasim | [TK]D-Fender: theoretically it shouldn't |
17:06.31 | alphaque | gtigene: then do that |
17:06.33 | gtigene | ManxPower: my phone transfers calls OK |
17:06.51 | [Jedi] | ok |
17:06.54 | [Jedi] | I have now zttool |
17:06.56 | [Jedi] | my alarms are RED |
17:06.58 | ManxPower | gtigene: If your phone does not support supervised transfers then there is no way to do supervised transfers in 1.0.x |
17:07.11 | ManxPower | [Jedi]: Red means "I don't see a line, it's a wireing problem" |
17:07.27 | ManxPower | Any time we get red alarms we call the telco and say "we have red alarms" and they fix it. |
17:07.48 | ManxPower | gtigene: and you need supervised transfers to work for call parking |
17:07.48 | _DAW | Hi All.. |
17:08.02 | gtigene | ManxPower: what is a supervised transfer? Its a Polycom phone, I just press the transf soft key. |
17:08.33 | ManxPower | gtigene: You have a polycom phone. You do NOT need "t" or "T" on your Dial line. |
17:08.43 | ManxPower | gtigene: What is the specific problem? |
17:08.55 | ManxPower | I manage 60 polycom phones and we use call parking all the time. |
17:09.11 | _DAW | [Jedi] - your prob is with T1/PRI? Try a loopback plug. |
17:09.12 | jeremywhiting | zttool doesn't show anything |
17:09.17 | gtigene | ManxPower: I looked up call parking in the Wiki and it said you park a call by pressing #. I press # and nothing happens. |
17:09.31 | ManxPower | gtigene: # is only for phones too stupid to have a transfer button |
17:09.53 | ManxPower | gtigene: transfer the call to the parking extension, listen to for the parking lot number, press transfer the second time to complete the transfer. |
17:10.07 | gtigene | ManxPower: thanks |
17:10.16 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
17:10.19 | [Jedi] | ManxPower: I am the telco |
17:10.34 | ManxPower | [Jedi]: then either you have a wireing problem or the port isn't provisioned on the PBX. |
17:10.43 | [Jedi] | ManxPower: the people who manages the phone switch to which my E1 is connected says my E1 is right |
17:10.58 | ManxPower | [Jedi]: then they are either lieing or you have a wireing problem. |
17:11.12 | ManxPower | perhaps you forgot to make a CROSSOVER cable when connecting to the PBX? |
17:11.47 | [Jedi] | the cable is right |
17:12.09 | [Jedi] | it's suposed to be right |
17:12.17 | [Jedi] | and my asterisk sends SABME messages to the poneswitch |
17:12.23 | ManxPower | [Jedi]: plug a loopback connecter into the E-1 line. If your techs can't see a loop then you can WITH NO DOUBT tell them the line is broken. |
17:12.26 | [Jedi] | so it seems the lowest-level is working fine |
17:12.42 | ManxPower | [Jedi]: LISTEN TO ME. Red Alarm means NOTHING AT ALL is working. |
17:12.57 | [Jedi] | ManxPower: but the remote switch says it is! that's the problem |
17:13.13 | ManxPower | [Jedi]: then you have a bad cable. |
17:13.23 | ManxPower | sounds like the TX pair from asterisk is right, but the RX pair isn't. |
17:13.25 | jeremywhiting | what is zttool supposed to show? |
17:13.26 | _DAW | [Jedi] loopback the cable to your * box |
17:13.28 | [Jedi] | uhm |
17:13.32 | *** join/#asterisk M_at (~matthewt@81-1-109-38.homechoice.co.uk) |
17:14.12 | gtigene | ManxPower: Thanks again that works fine |
17:14.16 | ManxPower | jeremywhiting: Sorry, for a TDM400P zttool does NOT show an error if the line is not plugged in. |
17:14.24 | ManxPower | I just tested it. |
17:14.27 | jeremywhiting | oh, it's just for pri cards, etc |
17:14.32 | jeremywhiting | got it, thanks anyway |
17:14.33 | ManxPower | sorry, bkw_ you are wrong. |
17:14.39 | bkw_ | about? |
17:14.44 | ManxPower | jeremywhiting: it's also for X100P. |
17:14.49 | gtigene | How do you listen in on SIP calls. |
17:14.52 | bkw_ | the x100p does |
17:14.53 | jeremywhiting | nope, I'm using a tdm04b |
17:14.55 | bkw_ | mine always has |
17:14.59 | JerJer | gtigene: chan_spy |
17:15.08 | ManxPower | bkw_: zttool provides no indication for TDM400Ps about if a line is plugged in. YES, it works for the X100P |
17:15.08 | bkw_ | I have never had a working tdm board |
17:15.09 | bkw_ | so I wouldn't no |
17:15.11 | gtigene | jerjer: thanks. |
17:15.16 | bkw_ | s/no/know/ |
17:15.22 | alphaque | JerJer: i heard (though i could be wrong) that chan_spy doesnt work in the present -STABLE |
17:15.33 | bkw_ | alphaque, talk to anthm |
17:15.33 | ManxPower | bkw_: Is it time to tell you to shut up and say you don't know what you are talking about? |
17:15.35 | JerJer | nobody should be running -STABLE |
17:15.41 | DarthClue | alphaque, use HEAD or talk to anthm. |
17:15.45 | bkw_ | ManxPower, what ever |
17:15.50 | alphaque | anthm: does chan_spy work with 1.0.6 ? |
17:15.55 | bkw_ | ManxPower, if it doesn't then I would consider that a bug |
17:15.57 | gtigene | alphaque, jerjer: I am running cvs head from about there weeks ago |
17:16.09 | ManxPower | bkw_: me too, actually. |
17:16.20 | ManxPower | bkw_: but zttool does not show any port specific information about that card. |
17:16.31 | alphaque | bkw_: does it work with 1.0.6 ? |
17:16.35 | DarthClue | alphaque: probably not, use HEAD or pay anthm to make it work with STABLE 1.0.9, 1.0.6 is too old. |
17:16.41 | bkw_ | ManxPower, it might print a warning if debug is up high enough |
17:16.42 | *** join/#asterisk carrar (tim@osburn.com) |
17:17.03 | alphaque | DarthClue: 1.0.6 is what freebsd's ports collection has |
17:17.36 | alphaque | DarthClue: like was discussed earlier, it aint broken for me, so no reason to upgrade to 1.0.9. will consider a bump to 1.2 when it comes though |
17:17.42 | bkw_ | ManxPower, you also have to realize I don't work with analog very much! |
17:17.55 | anthm | what's the dealio with the pettyness over who knows what they are talking about I conclude that you are both right depending on what card you are talking about |
17:18.11 | anthm | and we are not sure till we test both cards |
17:18.18 | bkw_ | exactly |
17:18.22 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
17:18.23 | anthm | I can attest that my x100p does have a red alarm |
17:18.31 | anthm | the other one I do not have |
17:18.32 | ManxPower | anthm: I only mention it, since when bkw_ was channeling Elvira he always told me to shut up and that I didn't know what I was talking about. |
17:18.34 | [Jedi] | uhm |
17:18.37 | [Jedi] | now I got RED/LB |
17:18.44 | [Jedi] | what's the difference between RED/LB and RED? |
17:18.51 | Hmmhesays | cause someone must always have the finer feathers |
17:18.51 | ManxPower | [Jedi]: no idea. |
17:19.30 | opus__ | rtfm? |
17:19.46 | anthm | I know read pertains to the physical |
17:19.46 | _DAW | [Jedi] RedLB = red alarm in loopback? |
17:20.10 | ManxPower | [Jedi]: Um, if you did what I suggested, you would not see that message. |
17:20.12 | anthm | no physical layer and yellow/blue is the logical |
17:20.17 | [Jedi] | _DAW but I don't have a loopback =????????? |
17:20.30 | ManxPower | [Jedi]: perhaps your "telco" is looping back. |
17:20.35 | [Jedi] | maybe |
17:20.39 | bkw_ | the telco might have it in LB |
17:20.39 | ManxPower | [Jedi]: call them up. |
17:20.53 | bkw_ | ManxPower, do tdm boards actually work? :P |
17:20.56 | ManxPower | [Jedi]: this is a good thing, if they are. |
17:21.13 | ManxPower | bkw_: They seem to, other than the odd issue with the some of the ports stop working until a reboot. |
17:21.18 | [Jedi] | no, they aren't in loopback |
17:21.24 | bkw_ | wired wrong? |
17:21.25 | [Jedi] | (they say they aren't in loopback) |
17:21.31 | ManxPower | [Jedi]: replace the connectors on the cable. |
17:21.32 | anthm | what is it hooked up with |
17:21.41 | [Jedi] | ok |
17:21.45 | anthm | try both a straight and a cross cable and see which one is better |
17:21.56 | ManxPower | [Jedi]: tell your telco to loop the line. They will fail. |
17:22.28 | anthm | you never know how it's comming in till you mess w it I've see side by side ports where 1 needs cross and the other straight |
17:22.36 | ManxPower | bkw_: I have to reboot our corporate PBX Asterisk Gateway every few days to avoid problems with the TDM400P. I'll be SO happy when we can yank the card in a week or two |
17:23.07 | *** join/#asterisk BMG (BMG@162.80-202-207.nextgentel.com) |
17:23.23 | ManxPower | [Jedi]: Let me know when you finally realize it's a wireing problem. |
17:23.32 | alphaque | ManxPower: amen to that. i've not liked the tdm400p's either |
17:23.44 | BMG | hey, got a problem with musiconhold, if we call the MOH extention number, that works, but we put a user on hold there is silence |
17:23.48 | alphaque | by comparison, the te410s we have are a gem |
17:24.16 | jeremywhiting | BMG, what kind of phones are you using? |
17:24.42 | anthm | I have 4 x100's and 1 olden days tdm400 in 1 box back to the aint broke dont fix it thing ;) |
17:24.45 | [Jedi] | incredible |
17:24.50 | [Jedi] | i've unloaded wct4xxp |
17:24.52 | [Jedi] | and reloaded it |
17:24.59 | BMG | jeremywhiting: cisco 7905G, SJ-Softphone and Cisco 7920 |
17:24.59 | [Jedi] | and the E1 goes up |
17:25.10 | Ariel_ | ok so how do I get this jbot for another channel? |
17:25.30 | [Jedi] | I have my E1 working |
17:25.35 | [Jedi] | and I've done nothing but reloading the driver! |
17:25.48 | alphaque | [Jedi]: without running ztcfg ?? |
17:25.52 | terrapen | EVERYBODY WANG CHUNG TONIGHT |
17:25.56 | M_at | Anyone used the new Grandstreams? GX200s |
17:26.01 | [Jedi] | I run ztcfg when I modprobe the driver, in /etc/modules.conf |
17:26.06 | jeremywhiting | BMG: you need to find out what the sip phone sends to the server when you press hold on the phone |
17:26.10 | jeremywhiting | I think |
17:26.22 | terrapen | today's music theme: 1980s party songs |
17:26.30 | terrapen | musical, even |
17:26.41 | opus__ | does anyone here use intercom? |
17:26.46 | ManxPower | <PROTECTED> |
17:26.55 | [Jedi] | ManxPower: why? they don't work well? |
17:26.59 | jeremywhiting | hey is Zap/3-1 port 3 on the tdm card? or is that indefinite |
17:27.00 | jeremywhiting | ? |
17:27.05 | BMG | jeremywhiting: ok, any idea how to do that ? |
17:27.07 | newmember | ManxPower: what are you moving too? |
17:27.10 | ManxPower | [Jedi]: you apparently missed my tirade about 2 mins ago about them. |
17:27.18 | ManxPower | newmember: T-1 cards and channel banks. |
17:27.26 | jeremywhiting | sip debug maybe, but I'm not sure exactly if that's the problem |
17:27.49 | jeremywhiting | I'm sort of new here too |
17:27.57 | jeremywhiting | only been at asterisk for about a couple months |
17:28.23 | BMG | ok :) we hear that the call is beeing placed at hold when we press hold, but just lack the music |
17:28.48 | BMG | anyone else got any clues ? |
17:29.18 | *** join/#asterisk pa (~Paolo@pa.user) |
17:29.38 | Ariel_ | BMG, is moh working on your system? did you install mpg123 |
17:29.52 | BMG | yes, if we call the extention then it works |
17:30.02 | anthm | use native moh perhaps? |
17:30.23 | BMG | thats what we are doing |
17:30.41 | anthm | the guy who asked about chanspy, it can be ported to stable but it will reqire consulting fee cos it's a pita |
17:31.17 | bkw_ | alphaque, was who asked |
17:31.44 | alphaque | anthm: that was me |
17:32.05 | alphaque | anthm: was just asking. i dont use chan_spy, though i am using the native moh patches which are part of chan_spy |
17:33.06 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
17:33.11 | yaaar | word |
17:34.33 | yaaar | anybody know why my cisco 7940 has no audio? I thought I got the channel setup about right, following the wiki. The phone can dial, and rings the phone it's calling, but neither party can hear the other. On the cisco handset I can only hear a soft 'click' about once every 2.5 seconds. ???? |
17:35.08 | yaaar | if i try to leave a voicemail on an asterisk mailbox it hangs up after a half-second saying 'no audio on <channel> |
17:35.11 | JerJer | yaaar: firewall and/or NAT is blocking RTP |
17:35.15 | _ioscanner | asterisk seem to die when I hang up a call. I see WARNING CDR on channel 'SIP/100-8c3c' not posted and same with lacks end |
17:35.16 | yaaar | nope |
17:35.19 | JerJer | yep |
17:35.28 | _ioscanner | Is this an Asterisk or mysql problem? |
17:35.39 | anthm | using chanspy or native moh as patches is not advised |
17:35.40 | SpaceBass | yaaar on the same lan as the * box? same subnet? |
17:35.47 | anthm | they are not complete |
17:35.48 | JerJer | _ioscanner: you are telling asterisk to not record a cdr |
17:35.49 | _ioscanner | seem that when it tries to write to mysql it dies |
17:35.54 | yaaar | JerJer: this machine is on a public subnet attached directly to the asterisk server via a switch with absolutely no firewall or nat in between. and it's going out on a zap channel |
17:35.59 | JerJer | iptables -L |
17:36.35 | _ioscanner | I should be loggin the cdr in mysql |
17:37.40 | yaaar | JerJer: it just shows the 3 chains (INPUT/OUTPUT/FORWARD) with "(policy ACCEPT)" after each one and no rules following. |
17:38.12 | [Jedi] | If I want to make an application which records voice, I should use EAGI? |
17:38.14 | [Jedi] | or what? |
17:38.16 | JerJer | then you have a codec problem |
17:38.25 | JerJer | are you looking at the whole picture in asterisk? |
17:38.38 | JerJer | [Jedi]: show application record |
17:38.55 | yaaar | JerJer: sorry, not sure what you mean? |
17:39.07 | yaaar | i've got asterisk running with about 7 v's and a c |
17:39.23 | yaaar | but it just looks like a normal call that's working fine |
17:39.27 | JerJer | and logger.conf ? |
17:39.37 | JerJer | allow=ulaw in general section of sip.conf? |
17:39.42 | JerJer | and the phone set to use ulaw |
17:40.21 | *** join/#asterisk |dennis| (~dennis@200.32.197.2) |
17:40.30 | yaaar | logger.conf has both console and messages set to 'notice,warning,error' |
17:41.22 | yaaar | allow=ulaw is in sip.conf |
17:41.27 | *** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com) |
17:42.01 | yaaar | hmmm. how can i check whether the phone is using ulaw? |
17:42.21 | jeremywhiting | yaaar: make a call and do sip show channels |
17:42.25 | _ioscanner | looks like mysql is having a problem. never mind |
17:42.26 | yaaar | cool |
17:42.32 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
17:42.53 | yaaar | yep format ulaw |
17:43.09 | Nugget | mysql is good at causing problems. :) |
17:44.25 | *** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
17:44.41 | *** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
17:46.00 | *** join/#asterisk bdunn (~bdunn@69.15.23.58) |
17:46.20 | ManxPower | the -users part of my .procmail rc: http://pastebin.ca/17555 |
17:46.27 | jeremywhiting | hi all, if I only have 2 lines coming in to a 4 port tdm04b where do I specify to only try those two ports for outgoing calls? |
17:46.46 | ManxPower | jeremywhiting: in your Dial line. |
17:46.55 | ManxPower | along with the group= in zapata.conf |
17:47.09 | terrapen | oh good god... |
17:47.10 | jeremywhiting | channel => 1-2 in zapata.conf? |
17:47.12 | terrapen | this song is hilarious |
17:47.35 | terrapen | "Conservative Christian, Right-Wing Republican, Straight White American Male" |
17:47.36 | ManxPower | along with a group=1 before the channel line. |
17:47.52 | ManxPower | Then you can Dial(Zap/g1/whatever) and it will pick the first port that's not in use by Asterisk |
17:48.45 | ManxPower | Apparently nobody cares about my procmailrc 8-) |
17:49.05 | anthm | try this |
17:49.06 | anthm | group=1,10,100 |
17:49.06 | anthm | channel => 1 |
17:49.06 | anthm | group=2,10,100 |
17:49.06 | anthm | channel => 2 |
17:49.06 | anthm | group=3,20,100 |
17:49.07 | anthm | channel => 3 |
17:49.09 | ManxPower | jeremywhiting: stop what you are doing. Go to voip-info.org. Spend the next week reading about Asterisk |
17:49.09 | anthm | group=4,20,100 |
17:49.11 | anthm | channel => 4 |
17:49.27 | anthm | g1-4 is indivdual |
17:49.32 | anthm | g10 us 1 and 2 |
17:49.36 | anthm | g 100 is all 4 |
17:49.43 | file | anthm: encore, ENCORE! |
17:49.47 | anthm | g20 is tge top 2 |
17:49.55 | yaaar | does anybody have a sample of a cisco 79xx sip.conf i can check out? sort of grasping straws here... |
17:50.15 | jeremywhiting | ManxPower: yeah, I have, just never read about digium cards and such cause we weren't planning to use them |
17:50.37 | jeremywhiting | but Junction Networks doesn't sound very good for some reason when we call, so we're trying this instead |
17:53.39 | jeremywhiting | is there a way to have asterisk cli initiate a call? |
17:54.13 | jeremywhiting | for debugging purposes only of course |
17:54.33 | file | app_changrab allows you to originate a call from the CLI if you wish, it's available at http://www.pbxfreeware.org/ |
17:54.33 | anthm | app_changrab available at a pbxfreeware.org near you |
17:54.35 | MikeJ[Laptop] | jeremywhiting, yes, app_changrab from pbxfreeware.com |
17:54.36 | MikeJ[Laptop] | hehe |
17:54.42 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
17:54.43 | MikeJ[Laptop] | echo. |
17:54.44 | file | dejavu |
17:54.53 | ManxPower | jeremywhiting: only if you have a soundcard set up on the server, and the developement headers for OSS or ALSA and you built Asterisk with those headers avainable. |
17:55.06 | ManxPower | jeremywhiting: or you could have looked it up on the Wiki: console phone |
17:55.23 | *** join/#asterisk salvini_fs (~felipesal@200165194043.user.veloxzone.com.br) |
17:55.38 | ManxPower | file: How do you run app_changrab? Just run "changrab" from the console? |
17:56.03 | MikeJ[Laptop] | for originate, you type originate, a channel, and a channel |
17:56.12 | file | ^^^ |
17:56.30 | los415 | u can always drop a call file in the outgoing folder |
17:56.39 | MikeJ[Laptop] | ie originate sip/blah local/1000@default |
17:56.41 | anthm | Usage: changrab [-b] <channel> <exten>@<context> [pri] |
17:56.42 | MikeJ[Laptop] | err |
17:56.44 | ManxPower | I wasn't aware apps could register console commands. I thought only res_, etc could. |
17:56.47 | MikeJ[Laptop] | y |
17:56.52 | Juggie | hrmf, cvs wont compile for me |
17:57.05 | Juggie | could be a cvs update issue, cleaning all source and redownloading |
17:57.11 | MikeJ[Laptop] | ManxPower, there is no diff between an app and a res.... or a chan for that matter, |
17:57.18 | file | just different naming convention |
17:57.24 | outtolunc | i just compiled about 20 minutes ago.. just a few issues but compiled |
17:57.24 | ManxPower | Juggie: I'll bet 1.0.9 will compile for you. |
17:57.25 | anthm | Usage: originate <channel> <exten>@<context> [pri] [callerid] [account] [timeout] |
17:57.26 | MikeJ[Laptop] | they can do whatever they want, just where you put em and what you call em |
17:57.43 | Juggie | ManxPower, probally, whats your point? |
17:57.46 | file | you can have an app_doitall that does a channel, exports functions everywhere, app, whateva |
17:58.25 | anthm | so originate Zap/g1/18005551212 1000@mycontext |
17:58.29 | MikeJ[Laptop] | you could have a MikeJ_ismyfavorite.c that does the same thing |
17:58.36 | anthm | to call someone and make them go to exten 1000 |
17:58.57 | cpatry | any chance to see it in head? |
17:59.00 | Juggie | yah its fine now, looks like cvs update just mucked something up. |
17:59.46 | anthm | well if you look at chan_sip and chan_zap it has cli commands in it too |
18:00.07 | anthm | cli commands are a way to expose the scope of your module to the outside |
18:00.26 | ManxPower | cpatry: if it's written by anthm it will never be in the standard Asterisk as far as I can tell. Is that correct, anthm? |
18:00.30 | MikeJ[Laptop] | and if it you don't like it.. rename it :D |
18:00.40 | *** join/#asterisk killerwhale (~killerwha@gocastlerock.com) |
18:00.50 | cpatry | ManxPower: wtf are ya talking about? |
18:00.57 | MikeJ[Laptop] | ManxPower, you like rubbing people the wrong way don' |
18:00.57 | anthm | hmm so things written by anthm never become std parts of asterisk ? |
18:00.59 | MikeJ[Laptop] | t you? |
18:01.00 | cpatry | there's many major contributions already in head by him. |
18:01.14 | MikeJ[Laptop] | isn't there a website of those? |
18:01.16 | bdunn | I have a two SIP phones - one at the office and one at the home office. When someone dials my extension, is it possible to have them both ring at the same time? |
18:01.17 | anthm | *shrug* |
18:01.45 | ManxPower | cpatry: In order for code to be included in Asterisk you MUST either give an unrestricted license to Digium OR you must assign them copyright to your code. Many people refuse to do that, so their work will never be part of Asterisk. |
18:02.05 | ManxPower | As I understand it, anthm does not disclaim any of these cool apps. |
18:02.18 | cpatry | ManxPower: take a look at http://www.cluecon.com/anthm.html |
18:02.37 | cpatry | what about chanspy? |
18:02.51 | cpatry | u really should take a look at that url manx. |
18:02.56 | mogorman | but everything lately has not been disclaimed |
18:02.59 | ManxPower | cpatry: that stuff happened a long time ago. |
18:03.03 | mogorman | and some taken out |
18:03.04 | ManxPower | I'm still waiting for anthm's response. |
18:03.17 | mogorman | but he has the choice to do that |
18:03.29 | ManxPower | anthm: will you disclaim app_changrab? |
18:03.41 | bdunn | Can anyone tell me how to make two extensions ring at the same time - simulring or whatever? |
18:03.48 | file | bdunn: & |
18:03.55 | file | bdunn: Dial(SIP/jcolp&IAX2/jcolp) |
18:03.59 | Juggie | he doesnt have to disclaim whats not in the tree. |
18:03.59 | file | will ring both at the same time |
18:04.08 | MikeJ[Laptop] | ManxPower, chanspy and app_dictate were both quite recent |
18:04.11 | *** join/#asterisk pabelanger (~a@67.71.252.98) |
18:04.14 | bdunn | file, Wow... that easy... thanks. |
18:04.16 | ManxPower | file: Oh, I repsect his refusal to disclaim some of his work. |
18:04.41 | ManxPower | I'm just saying that if he doesn't disclaim something it will never be in the Asterisk tree. |
18:04.51 | ManxPower | BTW, I thought bkw_ wrote chan_spy. |
18:05.00 | anthm | hard to say alot of the stuff I don't disclaim has already been refused |
18:05.02 | Juggie | i dont think he cares, thats why theres pbxfreeware.org |
18:05.24 | anthm | or i dont have the patience to modify it 1000 times |
18:05.29 | anthm | to make them happy |
18:05.38 | ManxPower | anthm: The second issue it prolly the more significant one. 8-) |
18:05.42 | killerwhale | hey all - Looks like I'm getting rid of TDMs and moving to Wildcard TE110P. Where can I information on how to do the following.... |
18:05.43 | killerwhale | 12 to data while passing all traffic through to the Asterisk Open Source PBX, which reliably routes the channels to their designated locations. This eliminates the need for an external router. |
18:05.57 | mogorman | asterisk has to have solid code |
18:06.01 | mogorman | and structure |
18:06.01 | killerwhale | I want to read-up when I get it. |
18:06.11 | mogorman | its a pbx |
18:06.14 | ManxPower | mogorman: Have you looked at the Asterisk souce code? |
18:06.20 | mogorman | yeah |
18:06.26 | Juggie | mogorman, look at app_voicemail |
18:06.27 | mogorman | a good bit |
18:06.35 | mogorman | lol that is the red headed step child |
18:06.40 | mogorman | that many have tried to fix |
18:06.48 | mogorman | and yet it grows bigger everyday |
18:07.31 | anthm | it's an interesting debate, should asterisk be designed to have loadable addons then insist on owning them all ? |
18:07.53 | anthm | does apache care if they own them all? |
18:08.25 | anthm | i'll gladly work on making the core stay the same long wnough to not break any modules |
18:08.44 | anthm | it's kind of a catch-22 no? |
18:08.49 | ManxPower | Asterisk has some significant issues, but it does seem to work amazingly well. |
18:08.58 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
18:09.24 | ManxPower | And with mark never being around to steer things in the right direction..... |
18:09.53 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
18:10.07 | ManxPower | as far as I can tell kpflemming is now steering Asterisk developement, but I'm sure that's not really the case. |
18:10.24 | anthm | precicely why we host a weekly call and are having a conference |
18:10.38 | ManxPower | anthm: is mark usually on those calls? |
18:10.42 | anthm | yep |
18:10.46 | ManxPower | nifty. |
18:10.49 | anthm | when he's in town |
18:10.52 | JerJer | stole the weekly conference call |
18:10.57 | mogorman | but work ever gets done on them? |
18:10.57 | ManxPower | Dude, he's never in town. |
18:11.12 | JerJer | he could call in, if properly motivated |
18:11.27 | ManxPower | JerJer: Exactly. |
18:11.29 | MikeJ[Laptop] | ManxPower, what's up with your attitude.. |
18:11.33 | anthm | anyone who wants to present an issue should attend cos him and kevin are there alot |
18:11.57 | MikeJ[Laptop] | if you are so anoyed about it, do somthing to change it |
18:12.02 | Katty | anthm: /he/ and kevin /is/ |
18:12.13 | MikeJ[Laptop] | I am soo sick of the #asterisk bitch and no action game |
18:12.18 | ManxPower | MikeJ[Laptop]: Perhaps low blood sugar, perhaps I'm just in a bad mood. |
18:12.31 | MikeJ[Laptop] | it's not just you. |
18:12.35 | anthm | Katty, irc is a grammar free zone |
18:12.39 | ManxPower | Perhaps I'm annoyed that 1.nextversion is taking so long. |
18:12.40 | Katty | anthm: ... |
18:12.45 | Katty | anthm: k |
18:12.49 | Hmmhesays | grammer/spelling |
18:12.50 | ManxPower | or more specifically the feature freeze, at least. |
18:12.59 | Juggie | ManxPower, i have a bet with drumkilla, 1.2 by july 31st |
18:13.01 | Katty | Hmmhesays: i'll grammer your spelling in a minute |
18:13.02 | MikeJ[Laptop] | ManxPower, well perhaps if you worked towards it actually happening it would sooner |
18:13.05 | Juggie | but its looking tougher and tougher to make it |
18:13.19 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
18:13.24 | ManxPower | OR perhaps it just seemed like a lot of this sillyness didn't happen when it seemed that mark was participating more. |
18:13.25 | Juggie | theres a case if redbull on the line :) |
18:13.41 | anthm | that's cute misspell grammar to tease us =p |
18:13.43 | MikeJ[Laptop] | ok, so theres your answer.. bitch, don't do |
18:14.05 | ManxPower | MikeJ[Laptop]: What do you suggest that does not require me converting my production systems to CVS-HEAD? |
18:14.09 | MikeJ[Laptop] | guidelines for 1.2 have been set, and are in process of happening |
18:14.14 | mogorman | be... |
18:14.18 | ManxPower | MikeJ[Laptop]: URL? |
18:14.22 | mogorman | ^_^ |
18:14.26 | MikeJ[Laptop] | URL what? |
18:14.34 | ManxPower | to the set guidelines? |
18:14.34 | MikeJ[Laptop] | I suggest you fix bugs. |
18:14.44 | MikeJ[Laptop] | there is no URL... |
18:14.58 | ManxPower | email message? Anything? |
18:15.01 | ManxPower | I'm curious. |
18:15.19 | ManxPower | MikeJ[Laptop]: You know I'm not a programmer. |
18:15.28 | MikeJ[Laptop] | the one from kevin in respose to you |
18:15.48 | MikeJ[Laptop] | the fact that there has been extended discussion both onthe dev list some time ago and on the dev calls |
18:16.05 | ManxPower | Ah, the goal of a feature freeze on x date and my suggestion that it be announced 30 days before the freeze happens? |
18:16.37 | ManxPower | I'm not sure I would call that "guidelines", but OK. |
18:16.52 | MikeJ[Laptop] | no one has ever said there would be a feature freeze |
18:17.03 | MikeJ[Laptop] | and there likely won't |
18:17.05 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca) |
18:17.09 | SarahEmm | r |
18:17.15 | SarahEmm | err |
18:17.17 | SarahEmm | hihi |
18:17.28 | anthm | everyone who wants stuff should attend the meeting where they can speak in person to address the entire developer base |
18:17.42 | anthm | its every thu |
18:17.54 | MikeJ[Laptop] | there will be a holdoff on invasive features as we fix as many non feature bugs as possible, then a RC branch, then 1.2 |
18:18.01 | MikeJ[Laptop] | I have told you this over and over again... |
18:18.12 | anthm | the most likely would welcome the imput |
18:18.12 | MikeJ[Laptop] | you seem to need it in writing |
18:18.33 | MikeJ[Laptop] | but for god sakes stop bitching about it.. |
18:18.42 | MikeJ[Laptop] | or do somthing... |
18:18.43 | anthm | cos the list and this irc is so overwhelming they probably dont even realize some of the stuff ppl bitch about |
18:19.00 | anthm | so the best policy is to get in line and bitch out loud |
18:20.38 | ManxPower | I just see Asterisk going the way of phpGroupware and that would REALLY suck. |
18:20.57 | MikeJ[Laptop] | then help |
18:21.09 | ManxPower | At least Asterisk doesn't have a twice yearly loss of source code 8-) |
18:21.16 | MikeJ[Laptop] | bug marshalls are always welcome |
18:21.25 | ManxPower | MikeJ[Laptop]: I simply believe that the problem is NOT a technical one. |
18:21.36 | MikeJ[Laptop] | like I said.. do somthing |
18:21.54 | ManxPower | MikeJ[Laptop]: I guess I could write up release guidelines. |
18:22.11 | ManxPower | But isn't that something the primary sponsor of asterisk should be doing? |
18:22.13 | MikeJ[Laptop] | you could do many things.. but youare too busy complaing to do anything at the momnet |
18:22.18 | cypromis | that is not enou8gh, you would have to enforce that spomeone reads them as well |
18:22.34 | ManxPower | cypromis: and follow them. Exactly. Only Digium can do that. |
18:22.38 | MikeJ[Laptop] | propose it, whatever |
18:22.40 | znoG | MikeJ is right, I don't whinge simply because I don't think I have much to offer back to the Asterisk project, and hardly the time either. |
18:22.56 | ManxPower | I'll stop complaining then |
18:22.58 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
18:23.01 | MikeJ[Laptop] | thank you |
18:25.22 | *** join/#asterisk virterm (~virterm@shiva.kanatek.com) |
18:25.50 | ronn | hi guys ... i have sipuras which fail to re-register after 40-50 minutes .. any one had similar problems |
18:26.07 | bkw_ | sounds like a deadlock |
18:26.14 | bkw_ | do calls totally stop working? |
18:26.33 | ronn | deadlock? the sipuras wouldn't register at all .. they just fail to register |
18:26.41 | ronn | get registration failed message |
18:26.43 | bkw_ | does asterisk continue to function? |
18:26.51 | ronn | yes.. asterisk works just fine |
18:26.53 | bkw_ | if you restart asterisk does it stop? |
18:26.57 | bkw_ | or re-register? |
18:27.24 | ronn | i wouldn't restart asterisk as it carriers other traffic at the time |
18:27.35 | ronn | i haven't tried restarting asterisk |
18:27.46 | bkw_ | how often are you registering your sipura? |
18:27.52 | bkw_ | and are you behind nat? |
18:27.58 | ronn | 3600 .. the deafult |
18:28.26 | ronn | yes behind nat |
18:29.01 | *** join/#asterisk mrgoby (~mrgoby@aa.linuxbox.com) |
18:29.18 | *** join/#asterisk guugmember (~guugmembe@200.6.235.232) |
18:29.32 | guugmember | hello guys, any comments about this product http://www.broad-tel.com/index_en.php |
18:30.52 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
18:33.36 | ender | anybody know how to boost the mic level on Sipura phones? |
18:34.44 | MikeJ[Laptop] | ender,phone or ata? |
18:35.11 | MikeJ[Laptop] | on the ata it is on the admin\advanced pagees, somthing weird like international or sonthing.. and it says gain |
18:35.29 | MikeJ[Laptop] | I would guess the phone has similar settings? |
18:35.33 | pdugas | ender: if it's an SPA-841, return the unit if you can They completely blow. |
18:35.41 | ender | MikeJ[Laptop]: phone |
18:35.49 | guugmember | any one has tested PA - 100 series from Broad-tel? |
18:35.54 | MikeJ[Laptop] | I only have that guess |
18:35.57 | ender | pdugas: it is an 841 and it will be returned eventually |
18:36.24 | ender | pdugas: however a user is stuck using it right now and I need to boost their mic level. |
18:37.01 | pdugas | ender: I've got one customer who bouht a batch of 6 then upgraded to latest firmware. They're close to useless. |
18:37.38 | pdugas | ender: you may look for older firmware and downgrade. I have one on the shop that works okay but has never been flashed. |
18:37.59 | ender | pdugas: so is there no way to adjust it? |
18:38.42 | pdugas | ender: recent version of fw had some gain adjustments. didn't have much luck myself. Not in front of one so I can't point to the exact setting. |
18:39.27 | ender | pdugas: k |
18:39.42 | pdugas | ender: luck! |
18:40.37 | *** part/#asterisk brookshire (~pfffft@207.111.174.1) |
18:40.41 | *** join/#asterisk brookshire (~pfffft@207.111.174.1) |
18:41.28 | ender | pdugas: thanks, I actually found something. |
18:41.44 | *** part/#asterisk brookshire (~pfffft@207.111.174.1) |
18:41.49 | *** join/#asterisk brookshire (~pfffft@207.111.174.1) |
18:42.02 | brookshire | i love it when xchat messes up |
18:42.21 | twisted[work] | brookshire, haha |
18:42.51 | ender | ah sweeet irssi |
18:42.54 | brookshire | hah.. EPIC 4 life! |
18:42.59 | brookshire | i don't know why i used xchat |
18:43.01 | brookshire | today |
18:43.16 | twisted[work] | i use xchat every day |
18:43.35 | brookshire | wierdo |
18:43.36 | brookshire | :D |
18:43.38 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
18:43.40 | twisted[work] | hah |
18:43.45 | twisted[work] | works great |
18:43.56 | brookshire | works great at crashing |
18:44.01 | shido6 | sunuva... |
18:44.03 | twisted[work] | shido6, uhm, good luck with that |
18:44.07 | twisted[work] | seeing as i'm not on a laptop |
18:44.30 | *** join/#asterisk meppl (~mephisto@A7d23.a.pppool.de) |
18:44.31 | twisted[work] | brookshire, i don't have any of those crashing issues |
18:44.46 | twisted[work] | you must be a nub |
18:45.05 | brookshire | nah.. i blame xchat |
18:45.21 | twisted[work] | i blame your inability to use xchat :P |
18:45.31 | ender | is there a way to bump up the audio at asterisk level? |
18:45.53 | shido6 | bump up the audio, eh? |
18:45.58 | shido6 | zap card? |
18:46.04 | twisted[work] | pump up the volume |
18:46.07 | shido6 | heh |
18:46.08 | brookshire | pump it up |
18:46.18 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
18:46.19 | twisted[work] | and dance |
18:46.21 | SpaceBass | hey folks |
18:46.21 | twisted[work] | dance |
18:46.35 | brookshire | haha.. well there is the bass |
18:46.41 | anthm | shido6, did JerJer tell you to talk at cluecon instead of him? |
18:46.42 | twisted[work] | lol |
18:46.42 | brookshire | we just need some treble |
18:46.53 | SpaceBass | when I do dial(SIP/mybroadvoiceaccount, <a number>) I get circuits busy from BV |
18:47.03 | brookshire | tit-tit-tit-ta-tit-tit-tit-ta |
18:47.04 | twisted[work] | well gee |
18:47.08 | shido6 | ahhr yoooh ghonna mess with meE on dis? cauze awf maye accent? |
18:47.19 | twisted[work] | SpaceBass, your syntax is wrong |
18:47.44 | anthm | ahem, shido6, did JerJer tell you to talk at cluecon instead of him? |
18:48.06 | SpaceBass | twisted[work] figured that :) what am I missing... feel like its glaring but I dont see it... tried dial(sip/bv|||<number>) too... |
18:48.25 | SpaceBass | i dont want to set a timeout or options |
18:48.42 | shido6 | anthm, yes - aye wiw tahlk in maye ahnold voice |
18:48.53 | Micc | SpaceBass, did you set the callerid properly? |
18:48.58 | shido6 | abowt voip frawd |
18:49.02 | anthm | then why is he pissed off that he is not talking? |
18:49.17 | SpaceBass | Micc i figured that was a moot point b/c BV doesnt accept outbound CID |
18:49.27 | anthm | he aparently feels he was the one who wanted to give the speech |
18:49.49 | shido6 | he can give the speech - its not something thats going to cause any havoc here |
18:49.54 | shido6 | I had to reserve the spot |
18:49.56 | Micc | SpaceBass, right if you don't set the fromuser to your broadvoice number it will give you a 500 error. |
18:50.05 | twisted[work] | SpaceBass, then try using the proper syntax |
18:50.08 | twisted[work] | it's on the wiki |
18:50.09 | anthm | well now he's pissed you may want to go work it out with him |
18:50.09 | twisted[work] | very simple |
18:50.16 | twisted[work] | and it's also in extensions.conf.sample |
18:50.26 | SpaceBass | thanks... |
18:50.30 | twisted[work] | dial(sip/provider/number) |
18:50.43 | *** join/#asterisk criptos (~criptos@201.138.231.189) |
18:50.46 | twisted[work] | note the / -- not a comma, a space, or a pipe |
18:50.51 | ender | shido6: no, sip phone w/ a shitty mic |
18:50.53 | Micc | SpaceBass, broadvoice works sometimes and doesn't other times. |
18:50.58 | shido6 | erf? |
18:51.00 | criptos | How to disable the callwaitting at the iaxy? I dont want call waitting... |
18:51.00 | shido6 | oh |
18:51.12 | brookshire | criptos: in the provisioning |
18:51.18 | SpaceBass | twisted[work] thanks, thats what I was looking for... no stranger to dial strings... looking for my typo... thanks for point it out |
18:51.33 | Micc | SpaceBass, so even if you get it all setup right, it depends on who is holding the bubble gum and duct tape on broadvoice systems. |
18:51.35 | file | brookshire: MATTTTTTTTTTTTTT! |
18:51.40 | brookshire | file! |
18:51.42 | file | hi. |
18:51.47 | twisted[work] | oh my god beck |
18:51.50 | twisted[work] | look at those inodes |
18:51.51 | SpaceBass | Micc I think there is some pixi dust involved too |
18:51.54 | file | omg Becky! |
18:51.56 | twisted[work] | *becky |
18:52.01 | twisted[work] | they are so big |
18:52.06 | file | they're like coconuts |
18:52.19 | twisted[work] | inodes? like coconuts? |
18:52.23 | file | yessss |
18:52.28 | brookshire | dang.. i really need to get gtkiaxyprov out |
18:52.32 | brookshire | TODAY I SAY! |
18:52.33 | brookshire | :D |
18:52.44 | file | I need to get my accounting stuff done, but I don't think I will |
18:52.49 | file | not today atleast |
18:52.56 | twisted[work] | heh |
18:53.05 | Micc | I'm still having dtmf issues with cvs head. I think I had better get the stable build so I can rule that out. |
18:53.28 | file | I have a feeling the box will go kaboom-like |
18:54.33 | guugmember | any one has tested PA - 100 series from Broad-tel? |
18:55.27 | SpaceBass | yaaar what version did you upgrade from? |
18:55.39 | SpaceBass | yaaar I haven't been able to go from 6.4 to 7.4 for some reason |
18:55.52 | twisted[work] | brookshire, i'll believe it when I see it |
18:56.01 | brookshire | twisted.. lol |
18:56.13 | brookshire | well... i just need to put it in cvs |
18:56.20 | brookshire | and then write the documentation |
18:56.26 | twisted[work] | do you have cvs commit? |
18:56.52 | twisted[work] | if so, do it! |
18:58.00 | InfraRed | where should my asterisk -> provider login information be stored? |
18:58.05 | *** join/#asterisk drbrown (~chatzilla@65.121.240.182) |
18:58.09 | *** part/#asterisk brookshire (~pfffft@207.111.174.1) |
18:58.16 | *** join/#asterisk brookshire (~pfffft@207.111.174.1) |
18:58.30 | drbrown | is CVS HEAD broken??? |
18:58.33 | drumkilla[work] | brookshire: it's already in CVS |
18:58.41 | Katty | borken???!!!!11oneoneone |
18:58.43 | brookshire | drumkilla!!! |
18:58.45 | brookshire | is it? |
18:58.46 | brookshire | where? |
18:58.52 | *** join/#asterisk krazykrab (~Krab@203.81.238.65) |
18:59.05 | drbrown | won't complile, it seems to get stuck on version.h |
18:59.18 | krazykrab | help needed on Asterisk@home .. All circuit BUsy now |
18:59.21 | twisted[work] | drbrown, is this a fresh checkout? |
18:59.22 | drumkilla[work] | drbrown: make clean |
18:59.38 | drbrown | yes, 2 miniutes ago on 2 different systems |
18:59.47 | drbrown | stuck in the same spot |
19:00.35 | krazykrab | anyone can help ? |
19:01.04 | drbrown | http://pastebin.ca/17562 |
19:01.27 | drbrown | That is the spot where it gets stuck on the compile, just keeps repeating itself |
19:02.38 | Katty | do do do ...can't ping this |
19:02.49 | twisted[work] | lol |
19:03.04 | twisted[work] | damn, you're right! |
19:03.33 | krazykrab | All Circuit Busy!! |
19:03.47 | twisted[work] | krazykrab, chances are that's a telco message |
19:04.01 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
19:04.03 | drbrown | should I file a bug report, or am I the only one having this problem?? |
19:04.19 | brookshire | no.. just talk with drumkilla |
19:04.23 | twisted[work] | lol |
19:04.24 | brookshire | complain to him, lol |
19:04.36 | drbrown | k |
19:04.40 | brookshire | i was just joking |
19:04.41 | brookshire | i don't know |
19:04.47 | brookshire | JOSH KNOWS! |
19:04.54 | twisted[work] | i do not |
19:04.57 | brookshire | do too! |
19:05.04 | file | I don't either |
19:05.05 | brookshire | he is the * of * |
19:05.10 | twisted[work] | no |
19:05.21 | file | the Josh club does not know. |
19:05.30 | M_at | Is the first * because you can't legally put the word there? |
19:05.34 | brookshire | well he knows, he just won't help you |
19:05.37 | Katty | does the peanut gallery know? |
19:05.42 | bkw_ | drumkilla you did get what I ment about adding that new cvs module to the cvsup server at digium |
19:05.47 | twisted[work] | let me check |
19:05.51 | bkw_ | it will get picked up by the mirror |
19:05.58 | krazykrab | twisted- this circuit busy thing coming again and again i this i have missed somthing |
19:06.03 | twisted[work] | nope... the peanut gallery (aka, my co workers) does not know |
19:06.09 | Katty | twisted[work]: k |
19:06.15 | drumkilla[work] | bkw_: no, we couldn't figure it out, haha |
19:06.23 | brookshire | yay! gtkiaxyprov |
19:06.25 | bkw_ | its in /etc/cvsup/sup |
19:06.26 | bkw_ | I think |
19:06.46 | drbrown | wonder why I can't compile |
19:06.51 | bkw_ | its a directory with per module with some files in it |
19:07.10 | bkw_ | go to the all one |
19:07.31 | twisted[work] | bkw_, it's /usr/local/etc/cvsup |
19:07.32 | bkw_ | add it to list.cvs |
19:07.39 | bkw_ | not the binary |
19:07.42 | bkw_ | the sup files |
19:07.55 | bkw_ | it might be in /usr/local/etc/cvsup/sup |
19:07.58 | bkw_ | not sure how he did that |
19:08.02 | twisted[work] | yeah |
19:08.05 | twisted[work] | that's wher eit is |
19:08.07 | bkw_ | but the all directory has a list.cvs |
19:08.11 | bkw_ | add it to the list |
19:08.15 | bkw_ | and the mirror will pick it up |
19:08.23 | bkw_ | note i say "the" mirror |
19:08.24 | bkw_ | the one and only |
19:08.44 | drumkilla[work] | bkw_: want a cookie? :p |
19:08.50 | bkw_ | look like krazykrab didn't get his help in time |
19:09.02 | Katty | aww |
19:09.02 | bkw_ | I can't do cookies |
19:09.03 | Katty | sniffle |
19:09.05 | bkw_ | cookies bad |
19:09.09 | Katty | poor wittle krab |
19:09.39 | file | my parents are on barbecuing spree |
19:09.43 | file | er on a |
19:09.48 | Katty | gosh |
19:09.59 | M_at | What are they bbqing? |
19:10.03 | file | burgers |
19:10.07 | file | the day before that it was hot dogs |
19:10.11 | Katty | veggie burgers? :> |
19:10.12 | denon | stir fry kitty cat |
19:10.24 | denon | haha, sorry kajtzu |
19:10.26 | denon | er Katty |
19:10.50 | drumkilla[work] | bkw_: i forwarded the info to malcolm, so we'll see |
19:10.50 | Katty | k, all better |
19:11.17 | *** join/#asterisk MicC_ (~sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com) |
19:11.37 | MicC_ | is there a newb channel? |
19:11.54 | anthm | #snipe |
19:11.54 | Katty | probably lots |
19:12.01 | twisted[work] | d'oh |
19:12.24 | MicC_ | just had a couple of SIP related questions |
19:12.48 | file | then ask 'em |
19:13.04 | MicC_ | thats what was hoping for |
19:13.16 | MicC_ | what is a good (free as in beer) IP softphone. |
19:13.21 | file | for what OS? |
19:13.27 | M_at | X-ten's X-Lite works for me |
19:13.57 | MicC_ | Winblows |
19:14.15 | file | X-Lite |
19:14.19 | MicC_ | thanks |
19:14.23 | bjohnson | I get "Maximum retries exceeded on call " but nothing shows up in show channels or anything. I've even restarted asterisk. What the heck could it be? |
19:14.25 | twisted[work] | free as in beer? |
19:14.28 | twisted[work] | beer is not free! |
19:14.36 | file | twisted[work]: poisoned beer is |
19:14.41 | Katty | boo, beer |
19:14.45 | Katty | oh |
19:14.45 | MicC_ | next question: I would like a couple of phone #s what is a good SIP provider that is relatively inexpensive? |
19:14.46 | twisted[work] | file, hushua |
19:14.47 | Katty | i was going to go do something |
19:14.55 | Katty | stop distracting me, silly irc |
19:15.03 | citats | mmmm beer |
19:15.14 | M_at | MicC_ : Are you using Asterisk? |
19:15.22 | MicC_ | yeap yeap |
19:15.23 | MicC_ | :P |
19:15.24 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net) |
19:15.30 | MicC_ | got it and AMP installed today |
19:15.32 | criptos | How Can I know what firmware version a iaxy is using? |
19:15.43 | MicC_ | AMP anygood btw? |
19:15.50 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
19:15.50 | MicC_ | I am a commandline man myself |
19:16.01 | M_at | You behind NAT? |
19:16.12 | MicC_ | yes...I can do 1 to 1 if I have to |
19:16.24 | *** join/#asterisk meppl (~mephisto@A7d69.a.pppool.de) |
19:16.28 | M_at | You may prefer to look for IAX termination then - works better than SIP where there is NAT |
19:16.36 | Hmmhesays | ugh, why do people use telephone dialers anymore |
19:16.51 | MicC_ | Hmmhesays: proof of concept |
19:16.52 | MicC_ | :P |
19:16.55 | *** join/#asterisk RedHatUser (~asaa@baydeinter-27-83.baydenet.com.br) |
19:16.57 | Micc | Micc_, are you mocking me? ;) |
19:17.07 | Malthus | lol |
19:17.09 | MicC_ | ug |
19:17.16 | Malthus | what's a telephone dialer |
19:17.16 | MicC_ | that a coincidence? |
19:17.38 | Hmmhesays | I'm thinking just to piss me off |
19:17.53 | Micc | I normally go by Mic but that is usually taken so I go by Micc. |
19:17.54 | MicC_ | M_at: ok...you know what my next question is...what is a good .... |
19:18.13 | MicC_ | I usually go by MicC but it was taken |
19:18.23 | Micc | yeah, hahaha, by me. |
19:18.38 | M_at | www.iaxprovider.net is a good starting point |
19:18.41 | Umaro | Hey guys, anyone worked with * on a SGI box, or signate's enterprise PBX thing? |
19:18.42 | Micc | Now I'm gonna have to make sure I take this name on all the big sites. :D |
19:18.46 | MicC_ | this is the first time this has happend |
19:18.59 | Umaro | supposedly it's open source, but they don't have a source release anywhere, of course |
19:19.03 | MicC_ | Micc: too late |
19:19.04 | MicC_ | :P |
19:19.08 | Micc | Micc_, and we ended up in the same channel. strange. |
19:19.14 | MicC_ | weird |
19:19.18 | MicC_ | you Irish? |
19:19.22 | brookshire | umaro?? |
19:19.30 | Micc | Nope, german and polish I think. |
19:19.37 | MicC_ | lol |
19:19.39 | Micc | mmm.. sounds like yummy saussage. |
19:19.45 | MicC_ | Mic is slang for Irish |
19:19.51 | MicC_ | Mac is scottish |
19:20.00 | Micc | Mine is short for Michael. |
19:20.01 | MicC_ | pikey=pikt...etc. |
19:20.06 | MicC_ | ah |
19:20.30 | MicC_ | M_at: Thanks for the help...one last question. What is a good IAX provider? |
19:20.32 | brookshire | we had it working on an altex here |
19:20.33 | Micc | Its a nickname. Started on my 12th birthday when I received a 2400 baud modem and signed up on my first bbs and couldn't yet type. |
19:20.43 | MicC_ | how old are you? |
19:20.57 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
19:20.58 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
19:21.03 | Micc | Its been with me for almost 16 years. As I am almost 28 now. |
19:21.20 | Umaro | brookshire: oh yeah? cool. |
19:21.21 | drumkilla[work] | there was a 'drumzilla' in here once |
19:21.26 | twisted[work] | yeah |
19:21.28 | twisted[work] | you scared him off |
19:21.32 | drumkilla[work] | :) |
19:21.34 | Malthus | lol |
19:21.55 | Umaro | brookshire: how was the performance? trying to figure out how much of signate's "enterprise pbx" is them actually making performance modifications to asterisk, and how much of it is that they just use SGI boxes |
19:22.33 | drumkilla[work] | twisted[work]: do you know if malcolm has to kill cvsup for the changes to take effect? |
19:22.43 | brookshire | umaro: i doubt they make zero changes to asterisk |
19:22.56 | drumkilla[work] | no, you doubt they make *any* changes |
19:23.05 | brookshire | yes |
19:23.11 | brookshire | i doubt they make *any* changes |
19:23.18 | brookshire | in fact i'm pretty sure of it |
19:23.25 | drumkilla[work] | nubs ... |
19:23.32 | brookshire | :D |
19:24.36 | Micc | Micc_, NuFone is a great IAX provider. |
19:24.42 | bdunn | I can't seem to work this one out... I have this in Extensions.conf, and I need it to ALSO ring ext 3004 when 3001 rings. Please help if you can. Thanks! |
19:24.43 | bdunn | exten => 3001,1,Macro(stdexten,3001,${BDUNN}0) |
19:24.43 | bdunn | exten => bdunn,1,Goto(3001|1) |
19:24.51 | MicC_ | MicC: HA! 30 years old used it on an Atari 800 BBS at 1200 baud |
19:24.57 | MicC_ | ph33r my l33t3ss |
19:25.23 | drumkilla[work] | fight!!!@!12! |
19:26.10 | M_at | Put it away |
19:26.20 | file[laptop] | noooooo it's so cheap |
19:26.59 | Micc | Micc_, you've got me beat there. Where are you from? |
19:27.30 | MicC_ | Saint-John, NB CANADA |
19:27.31 | MicC_ | :P |
19:27.38 | brookshire | oh no! |
19:27.41 | MicC_ | not exactly a technopolis |
19:27.46 | brookshire | two micc's |
19:27.54 | file[laptop] | Saint John? |
19:28.00 | file[laptop] | MicC_: I'm in Riverview |
19:28.07 | MicC_ | oh jesus |
19:28.13 | M_at | He wont save you |
19:28.19 | sivana | friggin beauty |
19:28.23 | file[laptop] | Dieppe is french |
19:28.35 | MicC_ | My dad lives in Moncton |
19:28.46 | MicC_ | do you know Darcy or Logan Cunning? |
19:28.48 | MicC_ | haha |
19:28.50 | M_at | Oh course Dieppe is french - it's in france |
19:28.53 | file[laptop] | can't say I do |
19:29.01 | MicC_ | lol...just checking. |
19:29.14 | SarahEmm | i love when people find out i'm from toronto canada. 'oh, do you know X? he lives in montreal!' |
19:29.30 | sivana | SarahEmm: ya, but do you know Jim Blumsen? |
19:29.45 | denon | SarahEmm: hey, I know someone from quebec, do you know her?! |
19:29.47 | denon | :) |
19:29.51 | sivana | he visited toronto once |
19:30.18 | SarahEmm | lol |
19:31.38 | MicC_ | ah |
19:31.49 | MicC_ | nufone only offers DID in michigan |
19:31.52 | MicC_ | :((((( |
19:32.34 | bdunn | I can't seem to work this one out... I have this in Extensions.conf, and I need it to ALSO ring ext 3004 when 3001 rings. Please help if you can. Thanks! |
19:32.36 | bdunn | exten => 3001,1,Macro(stdexten,3001,${BDUNN}0) |
19:32.37 | bdunn | exten => bdunn,1,Goto(3001|1) |
19:32.59 | bdunn | Seems like this should be so easy but I just can't figure it out. |
19:33.35 | *** join/#asterisk SteveL (~Steve@smtp.burlesonisd.net) |
19:33.54 | *** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr) |
19:33.56 | jhiver | hi all |
19:34.55 | *** join/#asterisk [Gator] (~qe2315us@66-118-23-3.texlinkcom.com) |
19:35.13 | jhiver | well, the chan looks pretty calm for now |
19:35.18 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net) |
19:35.23 | anthm | oh yeah? |
19:35.30 | anthm | emacs rocks! |
19:35.37 | denon | anthm: linux sucks! |
19:35.43 | jhiver | lol |
19:35.57 | anthm | solaris bay-bee |
19:36.07 | denon | anthm: ok, fine .. how about "vi kicks emacs' butt!" |
19:36.10 | jhiver | asterisk blows! (your mind, of course) |
19:36.20 | denon | solaris? oh come now .. even sun has given up on solaris |
19:36.28 | brookshire | hehe.. |
19:36.29 | anthm | dem fighting words |
19:36.32 | brookshire | asterisk blows (you) |
19:36.34 | brookshire | :D |
19:36.44 | jhiver | :) |
19:36.45 | drumkilla[work] | if only ... |
19:36.45 | anthm | emacs can pretend to be vi! |
19:36.51 | drumkilla[work] | we'd all be so set |
19:36.55 | jhiver | anthm: except slower |
19:36.57 | brookshire | haha.. russell would definately be a lucky man |
19:37.07 | Malthus | vi sucks |
19:37.08 | file[laptop] | Russell Wussell |
19:37.11 | Malthus | vim rocks |
19:37.13 | anthm | ESC-x ... vi-mode |
19:37.18 | jhiver | yeah, vim is good |
19:37.26 | file[laptop] | nooo ESC-x ... perl-mode |
19:37.31 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
19:37.45 | jhiver | I used to use emacs a hellofalot actually |
19:37.55 | jhiver | it was a bad habit from uni :) |
19:37.59 | jhiver | now I use only vim |
19:38.07 | jhiver | for a while I was using both :) |
19:38.14 | denon | I just use echo >> and never make mistakes |
19:38.21 | anthm | bah |
19:38.35 | eKo1 | sed for editing, echo for appending |
19:38.42 | denon | editing? wussat? |
19:38.55 | jhiver | yeah, emacs is good for heavy coding |
19:39.01 | anthm | if you use any of the features I added to asterisk you are using emacsified code mwa ha ha ha |
19:39.02 | jhiver | especially with the speedbar |
19:39.16 | bjohnson | vacuums suck |
19:39.18 | SpaceBass | eKo1 why not just ed |
19:39.20 | SpaceBass | :) |
19:39.26 | jhiver | but for config files and remote file editing, vim is just so cool |
19:39.41 | bjohnson | kate is great |
19:39.46 | Malthus | remote file editing? |
19:39.47 | brookshire | jove! |
19:39.50 | jhiver | it looks nice yeah |
19:39.51 | sivana | which Manager Proxy is the best? |
19:39.53 | brookshire | all your editors suck! |
19:39.55 | brookshire | MUAHAHAHAHAH |
19:39.57 | bjohnson | sftp + kate |
19:39.57 | anthm | see if ed can edit 4 files at once and let you edit the same file in 2 places at the same time =] |
19:39.58 | eKo1 | [master@dubraska ~]$ emacs |
19:39.58 | eKo1 | bash: emacs: command not found |
19:40.02 | jhiver | Malthus: like editing a file over ssh |
19:40.07 | bjohnson | all your editors are belong to us! |
19:40.09 | eKo1 | mwuahhahaha |
19:40.21 | file[laptop] | what a fight! |
19:40.27 | anthm | maybe cluecon should have code races! |
19:40.28 | brookshire | someone sip us up the bomb |
19:40.33 | MikeJ[Laptop] | heh |
19:40.40 | anthm | make and app that can ... blah ... GO! |
19:40.46 | jhiver | brookshire: LOL |
19:40.53 | Malthus | jhiver: you mean running vim on the local machine and editing the remote file, or running vim on te remote machine? |
19:41.05 | jhiver | Malthus: running vim on remote |
19:41.12 | Malthus | oh |
19:41.13 | eKo1 | load app_emacs.so |
19:41.20 | eKo1 | weeeee |
19:41.30 | Malthus | I thought there was a vim feature I didn't know about! |
19:41.57 | anthm | or how bout setup asterisk contest |
19:42.16 | *** part/#asterisk RedHatUser (~asaa@baydeinter-27-83.baydenet.com.br) |
19:42.19 | anthm | empty boxes with on os all the way to make a call to some sip addr to see who wins |
19:42.21 | eKo1 | the vi extensions in bash are much more powerful than emacs in my opinion. |
19:42.22 | cpatry | contest? who have the sexiest voice around ?:P |
19:42.37 | twisted[work] | me! |
19:42.43 | *** join/#asterisk rjreb (~rjreb@greatwall.amer.net) |
19:42.46 | anthm | can it play video games ? |
19:42.48 | anthm | and talk |
19:42.52 | anthm | and render images ? |
19:43.01 | anthm | send email |
19:43.08 | anthm | connect to aim and irc? |
19:43.10 | jhiver | eKo1: why would you want vi extensions on bash when there's perl? |
19:43.21 | anthm | yay for perl! |
19:43.30 | twisted[work] | anthm, an editor does NOT need to do all that shit |
19:43.37 | twisted[work] | but yes, VIM can indeed do most of that. |
19:43.42 | anthm | yes it does! |
19:44.04 | anthm | it's on page 1460 of the big book of what every editor should do |
19:44.08 | twisted[work] | lol |
19:44.10 | Silik0n | cpatry I do |
19:44.46 | cpatry | no, its when brian laugh! hehehe |
19:44.57 | anthm | which was drafted with emacs btw! |
19:45.12 | *** join/#asterisk reddawn (~reddawn@69.238.222.176) |
19:45.18 | jhiver | Silik0n: maybe you should do some asterisk sound files to prove it :) |
19:46.06 | jhiver | I'd love a suave: "Comedian Mail, would you like to listen to your messages, master" kind of voice ;-) |
19:46.08 | eKo1 | jhiver: uhm, 'set mode perl ' doesn't exist. |
19:46.12 | twisted[work] | you don't want that. |
19:46.13 | twisted[work] | no |
19:46.13 | twisted[work] | no |
19:46.18 | twisted[work] | bad touch |
19:46.51 | jhiver | what don't you want? |
19:47.32 | jhiver | eKo1: what's 'set mode perl' supposed to do? |
19:48.04 | anthm | i'd guess it would make it render perl code all pretty |
19:48.04 | eKo1 | Nothing, since it doesn't exists. |
19:48.28 | jhiver | lol |
19:48.34 | Corydon-w | But you can do :set syn=perl |
19:48.40 | jhiver | what would you want it to do? |
19:48.46 | jhiver | (if it existed) |
19:49.11 | Corydon-w | which changes the interpretation of the file, in terms of syntax color highlighting, to that of Perl |
19:49.14 | anthm | allow you to be able to enter any key on the keyboard mutiple ways? |
19:49.39 | jhiver | me I'd like 'set mode perl' to slap vim user if there is no 'use strict' and 'use warnings' statements at the top of the file :=) |
19:50.16 | *** join/#asterisk mrtwister (~user@cable-1-32.cgates.lt) |
19:50.23 | jhiver | like: "set mode perl" <enter> *SMACK* USE STRICT! moron! |
19:50.27 | *** join/#asterisk FaxTerix (~administr@lineaAK59.velocom.com.ar) |
19:50.28 | anthm | use YourOwnJudgement; |
19:50.33 | FaxTerix | Good days |
19:50.54 | jhiver | yo |
19:51.17 | jhiver | anthm: nonono |
19:51.25 | jhiver | use strict, use strict, use strict |
19:51.37 | Umaro | I'm going to have to agree with jhiver on this one |
19:52.03 | jhiver | anything else is pure and utter rubbish, and use strict; should be perl default behavior for fucks' sake |
19:52.35 | jhiver | I can't believe the amount of time i've lost because of no 'use strict' code |
19:53.07 | anthm | bah |
19:53.16 | Umaro | What's the most sane way for me to do silence detection in *? |
19:53.35 | Umaro | I've been trying to use BackgroundDetect, but it just takes an insane amount of resources |
19:54.45 | anthm | use Crutch; ? |
19:55.22 | jhiver | I didn't know this backgrounddetect app |
19:55.28 | jhiver | interesting! |
19:55.38 | FaxTerix | Does anybody have used TxFax? Because I'm having a problem and I can't figure out how to resolve it. |
19:56.04 | jhiver | out of curiosity, what are you doing that requires such detection? |
19:57.17 | Umaro | jhiver: answering machine detection. And before you get red in the face, this is only for calling existing customers to confirm appointments |
19:58.17 | jhiver | ? |
19:58.30 | jhiver | I'm not that judgmental :) |
19:58.41 | Umaro | jhiver: sorry, a lot of people are, I've found.. lol |
19:59.25 | jhiver | :) |
20:00.13 | eKo1 | funny, i've never encountered voip spam. |
20:00.17 | Silik0n | damned ricers |
20:00.20 | eKo1 | but then again, i don't live in north america. |
20:00.27 | jhiver | it's because it's not mainstream enough |
20:00.37 | mutilator | i don't get voip spam per say |
20:00.38 | jhiver | I guess this ought to happen with skype at some point |
20:00.47 | MicC_ | thanks for your help guys. |
20:00.48 | MicC_ | I am out |
20:00.51 | mutilator | but i do have 2 phone numbers i use when signing up for junk |
20:00.52 | jhiver | cya |
20:01.07 | mutilator | just never use them unless i need to activate something |
20:01.08 | gambolputty | how would any voip spam problem be solved? |
20:01.09 | tomtom_ | anyone has experience with channel grouping in zapata.conf? when i register to two groups (eg: group = 1,0) it doesn't seem to work ... |
20:01.20 | *** join/#asterisk So3kris (~jan-wille@soekris.xs4all.nl) |
20:01.28 | So3kris | hello |
20:01.43 | *** join/#asterisk Bicster (~Bicster@Bicster.active.supporter.pdpc) |
20:02.10 | So3kris | haha nice |
20:02.14 | So3kris | ik have a 4521 |
20:03.33 | wasim | (hic) |
20:03.45 | Bicster | hic |
20:03.53 | wasim | hail -drunks! |
20:03.58 | Bicster | those guys rule |
20:04.26 | Bicster | l33 with a capital T |
20:04.40 | denon | hey wasim, your country has Internet again! |
20:04.47 | jhiver | Umaro: http://www.voip-info.org/tiki-print.php?page=Asterisk+Wishlist search for app_machinedetect.c |
20:04.47 | Damin | Hmmm.. |
20:05.10 | wasim | denon: yes, we're back on the glass centric thing, and off that lousy e3 on satellite |
20:05.26 | denon | wasim: that outtage sounded pretty ugly |
20:05.28 | Damin | Can you set "port=" directives for each IAX peer? I.E. have one on port 4569 and another on say 5000? |
20:05.54 | Damin | I know you can define it globally... |
20:06.01 | Damin | But can you do it in a context? |
20:06.03 | wasim | denon: it was, and we're due for another one, for a week 18th-23rd |
20:06.20 | denon | wasim: that's when they're going to down it for the repairs? |
20:06.29 | denon | I heard they had to get it approved by the other countries first |
20:06.39 | Bicster | denon: that's when the department of homeland security will be taking it down to upgrade the carinvore servers |
20:06.45 | denon | hehe |
20:06.54 | Umaro | jhiver: is that better, od you think? |
20:06.55 | So3kris | i'm trying asterisk out for a couple of days but some questions. ik have the getting started for me and im wondering if you get only a email with the voice message of that you can listen your voice mail via the server. iḿ wondryy in some exampels the voice mailmailbox nummer is the same as the callerid and other exampels not |
20:06.59 | wasim | denon: yeah, they've patched it up temporarily, and then giving enough lead time to get alternates up |
20:07.28 | denon | wasim: get a boat and lay out a couple'a oc-192s :) |
20:07.33 | denon | you could make a fortune |
20:07.35 | emp | is *@home a good way to get started? |
20:07.43 | wasim | the worst of it was, it was just bloody 20 km off the coast, and it took them two weeks to get a ship in to fix it |
20:08.02 | wasim | denon: we're getting smw-4 sometime in oct to this single point of failure should go |
20:08.08 | denon | huh, 20km .. could almost send a diver |
20:08.46 | yaaar | Bicster: which one? |
20:08.47 | iCEBrkr | OK, there has to be SOMEONE that has tinkered with EAGI's and the audio stream... |
20:08.53 | jhiver | Umaro: no idea |
20:09.01 | jhiver | Umaro: just pointing you to it |
20:09.06 | Bicster | hehe yaaar ... mrs. wasim doesn't seem to like the idea of having other mrs. wasims around |
20:09.14 | Umaro | jhiver |
20:09.20 | Umaro | jhiver: er, oic |
20:09.25 | jhiver | it's me :) |
20:09.34 | FaxTerix | Did anybody have problems sending tiff images using TxFax? |
20:09.43 | *** join/#asterisk znoG (~gs@200.115.216.109) |
20:09.50 | *** join/#asterisk DannyF (~dannyf@h47n1fls32o865.telia.com) |
20:09.51 | yaaar | Bicster: you must not be too familiar with Islam |
20:10.02 | Umaro | jhiver: yeah, that's about the same as the backgrounddetect app |
20:10.10 | yaaar | (that was the joke) |
20:11.40 | tomtom_ | it seems that i can't use any other group number but 1 |
20:11.54 | yaaar | Bicster: ferocious....but dumb as a post |
20:12.23 | Umaro | jhiver: but thanks for finding it :) |
20:12.32 | Bicster | did the bugblatter beast make it into the movie? |
20:12.40 | wasim | there's a movie? |
20:12.52 | M_at | I believe it was mentionned |
20:13.17 | Bicster | wasim: it's been discussed at great length in -drinkers |
20:13.30 | Bicster | coppice hated it, nobody else saw it :) |
20:13.55 | M_at | It's great if you understand the true THHGTTG |
20:14.15 | M_at | Bicster: Reeally it's a 30 second AAC file |
20:14.17 | So3kris | mailbox=1000 is this a the dailing number ? |
20:14.31 | *** join/#asterisk santiago (~santiago@63.245.86.254) |
20:15.30 | Bicster | even worse |
20:16.07 | Bicster | I see people all the time who let their phone ring longer than necessary, just so they can annoy others with their 'cool' ringtone |
20:16.28 | M_at | Oh I don't - cant stand that |
20:16.30 | twisted[work] | my ring tone simply says "AFLAC!" |
20:16.36 | Bicster | twisted, awesome :) |
20:17.09 | Bicster | I was really pissed off that my last mobile phone didn't come with any non-musical ringtones. I had to hunt one down and install it, which took a few hours. |
20:17.11 | M_at | Get a decent symbian handset |
20:17.16 | M_at | It can say whatever you want |
20:17.39 | *** join/#asterisk albators (~albaneagr@host-84-9-255-27.bulldogdsl.com) |
20:17.52 | twisted[work] | i just made mine |
20:18.24 | twisted[work] | heh |
20:18.34 | tomtom_ | no one any clue about the group thing? |
20:18.48 | So3kris | !SGI++ |
20:19.08 | Bicster | SGI is still around? heh, stock price $0.59 |
20:19.09 | So3kris | SGI has a new 1u server |
20:19.21 | Bicster | why would anyone buy an SGI product when the company is clearly not long for this world? |
20:19.21 | twisted[work] | tomtom_, considering the vagueness of your problem, no |
20:19.38 | M_at | because they look cool |
20:19.46 | tomtom_ | twisted[work]: i shouldn't start pasting logfiles in here right away , now should i? |
20:19.52 | Bicster | well yeah .. the SGI 1600SW on my desk _does_ look cool :D |
20:19.56 | So3kris | the are cool and the are fast en good |
20:20.22 | So3kris | And the run unix |
20:20.23 | twisted[work] | tomtom_, no, but you could describe your problem in greater detail. |
20:20.25 | tomtom_ | the issue seems to be that if i use any other channel group in zapata.conf than g0 it doesn't work |
20:20.30 | So3kris | The real sgiÅ› than |
20:20.36 | twisted[work] | okay, what does it say? |
20:20.48 | tomtom_ | well, i'm getting an all circuits are busy when i dial out |
20:20.56 | tomtom_ | unaible to create zap channel |
20:21.09 | twisted[work] | you have to have group=x in zapata.conf |
20:21.26 | twisted[work] | where x is the group number, and it goes from above the first channel definition until it runs into another group line |
20:21.26 | shido | irix |
20:22.07 | tomtom_ | yeah, i have that .. when i put group = 0 and dial on g0 it works |
20:22.16 | tomtom_ | when i put group = 1 and dial on g1 it doesn't work |
20:22.19 | eKo1 | wasn't sgi doing something with * |
20:22.29 | tomtom_ | when i put group = 0,1 it works only on channel 1 |
20:22.49 | twisted[work] | pastebin your zapata.conf |
20:22.50 | tomtom_ | both on asterisk 1.0.7 and 1.0.9 |
20:22.57 | twisted[work] | because I know for a fact this works |
20:23.01 | tomtom_ | although it does work on similar systems |
20:23.10 | tomtom_ | yeah, it does .. only not on this system |
20:23.36 | tomtom_ | everyone is busy/congested at this time is what i get |
20:23.47 | *** join/#asterisk Inv_arp (~junya@adsl-3-251-24.mia.bellsouth.net) |
20:23.51 | twisted[work] | pastebin your zapata.conf |
20:24.08 | SpaceBass | anyone know a fix for a cisco phone that crashed during a firmware upload? |
20:24.45 | tomtom_ | twisted[work]: or shall i post it private? |
20:24.48 | Damin | SpaceBass: Get it replaced under SmartNet... |
20:24.55 | twisted[work] | pastebin your zapata.conf |
20:24.57 | Damin | SpaceBass: You did get SmartNet on it, correct? |
20:24.58 | SpaceBass | Damin was afaride of that |
20:24.58 | twisted[work] | pastebin |
20:24.59 | twisted[work] | pastebin |
20:25.03 | tomtom_ | yeah ok |
20:25.05 | tomtom_ | i get it |
20:25.13 | SpaceBass | Damin purchased smartnet... like 4 weeks ago.. vender is taking their time |
20:25.27 | SpaceBass | Damin its also still under warrenty... called cisco to open a case |
20:25.38 | Damin | SpaceBass: You can try rebooting it and see if it spews out any packets w/ Ethereal. It probably has some dying gasp recovery mechanism.. |
20:26.23 | SpaceBass | Damin will try that... thanks |
20:26.44 | SpaceBass | Damin not entirely sure what happened... was trying to update the logo and it locked up on getting an IP... then when I rebooted it just hangs |
20:26.51 | Bicster | SpaceBass: as in fish, or as in audio below 60Hz? |
20:26.51 | tomtom_ | twisted[work]: http://pastebin.com/pastebin.php?dl=312113 |
20:27.14 | SpaceBass | Bicster close... as in the musical instrurment(producing audio below 60hz) |
20:27.25 | Bicster | ok |
20:27.32 | twisted[work] | tomtom_, is this bristuffed? |
20:27.35 | Bicster | but they don't produce much noise in space ;) |
20:27.36 | tomtom_ | yeah |
20:27.41 | twisted[work] | heh |
20:27.44 | SarahEmm | ooh |
20:27.46 | twisted[work] | no guarantees there |
20:27.48 | SarahEmm | i thought it was space fish :) |
20:28.00 | tomtom_ | well i have systems with the exact same software image that do work :/ |
20:28.13 | twisted[work] | bristuffs changes zaptel iirc. |
20:28.28 | twisted[work] | in which case, i'm now lost |
20:28.39 | Bicster | bristuff fixes zaptel :P |
20:28.42 | *** join/#asterisk Caro_away (~caro@201.133.229.15) |
20:28.45 | twisted[work] | HAH |
20:28.48 | Caro_away | hi all |
20:28.49 | Bicster | hehe |
20:29.00 | Bicster | actually IIRC bristuff doesn't mess with zaptel |
20:29.11 | twisted[work] | chan_zap |
20:29.22 | Bicster | ok, it messes with that |
20:29.25 | SpaceBass | note: never ever order anything from pcuniverse.com |
20:29.37 | twisted[work] | yeah, and that's where his problems are going to come from |
20:30.40 | Caro_away | i have a little problem with tdm400p and te110p i have in my first pci the te110p and the second pci my tdm400 2 fxs 2 fxo in my zaptel.conf have fxo_ks=1-2 and fxs_ |
20:31.05 | Caro_away | in 2-3 but how i can configure the te110p like e1 |
20:31.17 | *** part/#asterisk Bicster (~Bicster@Bicster.active.supporter.pdpc) |
20:31.42 | Caro_away | because my tdm400 works fine but the te110p i can´t found info how configure in mode e1 |
20:32.33 | FaxTerix | Good bye |
20:33.55 | eKo1 | Caro_away: isn't it a jumper setting |
20:34.12 | albators | ca be jumper or software |
20:35.18 | twisted[work] | te110p has a jumper for t1/e1 |
20:35.20 | anthm | who want's res_monitorthatmuxes ? |
20:35.23 | twisted[work] | jumper ON means e1 |
20:35.26 | twisted[work] | jumper OFF means t1 |
20:35.34 | shido | modes |
20:35.35 | shido | dood |
20:35.36 | shido | modes |
20:35.37 | Caro_away | ah |
20:35.41 | shido | screw the jumper |
20:35.43 | Caro_away | i don´t check that |
20:35.52 | Caro_away | maybe is my mistake in this case |
20:36.09 | twisted[work] | shido, on the 110, you need to set the jumper. |
20:36.16 | *** join/#asterisk KarolM (asterisk@toronto-HSE-ppp4040107.sympatico.ca) |
20:36.18 | brookshire | oh.. haha |
20:36.25 | brookshire | thank god we have that documentation up |
20:36.25 | shido | you can modprobe it and change it from the command line |
20:36.38 | shido | or have I been hit in the head too many times |
20:36.44 | twisted[work] | shido, on the 410/405 yes, but on the 110, IIRC no |
20:36.50 | eKo1 | you can change jumper settings from the command line?! |
20:36.55 | shido | yeah eKo1 |
20:36.58 | brookshire | for the quad spans |
20:37.00 | Caro_away | but in zaptel.conf how i can put my conf is correct span=1,0,0,cas,hdb3 bchan=1-15 dchan=16 bchan=17-31 is correct |
20:37.01 | shido | I have a box in the philippines |
20:37.03 | brookshire | not the single one |
20:37.10 | shido | and I didnt want them to shut it down everytime - |
20:37.15 | Caro_away | yep is single e1 |
20:37.16 | mogorman | <PROTECTED> |
20:37.17 | shido | and found u can change the jumper settings |
20:37.19 | mogorman | dont be a nubb |
20:37.22 | mogorman | you can change both |
20:37.24 | mogorman | from source |
20:37.27 | brookshire | then wtf |
20:37.30 | mogorman | i mean the mod probe |
20:37.35 | brookshire | why are we making this stupid documentation then |
20:37.37 | mogorman | t1e1override |
20:37.41 | twisted[work] | mogorman, from source? |
20:37.43 | mogorman | is the flag |
20:37.51 | mogorman | well you could go edit the source |
20:37.57 | mogorman | and change default twisted ^_^ |
20:38.01 | brookshire | you guys need to tell jan what's up |
20:38.03 | SarahEmm | err |
20:38.03 | mogorman | but yeah i have typos a plenty |
20:38.07 | twisted[work] | that's a bit dumb, considering it's easy to swap the jumper |
20:38.10 | SarahEmm | why bother with a jumper if oyu can override it in software? |
20:38.13 | shido | no |
20:38.16 | shido | when u have a remote box |
20:38.21 | shido | and if you have someone "change" the jumper |
20:38.24 | shido | and have your box stolen |
20:38.26 | shido | that sux |
20:38.34 | shido | I would much rather do it from here remotely |
20:38.38 | shido | giveing NO one access |
20:38.41 | mogorman | and you can change it no matter where the jumper is |
20:38.47 | shido | yes |
20:38.51 | mogorman | people just like the ability to do it |
20:38.53 | mogorman | by hand |
20:39.02 | SarahEmm | err... |
20:39.05 | yaaar | hmmm. anybody know why calling my voip (asterlink) 800 number is just ringing without connecting to my * box? i'm looking at the console (with abou 7 v's) and it doesn't say anything at all. 'iax2 show registry' shows my registered to them.... |
20:39.12 | SarahEmm | how often do boxes change from T1 to E1 once put on site? :P |
20:39.33 | twisted[work] | apparently quite often |
20:39.34 | shido | if you are a carrier and your E1 gets cut off you have to quickly change to analog |
20:39.36 | mogorman | exactly |
20:39.42 | anthm | ask asterlink support, PAGING DARTH! |
20:39.45 | shido | and t1 channel abnks are easier to find |
20:40.34 | twisted[work] | do you not then still have to have someone on site? |
20:40.40 | DarthClue | yaaar: call me, sending you a pm now... |
20:40.48 | yaaar | k |
20:40.49 | twisted[work] | to change the equipment? |
20:40.51 | shido | yeah but not open the box |
20:41.01 | twisted[work] | but they could just as easily steal it |
20:41.04 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
20:41.09 | shido | the box is in an apparatus that allows you to use a keycode |
20:41.16 | shido | 3 to get to the box |
20:41.24 | shido | u just give them 1 to open the cage |
20:41.27 | shido | not the one to get the box out |
20:41.30 | shido | :) |
20:41.41 | shido | "plug this in and go home" |
20:41.52 | shido | paypal him a few bucks, done |
20:44.28 | lesouvage | tzafrir: The work for TBook is almost done. It took more time then expected. |
20:45.25 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
20:48.13 | yaaar | neat. thanks DarthClue |
20:48.46 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
20:48.57 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
20:49.13 | sivana | is the sip.conf username field a global variable in *? |
20:49.29 | SarahEmm | you're asterlink, DarthClue? |
20:50.07 | DarthClue | SarahEmm: i work for the dark side, we are many, many things. |
20:50.12 | anthm | see, we get bitched at for too much advert and every day ppl are suprised by stuff |
20:50.28 | SarahEmm | heh |
20:50.34 | MRH2 | is there any way to set ALERT_INFO on a Channel (when using a call file to auto-dialout) |
20:50.38 | SarahEmm | you need to offer 416/647 DIDs ;) |
20:51.00 | DarthClue | SarahEmm: we are working on it, but we have to negotiate with certain rebel forces to make that happen. |
20:51.28 | anthm | for the record cluecon IS asterlink and asterlink is run by anthm and DarthClue, bkw_ and file work with me. |
20:52.21 | MikeJ[Laptop] | SBC has more :) |
20:52.26 | SarahEmm | heh okay DarthClue |
20:52.34 | *** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
20:52.35 | SarahEmm | who are you, sivana? |
20:52.37 | mutilator | yep, i'm not sbc |
20:52.47 | sivana | I am... sivana :P |
20:52.50 | twisted[work] | sivana is sivana |
20:52.55 | sivana | hehe |
20:52.57 | SarahEmm | err |
20:52.59 | SarahEmm | what provider i meant ;) |
20:53.01 | sivana | voctel.com |
20:53.36 | sivana | but you need wholesale voip right? |
20:53.56 | SarahEmm | me? |
20:54.01 | mutilator | mine are only in michigan tho.. |
20:54.10 | sivana | ya |
20:54.14 | SarahEmm | a 647/416 DID and an 888 DID... i'm still looking around. |
20:54.25 | SarahEmm | 888/877/866/whatever |
20:54.28 | SarahEmm | vanity, tho |
20:54.31 | sivana | ya, we got them too |
20:54.39 | DarthClue | and Canada...must be usable in Canada. |
20:54.55 | twisted[work] | hah |
20:55.06 | sivana | ya |
20:55.46 | SarahEmm | yeah. |
20:55.54 | SarahEmm | 'k. i'll keep that in mind |
20:55.57 | SarahEmm | still looking around a bit. |
20:56.57 | Silik0n | who cares about canada? |
20:57.09 | MikeJ[Laptop] | canadians |
20:57.36 | pabelanger | Terrance and Philip do... |
20:57.39 | DarthClue | MustDie: you coming to cluecon? |
20:57.50 | MustDie | i never been to chicago |
20:57.52 | sivana | I am Canadian! |
20:57.54 | MustDie | so i will have a reason to go |
20:57.56 | MustDie | :P |
20:57.58 | Silik0n | MikeJ[Laptop] and canadians matter since when? |
20:58.14 | MustDie | Silik0n: since south park i guess |
20:58.15 | MustDie | :P |
20:58.16 | Silik0n | i mean they are famous for what Theo DeRaddt and a hockey strike? |
20:58.17 | sivana | in a bottle?!.... brilliant! |
20:58.19 | albators | anyone knows why i can't get caller id on my tdm card to my analog phone ? |
20:58.50 | pabelanger | Silik0n: don't forget beer. |
20:58.54 | albators | if i have call on hold, then i can see the caller id |
20:59.02 | Silik0n | oh yeah beer++ |
20:59.17 | pabelanger | albators: you line provisioned for the service? |
20:59.25 | albators | yes |
20:59.36 | twisted[work] | molson ice != beer |
20:59.42 | albators | i see the caller id in the debug |
20:59.48 | albators | but on phone nothing |
21:00.02 | Silik0n | hey hose hound get outta here eh |
21:00.07 | albators | and if i get the second caller , then on phone it will show the caller id |
21:00.14 | twisted[work] | Silik0n, WTF? |
21:00.27 | Silik0n | twisted[work] never saw Strange Brew? |
21:00.30 | twisted[work] | no |
21:00.35 | MustDie | twistah 1 |
21:00.36 | MustDie | !!! |
21:00.54 | Silik0n | 2 guys who's whole life is sitting around getting drunk and cracking on canadians |
21:00.59 | twisted[work] | oh |
21:01.00 | albators | :D |
21:01.01 | Silik0n | thats the plot of the entire movie |
21:01.05 | *** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
21:01.12 | twisted[work] | sounds like terrance and phillip |
21:01.14 | twisted[work] | only older |
21:01.16 | Silik0n | hah |
21:01.33 | albators | and other question, how to answer to the second call ;) since i never managed ;) |
21:01.49 | drbrown | has anyone had any problems with today's cvs HEAD compiling? |
21:01.55 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
21:02.04 | albators | i don't use HEAD since 3 days ago |
21:02.07 | DarthClue | yaaar: you coming to cluecon? |
21:02.10 | albators | since it stoped working! |
21:02.34 | drbrown | albators: does yours not compile either? |
21:02.56 | albators | it compiled but asterisk eat all my resurses :P |
21:03.06 | *** join/#asterisk opus_____ (~opus@dahphish.org) |
21:03.07 | terrapen | oh my god |
21:03.09 | opus_____ | hi |
21:03.13 | terrapen | look what i found on fark |
21:03.14 | terrapen | http://www.tufts.edu/vet/cfa/hoarding/FORMgallery.htm |
21:03.18 | albators | cpu 80% and asterusk not working :P |
21:03.42 | albators | eat cpu and ram, but not working :P |
21:04.10 | opus_____ | me hungry for cpu and ram soup |
21:04.15 | albators | ;) |
21:04.22 | albators | so about the caller id |
21:04.25 | albators | any idea? |
21:06.12 | *** join/#asterisk AST07070NYC (~spam@pool-68-239-181-169.nwrk.east.verizon.net) |
21:08.10 | AST07070NYC | hello, does anyone know of an example on the web which uses call files which pass data.... ie call NXXNXXXXXX and play file /var/foo.gsm ??? |
21:10.18 | *** join/#asterisk emp (~emp@70.57.239.37) |
21:10.26 | Druken | anyone here with a working AMP install that is willing to help me out? |
21:11.18 | emp | I just installed *@home and configured X-Lite, when it logs in, it gives "Login failed! Contact Network Admin. Your number is: 200" any hints? I'm just following the walkthrough at this point |
21:11.45 | ChkDigit | What is a complete list of the special extension names (s for the starting one, etc...)? |
21:12.13 | eKo1 | check the wiki |
21:12.25 | ChkDigit | I had hoped this would be quicker. |
21:13.06 | Hmmhesays | quintum cli is such a pain |
21:13.41 | eKo1 | ChkDigit: no, the wiki is quicker and safer |
21:13.48 | *** join/#asterisk data2 (~data@213.221.191.134) |
21:13.50 | eKo1 | quintum?! NO!! |
21:14.23 | eKo1 | I'm never using anything from them again. |
21:14.30 | Hmmhesays | oh yeah? why is that? |
21:14.59 | twisted[work] | quintums are fucking ridiculous to set up |
21:15.17 | eKo1 | yeah. |
21:15.24 | Hmmhesays | i'm not saying the setup is hard... it's really quite simple, but their cli is a bitch to write anything for |
21:15.35 | eKo1 | that, and the sip stack on the unit I bought sucks major donkey balls. |
21:15.41 | *** join/#asterisk santiago (~santiago@63.245.86.254) |
21:15.41 | DarthClue | ChkDigit: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf under predefined extension names. |
21:15.58 | Hmmhesays | it really depends on 1st or 2nd generation quintum and what firmware you have |
21:16.09 | eKo1 | I have the 1st gen. stuff |
21:16.40 | Hmmhesays | yeah that's why sip was an after thought for 1st gen, the second gen is a whole different animal |
21:16.52 | eKo1 | i bet the 2nd gen. stuff is way better. but after that experience, i'm not doing anything with quintum anymore. |
21:17.03 | twisted[work] | i'm running a 2nd gen device |
21:17.07 | twisted[work] | the cli fucking blows |
21:17.12 | twisted[work] | and the setup is still a bitch |
21:17.15 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
21:17.21 | Hmmhesays | naw, pretty simple once you get used to it |
21:17.25 | twisted[work] | i would not reccomend it to my worst enemy |
21:17.35 | Hmmhesays | it's no worse than cisco |
21:17.50 | twisted[work] | HAH |
21:17.55 | Micc | So is dtmf detection a problem for asterisk or do I just have something screwed up? |
21:17.56 | twisted[work] | cisco's cli at least gives you command help |
21:18.10 | twisted[work] | quintums says "refer to the user guide" |
21:18.19 | Micc | I've had the same intermittent problems with SIP and ZAP. |
21:18.20 | twisted[work] | and the users guide is so far from correct it's not funny |
21:18.32 | Hmmhesays | that is an inaccurate statement |
21:18.37 | twisted[work] | no it's not |
21:18.46 | MikeJ[Laptop] | one of my back burner projects was to re-work command completion and partial commands and context sensitive help in cli to be more cisco-like |
21:18.56 | Hmmhesays | sorry, the cli reference is accurate |
21:19.02 | twisted[work] | no it's not |
21:19.12 | Hmmhesays | if you are reading the one for your firmware revision it is |
21:19.15 | twisted[work] | most of the commands in the user guide that came with this thing do not exist |
21:19.42 | twisted[work] | as well as the ones on the website |
21:19.57 | Damin | twisted: Whatcha working on? |
21:19.58 | Hmmhesays | heh, name one that doesn't match on the cli reference |
21:20.16 | twisted[work] | Hmmhesays, kinda difficult seeing as it's been a month since i've touched the damned thing |
21:21.04 | Hmmhesays | I hear the same stuff about quintum all the time, they've got learning curve... so what |
21:21.13 | twisted[work] | that's not a learning curve |
21:21.19 | Hmmhesays | get the right cli reference and you'd be fine |
21:21.25 | twisted[work] | that's like saying the special olympians can compete in the olympics |
21:21.45 | Hmmhesays | sounds like you are complaining just to complain |
21:22.05 | twisted[work] | no, i'm stating my opinion on something that is a pain in the ass and has no accurate reference |
21:22.16 | Hmmhesays | your statement is not accurate though |
21:22.21 | twisted[work] | my statement is highly accurate. |
21:22.26 | twisted[work] | if you were here I could prove it |
21:22.38 | Hmmhesays | and you can't prove it now? |
21:22.56 | twisted[work] | uh, no, because the box is not set up any longer |
21:23.01 | twisted[work] | it's sitting here on my desk |
21:23.41 | Hmmhesays | so put an ip in it, update the firmware, grab the right cli reference and tell me what in the reference is not accurate. because I am quite curious |
21:23.55 | twisted[work] | Hmmhesays, i'm not going to the effort just to appease you |
21:24.05 | emp | what log file can i look in for * authentication messages? |
21:24.19 | twisted[work] | if you don't want to listen to my opinion then ignore it. |
21:24.46 | Hmmhesays | heh, honestly it seems more like you just don't like the fact that your statement could be inaccurate |
21:24.53 | ender | does the Polycom 301 actually have a mic built in for speaker phone? |
21:25.02 | data2 | if anyone can do a reasonable rate for a Sri Lanka Mobile? could they please msg me, ta. |
21:25.22 | twisted[work] | Hmmhesays, no, i don't like the fact that you can't just accept what i have dealt with as the truth |
21:25.49 | twisted[work] | end of conversation |
21:25.59 | twisted[work] | bring it up again and you will be quelched |
21:26.04 | Hmmhesays | heh |
21:26.38 | Hmmhesays | so kick me |
21:26.46 | eKo1 | fight fight fight |
21:27.11 | Hmmhesays | come on |
21:27.17 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
21:27.26 | eKo1 | all this antagonism over quintum... |
21:27.35 | Hmmhesays | heh, this isn't about quintum at all |
21:27.56 | Hmmhesays | that much is obvious |
21:32.56 | twisted[work] | oh, but it is. |
21:33.05 | Hmmhesays | heh, obviously you have deeper issues |
21:33.17 | *** mode/#asterisk [+q *!*@66.173.103.108] by twisted[work] |
21:33.30 | SwK[work] | CRAZED BUTT CLOWNS! |
21:33.37 | Assid | whats +q again? |
21:33.50 | twisted[work] | if you had spent the amount of time I have downloading manuals, trying different shit, and wasting time trying to figure out that piece of shit, you would understand where i'm coming from |
21:33.57 | twisted[work] | +q is quelch |
21:34.04 | twisted[work] | or quiet |
21:34.07 | Assid | aah |
21:34.08 | Assid | k |
21:34.08 | twisted[work] | however you want to look at it |
21:34.15 | Assid | i thought it was for the channel |
21:34.22 | Assid | didnt know you could do it per user |
21:35.30 | twisted[work] | telling someone they have 'deeper issues' when they are simply explaining the difficulty they've had with a product and their dislike for it is bad enough. taunting an op is just bad. |
21:36.03 | data2 | hmm, just checked my gateway box... asterisk 0.9.1 was just out last time i was on this channel :( |
21:36.05 | twisted[work] | Hmmhesays, i have nothing against you. we have had many another conversation/discussion that has been fine. when you try to tell me that i'm lying i get pissed off. |
21:36.07 | SwK[work] | but taunting ops and ircopers is fun sometimes |
21:36.31 | emp | do I need an outbound proxy IP for x-lite to do a echo test? |
21:36.35 | twisted[work] | SwK, heh, yeah, but basically calling them liars is one thing. |
21:36.43 | Assid | xlite is weird |
21:36.44 | SwK[work] | twisted[work]: LIAR! |
21:36.45 | twisted[work] | *one another |
21:36.51 | SwK[work] | heh |
21:36.55 | Assid | somehow everyone i call says they hear me funn |
21:36.57 | Assid | funny |
21:37.06 | twisted[work] | if you weren't sitting right behind me I might not have known you were joking |
21:37.08 | Assid | actually i dont know if its xlite.. or carrier |
21:37.27 | SwK[work] | Assid: heh xLite is ok but its not the best phone |
21:37.29 | Assid | your lying!! he aint behind you.. hes on the left of the chat screen! |
21:37.31 | *** mode/#asterisk [-q *!*@66.173.103.108] by twisted[work] |
21:37.36 | Assid | err. right.. |
21:37.38 | twisted[work] | Assid, lol |
21:37.40 | SwK[work] | what do you expect for a free version of a commercial product |
21:37.43 | twisted[work] | he's physically right behind me |
21:38.00 | Hmmhesays | heh, no I didn't say you were lying, I said your statement was inaccurate as of right now with the current firmware and cli reference |
21:38.02 | Hmmhesays | period |
21:38.12 | bkw_ | you tell em! |
21:38.16 | Assid | hahahaa |
21:38.17 | SwK[work] | Assid: actually twisted is correct we sit with about 15 feet between our backs |
21:38.18 | twisted[work] | hah |
21:38.31 | Assid | hrmm |
21:38.38 | *** join/#asterisk gst (~gst@85.124.173.170) |
21:38.45 | Damin | WTF piece of equipment are you guys talking about anyway.. |
21:38.48 | Assid | SwK[work]: what do you suggest i use? |
21:38.55 | SwK[work] | Assid: a hard phone |
21:39.00 | Assid | bah |
21:39.02 | Assid | not availableh here |
21:39.03 | twisted[work] | heh... at least |
21:39.05 | Assid | not for cheap anyways |
21:39.17 | Assid | everything costs a min. of over $100 |
21:39.17 | SwK[work] | no seriously tho... check the settings of your soundcard and mic etc... |
21:39.22 | twisted[work] | actually, xlite is fine, as long as your audio device is good and the machine can handle it |
21:39.26 | SwK[work] | yeah |
21:39.33 | twisted[work] | I use it occasionally on my mac |
21:39.46 | Assid | 3Ghz.. enough to voice chat man!!! |
21:39.46 | SwK[work] | crappy sound card, mic gain turned up too much etc can cause xlite headaches |
21:39.57 | Assid | hrmm |
21:40.00 | twisted[work] | Assid, haha.. yeah, that should be okay to handle the application |
21:40.24 | gst | is it somehow possible to reply in an AGI (or py-asterisk program) with a "user not found" (after an extension matches and the script is called)? |
21:40.28 | SwK[work] | call 800-555-1212 it'll talk to you |
21:40.31 | MikeJ[Laptop] | -q *!*@ |
21:40.38 | Assid | 800? |
21:40.41 | Assid | cool.. |
21:40.48 | MikeJ[Laptop] | -q *!*@* ..is it quiet in here? |
21:40.49 | SwK[work] | yeah 800-555-1212 its powered by tell me |
21:40.50 | Assid | isnt that bad use of office equipment? |
21:41.04 | Assid | no clue whats tell me |
21:41.09 | twisted[work] | MikeJ[Laptop], no |
21:41.15 | Assid | im guessing some knd of directory service? |
21:41.25 | SwK[work] | Assid: voice activated directory assistance |
21:41.28 | twisted[work] | tellme |
21:41.37 | twisted[work] | i'd use tell me |
21:41.40 | twisted[work] | 18005558355 |
21:41.46 | SwK[work] | Assid: ask for crazy stuff like "buttplugs" or "stained blue dress" |
21:41.46 | twisted[work] | you can actually have "conversation" with it |
21:41.49 | file | Hello, and thanks for calling Tell me! |
21:41.54 | file | Main menu. |
21:42.00 | Assid | yeah.. but i wanna know if someone else can hear me fine |
21:42.04 | bkw_ | here are some of your choices |
21:42.10 | bkw_ | stock quotes |
21:42.12 | file | god, we're such telephony geeks |
21:42.12 | bkw_ | weather |
21:42.14 | bkw_ | news |
21:42.25 | Assid | need human behind the network |
21:42.27 | *** join/#asterisk dysan (~ack@202.37.224.27) |
21:42.49 | SwK[work] | well seeing its voice activated and those systems sorta suck, if you get get a good response you'll know you are transmitting audio ok |
21:42.57 | SwK[work] | OR... call yourself on your cellphone |
21:43.03 | Assid | eeks |
21:43.06 | Assid | im not calling india!!! |
21:43.13 | *** join/#asterisk fugitivo (~ajf@158-190-126-200.fibertel.com.ar) |
21:43.15 | Assid | waaaay too expensive |
21:43.22 | SwK[work] | so you gotta US DID? |
21:43.22 | fugitivo | hello |
21:43.26 | Assid | yep.. |
21:43.33 | SwK[work] | or find dell's 800 number and call them |
21:43.36 | Assid | but .. will be funny to hear myself |
21:43.40 | SwK[work] | they have people there all the time |
21:43.58 | SwK[work] | or if you just want to test xlite there is a echo app for the dialplan |
21:44.03 | twisted[work] | Assid, simply call echo() then |
21:44.09 | twisted[work] | or whatever the echo app is |
21:44.10 | SwK[work] | loops whatever you say backto yourself with a pretty good delay |
21:44.11 | twisted[work] | i can't remember |
21:44.19 | SwK[work] | yeah what twisted said |
21:44.27 | MustDie | swk: i think Dell is a indian company, go and call them then ;) |
21:44.37 | SwK[work] | MustDie: hah |
21:44.38 | SwK[work] | I have |
21:44.41 | fugitivo | anyone is using the mp108 fxs? |
21:44.43 | twisted[work] | dell is not an indian company, although they outsourced my job a long time ago |
21:44.47 | twisted[work] | to... india. |
21:44.49 | SwK[work] | fugitivo: yes |
21:44.53 | SwK[work] | fugitivo: it works |
21:45.00 | SwK[work] | lemme guess DTMF problems |
21:45.11 | fugitivo | SwK[work]: it works fine? or some problems? |
21:45.24 | SwK[work] | it works fine |
21:46.09 | SwK[work] | comes with a not so standard setting for DTMF on RFC2833 (it comes default at 96 and should be 101) otherwise its ok |
21:46.20 | SwK[work] | comes in a 4 port and a FXS version too |
21:46.29 | SwK[work] | err FXO version |
21:46.38 | fugitivo | great, thanks |
21:46.42 | emp | is the * handbook valid with the current release? |
21:47.03 | Assid | hrmm |
21:47.04 | fugitivo | SwK[work]: how much did you pay for it? |
21:47.06 | Assid | echo does nothing |
21:47.08 | Assid | thats weird |
21:47.16 | Assid | exten => 399,1,Echo |
21:47.29 | SwK[work] | fugitivo: dont remember got it from voipsupply I think |
21:47.43 | SwK[work] | i dont order the stuff I just install it |
21:47.49 | fugitivo | ok, thanks |
21:48.05 | dysan | which config files do i need to edit to make outgoing calls using x-lite? i have 3 zap compatible cards installed and working |
21:48.46 | DarthClue | dysan: sip.conf and extensions.conf, check the wiki, if you need links, ask. |
21:49.08 | twisted[work] | http://www.voip-info.org/wiki-Asterisk |
21:49.47 | DarthClue | http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf and http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf |
21:51.08 | Micc | Sorry, maybe that wasn't that funny. |
21:51.27 | brimstone | i laughed |
21:51.45 | *** join/#asterisk Twister (Twister@216.30.232.108) |
21:52.03 | Micc | I mean, it wasn't that funny for twisted. |
21:52.26 | *** join/#asterisk talkwebhosts (~freenoder@c-24-127-182-46.hsd1.ca.comcast.net) |
21:55.22 | *** join/#asterisk pdugas (~pdugas@h79.95.40.69.ip.alltel.net) |
21:55.25 | Assid | okay 4.2/4.4 and 3.3/3.4 K/sec for GSM / ILBC |
21:55.32 | Assid | quality similar |
21:58.22 | eldu | is there a free video+voice sip client ? |
21:59.51 | So3kris | http://www.voip-info.org/tiki-index.php?page=VOIP+Phones#id216825 |
22:00.23 | eldu | my bad |
22:00.31 | eldu | i miss this link lol |
22:01.20 | So3kris | ? |
22:01.56 | MRH2 | ne1 know if I can set alert_info variable on a channel, or as an overall default? |
22:02.15 | eldu | no matter i know this wiki but i havent figured out this link |
22:03.35 | MRH2 | I can set in the dialplan fine |
22:04.54 | yaaar | ok guys. catch you tomorrow. thanks for the help DarthClue and happy birthday. |
22:05.46 | So3kris | Iḿ bizzy white the voicemail but something is nog fully clear. is mailbox nummer the nummber that you call toe listen the voicemail or a identique nummber that haves no connection white the nummbers that your dailing |
22:06.02 | *** join/#asterisk niZon (~ilt@S0106deadbeefbeef.wp.shawcable.net) |
22:08.57 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
22:09.57 | *** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155) |
22:11.00 | MRH2 | what I was hoping to do was dial out from channel to destination and setting the alert_info asterisk dials out to the 'originate channel' so that the phone placing the call will autopickup |
22:13.08 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
22:13.08 | *** mode/#asterisk [+o bkw_] by ChanServ |
22:14.06 | terrapen | i need a decent audio-in device for my powerbook |
22:14.11 | terrapen | the iMic just sucks ass |
22:15.06 | opus_____ | hey |
22:15.23 | opus_____ | in realtime, for sipusers => , how come I MUST have ipaddr= to the address that the phone is on? |
22:15.48 | opus_____ | I have 'host=dynamic' in my sql table, but, it wants ipaddr=192.168.0.247 and won't work without it |
22:17.59 | mariogamboa | i have a problem with my tdm22b i configure in zaptel.conf fxo_ks=1-2 and fxs_ks=3-4 but i don't recive tone for my analog phone in the port 2 what is missing? |
22:19.04 | angler_ | mariogamboa, have u configured zapata.conf and also loaded the wctdm driver? |
22:19.13 | mariogamboa | mmm no |
22:19.50 | mariogamboa | how i can do thant? |
22:20.32 | mariogamboa | in my zapata.conf i have context=from-pstn signalling=fxs_ks rxwink=300 |
22:21.03 | mariogamboa | and another options like usercallerid etc. i need to add somenthing more? |
22:21.47 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
22:21.48 | angler_ | zapata.conf signalling=fxo_ks channel => 1-2 signalling=fxs_ks channel => 3-4 |
22:22.06 | TripleFFF2sdf | does one know how to chage the which app which is refering to ? |
22:22.21 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
22:22.36 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net) |
22:22.42 | TripleFFF2sdf | EG: box that moh works.. # which mpg123 |
22:22.42 | TripleFFF2sdf | EG2 box that doesnt |
22:22.42 | TripleFFF2sdf | <PROTECTED> |
22:22.56 | angler_ | mariogamboa, after that load wctdm...... modprobe wctdm |
22:23.32 | twisted[work] | it's good to see more digium folk in here providing quick tech help |
22:23.37 | angler_ | :) |
22:23.42 | TripleFFF2sdf | heh |
22:25.02 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
22:25.21 | TripleFFF2sdf | same paths |
22:25.22 | drbrown | has anyone had any luck running asterisk as a non root user? |
22:25.27 | TripleFFF2sdf | just one is a symlink |
22:25.33 | TripleFFF2sdf | darn mpg |
22:25.47 | TripleFFF2sdf | no diff in asterisk conf.. no dif in mpg123 size not version |
22:25.49 | TripleFFF2sdf | nor perms |
22:25.53 | mariogamboa | what version of zaptel have you angler |
22:26.05 | TripleFFF2sdf | but mpg just exits on box |
22:26.18 | mariogamboa | because the wctdm module is loaded |
22:26.38 | angler_ | latest head cvs |
22:26.43 | mariogamboa | and no recive tone in the analog phone withh the modification in the zapata.conf |
22:26.44 | angler_ | run "ztcfg -vv" now |
22:27.05 | mariogamboa | yep |
22:27.33 | mariogamboa | i have channel 1-2 fxo 3-4 fxs 5-31 cas/user |
22:28.23 | angler_ | asterisk is started correct? |
22:28.28 | mariogamboa | yep |
22:28.34 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
22:28.46 | mariogamboa | i can recive the asterisk*CLI> |
22:28.51 | shmaltz | I get this message exactly every minute: |
22:28.52 | shmaltz | Jul 12 18:36:36 NOTICE[28197]: pbx.c:1689 pbx_extension_helper: Cannot find extension context 'INVALID' |
22:28.54 | shmaltz | why? |
22:29.05 | *** join/#asterisk bprice20 (~bprice20@cpe-24-194-236-170.nycap.res.rr.com) |
22:29.10 | RoyK | shmaltz: rtfm? |
22:29.24 | shmaltz | RoyK, ok, now what? I read it all |
22:29.44 | TripleFFF2sdf | shmaltz Add exten i,1,ahngup() |
22:29.56 | TripleFFF2sdf | i for invalid |
22:30.01 | shmaltz | TripleFFF2sdf, I know the solution, but why am I getting it? |
22:30.02 | TripleFFF2sdf | t for timeout |
22:30.12 | TripleFFF2sdf | coz something sending to inavalid exten |
22:30.18 | TripleFFF2sdf | a goto or somethign else |
22:30.22 | shmaltz | who? |
22:30.28 | shmaltz | I got no active channels |
22:30.44 | RoyK | shmaltz: pastebin your dialplan |
22:30.45 | TripleFFF2sdf | call you friend pastebin.. ask him to receive your dialplan ,, and tell him to come tell us the url to his party |
22:30.55 | TripleFFF2sdf | RoyK beat me to it |
22:31.35 | TripleFFF2sdf | so any reasin mpg123 now spawning on one box but ok on another ?\ |
22:31.43 | TripleFFF2sdf | same confs same exec |
22:32.13 | MikeJ[Laptop] | TripleFFF2sdf, if it is yesterday\today cvs head, update.. there were issues |
22:32.29 | shmaltz | RoyK, It's too long |
22:33.36 | TripleFFF2sdf | nah its a old one |
22:34.38 | TripleFFF2sdf | well one month |
22:34.40 | TripleFFF2sdf | max |
22:34.43 | TripleFFF2sdf | used to work |
22:34.46 | Juggie | yah eugh |
22:34.49 | Juggie | * crashes for me now |
22:34.50 | TripleFFF2sdf | just doesnt spanw mpg |
22:35.10 | mariogamboa | login root |
22:35.21 | mariogamboa | password |
22:35.45 | Juggie | for me * crashes when i put a call on hold |
22:36.11 | shmaltz | Ok, thaks guys |
22:36.15 | shmaltz | I know why |
22:36.38 | shmaltz | some stupid provider is trying to register or use our server, sip debug told me that |
22:37.25 | sivana | heh |
22:39.19 | Micc | What version of asterisk is nufone using? Aren't they using realtime? |
22:40.20 | TripleFFF2sdf | hehe |
22:40.27 | TripleFFF2sdf | anyone got a mpg idea ? |
22:40.32 | TripleFFF2sdf | unless i change the player |
22:40.35 | TripleFFF2sdf | but not good |
22:43.41 | mariogamboa | cool i really go to florida to take the asterisk tranning and dcap |
22:45.23 | twisted[work] | while you're there, don't forget to hide from the hurricanes |
22:45.25 | *** join/#asterisk heka (~heka@82.114.68.124) |
22:47.33 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
23:00.47 | mariogamboa | ok |
23:01.33 | litage | if you/your company needed to hire an intermediate+ linux/asterisk employee, what sorts of questions would you ask? |
23:02.28 | DougRoyer | To do what? |
23:02.48 | DougRoyer | Program? Configure? Compile? Add-On's? |
23:03.22 | litage | program, config, add-ons |
23:06.10 | *** join/#asterisk cinix (~ax@24-52-166-190.lndnnh.adelphia.net) |
23:06.11 | twisted[work] | litage, don't be a competetor ;) |
23:06.28 | DougRoyer | I think that asterisk development experience is rare. So you would want an experienced C/C++ developer on Linux with knowledge of SIP/ VOIP. |
23:06.37 | cinix | I am attempting to dial a local SIP client when I receieve an incoming call from my voip provider, and this is what I get: |
23:06.40 | cinix | Now forwarding IAX2/voicepulse-in-01@66.234.228.170:4569/3 to 'Local/matt@outgoing' (thanks to SIP/matt-9713) |
23:06.53 | cinix | Got SIP response 482 "Loop Detected" back from 24.52.166.190 |
23:06.54 | twisted[work] | oh wait |
23:06.55 | twisted[work] | you're in .au |
23:06.57 | twisted[work] | n/m |
23:07.12 | cinix | and then it makes some new local channel and tries to call that and everything falls apart :-/ |
23:07.37 | twisted[work] | cinix, turn off forwarding on your phone |
23:08.25 | twisted[work] | that means that whatever device is on SIP/matt sent a 302 Moved Temporarily or another forwarding code along with a URI |
23:08.46 | twisted[work] | asterisk will accept that URI and attempt to send it to that location |
23:08.58 | cinix | oh hmm.. thought I had disabled forwarding. lemme check |
23:09.04 | twisted[work] | which means SIP/matt is forwarded to uri: matt@<your * ip> |
23:09.44 | cinix | I had notransfer=1 set in sip.conf, I suppose that's not valid. |
23:09.48 | cinix | got it from iax.conf |
23:12.06 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
23:14.09 | cinix | okay you actually meant on the client, found the forwarding optinos but they aren't set |
23:14.44 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
23:17.06 | jeremywhiting | hi all |
23:17.29 | jeremywhiting | I'm having serious problems getting callerid on receiving calls to work with tdm04b |
23:19.15 | jeremywhiting | fsk_serie made mylen < 0 (-13) |
23:19.34 | jeremywhiting | CallerID feed failed: Success |
23:20.08 | jeremywhiting | I've seen a few posts like this in pastebin after google search, and on asterisk forums, but no solution |
23:21.57 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
23:22.15 | jeremywhiting | anyone ever experienced this? |
23:23.49 | MikeJ[Laptop] | jeremywhiting, there is a bug on a crash similar to this in mantis.. |
23:25.48 | jeremywhiting | mantis? |
23:26.15 | DarthClue | jeremywhiting: bug tracker. |
23:26.20 | jeremywhiting | it doesn't crash though |
23:26.25 | *** join/#asterisk dack (~dack@S0106000f664f0871.vc.shawcable.net) |
23:26.39 | DarthClue | may still be related. you running HEAD or STABLE? |
23:26.43 | jeremywhiting | it just doesn't get callerid on any zap channel is all |
23:26.45 | jeremywhiting | stable |
23:26.54 | jeremywhiting | well, latest gentoo I should say |
23:27.02 | DarthClue | 1.0.9 or something else? |
23:27.08 | jeremywhiting | 1.08 |
23:27.11 | jeremywhiting | .8 |
23:27.13 | dack | How come I can't seem to get the T extension to trigger on AbsoluteTimeout? It always just hangs up right away. |
23:27.29 | Qwell | dack: T or t? |
23:27.34 | dack | Qwell: T |
23:27.38 | dack | absolutetimeout. |
23:27.47 | DarthClue | jeremywhiting: you should upgrade to .9, .8 has a known caller id issue. |
23:27.55 | jeremywhiting | ok, thanks |
23:27.59 | Qwell | there was a bug in HEAD a few days/weeks ago, that had to do with exten t |
23:28.10 | Qwell | it affected me, and upgrading fixed it |
23:28.11 | dack | i'm on 1.0.8 |
23:28.18 | Qwell | then you should be ok, heh |
23:28.54 | DarthClue | 1.0.9 is the recommended stable |
23:28.55 | dack | I do "exten => T,1,Playback(beep)" |
23:29.09 | dack | but i get no beep on absolutetimeout |
23:29.25 | Qwell | perhaps its quiet? Have you tried a NoOp |
23:29.27 | Qwell | ? |
23:29.32 | DarthClue | dack: what is your absolutetimeout set to? |
23:29.39 | jeremywhiting | got it |
23:29.51 | jeremywhiting | gentoo dev in charge of asterisk is just a bit behind then I guess |
23:29.58 | Qwell | jeremywhiting: They always are. |
23:30.03 | Qwell | (no offense to the gentoo devs) |
23:30.21 | jeremywhiting | do I need libpri and asterisk-sounds, etc in line with it. e.g. 1.0.9 for it to work too? |
23:30.33 | Qwell | jeremywhiting: its recommended... |
23:30.39 | DarthClue | jeremywhiting: yes. |
23:30.41 | dack | DarthClue: 10 seconds |
23:30.44 | Qwell | overpaid? |
23:30.45 | jeremywhiting | ok, thanks |
23:30.51 | DarthClue | er..underpaid. |
23:30.55 | dack | gotta go, work on this later |
23:31.33 | *** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
23:32.12 | bdunn | If I want to record my own voice prompt - what should I search for on voip-info? |
23:32.29 | Qwell | record |
23:33.21 | bdunn | THanks.... I think that's the one this I didn't search on. :-( Duh |
23:34.36 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
23:34.37 | DarthClue | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Record |
23:35.46 | *** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net) |
23:35.49 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
23:35.51 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
23:36.06 | *** part/#asterisk Morex (~blah@host81-157-165-204.range81-157.btcentralplus.com) |
23:36.10 | Ariel_ | evening everyone |
23:36.27 | SarahEmm | hihi |
23:36.39 | brookshire | http://www.digium.com/downloads/gtkiaxyprov/ |
23:36.39 | SpaceBass | well after my cisco 7912 died during an update they are sending me another, no questions asked... thought that was nice |
23:36.43 | brookshire | yay! gtkiaxyprov |
23:37.03 | SpaceBass | hey Ariel_ |
23:37.51 | Ariel_ | SpaceBass, nice. I had a Polycom IP-500 speaker go out and they switch the phone with only me tell them the s/n. This was swapped out by Polycom directly. |
23:38.31 | SpaceBass | yeah, I had to tell cisco the s/n and no probs... pcuniverse.com on the other hand as still not registred my smartnets after 4 weks |
23:38.35 | SpaceBass | I'm about to call Visa |
23:40.49 | *** part/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
23:46.11 | *** join/#asterisk rv_weasel (~no@adsl-66-142-40-127.dsl.kscymo.swbell.net) |
23:47.09 | rv_weasel | can someone recommend a good IAX or sip phone for linux that is not harder to set than setting up all of asterisk |
23:47.39 | Ariel_ | rv_weasel, there is a linux version of xlite out there. |
23:48.20 | rv_weasel | sweet, ill look that up. thanks |
23:49.45 | SpaceBass | rv_weasel: for iax there is iaxcomm and gnonephone or something |
23:50.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
23:51.24 | rv_weasel | iaxcomm is RPMified for my distro!!! |
23:51.52 | Nugget | Linux is poo!!! |
23:52.28 | SpaceBass | yes, long live windows for workgroups |
23:52.52 | Hmmhesays | classic |
23:53.18 | Ariel_ | Nugget, ????? pooo?? oh you muct then be either an OSX or FBSD person. |
23:53.49 | Ariel_ | Commodore 128 (still have one) |
23:53.49 | rv_weasel | windows 3.1, without TCP/IP support!!! that is my fav |
23:54.02 | SpaceBass | I have my c64 |
23:54.21 | Nugget | I'm just in favor of using the right tool for the job. And there just aren't that many jobs where Linux fits that equation. |
23:54.31 | Nugget | zaptel is the only one I can think of off-hand |
23:54.42 | jeffgus | Nugget, iscsi server? |
23:54.49 | rv_weasel | i have calcs that are more powerfull than a C128 |
23:54.50 | jeffgus | with lvm backend? |
23:54.56 | jeffgus | and GFS? |
23:54.58 | Ariel_ | SpaceBass, I have an extra 1541 I need to sell Wonder how much I can get for it on Ebay. |
23:55.07 | SpaceBass | and my collection of original OS discs includes the first netware, os/2 warp... |
23:55.17 | rv_weasel | Nugget: how bout running the internat |
23:55.27 | Nugget | what the hell is an internat? |
23:55.27 | Ariel_ | Frist Netware I have one it's for the apple |
23:55.33 | SpaceBass | Ariel_: I've never looked up my IIe or IIgs or c64... they are just worth to much to me :) |
23:55.44 | SpaceBass | and by worht too much, I mean im lazy |
23:56.07 | rv_weasel | Nugget: only newbs freak about typos in chat. guess what that makes you |
23:56.28 | rv_weasel | which would explain you adversion to linux |
23:56.28 | Nugget | You're the guy who thinks that the internet runs on linux, but I'm the newb. I see. |
23:56.46 | SpaceBass | Ariel_: when I got my CNE 3.x there was still a netware client for apple... have that disc next to my netscape (mozilla) 0.8a disc (3.5 HD) |
23:56.54 | Nugget | linux could disappear today and "the internet" would be largely unaffected. |
23:57.03 | rv_weasel | Nugget: perhaps you should look at some data that is not published be MS |
23:57.18 | jeffgus | well the internet itself would be fine |
23:57.22 | Nugget | perhaps you should visit your ISP and see what they're running. |
23:57.24 | jeffgus | but a lot of web servers would disappear |
23:57.35 | rv_weasel | except for the fact that 60% or more of the websites would disappear |
23:57.56 | jeffgus | 60% is apache, not linux |
23:58.00 | SpaceBass | i will admit that i just tried linux on the desktop again, thinking I could salvage my intel workstation, and in a windows domain enviroment its just not ready... but its come a long way |
23:58.06 | jeffgus | but linux does run a good portion of apache servers |
23:58.09 | rv_weasel | ISP's cisco and the like. |
23:58.17 | SpaceBass | on the otherhand, I'd NEVER use a windows server for routing, nat, firewall... |
23:58.32 | kd5uzz | bx_dundi.c:1399: warning: implicit declaration of function `compress' |
23:58.32 | kd5uzz | pbx_dundi.c:1400: `Z_OK' undeclared (first use in this function) |
23:58.34 | Nugget | neither would I. |
23:58.36 | SpaceBass | so, I guess Im going to buy a powermac to replace my desktop |
23:58.49 | Nugget | I use routers for routing, as does most of the internet. |
23:58.53 | kd5uzz | any ideas? |
23:58.54 | rv_weasel | i am 100% linux now. except for my 166mhz laptop. that is win98 |
23:59.03 | SpaceBass | Nugget: what kind of routers? |
23:59.27 | Nugget | not linux boxes. :) |
23:59.39 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-209-168-226.dsl.sfldmi.ameritech.net) |
23:59.56 | jeffgus | FC "routers" run linux |