| 00:00.14 | alrs | er, 9112 |
| 00:00.18 | Qwell | http://www.ctiusa.com/solutions/customer-eng-solutions.asp |
| 00:00.18 | alrs | I don't know why I'm stuck o 9116 |
| 00:00.19 | Qwell | hotel phone |
| 00:00.20 | dFence | drmessano: the phones we have in our 2nd house are much more interesting for guests and the guests there are more likely to steal them (analog siemens phone with fancy lights) |
| 00:00.38 | dFence | guess how many have been stolen ever: 0 |
| 00:00.45 | drmessano | dFence: Excellent.. good luck |
| 00:01.00 | dFence | drmessano: the phones getting stolen is my least concern |
| 00:01.16 | JayTee52 | if you want to make sure none of your VOIP phones ever get stolen just buy Grandstreams. |
| 00:01.24 | dFence | lol |
| 00:01.27 | drmessano | Stolen isn't the problem |
| 00:01.33 | drmessano | Abuse is.. It's been said 3 or 4 times |
| 00:01.52 | dFence | same |
| 00:01.54 | drmessano | But again.. good luck on your install |
| 00:01.55 | alrs | drmessano: you're underestimating the 9112, they aren't flimsy like your linksys |
| 00:02.12 | drmessano | The linksys isn't flimsy |
| 00:02.15 | JayTee52 | so just buy 10 Grandstreams for the price of 3 good Polycoms and you'll do ok and then tack on 10 buck to the nightly room rates to cover the damage. |
| 00:02.58 | drmessano | But you're not gonna convince me a $50 SIP phone will last in a hotel |
| 00:03.42 | alrs | drmessano: the $50 one was analog, the 9112s are closer to $100 |
| 00:03.57 | drmessano | Have you ever tried to beat a hooker with a cordless phone? |
| 00:04.01 | alrs | and why you think a proven design like the Aastra (nortel) can't hack it is beyond me. |
| 00:04.13 | dFence | does anyone know a phone of the linksys spa922 caliber? (needed features: interal switch & poe) |
| 00:04.19 | drmessano | (nortel) <-- not impressed |
| 00:04.41 | alrs | I see decade-old Nortel handsets all over the place |
| 00:05.15 | drmessano | I see 15 year old NEC handsets all over the place too |
| 00:05.18 | drmessano | Doesn't make it right |
| 00:05.49 | dFence | drmessano: who's beating hookers!? |
| 00:05.59 | dFence | and why the hell did noone inform me!? |
| 00:06.49 | alrs | dFence: Uh, Aastra 9112i? |
| 00:06.51 | `Sauron | Because you're lame. |
| 00:06.58 | drmessano | If you smack a hooker with an Aastra, she's gonna shiv you.. Plain and simple |
| 00:07.33 | alrs | dFence: though it looks like you have to go up to the 9133i to get the eth bridge |
| 00:07.37 | dFence | hm... never tried an aastra to be honest... gonna add that to the list ;D |
| 00:08.58 | dFence | the 2nd eth-port is the n1 reason for the whole *-idea... would come way cheaper to equip every room with an ip-phone and extra ethernet port than a wifi-solution like that zyxel-stuff |
| 00:09.33 | drmessano | Or you could put in a few Cisco APs |
| 00:10.24 | dFence | drmessano: ill be leaving in september and once i'm gone that whole thing has to run by itself. i got panic-calls at 3am china-time because the router's lights were blinking ORANGE instead of green... |
| 00:11.22 | dFence | i worked my way through almost all major vendor sites, the only considerable solution is the hospitality-thing from zyxel for ~1200 bugs |
| 00:12.10 | ftp3 | anyone know someplace besides didx that I can get a good deal on wholesale usa dids? |
| 00:12.47 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:18.17 | dFence | when talking about POE does "inline" referr to "midspan", "endspan" or "go figure"? |
| 00:19.09 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 00:20.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:20.43 | dkwiebe | The Aastra phones aren't very heavy but they feel "good" |
| 00:21.17 | dFence | dkwiebe: is that also when whacking prostitutes or for regular use? :D |
| 00:21.46 | dkwiebe | dfence: prostitues are for whacking? lol |
| 00:21.55 | dkwiebe | dfence: regular use |
| 00:22.05 | dFence | dkwiebe: why, what did you think they're for!? |
| 00:22.24 | dFence | dkwiebe: ok... btw: aastra is not www.aastra.com, is it?! |
| 00:23.03 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-49-247.lns10.syd7.internode.on.net) |
| 00:23.05 | dkwiebe | yes it is |
| 00:23.18 | dFence | aaaa *panic* |
| 00:23.26 | dkwiebe | lol, what's up? |
| 00:24.49 | dFence | dunno the exact organization of aastra but in germany they're somehow related with detewe... had 2 detewe devices, both ended up costing us twice as much for maintenance as product-price itself |
| 00:24.56 | dFence | hm... should lay off the vodka |
| 00:25.35 | dkwiebe | like a maintenance agreement. You don't need those with their devices in Canada at least |
| 00:36.00 | unpaidbill | so do coupon codes actually exist for the digium store |
| 00:36.14 | unpaidbill | i want to get a discount on this here codec |
| 00:40.23 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 00:42.08 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ef5ed483b5168757) |
| 00:48.54 | unpaidbill | yea that's what i thought |
| 00:51.11 | drmessano | Did you just ask for a discount code for buying G729? |
| 00:51.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 01:01.01 | logi4023 | calls to my sip softphone are generating call progress (ringing signal) to the caller. Anyone knows why this is happening? |
| 01:01.10 | logi4023 | are not generating |
| 01:03.25 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 01:05.22 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
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| 01:14.23 | *** join/#asterisk VoiceCX (n=VoiceCX@216.10.136.139) |
| 01:15.40 | VoiceCX | do you guys ever think that someone may write a module for your trixbox or FreePBX to login to Sirius Radio Online to be used as a MOH module |
| 01:18.18 | *** join/#asterisk profounded (n=Bryan_Ru@c-68-82-34-163.hsd1.nj.comcast.net) |
| 01:20.57 | *** join/#asterisk GlobeTrotter (n=eric@196.40.26.98) |
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| 01:24.37 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.25) |
| 01:27.12 | profounded | hey quick question, i have a server setup in a remote location and i want to connect a sip phone to it. I would think it would be a matter of opening up port 5060 for udp and tcp and then port 1252 (which is the port that shows up when i run sip show peers).. what am i missing? |
| 01:28.03 | plik | 10000-20000 udp for rtp |
| 01:28.42 | profounded | thanks plik |
| 01:28.57 | plik | welcome |
| 01:35.28 | profounded | plik, the sip phone can be behind a firewall, its just the server that needs to have the ports open correct? |
| 01:35.37 | *** join/#asterisk geneg1 (n=gene@bas3-toronto01-1177779731.dsl.bell.ca) |
| 01:36.28 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 01:39.48 | lanning | profounded, yes, just the server (the sip phone's firewall must allow general outbound traffic, as usual) |
| 01:40.02 | profounded | thx lanning |
| 01:40.02 | lanning | also, you don't need port 1252 |
| 01:40.38 | profounded | figured, just added it because i was stuck, ty |
| 01:40.40 | *** join/#asterisk Frogzoo (n=Frogzoo@121.218.37.48) |
| 01:50.22 | ManxPower | VoiceCX: Most of us hate TrixBox and FreePBX |
| 01:53.21 | BeeBuu | hate for what? |
| 01:53.25 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 01:54.06 | ManxPower | BeeBuu: I'm sure each does for their own reasons. Personally, I hate them because their users come here and end up disrupting the channel when their users should be using the product support methods for the product they are using. |
| 01:54.24 | ManxPower | Also, only girlymen use a GUI |
| 01:54.48 | JayTee52 | lol |
| 01:55.56 | JayTee52 | I need help installing AsteriskNOW on Red Hat 3 but I can't find all my floppies, can someone help me? <<<< Joke example |
| 01:58.06 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 01:58.41 | ManxPower | Something like that. |
| 01:59.00 | JayTee52 | ManxPower, were you in here the other night for that? |
| 01:59.22 | outtolunc | nash! |
| 01:59.32 | JayTee52 | hehehe |
| 01:59.34 | ManxPower | There are many things to dislike about the GUI Asterisks, but the one that really gets my tail fur in an uproar is their users on this channel being offtopic. |
| 02:00.19 | ManxPower | If they don't want to support their product then they should close the channel and everyone can come here for all forms of Asterisk. But I'm pretty sure they don't want their channel flooded with Asterisk specific questions either. |
| 02:01.28 | JayTee52 | do the GUI based systems that are based on Asterisk still used the .conf files or have they replaced them with database or XML? |
| 02:01.30 | errr | Im having an issue with my voicemail. When it shows up in my email inbox the message attachment is a 0 size but it plays just fine and is not 0 size on the pbx |
| 02:01.33 | ManxPower | Lazyness is not a reason to come to a channel and ask off topic questions |
| 02:01.53 | JT | JayTee52: freepbx uses a sql backend which generates .confs |
| 02:01.57 | errr | any idea why I keep getting a 0 sized file in email instead of the actual message |
| 02:02.04 | ManxPower | JayTee52: Everyone that I've heard of has used complicated sets of include files and incredibly complicated scripts, macros, and AGIs |
| 02:02.37 | ManxPower | errr: Whatever is used to attach and build the message did something wrong 8-) |
| 02:02.45 | JayTee52 | my boss has a hardon for GUI's and keeps pushing me to find one for *. I keep telling him that it's not worth the effort. |
| 02:02.57 | ManxPower | I take it you have examined the full headers and MIME structure of the message? |
| 02:02.57 | errr | ManxPower: where can I kick it to make it do the right thing? |
| 02:03.14 | ManxPower | JayTee52: Is your boss a geek or a geek wannabe? |
| 02:03.53 | ManxPower | If so, challange him to build an Asterisk system prototype. |
| 02:04.01 | JayTee52 | my boss is an idiot that thinks Microsoft rules and is constantly bitching about their stock price. He wants to roll out Vista to everyone even though it won't run half our legacy apps. |
| 02:04.03 | ManxPower | Using A GUI |
| 02:04.16 | ManxPower | Then try to get all of the features of your current Asterisk to work in the GUI |
| 02:04.28 | JayTee52 | he also believes the world is 6000 years old and that dinosaurs coexisted with man. |
| 02:04.39 | ManxPower | JayTee52: He could either be an idiot or a geek wannabe |
| 02:04.52 | ManxPower | JayTee52: You are serious? |
| 02:04.59 | JayTee52 | his ego is constantly writing checks his intellect can't cash. |
| 02:05.11 | JayTee52 | yeah, he's born again hard core |
| 02:05.11 | ManxPower | Then perhaps he would take a challange |
| 02:05.29 | ManxPower | Honestly, I could not work with such a person. I'm a Devout Atheist. |
| 02:06.03 | errr | JayTee52: use freepbx to give it a gui then just do all the work in the conf files removing the files the gui will edit. If he just wants something to look at that will keep him busy ;) |
| 02:06.05 | JayTee52 | then we have half a thing in common, I'm just not devout about anything |
| 02:07.03 | JayTee52 | I wish sipX had support for IAX2, then I could have all my phones register to it and connect to Exchange UM and use * as my main server and PSTN gateway. |
| 02:07.35 | ManxPower | I'll tolerate people's beliefs but if they want to flash it in my face, I shall have a hard time keeping quiet. |
| 02:07.59 | errr | ManxPower: I dont know much about email headers but these seem to be right and there is a scetion for the msg001.wav in them |
| 02:08.02 | ManxPower | JayTee52: Why not use SIP |
| 02:08.15 | ManxPower | errr: pastebin the headers and I'll take a look |
| 02:08.18 | errr | ok |
| 02:08.25 | JayTee52 | he was born and raised in Indiana but want's to join the Sons of the Confederacy because he is such a civil war buff and refers to it as "The War of Northern Aggression" |
| 02:08.34 | JayTee52 | ManxPower, I do use SIP |
| 02:08.47 | ManxPower | JayTee52: I moved to the south from the north and MANY people down here call it that. |
| 02:09.16 | JayTee52 | I use SIP now to go between UM and * but I have to use sipX to transform UDP to TCP for UM because UM only speaks TCP. |
| 02:09.46 | ManxPower | Oh! Yes, I remember something like that being said somewhere. One of the stupidest things I've heard of in *years* |
| 02:10.35 | ManxPower | JayTee52: look at the 1.6 featureset. I'd not use it in production yet, but you can play with it and set up a prototype system. When 1.6 is stable, you will already know all you need to know about and I think 1.6 may support TCP |
| 02:10.55 | JayTee52 | if I could figure out how to do a transform from UDP to TCP using * 1.6 which supports SIP/TCP I'd do it to replace sipX |
| 02:11.23 | JayTee52 | ManxPower, I've already got a box setup with 1.6 but the docs are kinda scant |
| 02:11.51 | ManxPower | I can see the reason to use SIP/TCP. SIP is just call setup and teardown, that can be delayed slightly and should be reliable. |
| 02:12.09 | JayTee52 | according to the comments in the sip.conf file for 1.6 it does support sip tcp and tls |
| 02:12.18 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 02:13.42 | JayTee52 | I don't mind the Web GUI setup for sipX and I think it would be great for a non-technical person to use to administer the phone side of things but sipX won't use Digium hardware and I've already got 2 TDM04B cards and a TE212P card invested. |
| 02:14.11 | errr | ManxPower: http://fluxbox.pastebin.ca/1012305 |
| 02:15.05 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:16.42 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 02:17.05 | JayTee52 | I only have 4 more DVDs to watch to complete the entire 5 season Babylon 5 series. |
| 02:17.13 | gitguy | looks like the problems i had this morning with the pins and stuff were bandwidth problems |
| 02:17.34 | ManxPower | JayTee52: A nontechnical person will screw it up. Even if you train them, that training will eventually be lost along the long string of people that try their hand at admining the pbx |
| 02:17.39 | ManxPower | But I'm not cynical or anything |
| 02:17.53 | [TK]D-Fender | JayTee52, I borrowed the complete series and first 5 movies off a friend and had another rip it & I burned them. |
| 02:17.57 | gitguy | now i don't lose digits anymore with Read() and NoOp() |
| 02:18.04 | ManxPower | You have to *understand* telecom to admin any pbx |
| 02:18.04 | [TK]D-Fender | JayTee52, Working my way through 5 years of Andromeda right now :) |
| 02:18.45 | JayTee52 | oooh! I loved that show!!! |
| 02:18.51 | gitguy | ManxPower: was that for me? |
| 02:19.04 | [TK]D-Fender | JayTee52, I getting to. B5 sets the bar VERY high though... |
| 02:19.20 | JayTee52 | I got the first season of Babylon 5 used for 24 bucks then found the other seasons for 19.99 on sale at Best Buy |
| 02:19.40 | errr | ManxPower: so were the headers formed correctly? |
| 02:19.52 | JayTee52 | my all time favorite series only lasted 1 season. Firefly |
| 02:20.26 | [TK]D-Fender | JayTee52, thats next on my list to watch. Everyone says its awesome. |
| 02:20.42 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 02:21.01 | JayTee52 | [TK]D-Fender, the lead actor is a Canadian. Nathan Fillion, he's from Edmonton, Alberta |
| 02:21.37 | ManxPower | errr: If you use WAV49 format the files will basically be GSM (MUCH smaller) wrapped up in a .wav file that windows media players can play |
| 02:21.37 | errr | ManxPower: I tried using that the attachments are still 0 size |
| 02:21.51 | [TK]D-Fender | JayTee52, "thats nice" :) Don't care where they're from, only that they are good at what they were brought in to do. |
| 02:21.58 | ManxPower | errr: What mal reader are you using? |
| 02:22.07 | [TK]D-Fender | JayTee52, loved Farscape too.... miss that one. |
| 02:22.16 | JayTee52 | it's the characters and dialogue that make Firefly such an awesome show |
| 02:22.26 | errr | ManxPower: thunderbird. but even in the gmail web interface they are 0 size |
| 02:22.31 | JayTee52 | hehe, yeah. Friend of mine named her dog Frell |
| 02:23.18 | ManxPower | errr: I would have to refresh my limited memory of MIME but I don't know if you are supposed to have duplicate boundary tags, other than that it looks perfectly good to me |
| 02:23.54 | errr | ManxPower: if I use mutt from the cli to send the same voicemail it works fine and the message is not 0 size.. its only when asterisk sends it |
| 02:24.00 | errr | that it is 0 size* |
| 02:24.30 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:24.40 | ManxPower | Figure out what the difference between the two messages are |
| 02:27.03 | errr | ManxPower: the only diff is that the attachment is not 0 size so the header is much much larger sice it has all the data in it |
| 02:27.56 | *** join/#asterisk putnopvut (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 02:27.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 02:38.39 | *** join/#asterisk zackz (n=zack@64-136-221-136.dyn.everestkc.net) |
| 02:40.48 | zackz | is there a way i can make a "dialplan reload" reload like a | more in linux? |
| 02:42.44 | JayTee52 | zackz, why not just connect to the * server using putty or some other ssh client. The terminal window will be scrollable. |
| 02:43.30 | zackz | i cant scroll back far enough to see everything |
| 02:44.00 | rob0 | I don't think the asterisk console has a pager command. Increase your terminal's scrollback buffer. |
| 02:44.57 | errr | you could > the putput into a file |
| 02:45.01 | errr | output* |
| 02:45.19 | errr | use asterisk -rx then redirect to a file |
| 02:46.37 | zackz | errr: i did that but nothing shows up |
| 02:46.54 | profounded | im having a hard time getting asterisk to work with a remote sip phone. whenever i dial a number using the remote sip phone, i get an echo on the phone number i called. does sound like an rtp issue? i cant seem to configure port forwarding correctly if this is the case and im blaming it on my router |
| 02:47.20 | errr | ah you are right |
| 02:47.34 | zackz | i also tried piping it to tee but nothing hsows up that way either |
| 02:49.27 | zackz | and i odnt see anywhere in putty to increase the buffer DOH |
| 02:49.40 | profounded | putty can be increased |
| 02:49.54 | profounded | 1 sec |
| 02:50.39 | profounded | right click toolbar, change settings, window, and in there the line numbers |
| 02:50.58 | profounded | "lines of scrollback" |
| 02:51.45 | zackz | youre right |
| 02:52.16 | zackz | profounded: wha tkind of router do you have |
| 02:52.52 | ManxPower | Echo has nothing to do with any of that. Echo must be removed where the call is converted between PSTN/VoIP |
| 02:53.50 | errr | ManxPower: http://fluxbox.pastebin.ca/1012341 my logs seem to be indicating that app_voicemail may be trying to send a file that is not there instead of using the temp file that was actually created.. |
| 02:53.51 | profounded | versalink/westel clone d90-327... verison dsl router wifi point |
| 02:54.45 | zackz | ya i ManxPower is right |
| 02:54.50 | profounded | manxpower: i meant "echo test", when i speak into the phone that i called, my voice is repeated to me |
| 02:55.12 | profounded | i cannot hear the other party |
| 02:55.30 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:55.35 | lanning | you mean one way audio |
| 02:55.50 | profounded | yeah, it loops back |
| 02:56.25 | profounded | only for the receiving end, i get nothing for the party that made the call |
| 02:56.27 | lanning | are you calling an echo test extension? |
| 02:57.07 | profounded | no, works fine when i am inside the network, making me think its a port forwarding issue |
| 02:57.11 | zackz | one way audio is your router blocking rtp ports |
| 02:57.53 | drmessano | or NAT not set correctly |
| 02:58.18 | zackz | asterisk uses 10000-20000 for rtp i believe |
| 02:58.21 | zackz | udp |
| 02:58.26 | zackz | actually you can set it |
| 02:58.30 | profounded | the router first of all has a retarded dmz option that doesnt work, so i cant make use of that and use a better router/firewall |
| 02:58.34 | zackz | to whatever you want but that is default |
| 02:58.39 | drmessano | LOL |
| 02:58.47 | drmessano | editing rtp.conf is difficult? |
| 02:59.08 | profounded | i purposely set it to 10000-10010 in rtp.conf and port forwarded all those ports using udp just to test and still wont work |
| 02:59.13 | drmessano | forward 5060 UDP and whatever is in your rtp.conf |
| 02:59.23 | drmessano | Then set localnet and externhost/externip |
| 02:59.27 | drmessano | ~nat |
| 02:59.28 | jbot | well, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 02:59.51 | drmessano | ~sipnat |
| 02:59.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:59.56 | drmessano | There you go |
| 03:00.58 | profounded | drmessano, ill read into it, ty. i think im missing the set localnet and externhost/externip part of the equation. thx |
| 03:03.09 | drmessano | no probs |
| 03:04.34 | JayTee52 | well, I've been debating whether I should go to bed or install Red Hat 3 on my 486/33mhz system with 64MB of RAM and then compile Asterisk 1.2 on it. |
| 03:04.59 | drmessano | Got Floppy? |
| 03:05.09 | JayTee52 | got all my floppys :-) |
| 03:05.29 | VxJasonxV | I've created extensions by username. Is there a way, in the users.conf, to "alias" a numerical extension? |
| 03:05.38 | VxJasonxV | Of is that something that would have to be designed in the dialplan? |
| 03:05.55 | MooingLemur | dialplan. :) |
| 03:05.55 | VxJasonxV | s/Or/Or/* |
| 03:05.59 | VxJasonxV | :( |
| 03:06.11 | VxJasonxV | of or blah |
| 03:06.53 | VxJasonxV | that sucks, and is exceedingly annoying |
| 03:07.13 | rob0 | JayTee52: Bedtime for you young man. |
| 03:07.19 | JayTee52 | yep |
| 03:07.26 | JayTee52 | nite all |
| 03:07.30 | MooingLemur | VxJasonxV: I'd do it the other way around |
| 03:07.31 | rob0 | :) |
| 03:07.38 | VxJasonxV | hmmm |
| 03:07.42 | MooingLemur | extensions by number, and name as the alias in the dialplan |
| 03:07.49 | JayTee52 | a young man who can actually remember when Eisenhower was President :-) |
| 03:08.05 | MooingLemur | haha |
| 03:08.14 | JayTee52 | I turn 54 a week from this sunday |
| 03:08.29 | MooingLemur | so the 52 isn't quite right |
| 03:08.40 | JayTee52 | it was when I registered |
| 03:08.46 | MooingLemur | either as the year or age |
| 03:08.57 | MooingLemur | (I would have guessed the year) |
| 03:09.13 | JayTee52 | I wanted to just have jaytee like I do on Blitzed but Freenode ops are making me wait. the guy who has it registered hasn't logged in for over 12 weeks. |
| 03:09.36 | MooingLemur | hah |
| 03:09.43 | JayTee52 | 2 more days and I'm asking them again |
| 03:10.05 | JayTee52 | hey, I have lemurs at my work |
| 03:10.16 | MooingLemur | I bet they don't moo. |
| 03:10.20 | JayTee52 | ring tailed adn red ruffed |
| 03:10.25 | JayTee52 | nope they don't moo |
| 03:10.27 | rob0 | was born in Kennedy presidency |
| 03:10.40 | JayTee52 | he was a good President. Inspiring |
| 03:10.41 | MooingLemur | Carter :( |
| 03:10.47 | JayTee52 | not so much |
| 03:10.53 | MooingLemur | I'm young in this crowd :P |
| 03:11.04 | JayTee52 | nothing wrong with that |
| 03:11.39 | JayTee52 | just don't burn the candle at both ends all the time, try to save some wick for when you get to be my age |
| 03:12.46 | MooingLemur | I think my stress levels are reasonable most of the time :) |
| 03:12.52 | JayTee52 | anyhow, hope all of you have a pleasant evening and a great day tomorrow. Good nite! |
| 03:13.01 | MooingLemur | take care |
| 03:13.04 | drmessano | Ohhhhh "wick" |
| 03:13.05 | MooingLemur | d'oh |
| 03:13.09 | MooingLemur | hahah |
| 03:13.37 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 03:15.57 | zackz | hahahaha, i was watching FSN, and there was a message that came on that said "due to the length of the previous program, we now join the regular program already in progress" and it went to colored bars and tone |
| 03:17.56 | Braxus | lol |
| 03:18.45 | profounded | wow drmessano, that did it!! thank you very much! |
| 03:20.30 | zackz | i went through two different linksys routers trying to run sip over vpn tunnel, neither one worked, im never buying linksys again |
| 03:22.25 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583156.dsl.bell.ca) |
| 03:24.30 | mmlj4 | this is definitely a strange twist: VoIP over ham radio: www.arrl.org/qst/2003/02/VoIP.pdf |
| 03:26.02 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:26.36 | gitguy | can asterisk do sip over tcp? |
| 03:27.10 | mmlj4 | yes, if you like lag and stuff |
| 03:27.34 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 03:27.48 | bkw__ | mmlj4: wtf you talking about sip over TCP? and lag? you are talking out your ass |
| 03:27.59 | bkw__ | TCP is a MUST in the RFC not an optional thing |
| 03:28.27 | mmlj4 | bkw_: fragmentation for starters... there's a reason why UDP is preferred |
| 03:28.35 | bkw__ | fragmentation? |
| 03:28.44 | bkw__ | mmlj4: you seem to not know what you're talking about |
| 03:28.46 | mmlj4 | never mind |
| 03:28.56 | bkw__ | let me outline the dumbest thing in sip |
| 03:29.00 | bkw__ | the 64k thing |
| 03:29.03 | mmlj4 | one of use seems that way, yes |
| 03:29.10 | bkw__ | if the packet is over 64k it falls to TCP if it can't then it falls back to UDP |
| 03:29.21 | bkw__ | thats the dumb one |
| 03:29.26 | bkw__ | but their is NO lag on in TCP vs UDP |
| 03:30.15 | bkw__ | the other nice thing about TCP is the connection stays open |
| 03:30.25 | bkw__ | which can help with nat in some cases |
| 03:30.46 | *** join/#asterisk dawebber (n=dawebber@adsl-75-24-187-185.dsl.ltrkar.sbcglobal.net) |
| 03:31.00 | bkw__ | does asterisk do TCP yet? |
| 03:31.19 | bkw__ | I did know their was a patch for it but haven't been following to see if it was added |
| 03:31.33 | mmlj4 | gitguy: as i was saying, sure, SIP over TCP works, but you won't want to use it, the practical quality verses UDP just isn't there |
| 03:31.54 | bkw__ | again |
| 03:31.56 | bkw__ | talking out your ass |
| 03:32.12 | bkw__ | mmlj4: please go get a clue about it before you start yapping your trap |
| 03:32.37 | mmlj4 | bkw_: please list one streaming protocol in widespread use that does TCP |
| 03:32.53 | bkw__ | again you have no clue what you're talking about |
| 03:32.53 | mmlj4 | this is silly |
| 03:32.57 | bkw__ | SIP is Signaling |
| 03:32.59 | bkw__ | it goes over TCP |
| 03:33.02 | bkw__ | the Media is still UDP |
| 03:33.14 | mmlj4 | now you're starting to make sense |
| 03:33.22 | bkw__ | apparently you missed that fact |
| 03:33.34 | bkw__ | TLS is always TCP till DTLS is standard |
| 03:33.41 | drmessano | Who gives a crap about SIP quality? |
| 03:33.47 | drmessano | SIP is just signalling |
| 03:33.54 | drmessano | It can be so-so for all I care |
| 03:33.56 | bkw__ | drmessano: I think he was confusing media with signaling |
| 03:34.00 | drmessano | Give me reliable RTP |
| 03:34.06 | bkw__ | drmessano: then you need SCTP |
| 03:34.15 | bkw__ | but reliable media is one of those tricky things |
| 03:34.19 | bkw__ | you have to accept losses and move on |
| 03:34.27 | drmessano | Thats not my point |
| 03:34.51 | drmessano | Arguing over the quality of SIP is stupud |
| 03:34.55 | drmessano | stupid too |
| 03:35.02 | bkw__ | tcp vs udp isn't about quality |
| 03:35.18 | drmessano | [23:26] <mmlj4> bkw_: fragmentation for starters... there's a reason why UDP is preferred |
| 03:35.25 | drmessano | Thats not what I was reading |
| 03:35.46 | bkw__ | drmessano: read the RFC it has this really stupid rule about packets over the MTU |
| 03:35.56 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 03:36.02 | drmessano | I have better things to do than read the RFC |
| 03:36.15 | bkw__ | thats fine they are mind numbing |
| 03:36.23 | *** part/#asterisk dawebber (n=dawebber@adsl-75-24-187-185.dsl.ltrkar.sbcglobal.net) |
| 03:36.23 | bkw__ | but i you implement anything you had better know them well |
| 03:36.58 | drmessano | Deploying and implementing are two different things |
| 03:37.07 | bkw__ | well the other thing is TCP isn't optional |
| 03:37.12 | bkw__ | its a requirement .. aka a MUST |
| 03:37.22 | bkw__ | you can't optionally support TCP if you wish to be compliant |
| 03:37.23 | mmlj4 | yes, i knew SIP was just signalling, but yes, i got them confused, my bad |
| 03:37.32 | bkw__ | mmlj4: thats ok.. its late |
| 03:37.40 | drmessano | Hmm |
| 03:37.48 | drmessano | Not sure how that matters here |
| 03:38.06 | bkw__ | TCP has a few things going for it.. the reliable delivery is one |
| 03:38.14 | drmessano | Im pretty sure my phones don't care that Asterisk hasn't supported TCP until now |
| 03:38.15 | bkw__ | and sip info over TCP is going to be ordered properly |
| 03:38.16 | zackz | UDP is faster becuase it doesnt check to see if the packet was delivered |
| 03:38.30 | bkw__ | zackz: not 100% true |
| 03:38.38 | zackz | well, the protocol doesnt check at all |
| 03:38.38 | bkw__ | you can't tell speed differences in UDP vs TCP |
| 03:38.42 | zackz | you can do it in software |
| 03:39.00 | bkw__ | the bit thing is the connection being open and active all the time is a plus in some cases |
| 03:39.07 | bkw__ | and TCP is good over laggy satellite links |
| 03:39.12 | bkw__ | where UDP would fail |
| 03:39.29 | bkw__ | each has some advantages |
| 03:39.37 | bkw__ | having both is nice |
| 03:39.49 | zackz | udp is not generally used for data that needs to be 100% |
| 03:39.57 | bkw__ | that is true |
| 03:40.06 | bkw__ | but i'm not sure about you but I like my phone calls going thru 100% |
| 03:40.12 | bkw__ | :P |
| 03:41.49 | *** part/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 03:42.06 | zackz | you must not own a cell phone then :) |
| 03:42.30 | drmessano | He's a troll anyway |
| 03:42.51 | drmessano | I was waiting for the FreeSWITCH rhetoric |
| 03:43.48 | file | yawns |
| 03:43.55 | mmlj4 | who? not me, I hope |
| 03:44.48 | drmessano | No |
| 03:45.05 | drmessano | The one that just PM'ed me to warn me he has eyes everywhere |
| 03:45.17 | drmessano | and to "watch it" |
| 03:45.24 | file | drmessano: hrm? |
| 03:45.53 | drmessano | Yeah |
| 03:46.15 | rob0 | I'm an Asterlink customer, very pleased with it, so I stay out of the fights. :) |
| 03:46.27 | mmlj4 | ok, so s/SIP/IAX2/ and I'll restate my argument |
| 03:46.28 | drmessano | I was also told that this channel is logged publicly |
| 03:46.33 | drmessano | because, you know.. I never knew that |
| 03:46.36 | drmessano | Oh, wait.. |
| 03:46.59 | gitguy | didn't bkw contributed to asterisk? |
| 03:47.09 | rob0 | quite a bit IIRC |
| 03:47.10 | mmlj4 | it is logged, just about all networks/channels of consequence were stealth-logged for years |
| 03:47.12 | file | he did, long ago |
| 03:47.16 | drmessano | [23:41] <bkw__> troll I am not... better watch it.. I have eyes every where |
| 03:47.29 | jblack | so do potatoes. |
| 03:47.54 | drmessano | and it's "everywhere" |
| 03:47.57 | drmessano | one word |
| 03:48.01 | zackz | everwhare |
| 03:48.02 | rob0 | I will say it makes sense to me to use TCP for the control channel, but I don't know. |
| 03:48.14 | mmlj4 | rob0: I'd agree |
| 03:48.17 | drmessano | Well |
| 03:48.23 | drmessano | Asterisk has been around for........? |
| 03:48.30 | zackz | 1999 |
| 03:48.30 | file | awhile. |
| 03:48.32 | rob0 | the media pretty much HAS to be udp |
| 03:48.48 | Corydon76-dig | He unfortunately burned his bridges when he left, and only relatively recently has tried rebuilding them |
| 03:48.49 | drmessano | and I haven't heard a lot of cries of "ZOMGGGGG MY CALLS ARE DROPPING BECAUSE ASTERISK USES SIP UDP AND NOT TCP" |
| 03:48.53 | drmessano | So... |
| 03:49.15 | Corydon76-dig | Unfortunately, nobody wants want to be on the bridge again when he sets fire to it |
| 03:49.16 | drmessano | I welcome the addition of TCP to 1.6... but come on.. |
| 03:49.56 | drmessano | Acting like "Yay, Asterisk finally supports the RFC" really discounts the fact that Asterisk has been kicking ass for years |
| 03:50.23 | rob0 | googles for fire resistant hang gliders |
| 03:50.36 | Corydon76-dig | That's because as a design goal, we have interop above strict compliance |
| 03:51.11 | drmessano | Corydon76-dig: But aren't people whining about the RFC much more important than asterisk actually working?? |
| 03:51.25 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 03:51.47 | bkw__ | get this straight I don't care to rebuild any bridges with anyone.. just help people build really kick ass things related to telephony |
| 03:51.52 | bkw__ | NEXT!!! |
| 03:52.13 | bkw__ | I"m glad I burned them.. or C4'ed them. |
| 03:52.48 | bkw__ | anyway l8tr |
| 03:52.48 | *** part/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 03:53.13 | Corydon76-dig | and he wonders why people are reluctant to work with him |
| 03:53.30 | drmessano | O.o |
| 03:53.40 | Corydon76-dig | feeds the troll some more |
| 03:53.54 | file | is glad he defected |
| 03:54.03 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:54.22 | drmessano | just wants someone to build a good Win32 PBX system he can run on Windows 98 |
| 03:54.28 | rob0 | haha |
| 03:54.34 | drmessano | Can't we all just get along for a good cause? |
| 03:54.34 | rob0 | 3.1 |
| 03:54.53 | Corydon76-dig | thought Win98 and "good" were opposites... |
| 03:55.10 | zackz | win98 is better than vista |
| 03:55.15 | drmessano | lol |
| 03:55.22 | jblack | so, is bkw a loser now? |
| 03:55.30 | jblack | is he still contributing at all? |
| 03:55.42 | Corydon76-dig | He is not contributing to Asterisk at all, no. |
| 03:55.45 | file | he does not contribute any longer |
| 03:56.13 | rob0 | they're working on freeswitch? |
| 03:56.14 | Corydon76-dig | I wouldn't call him a loser, though. Diva, yes. |
| 03:56.27 | file | nods to rob0 |
| 03:56.33 | Corydon76-dig | Prima donna, you betcha. |
| 03:56.41 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 03:57.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:58.19 | *** join/#asterisk keith4__ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 03:58.37 | *** join/#asterisk paulproteus (n=paulprot@wide-rose.makesad.us) |
| 03:58.59 | keith4__ | so, I can get an extension to play out the console using ALSA, no problem. how do I play DTMF or say digiits or playback a gsm file out the console? |
| 03:59.01 | paulproteus | Is it possible to get a US-based service provider that allows me to receive SMS messages to a phone number served by my Asterisk? |
| 03:59.47 | *** part/#asterisk zackz (n=zack@64-136-221-136.dyn.everestkc.net) |
| 04:04.18 | *** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 04:05.09 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 04:05.12 | *** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 04:06.10 | logi4023 | calls to my sip softphone are not generating call progress to the caller. anyone knows why? |
| 04:06.23 | logi4023 | using dial() cmd. |
| 04:06.34 | logi4023 | softphone is xlite |
| 04:06.53 | Corydon76-dig | keith4__: just create a new extension with what you want to do |
| 04:07.10 | Corydon76-dig | keith4__: and you can create extensions from the CLI, too |
| 04:08.27 | Corydon76-dig | keith4__: "dialplan add extension....." |
| 04:12.11 | keith4__ | The extension alread exists. Currently, if I call '88', Console/dsp auto-answers. and anything I say into the phone, comes out line-out of the soundcard. All I have at extension 88 is Dial(Console/dsp) and then Hangup |
| 04:12.16 | keith4__ | this setup works fine |
| 04:12.50 | keith4__ | but I want to play a tone over Console/dsp too |
| 04:13.05 | Corydon76-dig | keith4__: Take a look at the AMI Originate command |
| 04:13.18 | Corydon76-dig | keith4__: or the CLI originate command |
| 04:18.17 | Nasra | any1 know of a video for instruction to install and running Asterisk from scratch with procedures since I am new ? |
| 04:19.17 | LARefugee | Nasra: See youtube |
| 04:19.36 | Nasra | youtube? |
| 04:19.45 | LARefugee | youtube.com |
| 04:20.05 | Nasra | well....let's just give a try then.... |
| 04:20.08 | Nasra | and thanks alot |
| 04:20.42 | mmlj4 | Nasra: or you can use trixbox or another of the cute-and-fuzzy distros that basically Just Work and are easy to administrate |
| 04:21.01 | [TK]D-Fender | video? Ancient useless junk. Only one I recall was a revision3 thing from over 3 years ago |
| 04:21.15 | Nasra | what I want is to learn wih my own mistakes from scratch |
| 04:21.21 | LARefugee | Here's one: http://youtube.com/watch?v=SQb71Y_X4yo |
| 04:21.33 | [TK]D-Fender | Read the book that people here kindly contributed 2 editions of and we can download freely. |
| 04:21.35 | Nasra | okay |
| 04:22.14 | Nasra | D-Fender I am reading the book that is why I don't participate in the chat just reading.. |
| 04:22.22 | Nasra | learning alot though |
| 04:22.59 | mmlj4 | reading is fundamental] |
| 04:23.06 | Nasra | and the needs more diagrams for newbies |
| 04:23.21 | drmessano | diagrams? |
| 04:23.37 | Nasra | like pictures etc.... |
| 04:24.15 | drmessano | Why does everyone find it necessary to define terms when the validity of the use of the term was questioned, not the meaning of it. |
| 04:24.53 | Nasra | drmessano: there are ppls like me.... |
| 04:25.04 | adeel | is it possible to hide the notify updates in the verbose CLI output? e.g.logger.c -- Extension changed 6014 new state Idle for Notify User 6036 |
| 04:25.26 | drmessano | lower your verbosity? |
| 04:25.35 | adeel | drmessano, i have it on 3 |
| 04:25.41 | drmessano | adeel: try 2 |
| 04:25.52 | drmessano | 2 < 3 |
| 04:26.33 | adeel | i wish there was a table that indicated at what verbose levels different components will log to |
| 04:26.46 | Nasra | mmlj4: just burned centOs 5.1 to test and learn....so I am getting there slowly....currently running Ubuntu 8.04 |
| 04:26.46 | adeel | i'm sure i can figure it out by diving/grepping through the source |
| 04:27.23 | adeel | drmessano, seriously? 2<3?? wow...that explains why i failed 2nd grade multiple times...i always thought 2>3 ....next thing you're going to say is that 1+1 doesn't equal 3! |
| 04:27.26 | adeel | =cp |
| 04:27.42 | keith4__ | ugh. centos |
| 04:28.03 | drmessano | adeel: I truly believe you |
| 04:28.12 | adeel | drmessano, hahahah |
| 04:28.17 | drmessano | ;) |
| 04:28.37 | drmessano | and for the record |
| 04:28.47 | drmessano | I don't think boolean terms came until 3rd grade |
| 04:28.59 | adeel | depends on the city/county/state |
| 04:29.05 | drmessano | Well... |
| 04:29.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 04:29.15 | adeel | i've gone to a lot of different schools |
| 04:29.34 | adeel | 10 different schools in 12 years |
| 04:29.41 | *** join/#asterisk Lin (n=igormorg@unaffiliated/lincity) |
| 04:29.57 | drmessano | In NJ, where the school system I was in was selected as one of the top 6 in the nation, it was 3rd grade... here in Georgia, which is 48th in Education, I think that was a college prep class in 11th grade |
| 04:29.57 | Lin | good morning(ugt) |
| 04:30.14 | adeel | hell, in 9th grade we were covering decimal places again and people were failing those tests |
| 04:31.07 | adeel | although, to be fair, that was in Canada |
| 04:31.40 | drmessano | In NJ, we had school system that had a school set up for all of the 6th graders in the township, so Jr High was 3 years, high school ended up being 3 years.. I moved here in 9th grade.. the first week.. I went from being a "senior" in Jr High to a real HS Freshmen... |
| 04:31.51 | drmessano | Well... First week I am here, we have an english test |
| 04:32.05 | drmessano | From the McGraw Hill 7th grade english book |
| 04:32.13 | drmessano | Said it at the bottom of the page |
| 04:32.16 | drmessano | I cried |
| 04:32.18 | adeel | hahaha |
| 04:32.47 | adeel | high school was a joke, and i was in the best school in the state |
| 04:33.03 | drmessano | I slept from 10th grade thru 12th grade |
| 04:33.13 | adeel | likewise |
| 04:33.29 | adeel | but even then, i STILL can't * to behave properly with DTMF |
| 04:33.31 | drmessano | I graduated with a 69.something average.. which was barely passing.. |
| 04:33.46 | drmessano | and now I make more money than most of the people I went to school with.... |
| 04:34.28 | adeel | no offense, but that typically isn't that difficult...hell call girls make more money than most people |
| 04:34.34 | drmessano | lol |
| 04:34.35 | adeel | like the chick with the mayor of new york |
| 04:34.43 | adeel | err governor |
| 04:35.22 | drmessano | Well, my inadequencies make it necessary for me to bolster my self esteem with comparison of monetary compensation |
| 04:35.34 | adeel | fair enough |
| 04:35.36 | drmessano | lol |
| 04:36.01 | adeel | seriously though....on a PURE SIP connection, * will randomly ignore some DTMF inputs |
| 04:36.07 | adeel | and will log that it's ignoring them |
| 04:36.45 | drmessano | Particular ITSP? |
| 04:36.50 | adeel | e.g. [May 6 13:22:56] DTMF[29746] channel.c: DTMF begin ignored '5' on SIP/6010-b68f8508 |
| 04:37.07 | adeel | well i don't think it's an ITSP problem |
| 04:37.29 | adeel | these are all being done on Polycom 601's or 330's with the latest firmware |
| 04:37.46 | adeel | using RFC2833 |
| 04:37.50 | file | what is the channel doing? |
| 04:38.20 | adeel | file, nothing out of the ordinary...there's nothing in the sip history or messages that seems incorrect |
| 04:38.32 | file | let me rephrase... |
| 04:38.34 | adeel | i think it's just that * ignores some inputs |
| 04:38.48 | file | is it two channels bridged together? it is in an IVR? |
| 04:38.48 | file | erm is it in |
| 04:38.58 | adeel | ohh, 2 channels bridged |
| 04:39.01 | adeel | outbound call |
| 04:39.09 | file | both sides set to dtmfmode=rfc2833? |
| 04:39.42 | adeel | one side is rfc2833 and one side is set to auto |
| 04:40.08 | adeel | but other DTMF signals are passed through |
| 04:40.11 | adeel | in the same call |
| 04:40.23 | file | are the DTMF digits being hit fast? |
| 04:40.45 | adeel | the users claim they're not |
| 04:41.00 | adeel | but i've looked at the minimum time, and it's 40 ms....there's no human way to do get that speed |
| 04:41.04 | adeel | unless polycom is buffering them |
| 04:41.13 | file | does it show it receiving an end DTMF? |
| 04:41.18 | adeel | yes |
| 04:42.18 | file | file an issue report with full log with the DTMF logging and I'll look at it tomorrow |
| 04:42.44 | adeel | sure |
| 04:42.46 | adeel | thanks |
| 04:43.58 | *** join/#asterisk s0lid (n=s0lid@210.213.198.7) |
| 04:47.59 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 04:48.23 | drmessano | I guess now is a good time to pack up the asterisk source and post it on Bittorrent |
| 04:49.04 | drmessano | Asterisk_PBX_SOURCE_CODE_LEAKED-ZoMg |
| 04:50.12 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:53.32 | *** join/#asterisk CaRb0n^ (n=playa@202.133.72.212) |
| 04:55.15 | [TK]D-Fender | drmessano, * has tons of leaks as it is... most of them memory.... |
| 04:55.31 | adeel | haha |
| 04:56.30 | drmessano | It also has limited hardware support.. for example, I still can't use my USR Robotics Sportster 14.4 WinModem as an FXO device |
| 04:56.33 | drmessano | That... is lame |
| 04:59.18 | keith4__ | any way to play audio out Console/dsp from the dialplan? |
| 04:59.31 | keith4__ | perhaps during a call to Console/dsp ? |
| 04:59.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:01.02 | adeel | there's supposed to be |
| 05:01.10 | unpaidbill | what the heck... how does asterisk pick the codec to use, is by the order they're allowed or is there something else? it's defaulting to ulaw if i put both ulaw and g729 in the sip config |
| 05:01.23 | unpaidbill | but if i have only g729 i cant call my other asterisk server because of no compatible codecs |
| 05:01.40 | unpaidbill | and i want it to use g729 when it dials to the voip provider :/ |
| 05:02.03 | unpaidbill | but fall back to ulaw when dialing that provider if the endpoint doesnt support g729! |
| 05:02.04 | adeel | unpaidbill, are you using sip? |
| 05:02.08 | unpaidbill | yeah |
| 05:02.16 | adeel | unpaidbill, the order in which they're listed in sip.conf matters |
| 05:02.28 | adeel | so try listing g729 before ulaw |
| 05:02.36 | unpaidbill | well that is confusing me because it always picks ulaw, and ulaw is listed after g729 |
| 05:02.43 | unpaidbill | if i only allow g729 it uses it |
| 05:03.43 | adeel | unpaidbill, i'd recommend double checking to make sure something else isn't over-riding the g729...otherwise, it could be the endpoint tries to force ulaw rather than g729 |
| 05:03.48 | keith4__ | adeel: can ChanSpy or ExtenSpy play sound out one end of a call? |
| 05:04.19 | adeel | keith4, i'm not sure i follow...'play sound out one end of a call'? |
| 05:04.20 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 05:04.43 | keith4__ | uh... let's say I pickup a zap channel and dial an extension that just dumps to Console/dsp, which is set to autoanswer |
| 05:04.55 | keith4__ | now I have a call to the soundcard's audio out port |
| 05:05.02 | adeel | sure, you can do that |
| 05:05.09 | keith4__ | I am doing that. it works perfectly |
| 05:05.31 | adeel | ok, so then what are you trying to do? |
| 05:05.52 | keith4__ | but now I want to, maybe... SayDigits() out the audio port |
| 05:06.02 | keith4__ | or, play an mp3 out the audio port |
| 05:06.48 | keith4__ | like, to announce a page or something |
| 05:06.58 | adeel | keith4, while a call is active? |
| 05:07.15 | keith4__ | yes |
| 05:07.24 | adeel | i mean, while you already have sound being dumped to Console/dsp? |
| 05:07.27 | keith4__ | or before it's active |
| 05:07.43 | adeel | if the soundcard is not being used, then you should be able to pipe anything you want to it |
| 05:07.57 | keith4__ | oh, like just use a system call? |
| 05:08.06 | drmessano | yes |
| 05:08.16 | keith4__ | hmmm |
| 05:08.17 | adeel | if it is in use, then you'll probably need to setup DMIX, if using alsa, |
| 05:08.25 | keith4__ | yes, using alsa |
| 05:08.30 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 05:08.39 | keith4__ | i was just about to ask about a possible contention problem... |
| 05:08.40 | [hC] | is it possible to get asterisk 1.2 chan_sip to listen on two ports? |
| 05:08.54 | adeel | otherwise you run into the typical contention problem |
| 05:09.28 | keith4__ | http://alsa.opensrc.org/DmixPlugin says dmix is enabled by default |
| 05:09.37 | adeel | keith4, but i'm not too sure how *s internals are in terms of console audio |
| 05:09.48 | *** join/#asterisk mikong (n=gones@203.193.37.251) |
| 05:09.50 | adeel | [hC], you can try setting the port= value again |
| 05:10.25 | adeel | keith4, best advice is to try it out...i don't really foresee why you shouldn't be able to do what you want |
| 05:10.27 | [hC] | adeel: that doesnt work. |
| 05:10.30 | [hC] | adeel: it just takes the first one. |
| 05:10.47 | adeel | [hC], hmmm....well unless you're able to bind to another IP then i don't think it can listen on 2 different ports |
| 05:10.57 | [hC] | iptables to the rescue. |
| 05:10.58 | [hC] | :) |
| 05:11.03 | adeel | [hC], yep |
| 05:11.44 | keith4__ | adeel: alright, thanks. I'll try it |
| 05:12.40 | adeel | np |
| 05:13.49 | mikong | HI , have somebody use the * for callback server ? |
| 05:14.15 | adeel | mikong, as a callback server? no...haven't used it as a server, but i have used the callback functionality and it works |
| 05:15.00 | keith4__ | what's a callback server? |
| 05:15.11 | adeel | you call the box, it hangs up and calls you back |
| 05:15.14 | mikong | How to do this ? the callback functional ? |
| 05:15.23 | keith4__ | ah, interesting |
| 05:15.37 | adeel | or can connect 2 different people together by initiating the calls |
| 05:15.40 | adeel | and then bridging it |
| 05:18.12 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 05:18.22 | mikong | But the * how to connect 2 different people ? |
| 05:18.53 | adeel | using the app_callback iir |
| 05:18.56 | adeel | er iirc |
| 05:20.21 | mikong | Is it include the normal version ? |
| 05:22.33 | adeel | should be |
| 05:25.47 | mikong | sure ? |
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| 05:37.45 | adeel | i don't see why it wouldn't...unless you didn't enable it while compiling |
| 05:37.45 | adeel | file, http://bugs.digium.com/view.php?id=12615 |
| 05:37.45 | adeel | thanks |
| 05:38.22 | unpaidbill | apparently 1.2.27 doesnt choose the codecs correctly |
| 05:38.54 | unpaidbill | it isnt honoring the priority i set in sip.conf general, but it works on a 1.4.19 server |
| 05:39.14 | drmessano | unpaidbill |
| 05:39.16 | mikong | thanks |
| 05:39.34 | drmessano | If you need g729 for a specifier peer, ONLY ALLOW G729 for THAT PEER |
| 05:39.41 | drmessano | Specific |
| 05:39.50 | drmessano | disallow=all and allow=g729 |
| 05:39.52 | drmessano | Simple |
| 05:40.33 | [hC] | anyone know of a sip client on osx that isnt as bad as x-lite? |
| 05:40.45 | [hC] | x-lite seems to not even send packets out anymore for some reason |
| 05:40.59 | drmessano | Maybe your OSX is broken |
| 05:41.03 | drmessano | Hard to believe, I know |
| 05:41.17 | [hC] | Its not likely. X-lite has a history of doing shit like this in osx |
| 05:41.24 | unpaidbill | drmessano no, that isnt it at all. |
| 05:41.25 | [hC] | xmeeting (sip client) works, but has no dtmf pad,. |
| 05:42.10 | drmessano | unpaidbill: What are you trying to do? |
| 05:42.35 | unpaidbill | im trying to set it to use g729 by default, but fail over to alaw if the endpoint doesnt support g729 |
| 05:42.38 | unpaidbill | disallow=all |
| 05:42.44 | unpaidbill | allow=g729,alaw |
| 05:42.53 | unpaidbill | in general, and in the specific config |
| 05:42.58 | unpaidbill | and it ALWAYS uses alaw |
| 05:43.08 | unpaidbill | but on 1.4, it goes by the order |
| 05:43.19 | unpaidbill | and for this specific machine i cant use 1.4 |
| 05:43.27 | drmessano | Try putting it on two lines? |
| 05:43.33 | jblack | <PROTECTED> |
| 05:43.39 | unpaidbill | yeah on 2 lines it doesnt do it either |
| 05:43.47 | unpaidbill | yes i can use g729 fine if it's the only codec i specify |
| 05:44.23 | unpaidbill | sip show settings... Codecs: g729,alaw,ulaw,gsm |
| 05:44.41 | drmessano | hmmm |
| 05:44.53 | unpaidbill | so i would assume that means priority for g729.. but no... 1.2 wont have it! |
| 05:45.04 | unpaidbill | unless there's something im missing that isnt needed in 1.4 |
| 05:45.55 | drmessano | Do you have the codec specified in some other place? |
| 05:46.35 | unpaidbill | it's in [general] and the user specific configuration |
| 05:46.38 | unpaidbill | in sip.conf |
| 05:46.41 | unpaidbill | and it's ordered the same way in both |
| 05:46.49 | unpaidbill | g729,alaw,ulaw,gsm |
| 05:47.04 | unpaidbill | im gonna buy another damn g729 license and test this on the 1.4 box |
| 05:47.12 | unpaidbill | stupid g729. |
| 05:47.14 | unpaidbill | hehe |
| 05:47.24 | drmessano | I dont ever specify a codec in a peer definition unless im overriding something |
| 05:47.36 | unpaidbill | well |
| 05:47.41 | unpaidbill | let me try removing it and see what happens |
| 05:47.53 | unpaidbill | i was just doing it because nothing else was working and i started going nuts with the config |
| 05:48.18 | [hC] | ok so i found x-lite's problem |
| 05:48.35 | [hC] | its sending my local ip in the sip headers instead of my public ip since presumably it coudlnt figure out what my public ip was. |
| 05:48.39 | [hC] | and there's no way to force set it |
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| 05:51.00 | unpaidbill | yea that didnt fix it |
| 05:51.11 | unpaidbill | i am a sad man. |
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| 06:40.18 | Lin | Hi there. Im having problems with SIP behind NAT ADSL routers. I have configured my router to FORWARD everything coming from 5060-5070 and 10000-20000 udp to my SIP Phone, but Asterisk keeps complaining about Maximum retries exceeded on transmission, testing with IAX2 works flawlessly. Anything else should be done? best regards. |
| 06:49.08 | drmessano | 1. Stay in channel more than 8 minutes.. this isnt Dell 24/7 tech support |
| 06:49.15 | drmessano | 2. ?????? |
| 06:49.32 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:09.09 | adeel | 2. if you want priority service, it's $50 bucks an hour |
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| 08:24.49 | tzafrir_home | OT: any idea how I can rename a page in voip-info.org ? |
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| 08:38.53 | E-bola | Can somebody help me figure out how to get the following functionality in my dialplan. I want a msg to be played when ppl call int that says "All lines are busy please wait, or press 1 to leave a message" which obviously should let the caller press 1 to go to voicemail or let him wait for somebody to answer the phone |
| 08:39.56 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 08:42.35 | mikong | Background() |
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| 08:45.05 | badcfe | hello. i would like to set a 420p wildcard in E1 mode by software (without touching the jumpers). how do i do this? |
| 08:45.36 | badcfe | the only configs for such a card is zapata.conf and zaptel.conf right? |
| 08:47.12 | E-bola | mikong: I tried using background but it never moves on, it just seems to wait for something to be entered forever |
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| 08:47.19 | E-bola | do i need to use some sort of timeout command or similar? |
| 08:48.12 | mikong | show your dialplan |
| 08:48.59 | E-bola | http://pastebin.com/m619aeba1 |
| 08:49.11 | E-bola | Its my first time trying to make an IVR sort of thing so it might be rubish |
| 08:49.56 | jblack | You're on the right track |
| 08:50.28 | jblack | I'd suggest adding: |
| 08:50.29 | E-bola | ive seen the ResponseTimeout and digittimeout commands, but im nto sure where to palce them |
| 08:50.40 | jblack | exten => s,n,WaitExten(20) |
| 08:51.14 | jblack | Also, instead of doing 2,1 2,2 2,3 so on, you can do 2,1 2,n 2,n 2,n ... Saves you from having to count. |
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| 08:51.34 | E-bola | ya but its easier to locate a line if you specify a number like if you need to make a goto |
| 08:51.38 | E-bola | atleast thats what ive experienced |
| 08:51.45 | E-bola | But where would you have me put the WaitExten? |
| 08:51.51 | E-bola | below background or above or? |
| 08:52.13 | krdian | is back. what's up? o/~ |
| 08:52.14 | jblack | below. |
| 08:52.16 | krdian | hi |
| 08:52.29 | jblack | Play your sound first, then wait for them to enter a digit. ;) |
| 08:52.45 | E-bola | jblack: so i change background to playback? |
| 08:53.17 | jblack | you can leave that as background. |
| 08:53.39 | E-bola | alright, ile test it now.... |
| 08:54.11 | krdian | is there any ip vide phone suports amr codec ? |
| 08:54.41 | krdian | s/vide/video/g |
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| 09:02.02 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 09:02.46 | Rico29 | hello ! does anybody know where I can find the second edition of the o'reilly book "asterisk the future of telephony" in pdf version ? |
| 09:03.13 | jblack | ~book |
| 09:03.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 09:03.40 | jblack | I really suggest you buy the book though |
| 09:04.34 | E-bola | jblack: thanks it works perfectly now :) |
| 09:04.43 | jblack | welcome. You're on the right track |
| 09:05.09 | jblack | I can tell you're reading the book. :) |
| 09:05.24 | Rico29 | thanks |
| 09:05.53 | jblack | I was referring to e-bola. Rico, I can tell you're working on getting the book. =) |
| 09:06.01 | jblack | Which is a great start to the day, indeed. |
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| 09:09.45 | E-bola | isnt reading the book |
| 09:09.45 | E-bola | hehe |
| 09:09.55 | E-bola | i did read most of it though about 1 year ago |
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| 09:12.18 | adeel | can someone explain to me what the hell this means, "You hereby grant Digium a perpetual, worldwide, royalty-free, irrevocable, non-exclusive, sublicenseable and transferable license under any patent You own or control, now or in the future, to make, have made, use, sell, offer for sale, or import Submissions or any modifications thereof, including without limitation any combinations of the Submissions or modifications thereof with soft |
| 09:12.18 | adeel | ware, technology or services of Digium or its affiliates.: |
| 09:12.57 | adeel | is that more or less saying that anything i ever make/patent can be used by Digium irrespective if it has nothing to do with *? |
| 09:13.01 | jblack | where did you read that? |
| 09:13.07 | adeel | http://bugs.digium.com/license_agreement.php |
| 09:13.30 | jblack | That's the license agreement for the bug tracker? |
| 09:13.38 | adeel | for submitting patches |
| 09:14.04 | jblack | Yes. That sounds right, then. |
| 09:14.23 | jblack | That's why a lot of the good stuff goes into callweaver, which is really promising, but never seems to make a release. |
| 09:14.35 | E-bola | lol |
| 09:14.40 | E-bola | thats an insane license agreement |
| 09:14.42 | adeel | eh, releases are overrated...so long as the thing compiles i'm fine |
| 09:14.53 | jblack | Some people (quite sensibly!) dont' want to give away all rights to their work. |
| 09:14.58 | E-bola | I dont think its even legal? |
| 09:15.11 | adeel | they might as well toss in that you're first born is legally digiums' |
| 09:15.11 | jblack | I never noticed the "any patent" part. |
| 09:15.14 | E-bola | if i understand it correctly you sign over all rights to anything you will ever make? |
| 09:15.20 | adeel | that's what i got out of it |
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| 09:15.27 | E-bola | It would more or less make you a slave of digium for life? |
| 09:15.27 | adeel | remind me to never submit patches to digium |
| 09:15.28 | E-bola | rofl |
| 09:15.30 | jblack | Yeah, it looks like a pretty wide carte blanche to me. |
| 09:15.54 | E-bola | Havent license agreements such as that and eula's etc |
| 09:16.02 | adeel | that's a nice way to kill an 'open source' project |
| 09:16.10 | E-bola | been voted invalid in courts numerous times? for the simple reason that nobody reads them? |
| 09:16.18 | jblack | I knew their submission agreement was onerous. I didn't realize it was that bad. |
| 09:16.19 | adeel | not that i'm aware of |
| 09:16.20 | E-bola | (except adeel) :) |
| 09:16.29 | adeel | aren't you glad that i do? |
| 09:16.35 | adeel | and i read them SPECIFICALLY for clauses like that |
| 09:16.36 | E-bola | Indeed I am :) |
| 09:16.46 | E-bola | I still dont think its legal though |
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| 09:16.54 | adeel | do you want to spend the time and money to find out? |
| 09:17.02 | jblack | Like I said, I knew they were nasty already, which is why I wouldn't submit the couple small patches I made to them. |
| 09:17.07 | adeel | begins the process to migrating to a 'safer' code base |
| 09:17.36 | adeel | i think i'll just start my own repo of patches that won't be subject to their rediculous license |
| 09:17.52 | adeel | something to add to my todo list |
| 09:17.53 | jblack | That's where callweaver came from, but they've diverged quite a bit. |
| 09:18.13 | adeel | i've got a couple of minor patches i'd submit...but not after reading that |
| 09:18.44 | E-bola | actualy thought asterisk was gpl.... |
| 09:18.45 | jblack | The fax stuff is in callweaver, there are conference apps built in that don't require zaptel... |
| 09:18.57 | jblack | asterisk is free software. |
| 09:19.04 | jblack | It's actually dual licensed, I believe. |
| 09:19.06 | adeel | E-bola, it's supposed to be....but digium owns the copyright and can change terms |
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| 09:19.33 | adeel | jblack, yes, but getting 3rd party supported programs for callweaver is a little troublesome |
| 09:19.40 | E-bola | So the sourcecode is GPL but patches you submit arent? |
| 09:19.42 | jblack | and yeah, it's gplv2 |
| 09:19.43 | E-bola | makes no sence... |
| 09:19.43 | adeel | any idea what bug tracker digium uses? |
| 09:20.00 | adeel | E-bola, well it states that digium can re-issue your patches to whatever license scheme they want |
| 09:20.04 | jblack | e-bola: That's something that confuses people. Licensing, and ownership, are two different things. |
| 09:20.37 | adeel | E-bola, the owner defines the license, and as such, can change it whenever they wish |
| 09:20.47 | jblack | If they accepted gpl'ed patches without taking ownership, then they wouldn't have the option to make a seperate release of asterisk, with a nonfree license. |
| 09:21.03 | adeel | jblack, not entirely |
| 09:21.08 | E-bola | adeel: well thats not possible |
| 09:21.13 | E-bola | you cant change the licens of GPL code |
| 09:21.23 | adeel | E-bola, i can if it's my code |
| 09:21.25 | E-bola | so if you submit GPL license patches |
| 09:21.25 | jblack | It's complicated enough for him without covering edge cases. |
| 09:21.30 | E-bola | digium cant use them? |
| 09:21.47 | adeel | E-bola, read the agreement, you sign over ALL rights and license terms to digium |
| 09:21.54 | adeel | digium will then define the terms of the patch |
| 09:22.06 | jblack | not in a differently licensed product, not without your permission, e-bola. |
| 09:22.10 | E-bola | thats bs |
| 09:22.14 | adeel | but from my understanding...if you RELEASE software under a certain license term, you cannot retroactively change that license |
| 09:22.18 | E-bola | how can digium proove it was the owner of the code who submitted it? |
| 09:22.40 | adeel | E-bola, with this clause: |
| 09:22.41 | adeel | You hereby grant Digium a perpetual, worldwide, royalty-free, irrevocable, non-exclusive, and transferable license to use, reproduce, prepare derivative works of, publicly display, publicly perform, distribute the Submissions, and to sublicense such rights to others. The rights granted may be exercised in any form or format, and Digium may distribute and sublicense to others on any licensing terms, including without limitation: (a) open |
| 09:22.42 | adeel | <PROTECTED> |
| 09:22.44 | jblack | Yeah. Post-fact revocation doesn't exist in the US, unless it's in the contract. Which it's not in the gplv2 |
| 09:22.53 | jblack | please don't flood. |
| 09:22.57 | adeel | sorry |
| 09:23.46 | jblack | And technically, you're not giving them full rights. |
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| 09:24.26 | adeel | well you're not transferring ownership |
| 09:24.34 | jblack | correct. |
| 09:24.43 | E-bola | doesnt matter |
| 09:24.45 | adeel | but you're giving them free reign to do with your work whatever they want, without having to pay for you |
| 09:24.55 | jblack | e-bola: Oh, trust me, it does. |
| 09:24.58 | E-bola | lol |
| 09:25.02 | E-bola | so if i submit the apache2 codebase |
| 09:25.05 | E-bola | digium then owns it? |
| 09:25.12 | adeel | no, it's not yours |
| 09:25.28 | E-bola | thats my point |
| 09:25.35 | E-bola | why would they think anything i submitted was mine ? |
| 09:25.47 | E-bola | if its gpl'd its likely its publicly availeble |
| 09:25.48 | adeel | E-bola, read the damn license and then ask questions |
| 09:25.52 | jblack | E-bola: Shhhhhhhhhhhh |
| 09:25.55 | adeel | it's the 4th paragraph that covers it |
| 09:26.06 | E-bola | It doesnt matter what it reads |
| 09:26.10 | adeel | yes it does |
| 09:26.11 | E-bola | its by logic not possible |
| 09:26.13 | adeel | it's a legally binding contract |
| 09:26.17 | E-bola | no its not |
| 09:26.18 | E-bola | thats my point |
| 09:26.21 | E-bola | where is my signature? |
| 09:26.23 | E-bola | they got no proof at all |
| 09:26.30 | adeel | your an absolute idiot |
| 09:26.35 | jblack | E-bola: You're right on the nose on you're thinking about third party software. That's gotten some projects that have dual licensed in the past before. |
| 09:27.00 | jblack | Forcing them to drop proprietary projects. |
| 09:27.10 | E-bola | adeel: you seem to think everything falls under american law, either that or you do not get it at all |
| 09:27.24 | E-bola | I cant sign shit by clicking a button |
| 09:27.27 | E-bola | if you think so ur an idiot |
| 09:27.40 | jblack | Actually, you can. |
| 09:27.48 | E-bola | Not where I live |
| 09:27.54 | E-bola | its not valid as a court contract |
| 09:27.57 | jblack | We're talking US laws, of course. |
| 09:27.58 | E-bola | thats been proven numberous times |
| 09:28.21 | E-bola | if thats the case in america their law is rediouclous, but then again that wouldnt supprise me. They have DMCA as well |
| 09:28.52 | jblack | The US unfortunately, based upon case law, decided that "shrink wrap" EULAS are legal contracts, which extends to active user consent. |
| 09:29.19 | jblack | The same thing that makes clicking "I agree" when you install american software legal in the US is the same thing that makes it legal on a website. |
| 09:29.46 | E-bola | I once saw a study of a test of the EULA system. It sid somewhere burried int he text that if you call the following number you would get 500$ in return. out of 50000 downloads only 2 people called |
| 09:29.52 | jblack | Anyways, going back to your more interesting question, what happens if I submit your GPL code to a project that demands I grant them non-exclusivity rights... |
| 09:30.20 | jblack | In the US, a contract is enforceable if a party agrees. Reading is not a requirement. |
| 09:30.42 | E-bola | Here it was decide that simply clicking "next" in an isntaller isnt agreeing |
| 09:30.46 | E-bola | since nobody ever reads it |
| 09:31.00 | jblack | And? |
| 09:31.05 | E-bola | Which makes sence, any country which doesnt work int hat way is obviously screwing their own citizens over |
| 09:31.48 | jblack | Unfortunately, most first world countries work exactly that way, and the trend is towards that behaviour, not away. You may not have those rights forever (but I hope you do) |
| 09:32.24 | jblack | And? government isn't about your rights. It's about protecting the power of those that have it |
| 09:32.37 | E-bola | Thats true if your a pessimist |
| 09:32.50 | jblack | regardless, it is what it is |
| 09:32.58 | E-bola | Some countries outside america actualy have working democracies :) |
| 09:33.15 | E-bola | without insane presidents and giant power lobbies |
| 09:33.37 | jblack | I'm not really interested in a chat about the flaws in the american (and british, and to a lesser extent, the entire european) legal system. |
| 09:33.51 | E-bola | lets shut it down then :) |
| 09:34.19 | jblack | Great. Now, if you want to get on your donkey, and go tilt at windmills, then I wish you godspeed, because the world can use more of those sorts. |
| 09:34.55 | jblack | but sitting here, fuming about the great arms swinging around, isn't gona do any good |
| 09:35.15 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 09:35.22 | E-bola | I just feel sory for the americans |
| 09:35.46 | jblack | There's places not so bad off, and others much worse. |
| 09:36.28 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 09:43.01 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.46) |
| 09:59.01 | badcfe | hello. i would like to set a 420p wildcard in E1 mode by software (without touching the jumpers) .. i hope its possible? how? |
| 09:59.40 | JT | i doubt it |
| 10:00.09 | Rico29 | does somebody knows how to write something on an IPphone lcd screen (thomson st2030) |
| 10:00.19 | Rico29 | looks stupid, i know |
| 10:00.21 | Rico29 | :) |
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| 10:31.20 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 10:31.27 | whymarkwh | hi all |
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| 10:49.07 | badcfe | in asterisk cli i do zap show status and the span i configured is OK but the IRQ listed is 0 .. does that mean its not up? |
| 10:54.27 | *** join/#asterisk MicW (n=Michael@p57AC8D7B.dip0.t-ipconnect.de) |
| 10:54.30 | MicW | hi |
| 10:54.53 | MicW | shoud it be possible to use video conferencing with asterisk and openwengo (as client)? |
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| 11:10.45 | Rico29 | MicW> i tried |
| 11:10.58 | Rico29 | but wengophone never registers on my asterisk |
| 11:11.04 | Rico29 | and i dont know why |
| 11:11.11 | Rico29 | but a friend of me did it |
| 11:17.48 | whymarkwh | badcfe: still here? |
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| 11:33.17 | tzafrir_home | badcfe, it means it does not lose interrupts |
| 11:33.38 | tzafrir_home | Which is generally a Good Thing |
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| 11:35.27 | *** part/#asterisk mikong (n=gones@219.133.4.75) |
| 11:42.15 | Mahmoud | any one knwos why can't I see "dialplan reload" ? |
| 11:42.19 | Mahmoud | using the latest SVN |
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| 11:45.41 | shasta | eeek, [May 6 10:15:42] NOTICE[14574] chan_sip.c: Peer 'øK´¶È^C°¶' is now Reachable. (8ms / 100ms) |
| 11:45.42 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-5b15ee08daecb8b6) |
| 11:45.50 | XnOSX | good morning! |
| 11:47.06 | XnOSX | anybody here have some problem with compile a zaptel driver with make b410p in debian 2.6.24-1-686? i have a problem for compile zapata |
| 11:47.23 | XnOSX | i need to install a b410p wildcard |
| 11:47.46 | XnOSX | here is the log http://pastebin.ca/1012613 |
| 11:49.24 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.221) |
| 11:50.33 | steliosk | XnOSX : Its not zaptel that has a problem its misdn |
| 11:59.07 | Rico29 | where can I find the asterisk headers ? |
| 11:59.29 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 12:02.44 | Rico29 | help please |
| 12:03.38 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:08.20 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:08.58 | ManxPower | Rico29: /usr/include/asterisk I believe. "find / -name asterisk.h -print" should tell you |
| 12:09.37 | Rico29 | i did it |
| 12:09.40 | Rico29 | i have the lib |
| 12:09.52 | Rico29 | but when i want to compile the addons, with many options |
| 12:09.55 | ManxPower | I don't think Asterisk contains any libraries. |
| 12:10.01 | Rico29 | it says : checking for asterisk.h... no |
| 12:10.10 | ManxPower | Rico29: you can't mix 1.2, 1.4, and 1.6 versions |
| 12:10.10 | Rico29 | (i use --includedir |
| 12:10.16 | Rico29 | i dont |
| 12:10.50 | whymarkwh | Skinny/SCCP Protocol is this just for cisco phones |
| 12:10.51 | whymarkwh | ? |
| 12:10.53 | ManxPower | What versions of Asterisk and Asterisk-addons do you have? |
| 12:11.04 | ManxPower | whymarkwh: yes. The protocol is owned and protected by Cisco. |
| 12:11.42 | whymarkwh | is there any other text to speech enjins that work with asterisk? |
| 12:11.57 | ManxPower | Other? Which one do you use now? |
| 12:12.04 | whymarkwh | none |
| 12:12.28 | ManxPower | There are at least two TTS engines that work with Asterisk. Festival and Cepstral |
| 12:12.39 | Rico29 | ManxPower> asterisk-1.4.19.1 asterisk-addons-1.4.6 |
| 12:12.54 | ManxPower | Try reinstalling Asterisk |
| 12:13.00 | Rico29 | sound qualioty with vestival is bad |
| 12:13.02 | Rico29 | festival |
| 12:13.26 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:13.29 | whymarkwh | ManxPower: witch is the better of the 2? |
| 12:13.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.04 | ManxPower | whymarkwh: I like Cepstral. It sounds much better than any other similar priced system. |
| 12:14.41 | whymarkwh | is it an asteriks addon? |
| 12:15.32 | ManxPower | No, it is a commercial product |
| 12:15.47 | whymarkwh | k thx |
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| 12:17.12 | lirakis_work | Rico29: there are TONS of configuration options, "voices" etc. with festival .. but its configured in Lisp ( oh dear god ) |
| 12:17.20 | lirakis_work | it is very very configurable though |
| 12:17.28 | lirakis_work | its ashame its not "nicer" by default |
| 12:19.31 | Rico29 | ManxPower> an idea for my pb ? |
| 12:22.04 | ManxPower | Rico29: Make SURE you have the |
| 12:22.23 | ManxPower | MOST RECENT Asterisk-Addons. There were significant fixes in the build process in the past few versions |
| 12:22.49 | shido6 | :) |
| 12:22.58 | shido6 | hey ManxPower |
| 12:22.59 | ManxPower | waves to Shido |
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| 12:29.11 | Rico29 | ManxPower> i downloaded them this morning |
| 12:29.25 | Rico29 | but i installed asterisk in non-root mode |
| 12:30.13 | ManxPower | Rico29: My servers are not exposed to the Internet in any way, so I've not felt the additional hassle of non-root is worth it on my systems. |
| 12:32.43 | Rico29 | :( |
| 12:32.46 | Rico29 | that sucks |
| 12:32.49 | *** join/#asterisk xenonex (n=xenonex@89.218.237.113) |
| 12:33.05 | Rico29 | it would help if I pastebin my ./configure result ? |
| 12:33.23 | *** join/#asterisk xenonex (n=xenonex@89.218.237.113) |
| 12:34.29 | ManxPower | IT might help someone, but not me |
| 12:34.37 | ManxPower | I only to the easy stuff for free 8-) |
| 12:35.08 | Rico29 | :) |
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| 12:36.04 | Rico29 | i realy dont understand why it doesn't work.... |
| 12:36.45 | jduggan | hey guys, are there any brits here using digium FXO (our specific model is TDM404B )? im having problems with the card |
| 12:38.19 | [TK]D-Fender | ~ask |
| 12:38.20 | jbot | extra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:42.16 | *** join/#asterisk Marquel (n=Marquel@port-418.pppoe.wtnet.de) |
| 12:42.22 | Marquel | morning |
| 12:42.25 | *** join/#asterisk nighty^ (n=nighty@KD124213240154.ppp-bb.dion.ne.jp) |
| 12:43.05 | x86 | morning |
| 12:43.14 | x86 | err.... moin ;) |
| 12:43.20 | x86 | wie gehts? |
| 12:43.43 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 12:45.32 | Marquel | moin x86 - gut und selbst? |
| 12:46.46 | Marquel | i have a little problem w/ an internal ZAP-channel. first and most important: call pickup by dialing default "*8" doesn't work. the phone just reports "not available". maybe something w/ my zap-channel configuration? |
| 12:47.26 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 12:49.35 | ManxPower | Marquel: What phone reports "unavailable"? |
| 12:49.40 | ManxPower | Analog phones don't do that |
| 12:50.14 | jduggan | wots the best to way to tell if my digium card is recognised and all ports working etc? |
| 12:50.22 | ManxPower | jduggan: use it |
| 12:50.41 | jduggan | ManxPower: well, i want to ensure the hardware is alright before i mess with the config, because its a config issue no doubt |
| 12:50.55 | ManxPower | waits to be told what actual card you have |
| 12:51.05 | jduggan | Digium TDM404B |
| 12:51.11 | jduggan | 4 port FXO |
| 12:51.16 | ManxPower | dmesg should tell you when the driver loads |
| 12:51.59 | badcfe | whymarkwh: yes im still here. and you? |
| 12:52.32 | jduggan | ManxPower: http://internetworkpro.org/pastebin/2445 |
| 12:52.53 | Marquel | ManxPower: an ISDN-phone, connected to a bri-card. |
| 12:52.57 | badcfe | tzafrir_home: thanks. but then i dont understand why i get "unable to create channel. reason 0" (circa) when i Dial(Zap/1) |
| 12:53.50 | jduggan | ManxPower: also http://internetworkpro.org/pastebin/2446 |
| 12:53.58 | [TK]D-Fender | jduggan: want to know if its ok? USE IT |
| 12:54.17 | ManxPower | Marquel: I don't know if whatever card you have supports *8, but if it does it would be configured for the driver |
| 12:54.51 | ManxPower | jduggan: read the error message. Do what it suggests. |
| 12:55.00 | Marquel | ManxPower: thx, that's something a can look up. |
| 12:55.00 | tzafrir_home | Marquel, something is wrong with the context? |
| 12:55.13 | ManxPower | jduggan: what color are the modules on the card? |
| 12:55.19 | tzafrir_home | *8 is implemented in the core of Asterisk , right? |
| 12:55.30 | tzafrir_home | It does work with analog Zap |
| 12:55.37 | jduggan | ManxPower: red |
| 12:55.39 | ManxPower | tzafrir_home: I thought so, but for example, chan_iax2 does NOT support *8 pickup |
| 12:55.40 | Marquel | if i get lost in a few moments, don't bother. i'll be back later. |
| 12:56.17 | ManxPower | jduggan: I don't see the card driver being loaded |
| 12:56.26 | tzafrir_home | Well, it seems to actually do something (but not the right thng) |
| 12:56.37 | ManxPower | zaptel is another required driver -- not hardware specific. |
| 12:56.40 | jduggan | ManxPower: hmm, i thought the zaptel driver was the card driver |
| 12:56.41 | jduggan | aha |
| 12:56.44 | jduggan | ok, one moment |
| 12:56.57 | ManxPower | jduggan: no, zaptel is the zaptel driver, your card driver would be listed in the Zaptel README. |
| 12:57.04 | jduggan | wctdm24xxp |
| 12:57.39 | ManxPower | jduggan: I strongly doubt that |
| 12:57.53 | jduggan | its been autoloaded, i shall read the readme |
| 12:58.26 | ManxPower | Next time read the README before coming here. |
| 12:59.09 | jduggan | ah, it was a pdf i read that suggested that was the correct module for the card |
| 13:00.07 | ManxPower | jduggan: Many idiots give advice. |
| 13:00.26 | ManxPower | I trust the docs located in the asterisk, zaptel source as the primary docs |
| 13:07.02 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net) |
| 13:07.55 | JayTee52 | mornin *'ers |
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| 13:16.03 | M1s3ry | jduggan, I may have missed earlier information, however is this the TDM400? or the TDM410? |
| 13:17.36 | ManxPower | jduggan: Digium TDM404B |
| 13:18.28 | *** join/#asterisk Marquel (n=Marquel@port-418.pppoe.wtnet.de) |
| 13:18.31 | Marquel | re |
| 13:20.11 | Marquel | tzafrir_home: what could that be? the context simply includes the context for sip-phones (working very well). |
| 13:21.00 | ManxPower | M1s3ry: IIRC, the 410B uses the tdm24xx driver, right? I know the 400B uses wctdm |
| 13:21.06 | *** join/#asterisk ididitwithsuse (n=koos@196.211.34.2) |
| 13:21.13 | M1s3ry | coorect |
| 13:21.17 | M1s3ry | correct* |
| 13:21.30 | M1s3ry | that's why i asked... I couldn't tell which one he had... |
| 13:21.35 | jduggan | wctdm24xxp <- thats what's loaded |
| 13:21.35 | ididitwithsuse | anyone usingxchat irc program? |
| 13:21.52 | tzafrir_home | Marquel, if you set verbosity to at least 3, what do you see in the CLI? |
| 13:21.54 | jduggan | is that right? |
| 13:21.55 | M1s3ry | wishes the naming scheme allowed for easier determination on which card a customer was using |
| 13:22.09 | tzafrir_home | ididitwithsuse, I am |
| 13:22.11 | ManxPower | jduggan: any digium driver will load even if the card does not exist in the system |
| 13:22.17 | M1s3ry | jduggan, what card is it? |
| 13:22.31 | jduggan | Found a Wildcard TDM: Wildcard TDM410P (4 modules) |
| 13:22.32 | Marquel | tzafrir_home: incoming call on zap (that's okay ), and an overlap dialed call coming from |
| 13:22.34 | jduggan | in dmesg |
| 13:22.35 | Marquel | *narf* |
| 13:23.06 | Marquel | tzafrir_home: incoming call on zap (that's the one to be picked up), and an overlap dialed call coming from the zap user in question (his MSN) to "<unspecified>". |
| 13:23.07 | ManxPower | jduggan: next time be more accurate. The TDM404B is a different card than the TDM404B you reported just a few mins ago |
| 13:23.37 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:23.46 | ManxPower | Sorry, the 414/410 |
| 13:23.49 | jduggan | ManxPower: i took the card details from the shop i bought it from.. i guess its actually a different model.. which i had no way to know until i loaded its module |
| 13:24.01 | jduggan | and dmesg told me |
| 13:24.23 | mwalling | hans, why did you kill your wife? |
| 13:24.29 | mwalling | dmesg told me to! |
| 13:24.32 | destructure | indeed |
| 13:24.34 | mwalling | i had to ! |
| 13:28.28 | [TK]D-Fender | Reiser performs well with fragmentation.... |
| 13:28.49 | [TK]D-Fender | </pun> |
| 13:28.56 | rob0 | ouch |
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| 13:29.54 | ManxPower | I used to have a client that was in criminal law defense. The only confidence I have in the prosecutor's office is that they will lie, cheat, and steal to win their case, regardless of if the defendant is innocent or not |
| 13:30.50 | ManxPower | In one case the person was on death row and the state had evidence that PROVED the person could not have committed the crime he was being executed for. |
| 13:30.53 | rob0 | agreed, I think lawyers tend to be the lowest form of human life, and prosecutors are the lowest form of lawyer |
| 13:31.21 | rob0 | too bad they can't be held accountable |
| 13:31.40 | ManxPower | Regardlesss of if you are pro or anti death penalty, I most people will agree that is terrible. |
| 13:32.05 | ididitwithsuse | what is considered the most resent stable version of asterisk? |
| 13:32.15 | rob0 | /topic |
| 13:33.53 | ididitwithsuse | ' |
| 13:34.33 | ididitwithsuse | . |
| 13:34.59 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:35.37 | [TK]D-Fender | "There were not big enough changes for Asterisk 1.6 to require a major ABI change release of libpri, so instead most of the 1.6 specific functions were back ported to the 1.4 branch of libpri (including BRI support, as well as a few other things such as TBCT for Q.SIG)" |
| 13:36.54 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:37.21 | tzafrir_home | Well, the ABI was technically officially changed: |
| 13:37.44 | tzafrir_home | in 1.4.3 the library had the SONAME 1.0 . In 1.4.4 it is 1.4 |
| 13:39.17 | [TK]D-Fender | tzafrir_home: but not "major" |
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| 13:41.26 | JayTee52 | rob0, did you ever hear this joke? Daughter: "Mom, can I get pregnant from having anal sex?" Mother: "Of course, dear! Where do you think all the lawyers come from?" |
| 13:41.39 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 13:42.02 | rob0 | lol, I have some lawyer friends to tell that one to! |
| 13:42.22 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 13:42.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:43.15 | JayTee52 | rob0, "Why don't sharks eat lawyers?" |
| 13:43.27 | rob0 | Professional courtesy. |
| 13:43.32 | JayTee52 | hehe |
| 13:44.22 | tzafrir_home | [TK]D-Fender, changing the SONAME is generally considered a major change |
| 13:48.21 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:48.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:50.50 | ididitwithsuse | ll |
| 13:52.38 | *** join/#asterisk shinao1 (n=shinao1@41.219.213.66) |
| 13:54.45 | ididitwithsuse | ll |
| 13:56.19 | *** join/#asterisk jeffspeff (n=jeff@c-68-53-81-14.hsd1.ky.comcast.net) |
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| 13:57.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:58.28 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
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| 14:00.33 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:02.49 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 14:03.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.205.41) |
| 14:03.36 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583659.dsl.bell.ca) |
| 14:03.41 | *** join/#asterisk zktech (n=BryantZ@216.109.196.34) |
| 14:04.07 | ididitwithsuse | in witch order is it best to install :zaptel ,, libpri,,addons,,asterisk or doesn,t it matter? |
| 14:07.05 | russellb | zaptel and libpri first (doesn't matter which order) |
| 14:07.07 | russellb | then asterisk |
| 14:07.07 | russellb | then addons |
| 14:08.43 | mort_gib | Anyone knows what chan_sip.c: Maximum retries exceeded on transmission means?? |
| 14:09.21 | russellb | means that the maximum number of retries on transmitting a packet has been exceeded ... |
| 14:09.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:09.41 | russellb | and the packet was never acknowledged by the other end |
| 14:09.50 | mort_gib | That's what I though, so network issue?? |
| 14:10.35 | zktech | bug 0009896 http://bugs.digium.com/view.php?id=9896 has been closed. I check the changelog for the 1.4.20.rc-2 and did not see it in there. I reall nead this in the branch releases. |
| 14:11.01 | zktech | How can I tell what the state is on the bug? Thanks |
| 14:11.40 | file | it was added to trunk and will be in 1.6.1 |
| 14:12.20 | zktech | What about 1.4 ? I am running |
| 14:12.26 | file | it will not be in there. |
| 14:12.39 | jasonwoot | hey, what's the ISBN number of the Book? |
| 14:12.40 | Juggie | its not a bug its a feature |
| 14:12.43 | shasta | anyone know if there was a bug, fixed between 1.4.17 and 1.4.20 that could cause such things? NOTICE[14574] chan_sip.c: Peer 'øK´¶È^C°¶' is now Reachable. (8ms / 100ms) |
| 14:12.44 | Juggie | ~book |
| 14:12.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:12.56 | `Sauron | ~buybook |
| 14:12.56 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 14:12.58 | jasonwoot | danke |
| 14:13.36 | zktech | How can I get a patch version into 1.4 right now I have one for 1.4.17 and I am stuck there until I can get something for the current branches. |
| 14:14.05 | ManxPower | zktech: if it did not make it into 1.4.x it never will. |
| 14:14.27 | ManxPower | you can manually apply the patch to your 1.4 if you want, or you can hope to find a 1.4 version on the bug file attachments |
| 14:14.35 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 14:15.21 | zktech | The version 12 of the patch worked on 1.4.17 but I could not get any of version of the patch to run on 1.4.19 or 1.4.20 rc |
| 14:15.38 | Tourinho | hello people.. if I have an application that receives a call, play prompts end dial to another place using SIP and bridge calls, should I have 2 g729 codecs license? |
| 14:17.43 | ididitwithsuse | where can i find a list of dependecies for asterisk? |
| 14:17.54 | [TK]D-Fender | ididitwithsuse: www.asterisk.org |
| 14:18.02 | [TK]D-Fender | Tourinho: 1 |
| 14:19.14 | zktech | Without the patch 0009896 I can not reboot any ATA that is secure. This is major for us. |
| 14:19.14 | zktech | When would it be likely that a version 1.6.1 would be available and stable for a production server? |
| 14:19.33 | file | we can not tell the future. |
| 14:20.11 | anonymouz666 | zktech: maybe when reach the version .20 |
| 14:20.24 | anonymouz666 | :P |
| 14:21.06 | zktech | anonymouz666 I is not in the changelog for .20 |
| 14:21.27 | Tourinho | [TK]D-Fender: thank you |
| 14:21.45 | anonymouz666 | zktech: I mean 1.6.20 |
| 14:21.56 | russellb | it's not going to ever be in 1.4 |
| 14:22.02 | russellb | it's a new feature, and was only merged into 1.6 |
| 14:27.03 | zktech | I will move to 1.6 then. Are the betas stable enough for production and am I likely to get hit with major dial plan incompatibilities? Where is the best source of info on making the jump from 1.4 to 1.6? |
| 14:27.35 | zktech | I am also using the TC400 cards is 1.6 likely to give me any issue there? |
| 14:29.08 | file | what you want is not yet in a 1.6 release |
| 14:29.33 | Nugget | built 1.6 yesterday |
| 14:29.54 | JayTee52 | has 1.6.0beta8 running on a test box |
| 14:33.34 | russellb | yay for people actually testing 1.6 |
| 14:35.35 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 14:36.27 | ididitwithsuse | if you have a system that already has asterisk installed suse you can select it from the instalation package how do update it to latest version, can one do this is the question? |
| 14:39.45 | russellb | that didn't make any sense |
| 14:39.54 | russellb | and is probably a #suse question or something similar |
| 14:44.01 | ManxPower | zktech: upgrade.txt in the asterisk source |
| 14:46.03 | zktech | ManxPower I am downloading it right now Thanks |
| 14:47.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:48.57 | ManxPower | zktech: There SHOULD be info for upgrading from 1.0.x to 1.2.x as well as from 1.2.x to 1.4.x. If there is not then download an asterisk 1.2 and look at that upgrade.txt too. |
| 14:49.18 | ManxPower | Things that were deprecated in 1.2 were removed in 1.4 and that can bite you if you don't have both upgrade.txt files. |
| 14:50.05 | ManxPower | I've lobbied to have both files included in 1.4 and 1.6, and both are in 1.6, IIRc, but I don't know about 1.4. |
| 14:50.11 | ManxPower | russellb: can you comment on that? |
| 14:50.34 | russellb | i have no comments on anything |
| 14:50.49 | russellb | i think they're all in 1.6 |
| 14:50.49 | *** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk) |
| 14:50.53 | russellb | don't think they're all in 1.4 |
| 14:51.56 | ManxPower | hands russellb some /* and */s |
| 14:52.37 | Tourinho | 1/1 encoders/decoders of 10 licensed channels are currently in use << this means that Im using 2 licenses? |
| 14:52.58 | file | you are using 1 license. |
| 14:53.26 | Tourinho | file: thanks |
| 14:53.35 | Marquel | tzafrir_home: no more ideas? |
| 14:58.52 | ManxPower | Tourinho: A license is for one encode, one decode simul |
| 15:01.10 | zktech | Is there a posted timeline for the 1.6 releases? |
| 15:01.21 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:04.38 | [TK]D-Fender | zktech: Never |
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| 15:06.13 | drfreeze | Hi |
| 15:06.57 | drfreeze | I have a T1 with an Adtran board that provides 8 phone lines |
| 15:07.07 | drfreeze | I route these lines to 2 TDM cards |
| 15:08.12 | drfreeze | Is it ok to also connect an analog phone to these lines? Currently, I have one line that I don't connnect to the TDM and use for other purposes. I am wondering if I can add it to Asterisk as well |
| 15:09.02 | XnOSX | anybody have a problem with the zaptel (make b410p) in debian unstable kernel 2.6.24? |
| 15:09.30 | [TK]D-Fender | drfreeze: Thats just retarded. Get a T1 card an connect it direct to * |
| 15:10.18 | drfreeze | [TK]D-Fender: at the time this was configured, the provider didn't support T1 card |
| 15:10.40 | [TK]D-Fender | drfreeze: and when the GOT the T1, why did they get a channel bank for it? |
| 15:10.41 | drfreeze | [TK]D-Fender: if it works, why call it retarded? |
| 15:11.13 | [TK]D-Fender | drfreeze: Spend money converting it BACK to analog only to take it into *. Thats like questioning why 10th generation photocopies suck. |
| 15:11.26 | drfreeze | [TK]D-Fender: I was all for the T1 card, but was told not to |
| 15:11.45 | [TK]D-Fender | drfreeze: See above. Wrong move. |
| 15:11.56 | *** join/#asterisk profounded (n=Bryan_Ru@c-68-82-34-163.hsd1.nj.comcast.net) |
| 15:12.09 | drfreeze | It's been a while, does the T1 support Fax? |
| 15:12.15 | jduggan | guys, when calling via the analogue fxo card, the sip client is hearing himself in the handset, like its looping back his own voice (doesnt seem like echo), what causes this?, if its sip-sip then its fine |
| 15:12.57 | [TK]D-Fender | drfreeze: T1 is just a carrier tech. |
| 15:13.21 | drfreeze | [TK]D-Fender: I need at POTS line for Fax |
| 15:14.12 | drfreeze | so, need to get a good analog signal somehow |
| 15:14.41 | tzanger | channel bank :-) |
| 15:14.43 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:14.49 | [TK]D-Fender | drfreeze: completely separate analog circuit direct from telco |
| 15:15.03 | drfreeze | [TK]D-Fender: you mean, like I am now doing. :) |
| 15:15.15 | drfreeze | well, that would be....retarded |
| 15:16.43 | drfreeze | Anyway, never got the question answered. I suppose the TDM card is smart enuf to recognize if an analog phone is in use on the same line |
| 15:17.16 | badcfe | my wildcard pri cpe span shows up with zap show status in * cli but it doesnt seem up. how do i verify on a lower level? |
| 15:18.14 | *** join/#asterisk drummond_ (n=rsd@h-67-103-23-130.phlapafg.covad.net) |
| 15:18.15 | drfreeze | badcfe: what does zap show channels say |
| 15:18.24 | badcfe | lszaptel also shows it as up (tho the channels are marked as Clear) |
| 15:18.51 | badcfe | drfreeze: pseudo pstn default |
| 15:19.20 | drfreeze | nothing, eh |
| 15:19.39 | badcfe | oh? |
| 15:19.51 | badcfe | pseudo is nothing .. ? |
| 15:20.08 | badcfe | its supposed to say something else huh? |
| 15:20.24 | drfreeze | badcfe: I have wildcard tdm, but it shows channels |
| 15:20.33 | drfreeze | what is a wildcard pri? |
| 15:20.52 | jasonwoot | changes topic to 'what should I get my mom for mother's day?' |
| 15:21.10 | badcfe | drfreeze: te420b |
| 15:21.39 | drfreeze | badcfe: I have ound a Wildcard TDM: Wildcard TDM400P REV I (4 modules) |
| 15:21.50 | drfreeze | what does zttool from the commandline say |
| 15:22.36 | badcfe | it opens a curses menu listing amond the other spans: OK T4XXP (PCI) Card 0 Span 1 |
| 15:22.38 | [TK]D-Fender | badcfe: .... and zapata.conf? |
| 15:22.42 | [TK]D-Fender | ~pb |
| 15:22.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:22.44 | [TK]D-Fender | ^^^^^^^ |
| 15:23.33 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net) |
| 15:23.52 | badcfe | http://yourbackuponline.net/pastebin/20080509172345.txt |
| 15:24.00 | ididitwithsuse | waht is the linux distro of choice for the instalation of asterisk if this is going to be my first install need to know whats best ? |
| 15:24.42 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:24.42 | *** join/#asterisk zktech (n=BryantZ@216.109.196.34) |
| 15:25.12 | badcfe | http://yourbackuponline.net/pastebin/20080509172504.txt |
| 15:25.13 | Nugget | linux is linux is linux. Use whatever you're most familiar with. |
| 15:25.17 | tzafrir_home | Marquel, again, what do you see in the CLI? |
| 15:25.23 | drummond_ | i rolled it out to CentOS |
| 15:25.33 | JayTee52 | I second what Nugget said |
| 15:25.35 | Marquel | tzafrir_home: incoming call on zap (that's the one to be picked up), and an overlap dialed call coming from the zap user in question (his MSN) to "<unspecified>". |
| 15:25.39 | tzafrir_home | ididitwithsuse, if there's a linux distro you're familiar with, use it |
| 15:25.44 | zktech | I got disconnected. Is ther a posted timeline for the 1.6 releases? Thanks |
| 15:26.21 | badcfe | [TK]D-Fender, drfreeze: and the net people tel me that this seems good but they say theres a protocoll alarm and ask me to verify that this is VN4 |
| 15:27.07 | JayTee52 | I've run the compiled version of 1.2 and 1.4 on RHEL 5, CentOS 5, Debian 4 and Ubuntu 6.06 and 7.10 server with no problems and the RPM builds from Livna on Fedora 6 and 7 with no problems. |
| 15:27.48 | drummond_ | i rolled it out to centos 5, but built it from scratch |
| 15:28.06 | JayTee52 | that's what my main production server is running now |
| 15:28.31 | JayTee52 | and my other * system is running RHEL 5 64bit |
| 15:29.27 | badcfe | [TK]D-Fender, drfreeze: does it seem correct? bu the way .. is there some low level debug possibilities at my side? |
| 15:29.37 | [TK]D-Fender | badcfe: Not sure. |
| 15:31.15 | tzafrir_home | Marquel, could you please enable debug logging in logger.conf and in the CLI and post the full log as well? |
| 15:32.33 | Marquel | tzafrir_home: that'll have to wait until tomorrow until i can reproduce the necessary circumstances. but i'll return with the logs. |
| 15:36.30 | drummond_ | jay, how isthe performance under 64bit? |
| 15:37.39 | Juggie | drummond_, we run all 64bit in prod, its fine. |
| 15:37.54 | JayTee52 | drummond, with a Quad Core Xeon and 4GB of RAM and 2 mirrored SAS drives it absolutely flies. |
| 15:38.05 | drummond_ | cool. |
| 15:38.39 | Juggie | Ya, we are running Dual Dual Core Xeons w/ 4gigs and Raid5 |
| 15:38.43 | Juggie | and its fast :) |
| 15:38.49 | JayTee52 | this is on a Dell 2950 and RHEL 5 64bit is what you get if you ask for RHEL preinstalled at the factory. If you want to run 32 bit you load it yourself. |
| 15:39.07 | drummond_ | i have it running on a POS dell dual dimension dual core machine, along with splunk, nagios, nfs, mysql, and a bunch of other stuff that hogs memory. i runs with out problems |
| 15:39.08 | *** join/#asterisk dFence (n=chatzill@ings-d93223ec.pool.mediaWays.net) |
| 15:39.25 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:39.45 | Zeeek | yo ho ho |
| 15:40.19 | badcfe | what is VN4? |
| 15:45.26 | Zeeek | The 3rd party ecosystem around asterisk is the subject of todays VoIP Users Conference live in 15 minutes. Join #voip-users-conference and talk/listen by reading http://x2z.eu |
| 15:46.09 | ftp3 | anyone know someplace besides didx that I can get a good deal on wholesale usa dids? |
| 15:50.04 | ftp3 | guess not :-D |
| 15:50.26 | ftp3 | how about a linux utility to join to mp3s for my asterisk moh |
| 15:50.35 | ftp3 | to=two |
| 15:50.49 | [TK]D-Fender | ftp3: Audacity, SOX, etc |
| 15:51.30 | *** join/#asterisk timgws (n=LivedTyp@202.172.97.51) |
| 15:52.10 | ftp3 | thank you :-) |
| 15:57.06 | Uatec | hi there |
| 15:58.05 | Uatec | i have a SIP session that's coming in to asterisk from our sip provider |
| 15:58.35 | Uatec | i now want to pass it over to my openser proxy |
| 15:58.38 | Marquel | cu all |
| 15:59.24 | Uatec | but when asterisk tries to send the invite to openser it sends the URI sip:012345678990@127.0.0.1 |
| 15:59.51 | Uatec | which is the number presented on the incoming call |
| 16:00.03 | Uatec | i need to override that as a URI and pass it my own details? |
| 16:00.06 | Uatec | how can i do this? |
| 16:00.14 | Mahmoud | Why can't I find "dialplan reload" command in the CLI? |
| 16:02.51 | Mahmoud | what is responsible to show "dialplan reload" command? |
| 16:03.20 | Uatec | type 'help' and see what's listed... |
| 16:05.28 | *** join/#asterisk RoyK (n=roy@cnbokcafe.uio.no) |
| 16:05.34 | [TK]D-Fender | Mahmoud: CLI is different between * versions. |
| 16:06.30 | Zeeek | Join #voip-users-conference and talk/listen by reading http://x2z.eu - see you there |
| 16:06.35 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:07.57 | Mahmoud | [TK]D-Fender, I'm using the latest SVN |
| 16:08.19 | [TK]D-Fender | Mahmoud: Of which branch? |
| 16:08.51 | Mahmoud | [TK]D-Fender, http://svn.digium.com/svn/asterisk/trunk/ |
| 16:09.00 | [TK]D-Fender | Mahmoud: Direct answer please. |
| 16:09.42 | [TK]D-Fender | Mahmoud: "looks" like 1.6. Indeed, type "help" and see what CLI options you've got. |
| 16:10.57 | Mahmoud | [TK]D-Fender, I have only "dialplan {set | show}" |
| 16:11.27 | [TK]D-Fender | Mahmoud: look harder. |
| 16:11.58 | Mahmoud | well, I have only two arguments for "dialplan" |
| 16:13.00 | Mahmoud | any modules to be loaded? currently autoload=yes in modules.conf |
| 16:14.31 | [TK]D-Fender | Mahmoud: try looking under "reload" |
| 16:16.00 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:16.47 | *** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 16:17.00 | Mahmoud | [TK]D-Fender, i see list of modules, the closest thing I see is extconfig |
| 16:17.19 | [TK]D-Fender | Mahmoud: "reload dialplan" |
| 16:18.03 | Mahmoud | [TK]D-Fender, reload d<tab> shows only dnsmgr and dsp (no dialplan) |
| 16:18.52 | Mahmoud | [TK]D-Fender, restarting asterisk, and then executing "dialplan show" doesn't show my dialplan |
| 16:19.48 | Mahmoud | this is all what I see by "dialplan show" http://pastebin.com/d5878b26b |
| 16:20.03 | Mahmoud | which is indeed not my dialplan |
| 16:20.26 | [TK]D-Fender | Mahmoud: You've wasted our time in describing it like you din't have the OPTION at all. |
| 16:21.01 | *** join/#asterisk NRich (n=NRich@72.37.252.50) |
| 16:21.12 | Mahmoud | [TK]D-Fender, well, I have "dialplan" but I don't have the "reload" argument. This is what I said and my appologies if I wasn't clear |
| 16:21.24 | [TK]D-Fender | Mahmoud: Go pastebin "cat /etc/asterisk/extensions.conf" and show us the perms on it as well. |
| 16:21.39 | Mahmoud | perms? |
| 16:21.43 | NRich | I have a machine with asterisk, it's using 97% cpu for unknown reasons.. can anyone help with this? |
| 16:21.47 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 16:21.52 | [TK]D-Fender | PERMISSIONS |
| 16:22.10 | JayTee52 | NRich if you run top what process is hogging the CPU? |
| 16:22.30 | NRich | asterisk |
| 16:22.41 | NRich | 4310 root 25 0 4932 1276 4524 R 94.1 0.6 1148:52 asterisk |
| 16:22.41 | Mahmoud | [TK]D-Fender, for testing, it's exactly copy and paste from configs dir |
| 16:22.52 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:22.58 | Mahmoud | [TK]D-Fender, permission: -rwxr-xr-x 1 root wheel |
| 16:23.01 | [TK]D-Fender | Mahmoud: I don't want your description, I want proof of tis condition. |
| 16:23.16 | JayTee52 | NRich, is this a production system or a test system? |
| 16:23.29 | NRich | production |
| 16:24.22 | Mahmoud | [TK]D-Fender, http://pastebin.com/m348fc370 |
| 16:24.33 | JayTee52 | NRich, what version of * and what linux distro and version? |
| 16:25.25 | NRich | this is a pbxtra box |
| 16:25.44 | [TK]D-Fender | Mahmoud: And you're running * as? |
| 16:25.51 | Mahmoud | [TK]D-Fender, root |
| 16:26.11 | [TK]D-Fender | Mahmoud: "module load pbx_config.so" |
| 16:26.51 | JayTee52 | NRich, I'd call Fonality |
| 16:27.11 | Mahmoud | [TK]D-Fender, done, so far no change, no output either |
| 16:27.11 | jjshoe_ | NRich Hey, I just sat down, developer at Fonality, what's up? |
| 16:28.24 | [TK]D-Fender | Mahmoud: pb "ls -l /etc/asterisk" |
| 16:29.35 | Mahmoud | [TK]D-Fender, http://pastebin.com/m797b7c0d |
| 16:29.48 | jjshoe_ | NRich feel free to pm me your four digit server-id and the issue you're having. |
| 16:30.31 | NRich | jjshoe_: message david kullman - dect on hud (im using his for barging) |
| 16:30.36 | [TK]D-Fender | Mahmoud: .... -rwxr-xr-x 1 root wheel 25331 May 9 19:56 extentions.conf |
| 16:30.44 | [TK]D-Fender | Mahmoud: I don't f'n think so... |
| 16:30.51 | [TK]D-Fender | :p |
| 16:31.01 | NRich | JayTee52: /whois NRich && internet whois my ip =] |
| 16:31.19 | JayTee52 | wtf? |
| 16:31.21 | Mahmoud | [TK]D-Fender, I don't get you? |
| 16:31.24 | [TK]D-Fender | Mahmoud: "extentions.conf" != "extensions.conf" |
| 16:31.29 | NRich | JayTee52: I am fonality |
| 16:31.39 | Mahmoud | [TK]D-Fender, LOL... |
| 16:31.59 | [TK]D-Fender | Mahmoud: BRILLIANT |
| 16:32.02 | JayTee52 | well, then I guess you don't need to call yourself then, do you? :-) |
| 16:32.14 | [TK]D-Fender | JayTee52: the voices need company sometimes... |
| 16:32.18 | Mahmoud | [TK]D-Fender, lol, man i spent hours troubleshooting this hehehe |
| 16:32.26 | NRich | I'm brand new here, they just gave me a task to debug and I'm trying my resources =] |
| 16:32.49 | NRich | the problem has already mostly been debugged by david kullman already |
| 16:33.39 | b11d` | Anyone here ever work with an NEC ElectraElite IPK system? |
| 16:33.48 | Qwell | NRich: #asterisk isn't one of your support resources. That's something you're going to need to learn pretty much immediately. |
| 16:34.04 | Mahmoud | [TK]D-Fender, working great.. thanks man! |
| 16:34.28 | jjshoe_ | Qwell I straightened it out with his boss :) |
| 16:34.35 | Qwell | jjshoe_: yeah.. |
| 16:35.08 | jjshoe_ | Qwell sorry buddy <3 |
| 16:35.23 | Qwell | meh, just.. yeah |
| 16:35.26 | tzafrir_home | well, it is a great support resource. If you don't rely on it as such... |
| 16:35.36 | Qwell | tzafrir_home: not when your job is supporting trixbox :) |
| 16:35.44 | Qwell | ie; paid support. That isn't going to fly |
| 16:36.01 | *** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net) |
| 16:36.19 | alrs | Rats, I was going to have NRich ask Kullmann about Sky Bed |
| 16:36.36 | jjshoe_ | alrs sky bed has been seperated. |
| 16:36.45 | jjshoe_ | how's it going lars? |
| 16:37.10 | alrs | Hanging out at my work-a-ma-job, Joel. |
| 16:37.29 | alrs | Just playing with zaptel-over-ethernet-over-MPLS |
| 16:37.57 | jjshoe_ | I'm playing with my rocket ship cup I got from dennys at lunch yesterday. Something tells me I'm having more fun. |
| 16:38.13 | alrs | You don't have a tattletale coffee cup? |
| 16:38.17 | jjshoe_ | "Tell him I said he's welcome to come over and rejuvenate Skybed" |
| 16:38.42 | jjshoe_ | I don't need a coffee mug from that bar when my girlfriend got that chick's number for herself that carlo's in love with. |
| 16:39.45 | JayTee52 | checks his IRC client to make sure he's in #asterisk and not #trixboxofficegossip |
| 16:39.48 | jjshoe_ | he died in a little in side that night :P |
| 16:40.23 | jjshoe_ | JayTee52 sorry, I hope I didn't interrupt any string of help you were giving? |
| 16:40.51 | jjshoe_ | it's basically dead quiet in here this morning. |
| 16:40.55 | JayTee52 | nope, just a joke |
| 16:41.12 | jjshoe_ | JayTee52 alrs used to work for us. just saying hey is all. |
| 16:41.39 | JayTee52 | like I said, I was joking |
| 16:41.46 | jjshoe_ | oh I know :) |
| 16:42.11 | [TK]D-Fender | Ok, question I asked a week or so ago : Looking for embedded systems similar to Soekris & Alix. Anyone got some links for me? |
| 16:42.35 | Qwell | [TK]D-Fender: gumstix |
| 16:42.37 | alrs | Now I face challenges such as http://www.pastebin.ca/1012858 |
| 16:42.41 | Qwell | too small? :) |
| 16:42.59 | [TK]D-Fender | requirements : 1 PCI minimum, 2 PCI is < 3 NICS. GBIT highly preferred |
| 16:43.37 | jjshoe_ | Qwell :P |
| 16:43.47 | Qwell | jjshoe_: ? |
| 16:43.55 | [TK]D-Fender | Qwell: Yes... too small... |
| 16:44.05 | jjshoe_ | Qwell gumstix. |
| 16:44.14 | Qwell | [TK]D-Fender: ever seen one of KrisK's gumstix Asterisk boxes? |
| 16:44.18 | jjshoe_ | alrs most retarded pastebin site I've ever seen. looks like ass in ie7. |
| 16:44.22 | [TK]D-Fender | Qwell: Nope. |
| 16:44.22 | Qwell | it's a pretty amazing sight |
| 16:44.51 | jjshoe_ | alrs enjoy you digium card. |
| 16:45.00 | alrs | all three of them |
| 16:45.07 | jjshoe_ | Qwell / [TK]D-Fender is astlinux any good? |
| 16:45.24 | Qwell | never actually used it, but I've heard lots of good things |
| 16:45.33 | Qwell | I know Kristian is quite competent |
| 16:45.53 | jjshoe_ | I was reading up on it, and I was very impressed |
| 16:47.33 | b11d` | TK.. just wanted to say, im quite happy and impressed with the SPA-8000 |
| 16:47.58 | [TK]D-Fender | b11d`: Glad to hear. |
| 16:55.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:58.01 | b11d` | I only wish I could set an NTP server on it.. thats the only thing I found lacking.. and thats not a big deal at all |
| 17:00.20 | JayTee52 | [TK]D-Fender, you might look at this: http://www.fit-pc.com/new/ |
| 17:00.49 | jjshoe_ | JayTee52 based on a mini/nano itx I presume? |
| 17:00.59 | JayTee52 | paperback size |
| 17:01.05 | JayTee52 | run Ubuntu or XP |
| 17:01.14 | JayTee52 | *runs |
| 17:01.34 | [TK]D-Fender | JayTee52: No PCI = no thanks |
| 17:01.46 | [TK]D-Fender | JayTee52: I want to build a router |
| 17:01.50 | JayTee52 | ah |
| 17:02.03 | JayTee52 | and you have to have embedded linux? |
| 17:02.14 | [TK]D-Fender | JayTee52: Many distros out there for this. |
| 17:02.28 | [TK]D-Fender | JayTee52: Soekris is an options, I'm looking for others. |
| 17:03.31 | JayTee52 | I haven't seen much in the way of SOC with a PCI connection. A router for VOIP? |
| 17:05.41 | *** join/#asterisk draygon (n=draygon@gateway5-pnap.exigo.com) |
| 17:13.47 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 17:16.47 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:22.39 | *** join/#asterisk RoyK (n=roy@cnbokcafe.uio.no) |
| 17:24.20 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 17:27.32 | jasonwoot | Anyone remember that guy the other day who was using Asterisk to control access to doorways? |
| 17:29.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:34.28 | DaveCanoe | can one forge a custom insert when using cdr_mysql or cdr_pgsql? |
| 17:34.45 | *** join/#asterisk DragoraN (n=dragoran@chello089173134033.chello.sk) |
| 17:34.55 | DragoraN | hi |
| 17:34.58 | [TK]D-Fender | JayTee52: Slap my Sangoma S518 for ADSL, 3-4 GBIT NICs (on-bourd preferred. Wifi would be nice, or MiniPCI to add, etc. |
| 17:35.11 | DragoraN | which SIP client i should use for my winmobile 6? can someone advice? |
| 17:35.19 | [TK]D-Fender | JayTee52: Frankly a NET5501 is jsut about perfect, but I'm looking for alternatives. |
| 17:35.50 | mwalling | jasonwoot: |
| 17:35.52 | mwalling | er |
| 17:35.54 | mwalling | .irssi/logs/6freenode/#asterisk/2008/05/06:08:52 < Madkiss> jasonwoot: basically I am trying to do this: My companies works on two different floors. every department has its own door, and you can open that door right now via the voip-phones and a wrapper script. i want to define two groups in the database: floor 1 and floor 2. whenever a member of one group calls numer "95", i want the door on the floor to be opened where the user is at that time. |
| 17:36.10 | mwalling | time stamp in EDT |
| 17:36.24 | *** join/#asterisk notbright (n=chatzill@unaffiliated/yourname/x-837320) |
| 17:37.29 | tzafrir_home | I got "strange disconnects" on outgoing Zap/FXO calls. Looking further at them I saw that the reason for disconnecting was the time-out parameter for Dial |
| 17:37.43 | JayTee52 | [TK]D-Fender, if I see anything else I'll let ya know |
| 17:38.03 | tzafrir_home | But the call has already started. Isn't an FXO call considered "answered" as soon as there's audio? |
| 17:38.28 | jasonwoot | TY mwalling |
| 17:38.39 | mwalling | thank grep :) |
| 17:38.49 | [TK]D-Fender | JayTee52: thx |
| 17:40.01 | [TK]D-Fender | tzafrir_home: its answered as soon as Zaptel can reserve the channel or until "callprogress=yes" is set and it stops "ringing" without giving a negative prorgess tone. |
| 17:40.49 | tzafrir_home | callprogress is not set |
| 17:41.16 | tzafrir_home | ah, there it is: answeronpolarityswitch=yes |
| 17:42.02 | mwalling | jasonwoot: you want the log from that day or do you have it? |
| 17:42.10 | tzafrir_home | wonders if there's any way to detect that one.... |
| 17:45.10 | notbright | Does anyone have an asterisk safe restart crontab srcript lying around somewhere? |
| 17:46.08 | DaveCanoe | sigh. I feel so invisible on this channel. |
| 17:47.16 | badcfe | anyone knows what a VNx is .. like a VN4 .. ? |
| 17:47.35 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:47.55 | notbright | DaveCanoe: I get that feeling myself sometimes. |
| 17:49.48 | *** join/#asterisk jackson__ (n=jackson@68-115-108-47.dhcp.roch.mn.charter.com) |
| 17:51.20 | DaveCanoe | does anyone know how to modify the insert that the CDR interface does to databases. In this case, I have multiple asterisk servers inserting CDRs into a database. I'd like one large CDR table, but I want to know _which_ asterisk server inserted the record. So I need each asterisk server to insert a static field (ie: asterisk_server_number or somesuch) |
| 17:55.50 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 17:57.47 | [TK]D-Fender | DaveCanoe: this is already detailed between the WIKI & BOOK. Go give them botha good read. |
| 17:59.07 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 18:06.03 | DaveCanoe | I'm on the sip wiki ... and I'm reading about cdr... and I don't see anything that implies I can modify the insert statement or how I should do it. Maybe you have a url I should look at? |
| 18:06.42 | bobbym | in this new asterisk release 1.6 the DeadAGI is going to be deprecated? |
| 18:12.54 | Juggie | the answer to that is no and yes |
| 18:13.35 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 18:15.44 | bobbym | Juggie: |
| 18:15.47 | bobbym | good ;) haha |
| 18:16.42 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 18:17.20 | JayTee52 | if I need to add 3 digits to the beginning of my dialed number over a PRI trunk do I just add the 3 digits in front of the dialed extension like this: Dial(Zap/g1/630${EXTEN}) ? |
| 18:17.43 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 18:19.55 | Madkiss | mwalling: what? |
| 18:20.07 | Madkiss | mwalling: i got it sorted out by now btw. |
| 18:20.21 | mwalling | 13:27 < jasonwoot> Anyone remember that guy the other day who was using Asterisk to control access to doorways? |
| 18:20.25 | Madkiss | mwalling: it's relatively simple after all |
| 18:20.34 | *** join/#asterisk musarati__ (n=musarati@p549247E3.dip.t-dialin.net) |
| 18:20.36 | Madkiss | mwalling: asteriskdb and the right magic in extension.ael do the trick. |
| 18:22.09 | Madkiss | mwalling: I am not sure I could offer what jasonwoot is looking for, tho. to open the door, i am calling /usr/local/bin/open_that_door_alread_stupid_culprit ... ; |
| 18:22.12 | Madkiss | ;) |
| 18:22.21 | jjshoe_ | JayTee52 yes |
| 18:23.15 | JayTee52 | jjshoe, thanks |
| 18:23.22 | mwalling | Madkiss: i just grepped the logs, although sounds cool |
| 18:23.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:23.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:23.46 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:23.50 | JayTee52 | I've been digging through the book looking for a way to set a 3 digit variable and concatenate with the $EXTEN variable but I can't find a specific example or a special function. |
| 18:23.57 | Madkiss | mwalling: it's simple mechanical stuff. we got a direct connection between the parport of our server and the door, and all i need to do is to call parport_ctrl with the correct parameters. |
| 18:24.19 | Madkiss | mwalling: i even extended the solution today so that any employee can use his company mobile phone to open a door. |
| 18:24.43 | Madkiss | mwalling: it's damn fscking cool, but as said -- the asterisk-part of it is really easy once you understood how asteriskdb works. |
| 18:25.08 | [TK]D-Fender | JayTee52: Dial(Zap/g1/${myareacode}${EXTEN}) |
| 18:27.47 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:28.04 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:31.05 | *** join/#asterisk dFence (n=chatzill@ings-d93223ec.pool.mediaWays.net) |
| 18:31.23 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:35.03 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:35.49 | JayTee52 | [TK]D-Fender, thanks!!! that's what I needed but couldn't find in the book whether I could just put the two together or whether I needed to use something like an & to join them |
| 18:36.05 | [TK]D-Fender | JayTee52: * variables ARE that dumb. its all just text |
| 18:36.32 | JayTee52 | [TK]D-Fender, and I'd much rather use a variable than hard code it even though the nxx # won't change |
| 18:36.51 | JayTee52 | hahahah, one of my coworkers switched the keycaps on my keyboard around. |
| 18:37.14 | JayTee52 | I think I'll just leave it since I'm a touch typist |
| 18:37.53 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 18:38.02 | *** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 18:38.11 | [TK]D-Fender | JayTee52: Do the same to him... and then remap them on an OS level to something ELSE completely different. |
| 18:39.16 | JayTee52 | hehehe, that's an excellent idea. We already put the BSOD screensaver on his computer and it took awhile for him to catch on. |
| 18:39.52 | [TK]D-Fender | JayTee52: OH.. and then really fuck with him and take a screeshot of his destop, and use that as a wall-paper and move all his files / icons off of it. |
| 18:40.12 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 18:40.13 | [TK]D-Fender | "I can't open my programs!!!!" |
| 18:41.41 | codefreeze-lap | DaveCanoe: hint: use cdr-adaptive; and see if CDR variables can get you where you want to go |
| 18:42.15 | codefreeze-lap | DaveCanoe: uh, sorry, cdr-odbc-adaptive, I meant |
| 18:43.08 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:44.55 | JayTee52 | way back in the late 80's I hacked all the intrinsic commands in command.com on my coworker's PC. duh intead of dir crpy instead of copy, etc. |
| 18:46.14 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:46.32 | [TK]D-Fender | JayTee52: Yup... ahh the good 'ole days |
| 18:46.39 | bobbym | JayTee52: how did you do that? |
| 18:47.09 | [TK]D-Fender | bobbym: file edit command.com |
| 18:47.13 | JayTee52 | bobbym, it was back in the dos days and I used Norton Editor |
| 18:47.18 | [TK]D-Fender | bobbym: You could search & replace those directly. |
| 18:48.20 | bobbym | [TK]D-Fender: really, i always wondered that if you want to chance those thing inside command.com you should decompile and compile again the com |
| 18:48.21 | bobbym | haha |
| 18:48.25 | bobbym | well good to know |
| 18:50.54 | DaveCanoe | codefreeze-lap: thx. |
| 18:53.40 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 18:55.18 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:55.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.17 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:01.21 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:07.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.18 | draygon | Hm. Is the originate feature removed from the newer asterisk version? |
| 19:12.39 | mintee | I need to GotoIf(Exist(soundfile)) |
| 19:15.25 | mintee | i understand that the EXISTS checks for a variable |
| 19:15.42 | mintee | but I need to check to see if a specific gsm file exists. |
| 19:16.16 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:16.51 | hmmhesays | STAT |
| 19:16.54 | mintee | heh, yep |
| 19:16.59 | mintee | thanks, just found it myself |
| 19:17.04 | *** join/#asterisk l2cache (n=chatzill@117.178.101.97.cfl.res.rr.com) |
| 19:17.27 | l2cache | Are there any good SIP providers that offer unlimited inbound minutes for a fixed rate per channel? |
| 19:19.01 | rob0 | well, there's ipkall, which might be good enough for your needs (DIDs in .wa.us) |
| 19:21.34 | *** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com) |
| 19:22.54 | l2cache | ok :) any others? |
| 19:23.38 | [TK]D-Fender | ~itsplist-us |
| 19:23.39 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:24.17 | l2cache | thanks |
| 19:25.01 | banzaika | anybody experienced a burst of noise (flat line) while on the phone ? |
| 19:26.09 | banzaika | mid-call |
| 19:30.22 | *** join/#asterisk fukz (n=basti@p5B06054B.dip.t-dialin.net) |
| 19:31.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:35.17 | hohum | how do I change the amount of time which goes by that asterisk waits for further DTMF digits before it times out? |
| 19:39.34 | banzaika | Set(TIMEOUT(digit)=10) |
| 19:40.45 | *** join/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net) |
| 19:40.46 | hohum | THANK OYUthanks |
| 19:41.07 | banzaika | np |
| 19:41.40 | LARefugee | I think it just hit me why I was having so much trouble registering to vonage and fwd through sip from behind my nat. |
| 19:42.36 | banzaika | round 2: anybody experienced a problem of an annoying sound in mid call (randomly) ? |
| 19:42.38 | *** join/#asterisk RoyK (n=roy@cnbokcafe.uio.no) |
| 19:42.46 | banzaika | like fax line |
| 19:43.16 | LARefugee | No one curious? |
| 19:43.22 | banzaika | go for it |
| 19:43.38 | LARefugee | Anyone use ethernet bonding on their server? |
| 19:44.01 | banzaika | bonding ? |
| 19:44.18 | LARefugee | two nics and the bonding kernel module. |
| 19:44.20 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:44.46 | nny_1 | has anyone implemented or tested a US reverse lookup system similar to http://www.monetra.com/~brad/callerid_shell.agi ? |
| 19:45.33 | LARefugee | guess not. I'll try googling. |
| 19:46.08 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:48.06 | banzaika | heh, bonding might be bad for registrations, imo |
| 19:48.11 | mercutioviz | Anybody know of Asterisk users in the central california? (Fresno, etc.) |
| 19:48.33 | mercutioviz | the central calif area, that is |
| 19:48.55 | LARefugee | banzaika: why do you say that? |
| 19:48.56 | rob0 | IIUC a bonded pair of interfaces would share a single IP address, each taking turns at the ARP layer. |
| 19:49.24 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:50.07 | LARefugee | rob0: Interesting. One of my nics is a realtek and I'm forced to use balanced-rr instead of balanced-alb. |
| 19:56.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.43 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:00.04 | LARefugee | Digium is taking forever to get back to me. Anyone ever have a problem with an X100M in the first port position on their TDM400P? Lots of noise on the channel? |
| 20:02.14 | LARefugee | Everybody playing hooky on Friday? |
| 20:02.54 | banzaika | sorry was afk |
| 20:03.20 | LARefugee | afk? I'm an irc noob. I figured out np. |
| 20:03.56 | banzaika | from what i read about bonding |
| 20:03.58 | banzaika | In Linux you can use the bonding kernel module for load balancing or Hot standby. The module combines multiple NICs into a single virtual interface. |
| 20:04.18 | banzaika | well that's different from rob0 said |
| 20:05.24 | banzaika | from my understanding you can use 2 physical layers to act as a single interface where one will take over if another is down |
| 20:05.48 | banzaika | i might be wrong though |
| 20:06.20 | rob0 | haha the only place I have a TDM400P, I no longer have telco service. But I did have to take out one of the FXS modules, I think it went south. |
| 20:07.26 | rob0 | Um, how does that differ from what I said? |
| 20:07.52 | Qwell | LARefugee: Have you called Digium support? |
| 20:08.06 | mintee | is there a function that will wait and listen to a key pressed without assuming it's an extension? |
| 20:08.07 | Qwell | oh, you said you did. nm |
| 20:08.08 | LARefugee | Oh I know that. I'm just wondering if anyone's had trouble with bonding. I'm going for throughput because I use the server for more than *. I have both nics plugged into the back of my buffalo router and peered with my vonage ata and a downlink to another switch. |
| 20:09.48 | LARefugee | rob0: Did you take it out of the first port? Did you hear rumbling, crackling on the channel? |
| 20:10.29 | LARefugee | Qwell: no e-mailed. Does IAXTEL work? |
| 20:10.50 | file | no, but you can directly call the PBX |
| 20:11.01 | banzaika | <LARefugee> whats your uplink speed ? |
| 20:11.29 | file | IAX2/guest@pbx.digium.com/s@default will call the main IVR |
| 20:11.33 | nny_1 | does this look correct exten => s,2,AGI(callerid_shell.agi|CALLERID(num)) |
| 20:11.50 | LARefugee | banzaika: no sure what you mean. From switch to switch is 100 mb full duplex. |
| 20:11.52 | nny_1 | they had AGI(callerid_shell.agi|${CALLERIDNUM}) as an example, but i don't think asterisk supports that |
| 20:12.06 | LARefugee | file: thanks. |
| 20:13.38 | banzaika | <LARefugee> you set as away, fyi |
| 20:14.35 | plik | hmmm, any suggestions how I can get rid of "__auto_congest: Auto-congesting call due to slow response" ? |
| 20:15.10 | LARefugee | blasted pidgin! |
| 20:15.10 | nny_1 | anyone wanna make a quick buck? |
| 20:15.29 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 20:15.34 | Katty | hai! |
| 20:15.48 | nny_1 | trying to implement the caller_id agi, a little unsure of some of the process, never used agi before.. I can pay someone to take a look and tell me what juju is missing |
| 20:15.49 | plik | hai2u kaitai |
| 20:16.08 | [TK]D-Fender | nny_1: exten => s,2,AGI(callerid_shell.agi|${CALLERID(num)}) |
| 20:16.15 | Katty | i has question about voicemail. |
| 20:16.17 | nny_1 | [TK]D-Fender: i'll try that thanks |
| 20:16.27 | Katty | there's all these audio files. unavailable, busy, and temp greeting. |
| 20:16.28 | [TK]D-Fender | nny_1: Don't forget to "reference" your function calls. |
| 20:16.37 | dkwiebe | plik: I belive you turn "qualify" off |
| 20:16.49 | dkwiebe | plik: it's been a while though. |
| 20:16.53 | [TK]D-Fender | mintee: "core show application read" |
| 20:16.57 | [TK]D-Fender | Katty: Mew. |
| 20:16.57 | Qwell | Katty: you can't. I'm guessing at the question. |
| 20:16.59 | plik | dkwiebe: thanks, I'll have a look |
| 20:17.05 | Katty | i see there are options to switch between the greetings, u, b, etc. |
| 20:17.06 | nny_1 | exten => s,2,AGI(callerid_shell.agi|${CALLERID(num)}) exten => s,3,NoOp(AGI Returned ${lookupname}) exten => s,4,Set(CALLERID(name)=${lookupname}) |
| 20:17.07 | Katty | oh :/ |
| 20:17.15 | Katty | really? :< |
| 20:17.18 | Qwell | Katty: continue though - you aren't asking what I thought you were :p |
| 20:17.20 | Katty | you can't do that? :< |
| 20:17.22 | Katty | oh! |
| 20:17.23 | Katty | phew. |
| 20:17.27 | Katty | /so/ |
| 20:17.30 | Qwell | unless you are, then no |
| 20:17.33 | Katty | how do i check what the phone is doing /first/ |
| 20:17.48 | Katty | on reject, go to voicemail(stuff,b) |
| 20:17.50 | Qwell | see the stdexten macro |
| 20:17.56 | Qwell | it does exactly that |
| 20:17.59 | Katty | jbot: stdexten? |
| 20:18.08 | Qwell | macro-stdexten in the sample config |
| 20:18.11 | [TK]D-Fender | nny_1: You might want to do some "lookup failure" safety checks, but thats otherwise sane. |
| 20:18.28 | [TK]D-Fender | Katty: "core show application chanisavail" |
| 20:18.30 | *** join/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net) |
| 20:18.40 | Qwell | [TK]D-Fender: don't even need to do that, if you're already dialing the phone |
| 20:18.53 | Katty | oooh |
| 20:18.54 | Qwell | (another assumption on my part, I guess) |
| 20:18.55 | Katty | i see how to do this |
| 20:19.03 | [TK]D-Fender | Qwell: depends on your definition of "busy". |
| 20:19.19 | [TK]D-Fender | Qwell: .... |
| 20:19.22 | [TK]D-Fender | ~assume |
| 20:19.23 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 20:19.51 | nny_1 | [TK]D-Fender: seemed to work, need to try it with a number that is listed by 411 google etc |
| 20:20.11 | Katty | what about this Temp Greeting, thing? |
| 20:20.16 | rob0 | My bum FXS was preventing the zaptel drivers from loading, so I had to remove it or not have zaptel. |
| 20:20.26 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
| 20:21.04 | [TK]D-Fender | rob0: foreign objects should not be plugged into your bum.... |
| 20:21.36 | [TK]D-Fender | Katty: "I'm on vacation and don't want to have to re-record my normal greeting when I come back" |
| 20:22.04 | Katty | [TK]D-Fender: yes, but is it... |
| 20:22.05 | Katty | u |
| 20:22.15 | Katty | or...b |
| 20:22.18 | Katty | or what is it lol |
| 20:22.24 | Katty | or does it just /play/ if it's there and ignores all others |
| 20:22.39 | LARefugee | rob0: fxs? That would be a s100m module. |
| 20:22.43 | [TK]D-Fender | Katty: it overrides b/u |
| 20:22.57 | Katty | cheers |
| 20:22.58 | Katty | i love it |
| 20:23.04 | *** join/#asterisk AsteriskRo (n=rosadesa@190.36.186.73) |
| 20:23.04 | *** part/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net) |
| 20:23.54 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 20:23.59 | DrkShadow | does anyone know the url/name of that packte5 software? I swear the name was "Packet5".. but I can't find anything on it. |
| 20:24.12 | *** join/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net) |
| 20:24.26 | rob0 | [TK]D-Fender: No wonder it didn't work! |
| 20:24.34 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 20:25.07 | LARefugee | digum shouldn't list iaxtel numbers on their website. It makes the rest of us "hope". |
| 20:25.46 | nny_1 | hmm agi script returns zero... |
| 20:25.57 | nny_1 | <PROTECTED> |
| 20:26.03 | *** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 20:26.16 | nny_1 | is there a way to debug the agi? |
| 20:26.27 | [TK]D-Fender | nny_1: "agi debug" |
| 20:26.35 | nny_1 | ha |
| 20:26.41 | nny_1 | yeah so damn intuitive it hurst ty |
| 20:26.43 | nny_1 | hurts too |
| 20:27.40 | [TK]D-Fender | ok, checkout time here, heading home. later all |
| 20:29.26 | unpaidbill | so im buying one of these handytone 286s |
| 20:29.30 | *** join/#asterisk ikevin (n=kevin@2001:5c0:99b9:0:20f:b0ff:fe4a:4589) |
| 20:29.36 | unpaidbill | i hope to be impressed. |
| 20:29.49 | LARefugee | is this chat archived? |
| 20:31.17 | banzaika | don't think so |
| 20:31.30 | banzaika | some clients do that by default |
| 20:31.42 | Qwell | ~logs |
| 20:31.43 | jbot | hmm... logs is apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 20:31.46 | Qwell | it is |
| 20:33.21 | banzaika | good to know |
| 20:33.25 | banzaika | thx |
| 20:33.45 | unpaidbill | ~handbook |
| 20:33.45 | jbot | [handbook] http://www.digium.com/handbook-draft.pdf |
| 20:34.17 | *** join/#asterisk wordzilla (n=me@122.109.126.193) |
| 20:34.27 | *** part/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 20:36.02 | Qwell | unpaidbill: you will be largely impressed |
| 20:36.03 | Qwell | ~gs |
| 20:36.04 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:36.09 | draygon | anyone feeling kind enough to help me with incoming call issue on asterisk? |
| 20:36.33 | LARefugee | ~digium |
| 20:36.34 | jbot | digium is probably the creator, primary developer, and maintainer of Asterisk. They have a full-time team dedicated to open source Asterisk development which carries the majority of the load. They also sell various hardware and software products, as well as provide support and development services. http://www.digium.com/ |
| 20:37.09 | LARefugee | jbot: I just got a GS bt200. I'm liking it. |
| 20:37.10 | jbot | You just got a GS bt200. I'm liking it.? |
| 20:37.10 | banzaika | draygon: post questions if people have answer you'll hear it |
| 20:37.34 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:38.42 | LARefugee | ~contact |
| 20:38.58 | LARefugee | ~digium support |
| 20:39.33 | LARefugee | I lost that connect to pbx.digium.com. I'll hack at it. file? |
| 20:40.35 | file | IAX2/guest@pbx.digium.com/s@default will call the main IVR |
| 20:40.42 | LARefugee | thanks! |
| 20:44.07 | rob0 | and without callerID ! :) |
| 20:44.15 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 20:44.15 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:46.04 | LARefugee | on teh phone with digium now. thx. |
| 20:50.01 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:50.09 | rob0 | I'm trying to decide between buying a FXO+FXS ATA, or moving my existing TDM11B. I don't have a dedicated machine to run as server, so it has to be shared on a dual-core AMD64 which is also functioning as a KDE workstation. |
| 20:50.55 | rob0 | I know the recommendation is no X11 on a zaptel+asterisk machine, is that still valid? |
| 20:52.19 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:52.24 | banzaika | X = bad on any server type machine. |
| 20:52.27 | danp | the recommendation is probably to not have things you don't really need |
| 20:53.12 | rob0 | haha |
| 20:55.35 | *** join/#asterisk mort___ (n=mort@user-54446aa1.lns4-c10.dsl.pol.co.uk) |
| 20:55.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.20 | *** join/#asterisk bakermd (n=none@38.104.0.102) |
| 20:58.33 | bakermd | Where should I ask questions regarding Polycom phones? |
| 20:59.30 | b11d` | here |
| 20:59.49 | b11d` | everyone here loves Polycom.. fire away :) |
| 20:59.49 | rob0 | or at Polycom support |
| 21:00.03 | b11d` | or your preferred Polycom vendor ;) |
| 21:01.10 | bakermd | IP 501 - bootrom 4.0.0.0423 - SIP 2.1.2.0078 :: Manually assigned IP address... can ping it from the network - it will not register. I would like to use the web interface for the phone, but it will not come up - as if it is not running |
| 21:02.43 | bakermd | I can do SIP tracing on my Asterisk box to get the registration working, but why would it not allow me to connect to the HTTP access? (I need to have 2 lines connecting to 2 servers, so at some point I will have to get to the interface...) |
| 21:02.48 | b11d` | i had that problem once, had to "reest to default" the configuration in the setup menu.. |
| 21:03.07 | b11d` | ~b11d |
| 21:03.08 | jbot | b11d is a constant source of misinformation... |
| 21:03.10 | b11d` | :) |
| 21:03.22 | bakermd | lol - okay ;) What is the more correct answer? |
| 21:04.34 | fskrotzki | you assigned the IP address but did you get the netmask correct? |
| 21:05.04 | bakermd | fskrotzki: Yes, simple class C.. 255.255.255.0 |
| 21:08.18 | *** join/#asterisk RoyK (n=roy@ti211310a080-7540.bb.online.no) |
| 21:08.24 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 21:08.33 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:09.41 | bakermd | So I guess I am confused - is using "reset to default" sound advice for fixing the IP 501 issue? (Pardon my ignorance - I do not know if jbot is actually a bot or not...) |
| 21:10.23 | [TK]D-Fender | jbot: are you a dog? |
| 21:10.31 | [TK]D-Fender | ~areyouadog |
| 21:10.32 | jbot | Bark! Bark! |
| 21:10.37 | rob0 | jdog! |
| 21:10.51 | bakermd | gotcha ;) |
| 21:12.53 | _ShrikE | ~botsnack |
| 21:12.53 | jbot | _ShrikE: :) |
| 21:13.03 | bakermd | b11d`: Reset local config or device setting? |
| 21:14.06 | bakermd | Resetting local config. |
| 21:14.38 | talntid | There was a problem connecting to mail.rtuinfo.com |
| 21:25.25 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 21:38.30 | *** join/#asterisk RoyK (n=roy@ti211310a080-7540.bb.online.no) |
| 21:45.12 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:47.58 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 21:50.56 | Tako-san | Anyone have a recommendation for a multi-port ATA device? In total we have 16 FXO ports that we want to connect to Asterisk via IP. |
| 21:51.07 | Tako-san | 16 FXS ports rather |
| 21:54.02 | [TK]D-Fender | Tako-san, 2 x Linksys SPA-8000 @ $230 ea |
| 21:55.15 | Tako-san | [TK]D-Fender: Really! You would recommend Linksys. Ok, if they will do the job. |
| 21:55.26 | rob0 | I'm trying to decide between buying a FXO+FXS ATA, or moving my existing TDM11B. I don't have a dedicated machine to run as server, so it has to be shared on a dual-core AMD64 which is also functioning as a KDE workstation. |
| 21:55.41 | Tako-san | [TK]D-Fender: Thanks. Will look into that product. |
| 21:55.49 | [TK]D-Fender | rob0, pile it on in good health. |
| 21:56.02 | rob0 | (I asked that during your drive home.) |
| 21:56.20 | [TK]D-Fender | rob0, My server hosts mail, web, ftp, routing, *, my 120" HT setup, and used to make me coffee. |
| 21:56.21 | rob0 | you think the TDM would be okay? |
| 21:56.38 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 21:57.12 | [TK]D-Fender | rob0, should be fine |
| 21:57.57 | rob0 | cool, thanks (it's worth a try anyway, and if I have trouble I can go with the ATA.) |
| 21:58.41 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 21:58.47 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 22:00.43 | muiro | pastebin seems to be... down. Any alternatives? |
| 22:01.02 | rob0 | ~pastebin |
| 22:01.03 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:01.05 | Tako-san | [TK]D-Fender: Are there other ATA devices you would recommend? Are there any shortcoming of the SPA-8000? You hve used it in a production environment yourself or have heard (good) things about it? |
| 22:01.25 | [TK]D-Fender | Tako-san, Multiple people here have used it and love it. |
| 22:01.46 | [TK]D-Fender | Tako-san, good size, flexible wiring, inexpensive. |
| 22:02.05 | Tako-san | [TK]D-Fender: Ok. The reason I am being a little cautious is when I hear the name "Linksys" I immediately think of consumer level routers/wireless access points. |
| 22:02.59 | Tako-san | [TK]D-Fender: But I am happy to hear the there are numerous people in this channel who are actively using the device and happy with it. Cheers. |
| 22:04.13 | muiro | Question: can I not compare a variable to "*" in GotoIf() or if() as such: http://rafb.net/p/GA2gXP42.html |
| 22:05.22 | [TK]D-Fender | muiro, Could be ok. that of course depends on the state of variables & priorities |
| 22:07.06 | muiro | [TK]D-Fender: everytime I try to make the comparison, I get this error: http://rafb.net/p/1KARsK53.html |
| 22:07.56 | [TK]D-Fender | muiro, And those single line warnings aren't doing any good. look at the BIG PICTURE. I jsut told you that the STATE of your variables can cause that to fail. |
| 22:08.09 | [TK]D-Fender | muiro, pastebin the ENTIRE deal at verbose 10 |
| 22:09.33 | muiro | same error at verbose 10 |
| 22:10.31 | muiro | looking up variable states on voip-info.org then I'll ask again with more info |
| 22:11.51 | *** join/#asterisk RoyK (n=roy@ip-42-56-149-91.dialup.ice.no) |
| 22:13.03 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 22:16.37 | muiro | [TK]D-Fender: you wouldn't be able to point me in the direction of any documentation regarding variable sattes would you? |
| 22:16.51 | muiro | [TK]D-Fender: or anything that would be pertinent |
| 22:16.52 | [TK]D-Fender | muiro, prove to me the variable isn't BLANK <- |
| 22:17.19 | rob0 | noop++ |
| 22:17.25 | [TK]D-Fender | muiro, because the way * would evaluate that would leave nothing on the other side of the "=" |
| 22:17.37 | muiro | [TK]D-Fender: I test for null directly before, but I'll throw a noop a little closer if it helps |
| 22:18.03 | [TK]D-Fender | muiro, pastebin the pwhole bloody thing. this 1 line at a time deal isn't getting you anywhere. |
| 22:18.20 | muiro | [TK]D-Fender: no need to be hostile |
| 22:20.39 | JayTee52 | muiro, he wasn't being hostile. |
| 22:21.14 | muiro | "bloody", but I'm the one basically begging for assistance so I can't really complain |
| 22:21.40 | JayTee52 | if he was actually being hostile he probably wouldn't be trying to help you |
| 22:22.09 | [TK]D-Fender | muiro : small tip then. When someone asks you for something to help you with a problem, that probably isn't a good time to start abstractly delcaring what is, and is not relevent. |
| 22:22.25 | [TK]D-Fender | muiro, Because if you knew.... you would have a problem. |
| 22:22.29 | [TK]D-Fender | wouldn't* |
| 22:22.32 | muiro | I didn't say it wasn't relevant. I'm currentl collecting the log |
| 22:23.40 | [TK]D-Fender | muiro, See I also don't know that your null test is any good either. A single error "message" doesn't prove much. so jsut place another call and pastebin the entire thing along with the dialplan that generated it. |
| 22:24.08 | muiro | I am |
| 22:24.14 | [TK]D-Fender | great |
| 22:25.48 | JayTee52 | wishes he had the BBC channel. |
| 22:26.32 | muiro | [TK]D-Fender: http://rafb.net/p/N8LCyv42.html |
| 22:27.21 | muiro | [TK]D-Fender: also earlier I null out MENU_CHOICE |
| 22:27.59 | muiro | [TK]D-Fender: I could put that in if you think it might be the problem |
| 22:29.47 | plik | JayTee52: which BBC Channel... depending where you are you can get some online at bbc.co.uk or zattoo.com |
| 22:29.53 | [TK]D-Fender | muiro, how about nooping it right before your test... |
| 22:30.30 | JayTee52 | I'm in Indiana, I'd have to pay extra to Comcast for it. |
| 22:30.33 | [TK]D-Fender | muiro, But this would be better regardless : exten => s,n,GotoIf($["${MENU_CHOICE}"="*"]?pds_main_menu,s,1) |
| 22:31.02 | plik | JayTee52: oh, those prolly won't work then - unless you can find a uk based proxy ;) |
| 22:31.28 | JayTee52 | I wanna watch this supposedly "cheesy" scifi show called Hyperdrive |
| 22:32.28 | plik | JayTee52: uknova will probably have torrents of it shortly |
| 22:32.55 | JayTee52 | plik, I'll keep my eye out for them, thanks! |
| 22:33.04 | plik | np :) |
| 22:33.29 | muiro | [TK]D-Fender: can do, making te call |
| 22:35.48 | muiro | [TK]D-Fender: earlier I attempted to double quote the "*", but adding the double quotes around ${MENU_CHOICE} fixed her up. Thanks for the debug. |
| 22:36.37 | *** join/#asterisk _polto_ (n=polto@elphelut.fttp.xmission.com) |
| 22:36.44 | _polto_ | hello all |
| 22:36.44 | muiro | [TK]D-Fender: I wasn't sure using double quotes would amount to much, asterisk not really doing much typing, at least not on the dialplan side |
| 22:36.51 | [TK]D-Fender | muiro, need it on both sides so that both have the "". its a literal char in the comparison, not an actual type-cast |
| 22:37.05 | muiro | [TK]D-Fender: gotcha, thanks much |
| 22:38.32 | _polto_ | can somebody help pls ? I am trying to configure a SIP trunk. My PBX is behind a GNU/Linux firewall with SIP helper. (it work for a SIP trunk provider). I am trying to connect a Sipura3000 located on the internet (real address, not NAT) as SIP trunnk. |
| 22:39.06 | [TK]D-Fender | muiro, You're welcome |
| 22:39.07 | _polto_ | [May 9 16:33:47] NOTICE[22459] chan_sip.c: Registration from 'elphelh <sip:lalala@192.168.1.16:5060>' failed for 'xxx.yyy.zzz.www' - No matching peer found |
| 22:39.18 | _polto_ | sorry |
| 22:39.30 | [TK]D-Fender | _polto_, undo any "sip helper", and follow this guide : |
| 22:39.32 | [TK]D-Fender | ~sipnat |
| 22:39.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:39.37 | _polto_ | this is the message I got in the Asterisk log. |
| 22:40.01 | [TK]D-Fender | _polto_, And its not ID-ing your device. looks like you didn't set it up right |
| 22:40.27 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:40.40 | _polto_ | [TK]D-Fender, on witch side? and why does it speak about the local address ? |
| 22:41.01 | [TK]D-Fender | _polto_, local address is because you didn't set up * to properly handle NAT. Follow the guide |
| 22:41.21 | [TK]D-Fender | _polto_, and the other issue could be either side. the problem is they don't both agree |
| 22:42.46 | _polto_ | [TK]D-Fender, thanks! |
| 22:42.50 | _polto_ | i am reading now. |
| 22:48.59 | *** join/#asterisk fluff (n=dune@snowflake.fluffigt.net) |
| 22:49.14 | _polto_ | do somebody know what can help ? sometimes my sipura3000 does not hangup the line properly and it's occupied. If I software reboot the Sipura the line is not released. I need to physically disconnect it and reconnect again. |
| 22:49.35 | _polto_ | Is there a way to release the line on sipura3000 remotely ? |
| 22:54.21 | [TK]D-Fender | _polto_, log into it and tell it to reboot. |
| 22:56.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583659.dsl.bell.ca) |
| 22:58.47 | _polto_ | [TK]D-Fender, I can log only with the web interface and if I apply changes and reboot, it does not hangup the line :( |
| 22:59.19 | [TK]D-Fender | <PROTECTED> |
| 23:05.08 | _polto_ | [TK]D-Fender, I mean if I call the line is occupied and the only way is to reboot sipura3000 physically |
| 23:06.04 | [TK]D-Fender | _polto_, But does the SPA *say* thats its busy? |
| 23:08.18 | _polto_ | [TK]D-Fender, how to know it ? |
| 23:08.31 | [TK]D-Fender | _polto_, look on the status screen |
| 23:09.54 | *** join/#asterisk mitcheloc (n=mitchel@216.207.245.1) |
| 23:11.44 | _polto_ | [TK]D-Fender, I suppose it's that : "Hook State:On" |
| 23:12.16 | *** part/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 23:12.49 | _polto_ | sorry, the PSTN line status is Hook State:Off. |
| 23:13.35 | _polto_ | [TK]D-Fender, PSTN State:PSTN Caller Accepted |
| 23:16.03 | *** join/#asterisk wordzilla (n=me@122.109.126.193) |
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| 23:23.01 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
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| 23:23.43 | Simon-- | anybody know which variables other than cid name, cid num are passed along an iax channel without patching in generic variable passing? |
| 23:23.57 | Simon-- | I don't feel like maintaining a patch just to pass a few other things.. |
| 23:30.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 23:48.54 | _polto_ | [TK]D-Fender, I think the issue with my trunk is just as stupid as port number. Sipura is connected is extension on port 5060, but use the port 5061 for the trunk. I never seen it before.. On my Asterisk box he message I see is : chan_sip.c: Registration from 'elphelh <sip:lalala@192.168.1.16:5060>' failed for 'xxx.yyy.zzz.www' - No matching peer found |
| 23:49.01 | _polto_ | port 5060 |
| 23:49.13 | _polto_ | may it be the reason ? |
| 23:50.03 | [TK]D-Fender | _polto_, could be. each peer needs its port set on those. |
| 23:50.17 | _polto_ | oh.. |
| 23:54.52 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |