| 00:01.39 | encode | somebody lied |
| 00:01.44 | encode | clearly < is < |
| 00:02.01 | C4colo | only if it is html encoded text |
| 00:02.09 | Katty | wickets. |
| 00:02.20 | Katty | wwhy didn't i name my dog Wicket? |
| 00:02.24 | Katty | that would have been just as cute. |
| 00:02.38 | encode | ~lart tzafrir_laptop |
| 00:02.38 | jbot | blasts tzafrir_laptop with a huge firehose then strangles tzafrir_laptop with it |
| 00:02.49 | encode | oh, it does different things. groovy |
| 00:03.14 | encode | well, thanks for your assistance Katty |
| 00:04.26 | encode | if i hang out and be quiet, will X-Rob and Qwell quit punching me? |
| 00:04.44 | Qwell | encode: maybe |
| 00:05.32 | X-Rob | encode: Maybe not. |
| 00:05.37 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 00:05.41 | X-Rob | You'll have to wait and see. |
| 00:06.19 | encode | ok |
| 00:06.44 | encode | props up a picture of himself in front of the computer and walks away |
| 00:09.18 | Katty | i really need to clean my house |
| 00:09.58 | seanbright | you can do mine next |
| 00:10.19 | seanbright | leave the dead hookers in the closet though, please. kthx. |
| 00:11.07 | unpaidbill | police notified |
| 00:11.36 | X-Rob | the police don't need any more dead hookers. Theyv'e got piles of their own. |
| 00:11.47 | *** join/#asterisk propellerhead (n=yogurt2u@host214.190-31-69.telecom.net.ar) |
| 00:13.00 | orkid | y do ppl kill hookers. wtf |
| 00:13.05 | Katty | can asterisk fax? |
| 00:13.09 | Katty | outgoing |
| 00:13.11 | Katty | not incoming |
| 00:13.28 | orkid | y do ppl kill hookers. wtf |
| 00:14.19 | C4colo | stop playing GTA and get back to work orkid |
| 00:14.41 | Katty | hmm. nevermind. |
| 00:14.44 | Katty | i don't want to fax |
| 00:15.12 | C4colo | good choice |
| 00:15.13 | seanbright | orkid: it's cheaper. |
| 00:15.28 | C4colo | faxing is antiquated and pointless |
| 00:15.40 | C4colo | it has been superceeded by newer protocols |
| 00:15.42 | X-Rob | stokes the boilers in his fax machine |
| 00:15.50 | C4colo | hahaha |
| 00:15.56 | Katty | i need something new |
| 00:16.00 | Katty | asterisk is getting boring |
| 00:16.29 | jaytee | I had fax working but then the wind died down and the sails are just hanging loose on their masts |
| 00:16.46 | X-Rob | Katty - Self Mutilation? That's always worth a laugh. |
| 00:17.03 | C4colo | only when it happens to other people |
| 00:17.08 | X-Rob | Exactly! |
| 00:17.14 | Katty | meh |
| 00:17.27 | C4colo | but you don't get the endorphin rush from it |
| 00:17.40 | X-Rob | Oh, I dunno. You'd be amazed at how hard you can laugh. |
| 00:17.43 | C4colo | I prefer spicy food for a good old-fashioned endorphin high |
| 00:17.52 | C4colo | good point |
| 00:17.58 | Katty | is there any fun new call management software like isymphony or fop |
| 00:18.15 | C4colo | I just use the CLI |
| 00:18.16 | jaytee | monast |
| 00:18.29 | Katty | oh? |
| 00:18.33 | C4colo | it is more exciting, with 50 simultanious calls it's like a game |
| 00:18.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:18.36 | Katty | i shall have to check that out |
| 00:18.45 | X-Rob | well there you go. anthm isn't dead. |
| 00:18.54 | X-Rob | 'lo! |
| 00:22.02 | Katty | oooh, an outlook dialing manager thingy |
| 00:23.03 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:27.52 | Katty | you know what sounds good? a bubble bath. |
| 00:28.21 | Katty | disappears |
| 00:32.31 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 00:33.18 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 00:35.19 | kerx | Hi everyone, if I'm doing a call transfer from an Asterisk system using a SIP broker, is there a way to send the party to a PSTN phone number eliminating the need for the asterisk system? |
| 00:44.08 | *** join/#asterisk ganeshix (n=chatzill@cpe-24-29-46-121.nycap.res.rr.com) |
| 00:44.58 | *** join/#asterisk riddlebox (n=james@ppp-70-242-131-65.dsl.stlsmo.swbell.net) |
| 00:45.46 | riddlebox | whats the best way to do a timeout in a auto attendant? |
| 00:46.21 | jaytee | Wait()? |
| 00:46.45 | riddlebox | jaytee, if I do that can a user press a key while it is waiting? |
| 00:47.41 | jaytee | riddlebox, perhaps if you explained in more detail what it is you're trying to accomplish it would be easier to advise the best course of action. |
| 00:49.27 | riddlebox | jaytee I want to have a user call in, have a background play, and if they dont press anything after 6-10 seconds, go to a general mailbox |
| 00:50.47 | jaytee | riddlebox, hang on a sec |
| 00:52.02 | riddlebox | jaytee, exten = s,3,Set(TIMEOUT(response)=5) |
| 00:52.02 | riddlebox | exten = s,4,Goto(default|201|1) |
| 00:52.02 | riddlebox | exten = s,5,Hangup |
| 00:52.06 | X-Rob | the 't' exten gets triggered after a timeout |
| 00:52.20 | X-Rob | exten = t,1,Goto(atimeout,s,1) |
| 00:52.23 | kerx | nice |
| 00:52.34 | kerx | You guys are good & fast |
| 00:52.43 | jaytee | riddlebox, look in the book on page 375 for a description of Background() |
| 00:53.16 | kerx | What is the best way to transfer an outbound caller to another PSTN Phone# through asterisk? |
| 00:53.39 | Katty | so, i had this fantastically devious idea mid-bubblebath. |
| 00:54.09 | Katty | i'm going to setup an extension to play a recording of the typical sex hotline IVR attendant |
| 00:54.21 | Katty | maybe even make a pretend ivr for it. |
| 00:54.35 | Katty | and make sure that in there, is something about being billed some absurd ammount of money per minute |
| 00:54.49 | Katty | and every time a telemarketer calls, i'm going to have the receiptionist send them there. |
| 00:55.33 | Katty | maybe find an audio snippet off youtube |
| 00:55.45 | Katty | with real bowchicabowow music in the background. |
| 00:56.10 | kerx | lol |
| 00:56.18 | kerx | good idea |
| 00:56.41 | jaytee | riddlebox, and you'd probably want to use WaitExten() as the next priority to wait for digits. The Background() app can play the menu choices and the WaitExten() will take the user input and then try to find a matching extension within the current context. |
| 00:56.49 | Katty | maybe use a real sex website to reference so it seems even more real |
| 00:57.17 | Katty | here we go. one stop phone sex shop |
| 00:57.26 | kerx | jaytee, do you know the issue i'm having? |
| 00:57.51 | jaytee | nope |
| 00:58.06 | Katty | thank you for calling the one stop phone sex shop. we offer low rates, mutiple billion options, and online indicators that tell you when your favorite /insert horrible world/ is available. |
| 00:58.11 | hardwire | its techno time |
| 00:58.11 | hardwire | damnit |
| 00:58.15 | Katty | s/billion/billing/ |
| 00:58.25 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 00:58.25 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 00:58.39 | jaytee | oh! the transfer thingy? good luck with that. You'd need a SIP Refer and getting that to work with authentication with your ITSP is gonna be a "fun time" |
| 00:58.56 | Katty | we offer many special discounts for our callers, 2 girl calls are also available... you know somethin real stupid but believable |
| 00:59.16 | kerx | thanks |
| 00:59.26 | kerx | and you guys have heck a lot of humour going on :) |
| 00:59.50 | Katty | if you're going to have a phone system that can make the cdrom eject everytime someone calls microsoft.com |
| 00:59.54 | Katty | you might as well have fun with it |
| 01:00.11 | jaytee | without a sense of humor most Asterisk hacks would have committed suicide by now, especially when you consider all the marvelous documentation at our disposal. |
| 01:00.12 | kerx | heck yea |
| 01:00.40 | kerx | so i need to have the asterisk server send a SIP Refer, then the caller will directly transfer to the PSTN # i send it to? |
| 01:00.49 | kerx | what kind of function in asterisk is this? I assume it's not Dial() |
| 01:00.50 | kerx | :) |
| 01:01.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d06a32bbfda3caaf) |
| 01:02.10 | jaytee | kerx, it's way over my head so I can't help ya with it. |
| 01:02.16 | kerx | k |
| 01:02.47 | jaytee | I only understand 2 Asterisk commands, Dial() and Hangup(). The rest are a mystery. |
| 01:04.05 | Katty | do what now? |
| 01:04.23 | Katty | what function, kerx? |
| 01:04.36 | kerx | Katty, well i'm currently doing the following: |
| 01:04.47 | kerx | I take an outbound call that was made through Asterisk to a regular phone # |
| 01:04.54 | Katty | nods |
| 01:04.54 | jaytee | wow, I'm watchin 2001: A Space Odyssey and I'd forgotten how much has changed. They just referred to the temperature of a hibernating human as 3 degrees centigrade. |
| 01:05.01 | kerx | and I want to transfer that caller to another phone number that is a regular phone # |
| 01:05.04 | kerx | I use Dial() |
| 01:05.14 | Katty | so |
| 01:05.15 | kerx | When I do this, it initiates another SIP connection, and does the entire thing via the * system |
| 01:05.21 | kerx | This causes double billing :-( |
| 01:05.32 | kerx | And more load on the asterisk system that is necessary |
| 01:05.37 | Katty | i'm not sure i get it |
| 01:05.44 | Katty | don't give me commands, pretend you're doing it |
| 01:05.48 | Katty | you pick up the phone and dial... |
| 01:06.07 | Katty | you dial one thing, and you want the server to dial something different? |
| 01:06.12 | kerx | * dials using SIP service provider to 818-555-1212 |
| 01:06.39 | kerx | Person at 818-555-1212 want's to speak to another person than she/he is currently speaking to |
| 01:06.57 | kerx | * transfer's the caller at 818-555-1212 out to 818-555-1313 |
| 01:07.02 | kerx | right now, I use the Dial() function |
| 01:07.03 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:07.05 | Katty | ah |
| 01:07.06 | Katty | so |
| 01:07.09 | Katty | that's their DIDs |
| 01:07.15 | Katty | what's their extension? |
| 01:07.29 | kerx | They don't have extension's, it's just a regular PSTN number's |
| 01:07.36 | kerx | the 818-555-1212 is a Customer |
| 01:07.47 | Katty | so two different servers? |
| 01:07.49 | kerx | the 818-555-1313 is a T1 Support Center |
| 01:08.03 | C4colo | I'm calling your customer right now to see if they want to switch providers =) |
| 01:08.04 | kerx | The * server is the Call Verifier that makes the initial outbound call to the customer |
| 01:08.23 | Katty | does the same asterisk server host both DIDs? |
| 01:08.38 | Katty | or are there two * servers involved? |
| 01:08.48 | kerx | Nope, * server only makes the initial outbound call |
| 01:08.59 | kerx | The other two are both PSTN |
| 01:09.07 | Katty | good luck with that |
| 01:09.15 | kerx | Hehe, yep!!!!! :) |
| 01:09.17 | jaytee | Katty, exactly what I said |
| 01:09.25 | Katty | now if it was IP |
| 01:09.27 | *** join/#asterisk nikko (n=nikko@adsl-074-182-164-013.sip.hsv.bellsouth.net) |
| 01:09.27 | kerx | Is it even possible w/ Analog phone's ? |
| 01:09.32 | Katty | no |
| 01:09.43 | jaytee | Zap has a transfer function |
| 01:09.45 | Katty | with an IP phone you could do some fun stuff |
| 01:09.57 | kerx | Katty, Yep, I'm going to have to bring in the IP layer, the problem is |
| 01:10.02 | kerx | The office only has DSL :-( |
| 01:10.07 | Katty | that's not a big deal |
| 01:10.14 | Katty | you can get sip trunks functional on DSL |
| 01:10.15 | kerx | I don't think I can setup for this customer 20 live agents taking calls through a DSL connection |
| 01:10.35 | kerx | 20 live calls w/ DSL @ 300-500kbps |
| 01:10.40 | jaytee | and internally you could use a SIP Refer header but trying to do that through an ITSP over a SIP trunk is going to be a major FAIL |
| 01:10.40 | Katty | ah |
| 01:10.41 | Katty | no |
| 01:10.45 | Katty | you'd need more like a 2x2 |
| 01:10.49 | kerx | Yep |
| 01:10.54 | kerx | 2mbps makes more sense |
| 01:11.04 | Katty | could try upgrading |
| 01:11.14 | kerx | They want to keep there T1, because they don't want to start experiencing jitter, and all the network problems that can begin happening |
| 01:11.20 | *** join/#asterisk chendy (n=chatzill@59.40.223.24) |
| 01:11.28 | *** part/#asterisk ganeshix (n=chatzill@cpe-24-29-46-121.nycap.res.rr.com) |
| 01:11.29 | kerx | I wouldn't have enough experience to tell them, don't worry about it! |
| 01:11.52 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:11.59 | kerx | Would they start experiencing drop calls, jitter, echo, etc.. etc.. w/ something like 20 agents on a 2x2 to a nice dedicated server w/ not too many hops? |
| 01:12.05 | Katty | anyway, analog phones are old and dumb |
| 01:12.18 | Katty | use 1 machine as host server |
| 01:12.18 | kerx | Yea, but somebody told me at a provider, they currently can do that w/ Analog phones |
| 01:12.22 | kerx | Blind transfer somebody |
| 01:12.25 | Katty | then open up firewall ports and connect all offices remotely |
| 01:12.48 | kerx | For example, you call the operator |
| 01:12.52 | Katty | that way no matter where you are, you just dial their extension, and POOF there they are |
| 01:12.55 | kerx | You tell her to transfer you somewhere |
| 01:13.00 | kerx | She transfer's you, then she leaves. |
| 01:13.05 | Katty | yep, that's blind |
| 01:13.12 | kerx | Blind transfer doesn't work with * ? |
| 01:13.16 | Katty | yes it does |
| 01:13.23 | Katty | attended transfer also works |
| 01:13.39 | kerx | It needs IP or can it blind transfer to another PSTN #? |
| 01:13.46 | Katty | both |
| 01:13.56 | Katty | you can take an incoming call |
| 01:13.57 | kerx | So, wouldn't that be my solution? |
| 01:14.00 | Katty | and blind transfer it out to a cellphone |
| 01:14.04 | Katty | but it will still hold two lines open |
| 01:14.18 | Katty | channel 1 incoming, channel 2 outgoing, pipe audio through server |
| 01:14.27 | kerx | What about : |
| 01:14.31 | Katty | drops both channels when call is over |
| 01:14.38 | kerx | Channel 1 Outgoing -> PSTN |
| 01:14.47 | kerx | Channel 1 Transfers the call to -> another PSTN |
| 01:14.47 | Katty | so, someone calls out. |
| 01:14.48 | *** join/#asterisk kisu (n=kkang@ip70-179-88-179.dc.dc.cox.net) |
| 01:15.00 | kerx | * server calls to 818-555-1212 |
| 01:15.05 | kerx | * server transfers call to 818-555-1313 |
| 01:15.09 | kerx | that's what happens :) |
| 01:15.22 | Katty | that doesn't make sense to me. |
| 01:15.36 | kerx | I use my asterisk server to call Ms. Katty |
| 01:15.39 | Katty | a sip phone dials out a channel to where? |
| 01:15.46 | Katty | okay so SIP -> Asterisk -> Telco? |
| 01:15.47 | kerx | I transfer Ms. Katty to my friend Joe Blow ;P |
| 01:16.08 | kerx | I want the initial call my asterisk server made to u, to go away, and now u are connected to Mr. Jow Blow ;P |
| 01:16.16 | Katty | nope |
| 01:16.20 | Katty | won't happen |
| 01:16.25 | kerx | fawk! |
| 01:16.29 | kerx | double billing it seems :-( |
| 01:16.35 | kerx | Does analog do it? |
| 01:16.40 | Katty | no |
| 01:16.43 | Katty | your server is the gateway |
| 01:16.56 | Katty | calls don't just magically disappear unless they are hungup on |
| 01:17.03 | kerx | Well, how does for example the Operator at AT&T transfer me to Business#1, and then she hangs up and I stay connected? |
| 01:17.12 | jaytee | ok, simple explanation! when he does this he's initiating the transfer IN ASTERISK internally. Asterisk maintains both the current connection and then bridges the call internally. He wants to transfer so that Asterisk and his 2 SIP trunks are no longer being used. This kind of transfer requires a SIP Refer to the first called party so that it initiates the call to the other party. |
| 01:17.12 | RypPn | its called route optimisation |
| 01:17.17 | Katty | just because she hangs up doesn't mean the channel hangs up |
| 01:17.48 | *** join/#asterisk chaozer (n=chaozer@c83-254-163-148.bredband.comhem.se) |
| 01:18.07 | kerx | jaytee, Correct on the dot, it sounds like based on your explanation that it IS POSSIBLE |
| 01:18.31 | Katty | neat. what's a SIP refer? |
| 01:18.39 | kerx | I donno |
| 01:18.49 | chaozer | Anyone present that uses the SIP realtime ifc in asterisk ? |
| 01:18.49 | jaytee | kerx, it is possible in many internal scenarios, the problem comes in with authentication to your ITSP and most of them will block the SIP Refer. |
| 01:18.55 | hardwire | die now |
| 01:19.11 | kerx | jaytee, So for example.... to understand clearly |
| 01:19.24 | Katty | hardwire: yes, i'm feeeling the same. |
| 01:19.27 | kerx | I use CompanyA and I get a SIP Trunk from them, that allows outbound calls |
| 01:19.34 | kerx | I use CompanyA to make those outbound calls |
| 01:19.55 | jaytee | Katty, I have Exchange 2007 UM setup at work. When I call my Exchange server and tell it to call a number in my Contacts or in the Directory it issues a SIP Refer request to my SIP phone. My SIP phone rings and I pick it up and it then dials the other party. |
| 01:19.55 | kerx | CompanyA needs to allow SIP Refer requests so that they can tell the outbound caller to connect to the other line directly ? |
| 01:20.04 | jeev | http://www.babble.com/CS/blogs/strollerderby/archive/2008/10/09/sarah-palin-s-high-school-grades.aspx lol she got like an 800 combined on SAT's |
| 01:20.53 | kerx | jaytee, was I correct on the above statement? |
| 01:22.44 | jaytee | kerx, yes and you need to find out how to make Asterisk issue a SIP Refer. It would involve using the SipAddHeader() application but I haven't been able to find any documentation on using that other than for Diversion or Notify requests. |
| 01:23.14 | kerx | Nice |
| 01:23.29 | kerx | Now I want to make sure you don't just mean parties who make the call can be transferred |
| 01:23.30 | Katty | neat\ |
| 01:23.47 | kerx | In my case I'm trying to figure out how to take a person who we initiated a call w/ and transfer that party to another PSTN# |
| 01:23.53 | [TK]D-Fender | kerx: first I seruosly doubt your provider will let you hand off 2 calls that go though them and free up your account for more incoming calls. |
| 01:24.24 | [TK]D-Fender | kerx: Second you'd be looking at "core show application transfer" |
| 01:24.30 | jaytee | I've been meaning to use wireshark to capture SIP requests to my phone while making a Click to Call or similar that uses SIP Refer to study the packet header info and try to figure out the correct syntax and header structure from that. |
| 01:24.39 | [TK]D-Fender | kerx: usually the only option is for * to remain in the middle |
| 01:25.07 | [TK]D-Fender | jaytee: You're thinking too hard... "transfer" |
| 01:25.44 | jaytee | [TK]D-Fender, won't that still keep * in the middle? |
| 01:25.50 | [TK]D-Fender | jaytee: No |
| 01:26.06 | jaytee | locally no, but if he does it with his ITSP? |
| 01:26.27 | kerx | As far as I what saw Gafachi is the provider I'm using, and in the ALLOW: on the SIP header's it stated REFER |
| 01:26.39 | kerx | So it might be allowed |
| 01:26.44 | kerx | I just have to figure out how to make it happen :) |
| 01:26.48 | [TK]D-Fender | jaytee: If both legs are SIP calls, they'll re-invte. Then if both legs are also with the same provider, it should recognize that and try to treat it like a 2BCT |
| 01:27.02 | *** join/#asterisk nikko (n=nikko@adsl-074-182-164-013.sip.hsv.bellsouth.net) |
| 01:27.08 | [TK]D-Fender | kerx: I HIGHLY doubt any vendor you'll find will support this |
| 01:27.39 | [TK]D-Fender | telcos often allow 2BCT over PRI |
| 01:31.29 | jaytee | ok, we're getting to the good part. HAL isn't going to open the pod bay doors. :-) |
| 01:31.29 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 01:32.23 | kerx | what is 2BCT? |
| 01:33.12 | [TK]D-Fender | kerx: 2 B-Channel Transfer |
| 01:33.48 | [TK]D-Fender | kerx: When you want to pass off 2 channels from a provider and have the telco reconnect them outside of your link to them |
| 01:34.02 | jaytee | so you can free up your B channels for other calls |
| 01:36.24 | jaytee | and of course if you ask your ITSP for something similar they'd say "Sure! No problem....and would you like pie with that?" |
| 01:36.26 | kerx | fawk |
| 01:36.48 | kerx | i'm sure there has to be a SIP provider that allows this |
| 01:36.56 | kerx | i'm going to test it, i'll let y'all know how it works out |
| 01:37.04 | kerx | What functions should I be looking at? |
| 01:37.07 | kerx | for the REFER? |
| 01:37.12 | jaytee | be sure to write a how-to and post it on the WIKI |
| 01:37.31 | kerx | jaytee, lol, i promise you i will if i can get this figured out |
| 01:37.36 | [TK]D-Fender | kerx: You seem to miss the poitn. They want to make money off you. Offering this service at no extra cost is extremely unlikely. |
| 01:37.53 | [TK]D-Fender | kerx: and I TOLD you the function for this. Pay attention |
| 01:37.56 | kerx | [TK]D-Fender, yeah, I understand that. I'm going to be blind at that for now, until I see it fail |
| 01:38.07 | kerx | [TK]D-Fender, I'm sorry, let me read up in our chat logs |
| 01:38.22 | kerx | 'core show application transfer' |
| 01:38.30 | [TK]D-Fender | wonders why he bothers. You can't even just hand out the answer any more... |
| 01:38.34 | kerx | That looks like a rasterisk console statement? |
| 01:38.59 | kerx | Not sure what you mean by "You can't even just hand out the answer any more..." |
| 01:39.00 | kerx | ? |
| 01:39.48 | [TK]D-Fender | now wonders why they also can't get sarcasm about all the other things they don't get.... then again at least that shows symmetry. |
| 01:39.58 | jaytee | hahahahaaa |
| 01:40.22 | kerx | sighs. |
| 01:40.32 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 01:40.42 | jaytee | is reminded of the title of a Rod Stewart album, "A nod is as good as a wink to a blind horse" |
| 01:47.22 | murdock_ut | Come on my little atom processor.....compile |
| 01:48.33 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 01:48.33 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 01:49.10 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 01:49.10 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 01:50.25 | *** join/#asterisk chendy (n=chatzill@219.134.30.43) |
| 01:59.47 | chaozer | hmm .. Im getting the following in my log whe trying to login via SIP: |
| 01:59.53 | chaozer | [Oct 24 03:47:08] DEBUG[1762] db.c: Unable to find key 'adamw' in family 'SIP/Registry' |
| 02:00.45 | *** join/#asterisk gones (n=gones@61.141.80.81) |
| 02:00.47 | chaozer | anyone got any idea ? ;) |
| 02:02.50 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:02.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:04.18 | chaozer | oh well.. this sucks :) |
| 02:10.42 | thehar | anyone around? *whistle* |
| 02:10.50 | drmessano | no |
| 02:10.54 | thehar | haha |
| 02:11.06 | jaytee | nope, no one's here |
| 02:11.12 | thehar | trying to find a script to do some call load generation testing |
| 02:11.14 | thehar | over pri |
| 02:11.51 | drmessano | WTF is a "rasterisk" |
| 02:11.59 | drmessano | [21:38] <kerx> That looks like a rasterisk console statement? |
| 02:12.03 | drmessano | He's used that TWICE |
| 02:12.10 | jaytee | drmessano, I was wondering that myself |
| 02:12.17 | kerx | rasterisk = asterisk -r |
| 02:12.29 | drmessano | How does that even make sense? |
| 02:12.33 | kerx | yeah, i might have sounded really stupid tonight |
| 02:12.36 | LiNeTuX_Home | drmessano: I think that's Astro from the Jetsons' way of calling Asterisk |
| 02:12.37 | kerx | hope you guys get a kick out of it |
| 02:12.51 | drmessano | vasterisk is asterisk -v? |
| 02:12.58 | jaytee | oh, so now we're making up new words? how about wonklybeeble? sounds like a good one to me. |
| 02:14.05 | drmessano | jaytee: I just ran coresterisk sipshow magoodle, and I don't see my wonklybeetle.. Shall I binpaste my figcondoodle? |
| 02:14.26 | drmessano | Correct answer: Meep Meep |
| 02:14.49 | jaytee | sure! why the frigtrixbox not? |
| 02:14.59 | kerx | hey didn't the author create the word DUNDI |
| 02:15.01 | kerx | hahahahahahahahaha |
| 02:15.03 | kerx | get a kick out of that |
| 02:15.16 | kerx | c'mon laugh out loud friend's |
| 02:15.29 | kerx | even start rolling on the floor maybe |
| 02:15.36 | jaytee | which author? Hemingway? J.D. Salinger? |
| 02:15.45 | kerx | yeah, exactly that one |
| 02:15.46 | kerx | u got it :) |
| 02:15.49 | kerx | rotfl |
| 02:15.50 | kerx | lol |
| 02:15.57 | kerx | kicks and starts dancing to the macarena |
| 02:16.21 | LiNeTuX_Home | only laughs when the nice people in the box laugh |
| 02:16.43 | jaytee | I have never and will never under no circumstances whatsoever dance the macarena. |
| 02:16.59 | drmessano | I only laugh when Jesus tells me to.. and right now, he's telling me angry things. Lots and lots of angry things/ |
| 02:17.27 | LiNeTuX_Home | drmessano: don't forget the tear gas |
| 02:18.02 | jaytee | drmessano, does he listen if you talk back? if he does, tell him I said, "The Powerball number were wrong, biatch!" |
| 02:18.11 | jaytee | numbers |
| 02:18.39 | drmessano | jaytee: You made him angrier. When he finishes his jello sandwich, he said you will perish. |
| 02:18.43 | lanning | has to go talk to the king of the potato people... |
| 02:19.05 | LiNeTuX_Home | Hurrah! The yellow light has come to save us! |
| 02:19.42 | jaytee | drmessano, tell him I called him an incompetent diety and a lousy absentee landlord and then challenged him with "Bring it on!" |
| 02:20.14 | drmessano | I like candles and things that are burny. I can feel their hate. I like feeling the hate, and hearing the screams. |
| 02:20.17 | LiNeTuX_Home | smells smoldering wood |
| 02:20.17 | jaytee | might as well worship the damn Claw machine in the supermarket for all the good it'll do. |
| 02:20.28 | *** join/#asterisk srf21c (n=srf@ip98-165-60-42.ph.ph.cox.net) |
| 02:20.34 | *** part/#asterisk srf21c (n=srf@ip98-165-60-42.ph.ph.cox.net) |
| 02:21.05 | drmessano | Oh heh, sorry.. back to talking about Asterisk.. I need to finish this hole I am making in my leg to get the steam out. |
| 02:21.54 | [TK]D-Fender | jaytee: Watched Toy Story a few too many times? |
| 02:22.10 | jaytee | just once, maybe twice |
| 02:22.34 | [TK]D-Fender | The Claw has chosen! |
| 02:22.41 | jaytee | but the only real talent I've discovered I have is actually winning at one of those claw machines. |
| 02:22.42 | *** join/#asterisk lolipops (n=lolipops@modemcable238.118-82-70.mc.videotron.ca) |
| 02:22.47 | LiNeTuX_Home | i think i can re-enact the entire movie now |
| 02:23.09 | jaytee | so I ended up in a lousy paying job with little money and a closet full of stuffed toys. |
| 02:23.23 | lolipops | i have a problem with asterisk 1.4.22 w/ imap voicemail. when using VoiceMailMain, if the call is terminated before a valid user is specified, asterisk crashes. |
| 02:23.24 | drmessano | Oh admit it.. Jaytee likes to watch Toy Story on the DVD player in his chevy van with melted milk duds in his pocket and a warm bottle of cheerwine. |
| 02:23.53 | jaytee | I have a good friend who can recite verbatim the entire script of Alice's Restaurant |
| 02:23.53 | LiNeTuX_Home | cheerwine? is that worse than MD 20/20? |
| 02:24.11 | drmessano | _not parked outside of a elementary school either_ |
| 02:24.28 | drmessano | Dreaming about having a ferris wheel on his front lawn... |
| 02:25.19 | jaytee | I'm pretty good with The Further Adventures of Nick Danger by Firesign Theatre |
| 02:26.44 | drmessano | My father used to make us listen to "Dont crush that Dwarf, hand me the pliers" somewhat endlessly |
| 02:28.15 | jaytee | "Why, that's nothing but a two bit ring from a Crackerback Jox!" "I'll sell it to you for five thousand dollars?" "Five thousand! What kind of a fool do you take me for?" "First class!" |
| 02:28.56 | LiNeTuX_Home | mmmm. Raffi. |
| 02:30.53 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:35.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:42.15 | chaozer | beats his head against the wall. |
| 02:42.32 | chaozer | im going to loose it. |
| 02:42.43 | jaytee | the other wall is softer |
| 02:43.54 | hesco | might there be a clean-samples target in the 1.6.01 Makefile? My cdr_pgsql was working until I ran make samples multiple times. Somehow I've got to clean out this mess and start over. |
| 02:49.23 | *** join/#asterisk axisys (n=axisys@117.18.231.53) |
| 02:50.50 | chaozer | ths so weird. |
| 02:51.17 | chaozer | i can do 'realtime load sipusers name adamw' in the CLI and it works just fine |
| 02:51.59 | chaozer | but registering doesnt work ... |
| 02:53.11 | *** join/#asterisk LemensTS (i=LemensTS@adsl-70-238-149-98.dsl.stlsmo.sbcglobal.net) |
| 03:12.40 | *** join/#asterisk CrazyTux (n=brandon@user-vcauq7r.dsl.mindspring.com) |
| 03:22.21 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.118) |
| 03:23.52 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 03:34.40 | *** join/#asterisk slingr (i=santas@will.one.day.hack-the-pla.net) |
| 03:39.42 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-254-206.dsl.pltn13.sbcglobal.net) |
| 03:48.17 | slingr | is there a way to set asterisk realm in sip.conf for a specified trunk? |
| 03:49.51 | hesco | what does this mean? res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!, a Notice thrown to the console. |
| 03:52.56 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 04:01.34 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 04:01.40 | kerx | Hey y'all I'm back after cooling myself down |
| 04:01.58 | kerx | Sorry for what happened earlier today in the channel, both the ignorant conversation, and the stuff I pulled afterwords |
| 04:02.00 | kerx | My bad! |
| 04:02.17 | kerx | I anyways tried the REFER, and would like to share w/ you my experience :) |
| 04:02.27 | kerx | Anyone around to see the logs :) ? |
| 04:03.11 | kerx | http://pastebin.ca/1235373 |
| 04:04.16 | *** join/#asterisk Gopher_77 (n=jim@akro-pool1-cs206.pool.dslohio.net) |
| 04:12.58 | hesco | will a make clean remove anything from /etc/asterisk ??? |
| 04:14.30 | kerx | no |
| 04:14.30 | hesco | hiding out are we? |
| 04:14.58 | *** part/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 04:15.01 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 04:15.46 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 04:15.47 | hesco | kerx: is there a question you have about those logs? |
| 04:17.20 | kerx | hesco, It looks like the provider s telling me to go away or something |
| 04:17.26 | kerx | I'm not sure what it is exactly though |
| 04:18.00 | kerx | what do u think? |
| 04:19.18 | [TK]D-Fender | kerx: what do you mean "what it is"? |
| 04:20.21 | kerx | Oh, :) well it looks like what you mentioned earlier is correct |
| 04:20.30 | [TK]D-Fender | Imagine that... |
| 04:20.43 | kerx | They gave me a SIP/2.0 405 Method Not Allowed |
| 04:20.52 | kerx | I just wanted to be 100% sure that, that is the case |
| 04:21.02 | kerx | I used the Transfer() method, and the dialplan is exactly: |
| 04:21.03 | [TK]D-Fender | kerx: 405 is SIP for GTFO :) |
| 04:21.33 | kerx | exten => 1,1,Transfer(SIP/gafachi/18183453045) |
| 04:21.43 | kerx | so I wanted to call that PSTN# using that SIP |
| 04:21.48 | kerx | :-( |
| 04:21.54 | kerx | Anyways, yeah, you were 100% right |
| 04:22.08 | kerx | But I'm weirded out how other people provide those features |
| 04:22.41 | kerx | I noticed this: http://bugs.digium.com/view.php?id=3554 |
| 04:22.59 | [TK]D-Fender | kerx: And I want a million dollars. You just don't see ITSP's lining up to make ME happy either. |
| 04:23.25 | kerx | [TK]D-Fender, It's a patch for doing exactly what you mentioned 2BCT |
| 04:23.28 | [TK]D-Fender | kerx: They are there to make MONEY off you. Letting you hand off calls and pass to 3rd parties, etc isn't in their fiscal interest |
| 04:23.29 | *** join/#asterisk axisys (n=axisys@117.18.230.23) |
| 04:23.49 | kerx | oh roger that, but you know... there's always ways :) |
| 04:23.55 | kerx | if u push hard enough |
| 04:24.08 | kerx | "Two-B-channel transfer with Zap; note that 2BCT service just exists for Lucent 5ESS switches, then a regular Dial() on the same PRI span should invoke 2BCT; "transfer" keyword in zapata.conf enables/disables 2BCT on channels " |
| 04:24.11 | kerx | That's a quote from the patch |
| 04:24.17 | [TK]D-Fender | kerx: Sure... find someone willing to take enough money from you to do it. But then again that money could have bought better ways. |
| 04:24.34 | kerx | What are those "better ways" ? :P |
| 04:25.09 | [TK]D-Fender | kerx: The patient is dead, put the defibrillator down there's a coal mine cashing in with every jolt... |
| 04:25.42 | [TK]D-Fender | kerx: pay for mor channels, let them sit in Dial like the rest of us <- |
| 04:25.56 | kerx | hehe |
| 04:26.24 | kerx | well the customer right now i'm trying to get is currently using analog system that doesn't charge for the (double billed) minutes |
| 04:26.34 | kerx | they are doing call transfer's w/ some technology, not sure what exactly... |
| 04:26.55 | kerx | only way to help the customer is to be able to do this for them, otherwise they won't even consider my services for voip |
| 04:26.57 | kerx | so .... |
| 04:26.58 | *** join/#asterisk jameswf-home (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 04:27.01 | kerx | <- loses customer :P |
| 04:27.12 | kerx | sad, but i guess it has to happen |
| 04:27.57 | [TK]D-Fender | kerx: So maybe, just MAYBE you don't have a leg to stand on or a viable business model. That's what we call CAPITALISM :) |
| 04:28.16 | kerx | possibly |
| 04:28.16 | [TK]D-Fender | kerx: BTW, that'll be 50$ for this introductory business course ;) |
| 04:28.32 | kerx | lol, send me the course objective and i will consider :P |
| 04:30.26 | [TK]D-Fender | RP2008!!! |
| 04:30.33 | [TK]D-Fender | runs around in circles |
| 04:31.36 | lolipops | [TK]D-Fender, any application to check if a voicemail inbox exists? |
| 04:33.44 | [TK]D-Fender | lolipops: "core show function MAILBOX_EXISTS" |
| 04:33.58 | lolipops | thanks. |
| 04:34.55 | jameswf-home | nader |
| 04:39.39 | [TK]D-Fender | ~book |
| 04:39.40 | jbot | rumour has it, book is 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 |
| 04:39.41 | [TK]D-Fender | ~101 |
| 04:39.42 | jbot | [101] Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 04:41.48 | jameswf-home | ~202 |
| 04:41.54 | kerx | ~102 |
| 04:41.55 | jbot | i guess 102 is #asterisk |
| 04:42.00 | kerx | heh |
| 04:43.55 | jameswf-home | ~103 |
| 04:44.05 | jameswf-home | ~404 |
| 04:44.05 | jbot | Could not find 404. Maybe you misspelled it? |
| 04:44.32 | jameswf-home | ~404 is <reply>ERROR 404 answer not found |
| 04:44.33 | jbot | ...but 404 is already something else... |
| 04:45.22 | kerx | hehe |
| 04:48.01 | [TK]D-Fender | ~404 |
| 04:48.02 | jbot | ERROR: 404 answer not found |
| 04:48.35 | [TK]D-Fender | pets jbot |
| 04:51.00 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-196.phlapa.east.verizon.net) |
| 04:51.25 | carrar | [TK]D-Fender, know of a easy eas way to delete a sip header? |
| 04:52.24 | [TK]D-Fender | carrar: ^H^H^H^H^H^H^H^H^H^H^H |
| 04:52.29 | carrar | haha |
| 04:52.36 | carrar | while in a dialplan |
| 04:52.40 | [TK]D-Fender | carrar: translation : No |
| 04:53.09 | carrar | need a SIPDelHeader |
| 04:53.13 | [TK]D-Fender | carrar: * isn't a proxy and its almot miraculous that it even lets you look at them let alone add to them |
| 05:07.02 | jameswf-home | ~gpllaw |
| 05:07.03 | jbot | Gpllaw wears tights and a cape and enforces the GPL and trolls violators to their core |
| 05:13.21 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 05:16.14 | denon | so .. BSA, without a budget |
| 05:19.57 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-75-251-168.myvzw.com) |
| 05:20.12 | Gopher_77 | can someone help me set up alsa drivers? |
| 05:20.46 | Gopher_77 | I have a Tiger3XX Modem/ISDN interface |
| 05:21.16 | Gopher_77 | amixer shows Phone and Line |
| 05:21.20 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:21.41 | drmessano | Damn, I am so dumb |
| 05:21.46 | drmessano | I just killed a PAP2 |
| 05:22.35 | [TK]D-Fender | *yawn* back later... |
| 05:25.12 | jameswf-home | drmessano: you connected it to vonnage? |
| 05:30.40 | drmessano | Worse |
| 05:31.05 | drmessano | I upgraded the FW to 3.1.22, then factory reset it, just to see if it would revert back to Vonage settings |
| 05:31.12 | drmessano | It did, and it pulled down the XML |
| 05:31.30 | drmessano | I am able to factory reset it |
| 05:31.39 | drmessano | But I think the upgrade rule is gonna screw me |
| 05:31.51 | drmessano | Well, wait |
| 05:31.52 | drmessano | No |
| 05:32.28 | drmessano | If I can factory reset it, I still own it |
| 05:33.01 | drmessano | Hmm |
| 05:33.07 | drmessano | its not seeing my TFTPd tho |
| 05:34.50 | jameswf-home | wow |
| 05:34.51 | jameswf-home | <PROTECTED> |
| 05:35.31 | Gopher_77 | lol |
| 05:35.41 | *** join/#asterisk CrazyTux (n=brandon@user-vcauq7r.dsl.mindspring.com) |
| 05:35.47 | WimpMan | Is that M$s new secutity strategy? |
| 05:37.05 | jameswf-home | it was in a email from one of the "security and penetration" folks on a linux list |
| 05:43.38 | *** join/#asterisk gones (n=gones@61.141.80.81) |
| 05:46.11 | *** join/#asterisk ant (n=ant@67-207-113-226.static.wiline.com) |
| 05:54.35 | orkid | i have now setup the 3102 with asterisk. i'm trying to get a 'testfeature' application map to work. when i call extension 123, i set ____DYNAMIC_FEATURES=testfeature, dial sip/2000 (the phone on the fxs port of the 3102). so i call 123 from the softphone, and try #9 (should activate testfeature), but it the dtmf characters just go through to the other phone... i try both from the softphone and the phone on the fxs of the 3102, and the tones go through |
| 05:54.42 | orkid | what could it be ? |
| 05:56.42 | orkid | and btw, les.net still hasn't gotten back to me. iirc they said theyd get back in 24 hours, on their contact page |
| 05:58.14 | orkid | hmm, magically, it works now |
| 05:58.59 | *** join/#asterisk axisys (n=axisys@117.18.230.23) |
| 06:10.33 | *** join/#asterisk nuonguy (n=john@c-71-198-1-139.hsd1.ca.comcast.net) |
| 06:14.13 | hesco | OK, I just completed a make clean; make menuselect; make; make install and I'm still getting these cdr_pgsql errors: Reason: ERROR: column "calldate" specified more than once |
| 06:14.40 | hesco | pb |
| 06:14.45 | hesco | ~pb |
| 06:14.45 | 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 |
| 06:15.39 | hesco | full set of errors at: http://pastebin.ca/1235431 |
| 06:18.05 | hesco | I tested those connection parameters by hand and I know they work. |
| 06:32.31 | hesco | more notes on the subject: http://pastebin.ca/1235438 |
| 06:34.35 | *** part/#asterisk CrazyTux (n=brandon@user-vcauq7r.dsl.mindspring.com) |
| 07:00.36 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 07:05.01 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:06.11 | orkid | hmm, how the heck do i get asterisk to craft a sip packet to send a hookflash... |
| 07:06.38 | orkid | i only see sipaddheader(), and that adds a header to the first invite message sent |
| 07:06.43 | orkid | according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPAddHeader |
| 07:08.46 | orkid | or.. maybe sendDTMF and SIPDtmfMode |
| 07:25.50 | *** join/#asterisk philipp64 (n=chatzill@63.224.43.239) |
| 07:27.04 | orkid | flashing sucks :P |
| 07:30.13 | WimpMan | correct |
| 07:32.14 | orkid | chan_sip.c:3978 sip_indicate: Don't know how to indicate condition 9 |
| 07:32.15 | orkid | :( |
| 07:39.47 | orkid | is there any hope ? |
| 07:40.02 | orkid | seems like asterisk 'senses' the flash... 'which from reading online is condition 9 |
| 07:40.23 | yang | can register => string be used crypted together with md5secret= ? |
| 07:40.24 | orkid | or maybe not... |
| 07:41.53 | orkid | what am i even talking about... i set an applicationmap to SendDTMF on #9, |
| 07:42.01 | orkid | SendDTMF(F) |
| 07:42.16 | orkid | since (Flash) gave me 'invalid characters' for l s h |
| 07:42.50 | orkid | so i figured F is the one I wanted.. so that worked fine and dandy, but now the warning of not knowing how to indicate condition 9 |
| 07:43.57 | orkid | http://bugs.digium.com/view.php?id=12999 |
| 07:44.09 | orkid | where is drmessano |
| 07:50.10 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-7ae78b4ea84c574e) |
| 07:55.16 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 07:58.36 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:02.01 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 08:05.03 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:06.14 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 08:07.29 | phpboy | Hey all, I'm having a problem with Asterisk 1.4.21.2, at random on any type of call (IAX2, SIP or ZAP) the person you're calling will no longer transmit voice, can anyone think of a general reason why this would happen? |
| 08:09.30 | orkid | wild guess (not sarcastic): something to do with nat connection timeout? |
| 08:12.16 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 08:13.01 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 08:13.24 | miloux | ~book |
| 08:13.25 | jbot | methinks book is 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 |
| 08:13.28 | phpboy | orkid: it's on a local network |
| 08:13.31 | phpboy | ? |
| 08:13.31 | kippi | hey |
| 08:13.45 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 08:14.17 | kippi | I need to create flow diagrams of my asterisk config, how contexts link etc, has anyone done this before ? or has a good idea of how to do this? |
| 08:14.49 | encode | kippi: tried mspaint? |
| 08:15.51 | kippi | need to be able to take the asterisk configuration I have and then dump it into a application, something like doxgyen |
| 08:19.15 | orkid | hohoho wow |
| 08:21.21 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:22.25 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:27.05 | aiksa[LV] | morning! |
| 08:27.39 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 08:28.12 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 08:29.07 | aiksa[LV] | perhaps anyone else has done something similar and can share their experince: i want to automatically asign status for a number if a number called was out of coverage. |
| 08:29.18 | aiksa[LV] | the connection to PSTN - over PRI |
| 08:29.35 | aiksa[LV] | now - the operators wont signal this by q931 messages |
| 08:29.56 | aiksa[LV] | and generaly give an early audio describing the status |
| 08:30.39 | aiksa[LV] | is there a possibility to match this early audio to a saved recording (with certain fault tolerance) and make an appropriate call routing decision? |
| 08:31.17 | aiksa[LV] | i meant difference threshold with "fault tolerance"? |
| 08:31.38 | aiksa[LV] | and all of this - w/o getting too dirty with the channel driver? |
| 08:31.39 | aiksa[LV] | :) |
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| 08:50.54 | kaldemar | kippi: make a script that makes a dot file based on your dialplan. then you can use graphviz to create the diagram. |
| 08:52.13 | kippi | where can I find information on how to make a dot file etc |
| 08:52.24 | kaldemar | google it. |
| 08:53.21 | kippi | ok |
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| 08:54.54 | *** join/#asterisk oej (n=olle@193.136.47.146) |
| 09:03.09 | kippi | thanks |
| 09:17.39 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:21.49 | *** join/#asterisk XnOSX (n=XnOSX@212.145.55.118) |
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| 09:44.38 | *** join/#asterisk asim- (n=sim@gateway1.beatthatquote.com) |
| 09:44.57 | asim- | hi, need some help |
| 09:45.11 | asim- | multiple people can login to sip accounts on my asterisk 1.6 |
| 09:45.16 | asim- | how would i disable that? |
| 09:51.41 | *** join/#asterisk jameswf-home (n=james@ip68-109-175-189.ph.ph.cox.net) |
| 09:51.49 | IsUp`afk | what exactly you mean asim-? |
| 09:52.08 | *** join/#asterisk XnOSX (n=XnOSX@212.145.55.118) |
| 09:52.54 | asim- | i mean |
| 09:53.01 | asim- | more than one person can log into a single sip account |
| 09:53.03 | asim- | at the same time |
| 09:53.15 | asim- | imagine two people could be logged into the same msn account :| |
| 09:53.37 | IsUp | host=192.168.x.x -> only 192.168.x.x can connect to sip account |
| 09:54.19 | asim- | yea but what if ip's arent static |
| 09:55.12 | kaldemar | with permit you can define a block. |
| 09:55.23 | IsUp | well, i am not using SIP so much. maybe someone can help to you around here. |
| 09:55.31 | kaldemar | set up separate accounts. |
| 09:55.43 | kaldemar | change the secret |
| 09:56.00 | asim- | there are seperate accounts. multiple sip accounts. but someone tested and noticed two people could login to the same account |
| 09:56.11 | kaldemar | why do you have more than one client using the same information? |
| 09:56.17 | asim- | we are testing |
| 09:56.32 | asim- | someone logged in on a laptop and desktop |
| 09:56.37 | kaldemar | use secrets and don't give them out to everyone. |
| 09:56.37 | asim- | and saw that they were logged in on both |
| 09:56.41 | asim- | .... |
| 09:56.43 | kaldemar | you don't have a problem. |
| 09:56.46 | asim- | eh i do |
| 09:56.53 | asim- | i dont want to be logged in in two places |
| 09:57.05 | kaldemar | then don't do it. |
| 09:57.09 | asim- | .... |
| 09:57.11 | asim- | helpful thanks |
| 09:57.22 | kaldemar | and there is no such thing as a login, you're talking about registering. |
| 10:03.51 | kaldemar | the whole idea of a dynamic host and registering is that you can move. hence registering from different addresses is possible. the calls will however go to only one place. |
| 10:08.16 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
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| 10:45.22 | jblack | Ohhhh. futures trading limited this morning. |
| 10:58.48 | ITguru | If wanrouter says connected, does that means my device works? |
| 11:00.10 | IsUp^afk | ITguru, it means your link is OK |
| 11:00.16 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 11:00.20 | IsUp^afk | i mean physical connection |
| 11:00.28 | ITguru | IsUp^afk, which is a good start... right! |
| 11:00.28 | IsUp^afk | check with: wanpipemon -i w1g1 -c Ta |
| 11:00.36 | ITguru | is in the middle of a baptism by fire |
| 11:00.41 | IsUp^afk | if theres any alarms or any strange thing in RX level |
| 11:01.32 | ITguru | No alarms today! |
| 11:01.40 | IsUp | great |
| 11:01.43 | ITguru | I just want to configure my damn lines |
| 11:01.58 | IsUp | so, you can :D |
| 11:02.23 | ITguru | IsUp, I'm trying here! But I'm not 100% sure what I'm doing! |
| 11:02.34 | ITguru | I've got FreePBX configured, and up and running |
| 11:02.36 | IsUp | okay, what you trying to do? |
| 11:02.51 | ITguru | I want to get my incoming calls working first |
| 11:03.41 | IsUp | using PRI or what? |
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| 11:18.48 | ITguru | back! |
| 11:19.03 | ITguru | sorry, I accidently rebooted the remote server, and not my laptop! |
| 11:20.14 | IsUp | wellcome back |
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| 12:11.20 | *** join/#asterisk marcus78 (n=marcus78@58.59.199.51) |
| 12:11.57 | marcus78 | Question! is it possible to call an agent that is in queue but is not in call |
| 12:16.17 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:16.27 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:17.40 | chaozer | marcus78, yes |
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| 12:35.11 | ITguru | When I try to make outgoing calls, I get the message all circuits are busy now |
| 12:38.55 | [TK]D-Fender | yup... don't get your answer in < 3 minutes, then just run away.... |
| 12:39.13 | *** join/#asterisk Dream_th (n=dream_th@c-516370d5.01-19-6b736810.cust.bredbandsbolaget.se) |
| 12:39.14 | *** join/#asterisk dan01 (n=me@cpe-24-92-241-92.twcny.res.rr.com) |
| 12:39.37 | Dream_th | hi |
| 12:40.33 | Dream_th | what kind of information do i need from my provider so i could setup sip trunk |
| 12:41.34 | dan01 | Hi, is anyone familiar with adding a SIP phone to Fonality (PBXtra), without paying them? Even a pointer into the right direction (as to what files to edit, already tried sip.conf, but no luck), or an active fonality forum would be great. |
| 12:42.18 | [TK]D-Fender | Dream_th: user, pass, host, allowed codecs. |
| 12:42.42 | russellb | dan01: not using it so that you don't have to worry about it? :) |
| 12:42.47 | russellb | that's my recommendation! |
| 12:43.23 | [TK]D-Fender | dan01: You saying their GUI won't let you add another phone due to licensing limits? |
| 12:43.24 | dan01 | Well I didn't pick the fonality solution, I would have gone plain asterisk myself, but alas, I am just a guy trying to make things work ;) Company bought polycom soundpoint 320 phones. |
| 12:43.34 | dan01 | If you want to add phones, you have to pay them. |
| 12:43.45 | dan01 | They allow you to do it manually, but won't tell you how, as it isn't officially supported. |
| 12:43.57 | [TK]D-Fender | dan01: Then you bought a dead end, so you'd best pick out a few more caskets.... |
| 12:44.33 | dan01 | I agree, I was asked to try to make this work, before they send the phones back (they ordered without consulting me) |
| 12:45.20 | dan01 | The phone is talking to the PBX, I just keep getting the IVR system, no matter what extension I dial or if I try to dial out using 9. profile in sip.conf matches the existing phones that do work. |
| 12:45.31 | [TK]D-Fender | dan01: Somebody should go for training to understand what they bought... |
| 12:48.49 | Dream_th | [TK]D-Fender: i am trying very different trunk configs but im unable to make it work, im in contact with the provider and they are willing to help me except i need to be specific what kind of information (settings) do i need in order to correctly configure my trunk |
| 12:49.20 | *** join/#asterisk genin (i=benny@ANice-252-1-41-132.w82-122.abo.wanadoo.fr) |
| 12:49.26 | genin | alo |
| 12:49.27 | [TK]D-Fender | Dream_th: and I just told you them. |
| 12:49.46 | genin | anyone with asterisk experience and C experience want a job in france? |
| 12:49.59 | Dream_th | [TK]D-Fender: that information i already have |
| 12:50.42 | [TK]D-Fender | Dream_th: well until you show us your configs and SIP debug of the failures we can't begin to guess what is wrong with it. Pastebin is your friend... |
| 12:50.44 | [TK]D-Fender | ~pb |
| 12:50.44 | 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 |
| 12:52.49 | dan01 | [TK]D-Fender, does the issue I mentioned above sound like a typical misconfiguration issue, or is it too vague? |
| 12:53.27 | [TK]D-Fender | dan01: Who know what kind of extra checks they do to validify this new SIP peer... can be dialplan issues, can be base phone config issues, etc |
| 12:53.47 | [TK]D-Fender | danYou're FUBAR'd and nobody is going to want to go through the effort of trying to beat it into compliance. |
| 12:54.10 | [TK]D-Fender | dan01: Send them back or pay Fonality. Thats the business decision that was made. Time to live with it. |
| 12:54.44 | dan01 | I know, totally agree, but I personally wanted to find out anyways. I was surprised to hear that fonality won't let you add phones yourself, so I had to check it out. |
| 12:54.50 | genin | when i say that we have all of our clients using g711 |
| 12:54.56 | genin | does that make anyone want to laugh |
| 12:55.41 | Dream_th | [TK]D-Fender: http://pastebin.com/me6ede74 |
| 12:56.01 | [TK]D-Fender | genin: No... some of us LIKE quality... |
| 12:56.24 | genin | when i say 8 calls simulataneously from an adsl line? |
| 12:57.01 | coppice | maybe he's referring to the way G.711 sounds like penis in Swahili |
| 12:57.02 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:57.12 | *** join/#asterisk gcbirzan (n=gcbirzan@pida/gcbirzan) |
| 12:57.15 | genin | heh |
| 12:57.20 | creativx | have you ever heard a penis. |
| 12:57.33 | genin | when it is plunging into a pussy yes |
| 12:57.40 | genin | it is a quite comforting sound |
| 12:57.45 | jaytee | wow, looks like I got here just in time :-( |
| 12:57.50 | coppice | yeah. it goes "swiiiiiiiiiiiiish" |
| 12:57.50 | creativx | hehe |
| 12:57.51 | creativx | penis time |
| 12:57.55 | genin | HEH |
| 12:57.58 | [TK]D-Fender | ... |
| 12:58.12 | genin | we are working out using 729 |
| 12:58.15 | creativx | morning [TK]D-Fender :) |
| 12:58.21 | feeds | says OMG |
| 12:58.22 | genin | digium trancoding cards |
| 12:58.25 | [TK]D-Fender | genin: I could pass 8 calls on G.711 on my DSL connection. |
| 12:58.46 | genin | yes it works pretty well but sometimes i have clients whos calls cut |
| 12:58.51 | genin | or quality drops |
| 12:59.06 | [TK]D-Fender | genin: then someone had better examine their approach |
| 12:59.12 | genin | i think 729 would be better |
| 12:59.31 | genin | maybe it has to do with the fact a client might be running emule on the same line during |
| 12:59.49 | genin | 8 calls is 512k up so it doesnt leave much room for takign the bandwidth |
| 13:00.31 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 13:00.35 | coppice | its more like 650k |
| 13:00.45 | genin | isnt it 6' |
| 13:00.50 | genin | 64k per call |
| 13:00.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:00.59 | [TK]D-Fender | genin: Yes, that kind of behavior is what we telephony-informed types would call "stupid" |
| 13:01.00 | coppice | + RTP overheads |
| 13:01.05 | genin | hah |
| 13:01.07 | genin | ah okay |
| 13:01.15 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 13:01.21 | genin | we are looking into SER with an rtp proxy |
| 13:01.34 | [TK]D-Fender | genin: Won't change anything... |
| 13:01.41 | genin | not on the clients side |
| 13:01.55 | genin | someone told me gsm is the industry standard |
| 13:02.03 | genin | and better quality than g729 |
| 13:02.18 | genin | i havent used it tho, mainly i dont think our client side gateways support it |
| 13:02.39 | [TK]D-Fender | genin: GSM is considered slightly lower for voice than G.729 by a fair number of people, but overall pretty close. GSM is also larger however |
| 13:02.52 | genin | ah cool |
| 13:02.56 | genin | thnx |
| 13:03.02 | *** part/#asterisk dan01 (n=me@cpe-24-92-241-92.twcny.res.rr.com) |
| 13:03.18 | coppice | GSM 06.10 is such a great codec, the GSM networks no longer use it |
| 13:03.57 | genin | there are different kinds of 729 as well right |
| 13:04.02 | genin | a and b |
| 13:04.24 | IsUp | whats difference between codecs? i mean, should i use G729 or GSM? how can i select the best? |
| 13:04.35 | genin | perfect timing |
| 13:04.39 | genin | with the question |
| 13:04.40 | genin | heh |
| 13:05.00 | coppice | more G.729 is the A kind. Its a stripped down lower quality version of G.729. G.729B is an add on to give it VAD |
| 13:05.12 | genin | VAD? |
| 13:05.12 | coppice | s/more/most |
| 13:05.38 | [TK]D-Fender | ~vad |
| 13:05.39 | jbot | vad is, like, Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 13:05.56 | genin | ah silence suppression |
| 13:06.01 | coppice | VAD is not silence suppression |
| 13:07.08 | genin | the admin here was saying if our clients sends our asterisk a call in 729a but the provider we send it to accepts 729b that we will have a huge problem |
| 13:07.23 | genin | an asterisk with a digium transcoding card |
| 13:07.58 | Dream_th | [TK]D-Fender: anything :S |
| 13:08.17 | coppice | genin: these things get negotiated, and should not be a problem |
| 13:08.30 | genin | hrm |
| 13:08.33 | genin | interesting |
| 13:08.36 | [TK]D-Fender | Dream_th: Why am I seeing an ATA's user-agent in there? Next, no provider should have "nat=yes" in the peer |
| 13:09.02 | [TK]D-Fender | Dream_th: And don't mask IP's in pastebins |
| 13:09.16 | Dream_th | it has ata's useragent because it works with that ata |
| 13:09.31 | Dream_th | i'll remove nat=yes, its just a configuration some other one gave me |
| 13:09.40 | [TK]D-Fender | Dream_th: and you're trying to get around them and use *? |
| 13:10.12 | *** join/#asterisk [simux] (n=chatzill@189.85.128.10) |
| 13:10.13 | Dream_th | im in contact with them and they will provide me with any information i need |
| 13:10.19 | Katty | stretches |
| 13:10.21 | Dream_th | to setup this trunk |
| 13:10.40 | jaytee | does deep knee bends |
| 13:10.41 | [TK]D-Fender | Dream_th: Next you show a register's SIP debug failing but only show a peer. I am beginning to wonder how much else is wrong. |
| 13:11.18 | Katty | consumes redbull |
| 13:11.22 | jaytee | "crack.....snap.....crack....crunch.....skrrrsssshhhh" |
| 13:11.25 | IsUp | hey Katty |
| 13:11.26 | jaytee | ow |
| 13:11.38 | Katty | IsUp: goodyawnmorning |
| 13:11.43 | Katty | [TK]D-Fender: mew. |
| 13:11.58 | [TK]D-Fender | Katty: Mew. |
| 13:12.10 | Dream_th | [TK]D-Fender what excactly do you need me to show you, because i really dont know |
| 13:12.39 | [TK]D-Fender | Dream_th: you don't have a call attempt in there. You show SIP debug of a register. The peer entry has nothing to do with that. |
| 13:12.39 | Katty | i could use a Clue Muffin for breakfast. |
| 13:12.47 | Katty | jbot: ClueMuffin |
| 13:12.47 | jbot | [~cluemuffin] A perfect blend of bran & ClueBat (tm). Not to be confused with the Chinese Fighting Muffin. |
| 13:12.50 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:13.14 | [TK]D-Fender | Wow forgot about that one :) |
| 13:13.52 | russellb | very nice, indeed |
| 13:13.57 | russellb | pokes Katty and runs |
| 13:14.37 | coppice | here in china we generally go for the american blueberry or plain english muffin |
| 13:15.04 | Dream_th | [TK]D-Fender: http://pastebin.com/m1e9d0feb |
| 13:15.19 | gcbirzan | Is there a way to get out of a queue, while you're waiting? (The pressing * to end the call only works after somebody answers) |
| 13:16.51 | [TK]D-Fender | gcbirzan: "context=" for your queue definition. |
| 13:17.00 | [TK]D-Fender | gcbirzan: its documented in the smeple config |
| 13:17.11 | [TK]D-Fender | sample* |
| 13:17.41 | [TK]D-Fender | Dream_th: it isn't even trying to use that peer which of course we don't even see. |
| 13:17.50 | gcbirzan | [TK]D-Fender: Oh, indeed. Missed that. Thanks! |
| 13:17.54 | Katty | russellb: UNSOLICITED POKING |
| 13:18.07 | [TK]D-Fender | Dream_th: and you're shoing retransmits. Makes me wonder if you've got things forwarded properly |
| 13:18.15 | Katty | hugs russellb |
| 13:18.19 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 13:18.50 | Dream_th | i just pasted what i got on cli |
| 13:18.55 | Dream_th | during the call test |
| 13:19.18 | Dream_th | btw the trunk name is called IPKO |
| 13:19.36 | [TK]D-Fender | Dream_th: Well we've never seen a SIP peer to match the one named in there. |
| 13:19.48 | [TK]D-Fender | Dream_th: And what have you got forwarded to your * box? |
| 13:19.51 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:19.51 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:20.19 | Dream_th | [TK]D-Fender: what do you mean by what i have forwared to my box |
| 13:20.22 | [TK]D-Fender | Dream_th: Because we also never even see your provider answering you back. |
| 13:20.30 | [TK]D-Fender | Dream_th: PORTS.... |
| 13:20.34 | Dream_th | its on public ip |
| 13:20.44 | [TK]D-Fender | Dream_th: Check your firewall on the box then. |
| 13:20.52 | [TK]D-Fender | Dream_th: and any routers in the way |
| 13:21.00 | Katty | Zeeek! |
| 13:21.10 | Dream_th | also just for your information the provider uses Netcentrex softswitch |
| 13:21.17 | Katty | hugs Zeeek |
| 13:21.22 | Zeeek | {{{{{{{Katty}}}}}}} |
| 13:21.27 | Katty | Zeeek: how beith? |
| 13:21.33 | Katty | Zeeek: did you know it's friday? |
| 13:21.41 | Zeeek | ich bin ein Berliner |
| 13:21.43 | [TK]D-Fender | Dream_th: Doesn't seem to matter what they use... you aren't getting an answer to your register attempts |
| 13:21.43 | *** part/#asterisk [simux] (n=chatzill@189.85.128.10) |
| 13:21.48 | Katty | yo are a donut?! |
| 13:21.58 | lmadsen | donuts do not exist |
| 13:22.00 | Zeeek | yes, with creamy filling |
| 13:22.00 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 13:22.03 | Katty | hot. |
| 13:22.05 | Katty | i'll take two! |
| 13:22.07 | Katty | hugs anonymouz666 |
| 13:22.13 | Katty | with milk please. |
| 13:22.15 | Zeeek | and I have two, so that's perfect |
| 13:22.26 | Zeeek | er, only rice milk today |
| 13:22.33 | Katty | rice milk is sweet (= |
| 13:22.40 | Zeeek | yes. |
| 13:24.27 | Dream_th | [TK]D-Fender: the iptables on the box is empty and as i said its on public ip |
| 13:25.35 | [TK]D-Fender | Dream_th: maybe the remote host is wrong... you aren't getting an answer. If your user, etc was wrong at least you'd get a complaint back. You are getting absolutely nothing. That means either networking is in the way, or your host is wrong, or they are down. |
| 13:26.16 | Dream_th | [TK]D-Fender when i put this information on grandstream ata it works with host, user, pass |
| 13:26.18 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 13:26.57 | [TK]D-Fender | Dream_th: This isn't a debate unfortunately. No answer to a SIP communication attempt can only men one of those things. |
| 13:27.23 | [TK]D-Fender | mean* |
| 13:27.49 | Katty | lmadsen: do we hug? |
| 13:28.21 | Qwell | Katty: you don't want to catch anything |
| 13:28.29 | Katty | gosh. |
| 13:28.50 | lmadsen | Katty: ummm.... yes! |
| 13:28.54 | lmadsen | hugs@katty.com |
| 13:29.18 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 13:29.19 | Katty | geek. |
| 13:29.25 | lmadsen | :D |
| 13:29.26 | lmadsen | totally |
| 13:29.39 | lmadsen | at least you didn't call me a nerd... thems be fightin' werdz! |
| 13:30.06 | [TK]D-Fender | 'course where we comes from, all werds is fightin werds! |
| 13:30.19 | Katty | lmadsen: http://www.motivatedphotos.com/?id=652 |
| 13:30.35 | gcbirzan | Qwell: From you, chances are one could catch literally anything :-P |
| 13:30.38 | [T]ank | I am searching for a solution to faxing over asterisk. I see that some people are doing it using an ata and g711u. That configuration is not working for me... any one here have a way to do it without a separate fax server like hylafax? |
| 13:30.39 | lmadsen | [TK]D-Fender: we don't take too kindly to yer type around here! |
| 13:30.57 | [TK]D-Fender | changes quickly to Helvetica |
| 13:30.58 | lmadsen | Katty: lol |
| 13:30.59 | Katty | lmadsen: i do. |
| 13:31.00 | Qwell | gcbirzan: careful, or I'll bring up your goat fetish... |
| 13:31.07 | Katty | lmadsen: fender is super duper. |
| 13:31.21 | lmadsen | really? |
| 13:31.24 | lmadsen | weird |
| 13:31.31 | gcbirzan | Qwell: I was in the UK this week, I'm thinking of going for sheeps. |
| 13:31.45 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 13:31.45 | Katty | maahhhhh |
| 13:31.55 | Katty | i would mind a few pet sheeps. |
| 13:32.04 | Katty | s/would/wouldn't/ |
| 13:32.12 | Katty | jbot: thank you, dear. |
| 13:32.12 | jbot | pas de quoi, Katty |
| 13:32.23 | genin | pas de quoi |
| 13:32.25 | genin | heh |
| 13:32.28 | genin | its a french bot |
| 13:32.40 | coppice | what do you call a Welshman with a thousand lovers? |
| 13:32.42 | coppice | A shepherd |
| 13:32.46 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 13:32.46 | genin | heh |
| 13:32.56 | Zeeek | jbot: tu devrais aller te faire avoir chez les Grecs |
| 13:33.12 | Katty | Zeeek: jbot doesn't know much french. try [TK]D-Fender |
| 13:33.21 | [TK]D-Fender | jbot: t'es une maudite crise de vidange! |
| 13:33.26 | Zeeek | You try [TK]D-Fender |
| 13:33.34 | lmadsen | we need a translator bot |
| 13:33.35 | genin | jbot: k est ki space mehc |
| 13:33.36 | [TK]D-Fender | See? Stunned SILENT |
| 13:33.46 | genin | apparement |
| 13:34.06 | Katty | Zeeek: okay |
| 13:34.09 | genin | qui voudrais un boulut sur la cote d azur? |
| 13:34.15 | genin | boulot |
| 13:34.22 | genin | avec un appartement |
| 13:34.22 | Zeeek | pas moi, trop de touristes |
| 13:34.24 | Katty | [TK]D-Fender: parlez lentement--je ne comprends pas |
| 13:34.25 | Zeeek | http://www.voipfraud.net/en/node/457 |
| 13:34.39 | genin | il faut connait C aussi |
| 13:34.40 | Zeeek | this'll make your day ^^^^ |
| 13:34.43 | genin | et asterisk bien sur |
| 13:35.25 | Katty | je m'appelle au revoir! |
| 13:35.26 | Zeeek | [TK]D-Fender: did you see the little app that does a slide show on your Polycom? |
| 13:35.31 | genin | ou bien si klk connait un bon programeur qui veux demanager ici |
| 13:35.39 | UnixDawg | point |
| 13:35.52 | Zeeek | genin poste sur http://asterisk-france.net |
| 13:35.56 | [TK]D-Fender | genplus comme faites du creme sure, mem avec du frommage gratinee la-tu ;) |
| 13:35.59 | genin | cool merci |
| 13:36.47 | Katty | [TK]D-Fender: what would 'gosh' be in french? |
| 13:36.53 | Katty | [TK]D-Fender: or something similiar |
| 13:37.17 | genin | mon dieu |
| 13:37.22 | genin | that is my god |
| 13:37.34 | genin | or merde is good |
| 13:37.36 | stintel | :P |
| 13:37.37 | genin | like |
| 13:37.38 | genin | merde |
| 13:37.39 | Qwell | mon dieu is your god? |
| 13:37.44 | genin | bon dieu |
| 13:37.48 | Katty | is that something like fondu? |
| 13:37.48 | genin | good god |
| 13:37.57 | genin | fondre meanss |
| 13:38.00 | genin | to melt literally |
| 13:38.09 | genin | fondu means melted |
| 13:38.17 | genin | j ai fondu mon ordinateru |
| 13:38.22 | genin | i melted my computer |
| 13:38.36 | Katty | neat. |
| 13:38.43 | genin | where u from katty |
| 13:38.48 | [TK]D-Fender | As in : Il faut faire fondre la frommage au fond ;) |
| 13:38.51 | stintel | mais qu'est-ce que tu dis :P |
| 13:39.14 | jaytee | ok, so there's cheese in there, I know that much |
| 13:39.32 | [TK]D-Fender | jaytee: So basically... business as usual ;) |
| 13:39.38 | Katty | genin: Ganymede |
| 13:39.41 | genin | it has to melt the cheese to the back |
| 13:39.42 | genin | heh |
| 13:39.47 | genin | ganymede |
| 13:39.50 | genin | is that like in texas |
| 13:40.03 | jaytee | there's a place in town here called Schaeffer's that has awesome fondue and a wine list a mile long. Some of the wines go for hundreds a bottle. |
| 13:40.23 | lmadsen | likes wine |
| 13:40.57 | Katty | genin: that's about 1070400km from Jupiter |
| 13:40.59 | jaytee | Ganymede is one of the 4 Galilean moons of Jupiter. The others are Callisto, Io and Europa |
| 13:41.00 | genin | missouri |
| 13:41.20 | genin | the show me state |
| 13:41.23 | genin | nice |
| 13:41.44 | lmadsen | shows you |
| 13:41.48 | gcbirzan | Katty: So, that's in Canada, then? |
| 13:41.48 | Katty | lmadsen: the really really sweet wine is nice. |
| 13:41.57 | genin | so you from one of jupiters moons |
| 13:41.57 | lmadsen | Katty: oh, I don't like sweet drinks |
| 13:41.58 | Katty | gcbirzan: sigh. |
| 13:41.58 | cesar_CR | hello guys I have a AEX2400 card with FXO ports, can I set all those lines in a trunkgroup for asterisk to pick an available line in case of congestion ? |
| 13:42.00 | jaytee | ya gotta be careful who you show though, make sure they're not underage |
| 13:42.01 | genin | but using a bnc in missouri |
| 13:42.07 | Katty | gcbirzan: Ganymede is a moon of jupiter. |
| 13:42.07 | genin | cool |
| 13:42.16 | lmadsen | cesar_CR: yes you can! look for 'group=' |
| 13:42.27 | lmadsen | group=1 |
| 13:42.31 | jaytee | cesar_CR, yeah. Which version of Asterisk are you running? |
| 13:42.35 | lmadsen | channels => 1-24 |
| 13:42.47 | lmadsen | exten => _X.,1,Dial(Zap/g1/${EXTEN}) |
| 13:44.06 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 13:44.22 | cesar_CR | lmadsen, ok |
| 13:44.30 | cesar_CR | jaytee, 1.6 |
| 13:45.27 | jaytee | ok, then you would need to change the syntax in lmadsen's example to Dial(DAHDI/g1/${EXTEN}) instead. |
| 13:45.44 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 13:45.49 | Katty | [TK]D-Fender: hey, i found your daughter. |
| 13:45.51 | Katty | [TK]D-Fender: http://d.imagehost.org/view/0057/evil.jpg |
| 13:45.51 | jaytee | and there's example in the sample config files |
| 13:46.03 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 13:46.36 | cesar_CR | ok I got it guys thanks ! |
| 13:48.57 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 13:48.57 | jaytee | wow, DOW is down over 400 on the open |
| 13:49.02 | Katty | jaytee: yeah i was reading that on reddit this morning |
| 13:49.23 | jaytee | cranks up the Guns and Roses......"Welcome to the jungle, we got fun and games......" |
| 13:49.52 | jaytee | Katty, oh! so you're the one who uses reddit. I was wondering who that was :-) |
| 13:50.11 | Katty | everyone uses reddit, dear. |
| 13:50.16 | [TK]D-Fender | Damn Canadian dollr is crashing hard against USD |
| 13:50.34 | lmadsen | sweet! |
| 13:50.45 | lmadsen | doesn't quite understand why that is such a bad thing |
| 13:50.51 | lmadsen | it's good for the manufacturing industry |
| 13:51.03 | lmadsen | and great for those of us in Canada paid in USD |
| 13:51.40 | gcbirzan | Katty: I kind of knew that. I was being facetious |
| 13:51.59 | *** join/#asterisk brettnem (n=brettnem@user-387oe5t.cable.mindspring.com) |
| 13:53.03 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:53.15 | Katty | gcbirzan: (= |
| 13:53.24 | Katty | jaytee: did you see the psycho girlfriend car bashing yet? |
| 13:53.39 | jaytee | no? is that on reddit? |
| 13:53.41 | *** part/#asterisk rnovotny22 (n=rnovotny@71-220-107-37.mpls.qwest.net) |
| 13:53.44 | Katty | yeah |
| 13:54.02 | Katty | jaytee: http://www.flickr.com/photos/17680179@N06/sets/72157607800394250/ |
| 13:54.27 | *** join/#asterisk korihor (n=korihor@200-71-160-128.genericrev.telcel.net.ve) |
| 13:54.28 | russellb | Katty: you did that, didn't you |
| 13:54.40 | Katty | russellb: i don't have that much energy. |
| 13:54.46 | russellb | touche |
| 13:54.49 | Katty | ;) |
| 13:54.51 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 13:55.03 | jasonwoot | woot |
| 13:55.13 | Katty | russellb: i have been known to write Wash Me in the dust tho |
| 13:55.45 | *** join/#asterisk robevans (n=robevans@195-78-16-190.fibertel.com.ar) |
| 13:55.59 | russellb | ha, yes, a classic. |
| 13:56.08 | *** part/#asterisk robevans (n=robevans@195-78-16-190.fibertel.com.ar) |
| 13:56.25 | [TK]D-Fender | Katty: It was YOU!?!?!? |
| 13:56.33 | Katty | [TK]D-Fender: yes, dear. the truth comes out. |
| 13:56.42 | jasonwoot | anything scandalous happen at astricon? spill |
| 13:57.02 | Zeeek | does anyone listining design, develop or sell hosted pbx services? |
| 13:57.04 | Katty | what happens at astricon, stays at astricon. |
| 13:57.14 | jasonwoot | c'mon, it's fun friday |
| 13:57.19 | Katty | ;) |
| 13:57.37 | russellb | has some dirt on file from astricon |
| 13:57.38 | Katty | Zeeek: we sell hosted services. |
| 13:57.46 | Zeeek | you do? |
| 13:57.50 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 13:58.02 | Zeeek | I want a new feature |
| 13:58.13 | file | pushes russellb |
| 13:58.13 | lmadsen | has some dirt on himself at astricon of yore |
| 13:58.16 | [TK]D-Fender | Katty: Exceptfor that last little gift that goes away after a boatload of penicillin ;) |
| 13:58.25 | russellb | lmadsen: +42 on that ..... |
| 13:58.27 | Katty | Zeeek: don't we all. |
| 13:58.28 | Zeeek | So, the customer can configure services on the web, right? |
| 13:58.35 | russellb | lmadsen: :-X ! |
| 13:58.35 | lmadsen | russellb: mad points! |
| 13:58.43 | Katty | [TK]D-Fender: luckily, i don't come with gifts like that! |
| 13:58.43 | lmadsen | russellb: you shut your filthy mouth! :) |
| 13:58.46 | Zeeek | like say, where an extension is routed, the dely to go to vmail... right? Yes? |
| 13:58.53 | Katty | [TK]D-Fender: my gifts consist of cookies, muffins, and possibly OJ |
| 13:58.57 | russellb | lmadsen: I meant me, not you. But yeah, I guess I have dirt on you, too. |
| 13:59.01 | Hertzy3 | Does anyone know if there is a way to block a specific phone number from calling your asterisk server? |
| 13:59.14 | lmadsen | russellb: heh... you only have rumours! |
| 13:59.24 | [TK]D-Fender | Hertzy3: Call the telco. |
| 13:59.25 | Zeeek | Katty: can you, huh, huh? |
| 13:59.39 | lmadsen | Hertzy3: call the telco to block there, or do CID pattern matching |
| 13:59.48 | Katty | app_lookblacklist |
| 13:59.56 | Zeeek | all assuming CID is there |
| 13:59.57 | lmadsen | exten => _X./4165551212,1,Playback(go-away) |
| 14:00.04 | Katty | Zeeek: no. |
| 14:00.13 | Zeeek | no web interface? |
| 14:00.17 | Zeeek | that way too hosted! |
| 14:00.21 | [TK]D-Fender | ~zeeek |
| 14:00.21 | jbot | somebody said zeeek was someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 14:00.21 | Katty | do i look like a vending machine? |
| 14:00.45 | Katty | Zeeek: our idea of 'hosting' is to nat a few polycoms |
| 14:00.55 | jasonwoot | I wish I had a real telco... we drive a dump truck full of money to their door each month, but they told me they can't block calls |
| 14:00.56 | Zeeek | Katty: I don't know, I didn't fall in love with your looks, only your mind and good nature |
| 14:01.04 | Katty | Zeeek: harsh. |
| 14:01.10 | [TK]D-Fender | Katty: So if I push your buttons, you'll give me some "sugar"? ;) |
| 14:01.18 | Katty | [TK]D-Fender: more like a loose tooth. |
| 14:01.20 | lmadsen | ~drumkilla |
| 14:01.21 | jbot | [drumkilla] Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>, or someone who should be ph33r3d, or russellb |
| 14:01.25 | Zeeek | the quote? That was when I was a young buck |
| 14:01.29 | Katty | [TK]D-Fender: not keep on the unsolicitied button pushing. |
| 14:01.33 | Katty | [TK]D-Fender: s/keep/keen/ |
| 14:01.53 | russellb | lmadsen: old school |
| 14:01.59 | Zeeek | I WANT CONFIGURABLE hosted pbx |
| 14:01.59 | lmadsen | totally |
| 14:02.01 | lmadsen | ~blitzrage |
| 14:02.02 | jbot | [blitzrage] a super cool fellow |
| 14:02.02 | Katty | i don't even have buttons today :< |
| 14:02.06 | lmadsen | ~lmadsen |
| 14:02.06 | jbot | you are probably dreamy http://blogs.dfw.com/startle_grams/images/leif75_4.jpg |
| 14:02.10 | lmadsen | lol |
| 14:02.30 | Zeeek | ~fortunecookie |
| 14:02.37 | Katty | welcome to the 79s. |
| 14:02.39 | Katty | 70s |
| 14:02.45 | Zeeek | "I was once sad cause I had no shoes..." |
| 14:02.47 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.50 | Katty | hey _ShrikE |
| 14:02.54 | Zeeek | "then I met a man who was happy with no head" |
| 14:03.01 | _ShrikE | hugs Katty |
| 14:03.06 | Katty | hugs _ShrikE |
| 14:03.16 | _ShrikE | how'w that pup? |
| 14:03.20 | _ShrikE | err how' |
| 14:03.21 | Katty | growin like MAD |
| 14:04.03 | Katty | _ShrikE: http://www.flickr.com/photos/izaah/2964950397/in/set-72157608195837483/ |
| 14:04.04 | Zeeek | anyway, ever since they blogged about the app to show Flickr photos on your Polycom, the server is down |
| 14:04.34 | Katty | i should just make me a digital picture frame. |
| 14:04.39 | Zeeek | http://bit.ly/3blPWW |
| 14:05.42 | Katty | that might come in handy someday |
| 14:08.36 | Zeeek | Katty: whaazzzat? |
| 14:08.38 | Katty | Zeeek: so all the phones would access the same group of photos? |
| 14:08.43 | Katty | since it's in sip.cfg |
| 14:08.52 | Zeeek | no it's in the microbrowser of each phone, silly |
| 14:08.58 | Zeeek | NO NOTY SIP |
| 14:09.20 | Zeeek | OMG Katty, you didn't listen tolast weeks VUC: "How to configure your Polycom" |
| 14:09.39 | Zeeek | the one where it says "DOn't screw with sip.cfg, you'll have to change every time you update" ? |
| 14:10.05 | Zeeek | no this is a setting on the phone's own file |
| 14:10.08 | Katty | oh, right. |
| 14:10.17 | Katty | Main Browser Home i presume |
| 14:10.20 | Zeeek | And it shoots at a parrticular Flickr account |
| 14:10.28 | Zeeek | no IDLE thingie |
| 14:10.48 | Zeeek | I actually put a photo of the phot somewhere... |
| 14:11.30 | Zeeek | http://bit.ly/ipslides <--- here ya go |
| 14:11.56 | Zeeek | hmmm must have expired |
| 14:12.10 | Katty | wonders |
| 14:12.23 | Katty | googles |
| 14:12.45 | Zeeek | here it is http://bit.ly/1CPmqo |
| 14:13.12 | Zeeek | like I said, I think there server is now down though |
| 14:13.25 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-d2222bd4cc95454f) |
| 14:13.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.32 | Katty | Zeeek: so if i edit the idle bit (which we currently have our logo in the main sip.cfg) |
| 14:13.35 | kippi | on asterisk 1.6 is there away to busy out channels on the isdn, as i know you can't do this on 1.4 |
| 14:13.47 | Katty | Zeeek: what's the individual config file for the phone? |
| 14:14.09 | Katty | MAC.cfg? |
| 14:14.14 | Zeeek | Katty: it won't work because the server given is not working now. Also, you need to allow Flickr to show the photos thru their API |
| 14:14.20 | Zeeek | ya |
| 14:14.42 | Katty | do those settings override the main sip.cfg? |
| 14:15.40 | Zeeek | I believe it should yes |
| 14:15.45 | Katty | mkay |
| 14:16.15 | Zeeek | What I need to do is get a copy of the scrip though because as I say, the server given in the example is down now |
| 14:17.27 | Zeeek | maybe one of the thousdaznds in this channel has the script |
| 14:17.28 | Katty | is surprised by MAC.cfg |
| 14:17.40 | Katty | wow, it's mostly empty |
| 14:18.12 | Katty | i can i just copy and rename sip.cfg and make some changes? |
| 14:18.17 | Zeeek | the phone doesn't do any images becsides BMP or something so the conversion is on th efly. I guess if I wasn't so lazy I'd look at the php functions, I think it's not hard top do the conversion |
| 14:18.54 | Zeeek | [TK]D-Fender: is the resident evil^H^H^H expert on Polycoms here |
| 14:19.03 | jer | [TK]D-Fender, i know, i'm making a small fortune currency trading against it... just getting my money back into Canadian dollars is going to have to wait until it's way back up |
| 14:19.35 | *** join/#asterisk ElSonico (n=tav@KMMCDXXVI.gprs.saunalahti.fi) |
| 14:21.26 | *** join/#asterisk af_ (n=getsmart@88-149-230-89.dynamic.ngi.it) |
| 14:23.06 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 14:24.45 | Katty | reboots phone |
| 14:25.21 | seanbright | reboots Katty |
| 14:25.59 | jaytee | Katty, is this a Polycom? |
| 14:27.54 | Katty | jaytee: yes. |
| 14:28.06 | Katty | Zeeek: copying sip.cfg and renaming it made the phone asplode :< |
| 14:28.25 | Katty | Zeeek: so i made a copy of 0000etc.cfg, and pasted the <IP_500> section into it |
| 14:28.30 | Katty | dunno if that's right or not |
| 14:28.36 | Katty | never used individual cfgs before. |
| 14:29.16 | jaytee | Katty, I highly recommend you read this white paper. http://www.polycom.com/common/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf |
| 14:29.59 | jaytee | it clarifies the provisioning setup much better than the SIP Admin guide does. |
| 14:30.58 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 14:32.35 | Katty | oooh |
| 14:32.39 | Katty | that makes more sense |
| 14:33.46 | Zeeek | Actually I mis-stated the Mac thing, I use the system they describe here. phone_xyz.cfg |
| 14:34.58 | Katty | reboots again |
| 14:35.15 | Katty | i told MAC.cfg to reference MAC_phone1.cfg and sip.cfg |
| 14:35.23 | Zeeek | the call to phone_xyz.cfg is in the MAc.cfg file |
| 14:35.37 | Katty | and then in MAC_phone1.cfg and pasted the IP_500 block from original sip.cfg, with my changes |
| 14:36.36 | Katty | Zeeek: you might have to pastebin your phone_MAC.cfg for me |
| 14:36.53 | Zeeek | naw, you don't want that |
| 14:37.07 | Katty | bummer. it's not working. |
| 14:37.16 | Katty | in fact, after those changes, my phone doesn't know what extension it is. |
| 14:37.24 | Katty | fun |
| 14:37.45 | Zeeek | oh; Not so good. |
| 14:38.02 | Katty | i've no idea what i'm doing hehe |
| 14:38.10 | Zeeek | lucky you backed up all the old files |
| 14:38.39 | Katty | yep |
| 14:38.44 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 14:39.01 | Katty | however. |
| 14:39.09 | Katty | i could just copy sip.cfg and rename it sip-MAC.cfg |
| 14:39.21 | Katty | and then have MAC.cfg reference sip-MAC.cfg and take out the phone1 reference |
| 14:40.31 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:42.36 | Zeeek | serious migrane material in that last idea |
| 14:43.04 | Katty | :< |
| 14:44.09 | Zeeek | Katty: here's the line from MacAddr.cfg that counts: |
| 14:44.14 | Zeeek | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone_zeeek650.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/> |
| 14:44.55 | Zeeek | You're actually supposed to use the directories but even I am not that geeky |
| 14:44.57 | Katty | can you pastebin your phone_zeeek650 file? |
| 14:45.07 | Zeeek | not for a network with two Polycoms |
| 14:45.09 | Katty | that's the bit i'm having problems with |
| 14:46.01 | Katty | i don't know what all i need to include |
| 14:46.03 | *** join/#asterisk FarrisG (n=FarrisG@pool-71-164-195-61.dllstx.fios.verizon.net) |
| 14:46.07 | kippi | on asterisk 1.6 is there away to busy out channels on the isdn, as i know you can't do this on 1.4 |
| 14:46.24 | Zeeek | it has all the <reg reg.1. stuff with names and passwords |
| 14:46.37 | FarrisG | Can anyone point me in the right direction for figuring out how Chanspy works? After an update, it no longer does what it used to do. |
| 14:46.38 | Katty | just put in DELETED in that spot then |
| 14:46.43 | Zeeek | I will PM you Katty (aka get a room) |
| 14:46.46 | Katty | kk |
| 14:46.51 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 14:47.04 | FarrisG | It used to cycle through existing channels when I hit *, but now it just cycles through ONE channel, no matter how many are active |
| 14:52.18 | magronez | is away: almocar |
| 14:53.36 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 14:54.31 | *** join/#asterisk deStone (n=deStone@unaffiliated/destone) |
| 14:55.56 | *** join/#asterisk ElSonico (n=tav@KMMMDCCCLXXII.gprs.saunalahti.fi) |
| 14:56.21 | deStone | First off, I am a complete moron when it comes to all things phone-related -- so bare with me. We currently have an ivr setup using VMMI and some other mailbox software. We are having an entirely new system brought in (evolveip.net) but we need to retain the IVR functionality (enter your account number, your balance is ..., your last three transactions were...) |
| 14:56.56 | deStone | in asterisk -- is there a quick and dirty way to line up data and setup an ivr? |
| 14:57.34 | genin | wow you are such a moron |
| 14:57.36 | genin | heh |
| 14:57.36 | genin | jk |
| 14:57.41 | genin | so am i, its cool |
| 14:58.25 | deStone | :\ |
| 14:58.27 | deStone | yeah |
| 14:58.41 | deStone | i know its 100000x more time complex then I want it to be |
| 14:58.44 | genin | i have no idea what you are talking about even |
| 14:59.04 | genin | i just found out what asterisk was in feb |
| 14:59.06 | genin | this year |
| 14:59.19 | deStone | ah |
| 14:59.25 | genin | learning from a norweigian, teaching me in french when my native language is english |
| 15:00.31 | Zeeek | genin, what is the meaning of all the gibberish? |
| 15:00.40 | jaytee | I learned nuclear engineering from a Latvian who taugh me in Esperanto |
| 15:01.41 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 15:06.48 | jaytee | hmmm, why is that core temperature gauge in the red? |
| 15:07.00 | aiksa[LV] | jaytee: wow |
| 15:07.13 | aiksa[LV] | a Latvian teaching nuclear engineering |
| 15:07.34 | aiksa[LV] | makes me feel proud |
| 15:08.17 | jaytee | Wall Street analyst: "the market is making a minor adjustment." News Media: "THE SKY IS FALLING, THE SKY IS FALLING!!!" |
| 15:08.35 | aiksa[LV] | strange nevertheless, Lithuanians had the infrustructure for that |
| 15:08.57 | jaytee | anyone here use Hylafax? |
| 15:09.07 | *** join/#asterisk wonderworld (n=ww@ip-62-143-38-55.unitymediagroup.de) |
| 15:09.26 | wonderworld | hey, is there a way to play a beep sound on an existing channel from the cli? |
| 15:17.00 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:17.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:18.20 | Zeeek | As always on Friday, we'll be convening on #voip-users-conference in about 30 minutes |
| 15:18.47 | Zeeek | http://voipusersconference.org gives all the info to call in via SIP or PSTN |
| 15:18.57 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 15:19.29 | Zeeek | russellb will be expounding on a whole bunch of new great stuff about asterisk 1.6, life and everything |
| 15:19.30 | aiksa[LV] | jaytee: - yeah, I use hylafax |
| 15:21.34 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 15:22.42 | jaytee | aiksa[LV], do you like it? any headaches with it? |
| 15:22.48 | Qwell | Zeeek: yeah, you might want to run that by him first |
| 15:23.48 | Zeeek | naw, he's cool |
| 15:24.40 | Zeeek | Qwell will be speaking on various issues about asterisk, security and life, the universe and everything |
| 15:24.58 | *** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
| 15:25.17 | Zeeek | Paris Hilton will be asking for help setting up her home pbx |
| 15:25.37 | Qwell | Zeeek: good, tell her I'll help for $250,000/hour, 3h minimum |
| 15:25.58 | Zeeek | she has a different idea of "Value added" ™ |
| 15:25.59 | russellb | tell her I'll help for ... nevermind |
| 15:26.06 | Qwell | O.o |
| 15:26.10 | smach | hi folks, I was wondering should a sip proxy answer a sip request on the port used by the client or on the port mentioned on the contact header ??? |
| 15:29.26 | Zeeek | russellb: the answer my friend, is... |
| 15:29.50 | lmadsen | blowing in the wind! |
| 15:30.02 | Zeeek | I get email daily from Paris, with photos and everything. She's so cool. A real classy lady. |
| 15:30.10 | Zeeek | lose the wind part |
| 15:31.21 | *** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
| 15:33.04 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:33.49 | Katty | hmm |
| 15:33.52 | rwaite | hi all |
| 15:34.18 | rwaite | anyone here have any opinions on digium's hpec? |
| 15:34.55 | *** join/#asterisk remont (n=reMont@200.73.192.226) |
| 15:35.26 | *** join/#asterisk fred-tmft (n=fred-tea@ip70-171-36-194.ga.at.cox.net) |
| 15:35.27 | jaytee | hpec? hardware echo cancellation? |
| 15:37.27 | rwaite | well, there is a software one |
| 15:37.47 | rwaite | i have a tdm400p so the h/w canceller won't work on it |
| 15:38.47 | lmadsen | jaytee: hpec is software echo can |
| 15:38.57 | jaytee | oh, ok. HPEC, high performance embedded computing |
| 15:39.29 | rwaite | high performance echo canceller, sorry |
| 15:40.43 | jaytee | I have the hardware ec on my TE212P so I don't use the software solution |
| 15:41.47 | jaytee | funny how HPEC has two distinctly different meanings depending on whether you're talking VOIP and Digium or computing at MIT. :-) |
| 15:44.00 | *** join/#asterisk kannan (n=kannan@123.201.136.118) |
| 15:44.03 | kannan | hello all |
| 15:45.50 | rwaite | i think i'm going to grab a couple licenses and give it a go. anything is better than what we have now, i'd think |
| 15:46.28 | Zeeek | see you on the conferenc echannel #voip-users-conference or the call http://bit.ly/voip |
| 15:46.30 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:48.47 | *** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
| 15:50.05 | *** join/#asterisk beek_ (n=klinebl@pool-96-245-14-102.phlapa.fios.verizon.net) |
| 15:51.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:51.31 | [TK]D-Fender | rwaite: If your card is under warranty you are entitled to some free licenses for it. |
| 15:51.38 | [TK]D-Fender | rwaite: Barring that, try OSLEC first |
| 15:52.06 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 15:56.59 | carrar | MORNING!! |
| 15:57.26 | carrar | ohayoooooooooooo |
| 15:58.37 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:58.50 | *** join/#asterisk synchris (n=synchris@athedsl-81668.home.otenet.gr) |
| 15:58.56 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 15:59.00 | [TK]D-Fender | gozaimasu? |
| 15:59.25 | carrar | hai! |
| 15:59.38 | carrar | I'm of the informal variety |
| 16:00.22 | *** part/#asterisk fred-tmft (n=fred-tea@ip70-171-36-194.ga.at.cox.net) |
| 16:00.36 | rwaite | [TK]D-Fender: i will, thanks |
| 16:00.37 | carrar | been hittin Japan every year for the last few years, lovin it |
| 16:02.50 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 16:06.34 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:07.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:08.31 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-9221291fd05c4267) |
| 16:09.54 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 16:12.12 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 16:12.14 | casix | hello |
| 16:13.17 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:13.36 | cesar_CR | hello anybody expert in upgrading a cisco phone firmware ?? |
| 16:14.18 | casix | I have a problem connecting an asterisk with a ser server with a sip trunk. I need to make a trunk for incoming calls and another for outgoing calls because I need to use differents codecs. I have created a peer and a user with the same host option and different user and password but all, incoming and outgoing calls, use the same definition. How can I make it work? |
| 16:14.26 | jameswf-home | I am an expert at performinf velocity testing of cisco products |
| 16:14.29 | tzanger | haha |
| 16:14.41 | *** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net) |
| 16:16.01 | cesar_CR | :) |
| 16:22.52 | *** join/#asterisk onats (n=onats@unaffiliated/onats) |
| 16:24.45 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:26.30 | *** join/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 16:27.40 | damnpoet | hi there! could anyone tell where can i find information for the integration of a panasonic TDA200 with asterisk? |
| 16:27.56 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 16:29.46 | damnpoet | anyone? |
| 16:34.51 | [TK]D-Fender | damnpoet: No such thing |
| 16:35.23 | [TK]D-Fender | damnpoet: Whatever means you have of getting the Pana to talk to * has nothing to do with what it'll take to set up * to do whatever you will have it do |
| 16:35.51 | [TK]D-Fender | DamnIf it can talk SIP, then all * needs is a basic peer entry similar to setting it up with any other ITSP. The rest is up to what you want * to do for you |
| 16:35.59 | carrar | cesar_CR, I've done my fair share of cisco 7900 firmware upgrades |
| 16:36.28 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 16:36.29 | [TK]D-Fender | damnpoet: If you wire it over T1, analog, etc, then its jsut like plugging it into telco wiring and again the word "integration" doesn't really exist |
| 16:40.47 | damnpoet | [TK]D-Fender: thanks alot, but i`m a newbie on this subject, could you point me in the right direction for finding info about a tda200 and * working on the same network |
| 16:41.16 | [TK]D-Fender | damnpoet: How are you intending on connecting them? |
| 16:42.18 | damnpoet | [TK]D-Fender: * for the internal network and the pbx for analog conections, is that what you mean? |
| 16:44.04 | casix | I have a problem connecting an asterisk with a ser server with a sip trunk. I need to make a trunk for incoming calls and another for outgoing calls because I need to use differents codecs. I have created a peer and a user with the same host option and different user and password but all, incoming and outgoing calls, use the same definition. How can I make it work? |
| 16:44.05 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-75-251-168.myvzw.com) |
| 16:44.17 | onats | hi, i need some help. I am trying to connect two soft phones, one is outside the network, the other is in my lan. the ports have been forwarded as follows: 5060-5072 udp, 4569-4569 udp 10001 20000 udp. i also edited rtp.conf to have rtpstart=10001,rtpend=20000. then sip_nat.conf to: nat=yes,externhost=<ddns address>,externrefresh=28800,localnet=<internal lan subnet>/255.255.255.0 |
| 16:49.00 | ManxPower | onats: You only need 5060/UDP and 10000 - 20000/UDP (or whatever you have in rtp.conf) 4569 is IAX2 and 5061-5072 are not used for SIP. |
| 16:50.32 | onats | ManxPower, thanks for the response. my problem is that i there is no audio coming in on both sides.. |
| 16:50.48 | ManxPower | onats: Check your firewall or NAT settings on the router. |
| 16:51.44 | [TK]D-Fender | damnpoet: I mean how is the Pana going to talk to *? |
| 16:52.02 | ManxPower | BTW, does the one that is on your LAN work? |
| 16:52.25 | [TK]D-Fender | onats: you must have "canreinvite=no" |
| 16:52.51 | onats | ManxPower, i have set the port forwards on my router already. firewall should automatically follow.. |
| 16:52.52 | casix | quit |
| 16:52.55 | casix | ups |
| 16:52.56 | casix | :P |
| 16:52.59 | onats | D-Fender, where is that setting found? |
| 16:53.16 | ManxPower | onats: You must not have read the ~sipnat document. |
| 16:53.17 | ManxPower | ~sipnat |
| 16:53.18 | 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 |
| 16:53.25 | ManxPower | It should tell you all you need to know. |
| 16:53.48 | [TK]D-Fender | ManxPower: Of course not... you know what he's using.... |
| 16:53.49 | onats | thank you |
| 16:53.51 | ManxPower | onats: what the heck is sip_nat.conf. It's not a standard Asterisk file. |
| 16:54.19 | onats | manxpower, i'm using trixbox. but i would think concepts are similar? |
| 16:54.21 | Qwell | ~freepbx |
| 16:54.22 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:54.31 | onats | is there something like trixbox on asterisk? |
| 16:54.39 | Qwell | ...what? |
| 16:54.43 | ManxPower | onats: Well there is 10 mins of my life I'll never get back. |
| 16:54.51 | ManxPower | onats: talk to the trixbox people. |
| 16:54.52 | [TK]D-Fender | ~whee |
| 16:54.53 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 16:55.24 | onats | sorry about that... |
| 16:55.43 | ManxPower | onats: almost nothing we say will apply to Trixbox. |
| 16:55.59 | Qwell | unless we say something like "Some Asterisk distros really suck." |
| 16:56.03 | onats | ManxPower, ok. im just initially playing with it... |
| 16:56.25 | [TK]D-Fender | onats: I told you the setting to look for. Go hunt through your GUI to find how to set it |
| 16:56.44 | onats | D-Fender.. ok will look it up |
| 16:57.10 | damnpoet | s |
| 16:59.05 | ManxPower | [TK]D-Fender: I audited the Asterisk Fast Start class at Digium this week. It is a good class for Asterisk noobs. |
| 16:59.39 | ManxPower | They gave the people that paid for the class a TDM411B, a TE100P, and a Polycom phone. |
| 16:59.45 | Qwell | ManxPower: O.o |
| 16:59.54 | Qwell | ManxPower: and you didn't come upstairs? |
| 17:00.04 | Carlos_PHX | Just switched my desk phone from a Polycom 650 to a Linksys 962. Wow, what an improvement, never realized the extreme suckage that is Polycom until I tried something better. |
| 17:00.15 | ManxPower | Qwell: I'll be auditing more classes. |
| 17:00.16 | [TK]D-Fender | LOL |
| 17:00.22 | [TK]D-Fender | Carlos_PHX: Hilarious... |
| 17:00.42 | ManxPower | Carlos_PHX: Please put down the bong and step away from the computer. |
| 17:00.42 | [TK]D-Fender | Carlos_PHX: Considering anything an "upgrade" from an IP 650.... |
| 17:01.17 | magronez | is back |
| 17:01.25 | Carlos_PHX | I'd happily trade all the Polycoms in our deployment for Linksys. |
| 17:01.28 | [TK]D-Fender | Carlos_PHX: Don't think you'll find a preson in this room who isn't laughing at the thought of that.... |
| 17:01.41 | Qwell | Carlos_PHX: What Polycoms? |
| 17:01.44 | Carlos_PHX | Huh, interesting. Used both? |
| 17:01.48 | Qwell | That can be arranged. |
| 17:01.52 | [TK]D-Fender | Carlos_PHX: Perhaps you can tell us what you liked and disliked about each to come to that conclusion... |
| 17:01.59 | Carlos_PHX | We have anything from 500s and up. |
| 17:02.00 | [TK]D-Fender | Qwell: IP 650 <--- |
| 17:02.17 | Carlos_PHX | From the service side, they take forever to boot. |
| 17:02.45 | Carlos_PHX | User side is harder to say, just overall usability, number of available buttons on the add-on console, things like that. |
| 17:02.45 | Qwell | Carlos_PHX: How many 650s? |
| 17:02.47 | onats | which asterisk version should i get started with? something just for home use? |
| 17:02.54 | [TK]D-Fender | Carlos_PHX: On the same side, you don't HAVE to reboot Polycom's all the time. |
| 17:02.56 | ManxPower | Carlos_PHX: If you have to keep rebooting your phones then you have serious issues unrelated to your phones. |
| 17:02.59 | Carlos_PHX | Qwell: I have no idea, not that many I'm sure. |
| 17:03.03 | *** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn) |
| 17:03.09 | Qwell | find out - like I said, that can be arranged. |
| 17:03.32 | Carlos_PHX | [TK]D-Fender: They have to be rebooted the same number of times as the Linksys, when we make changes and fine tune settings. |
| 17:03.37 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 17:03.48 | [TK]D-Fender | Carlos_PHX: I can't imagine a more usable phone than a 650, the only point I can agree on is the side-car button count. |
| 17:03.51 | ManxPower | Qwell: It is pretty obvious that the person that wrote the class lessons and slides never worked in Asterisk tech support. 8-) |
| 17:03.54 | Carlos_PHX | I can test a setting in 10 seconds vs. 7 minutes. |
| 17:04.06 | [TK]D-Fender | Carlos_PHX: How often do you have to actually change people phones? |
| 17:04.20 | [TK]D-Fender | Carlos_PHX: And you've done something wrong if it takes 7 minutes. |
| 17:04.27 | [TK]D-Fender | Carlos_PHX: Mine boot in 2 or less. |
| 17:04.34 | Carlos_PHX | Interesting perspective, I had that too until I tried the other on my own desk. |
| 17:04.47 | hesco | ~pb |
| 17:04.48 | 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 |
| 17:05.09 | Carlos_PHX | We're a service provider, so we're changing or installing new phones every day. |
| 17:05.27 | [TK]D-Fender | Carlos_PHX: Installing new phones is no issue, 1st boot, end of story. |
| 17:05.31 | Carlos_PHX | Stopped selling the Polycoms long ago, but never got around to changing my own. |
| 17:06.09 | jaytee | Carlos_PHX, how much an ounce does that Kronik you're smoking cost? |
| 17:06.19 | Carlos_PHX | That's true for basic functions but changes require a reboot in most cases. |
| 17:06.37 | Carlos_PHX | Heh, I see there are Polycom fans here. Have you had a 962 on your own desk for a while? |
| 17:06.44 | [TK]D-Fender | Carlos_PHX: What do you actually have to change once the identity is set? |
| 17:07.03 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 17:07.08 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 17:07.17 | Carlos_PHX | Anything from ringers to the button assignments, and much more. |
| 17:07.31 | [TK]D-Fender | Carlos_PHX: Button assignments? Which? |
| 17:07.39 | Carlos_PHX | The add-on console. |
| 17:07.57 | [TK]D-Fender | Carlos_PHX: Who needs to reboot? Add them direct and you don't reboot. |
| 17:08.08 | Carlos_PHX | Hard to do when you're an ITSP. |
| 17:08.15 | Carlos_PHX | Unless you wanna go on site. |
| 17:08.26 | [TK]D-Fender | Carlos_PHX: Thi si user-grade stuff. its in the USER guide. |
| 17:08.43 | Carlos_PHX | We don't work that way. |
| 17:08.49 | [TK]D-Fender | Carlos_PHX: The idiot who can't add a contact to their directory doesn't deserve a phone. |
| 17:08.52 | ManxPower | Carlos_PHX: HUH? You set up the phones to check their config file for updates once per night and download it's new config and reboot when not in use. |
| 17:09.14 | Carlos_PHX | We've had partial success with that, however when we want a change we want it to happen now. |
| 17:09.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.52 | [TK]D-Fender | Carlos_PHX: Its pretty instant when you do it on the phone, but if you want to micro-manage chump directory setting, then you should look at your business approach |
| 17:10.06 | Carlos_PHX | But hey, put both on your desk for a week if you're a heavy phone user and see what you think. |
| 17:10.20 | Carlos_PHX | My 650 is going on eBay or whatever. |
| 17:10.25 | [TK]D-Fender | Carlos_PHX: But if thats your greatest concern then hey, have fun with Linksys. I'd much rather go for Polycom quality |
| 17:10.41 | Carlos_PHX | The quality is a good theory, haven't seen it yet. |
| 17:10.55 | Carlos_PHX | The speakerphone IS better. |
| 17:10.58 | Carlos_PHX | Way better. |
| 17:11.00 | Carlos_PHX | Never use it. |
| 17:11.15 | [TK]D-Fender | Carlos_PHX: Anyway, glad you're happy with them. |
| 17:11.45 | Carlos_PHX | I had no idea others were so partial, so just tossing it out here to see what others think. It was educational. :-) |
| 17:11.50 | [TK]D-Fender | Carlos_PHX: I'd hate to have to centrally manage every little thing all the time for clients like that. |
| 17:12.09 | Carlos_PHX | It's core to our business model. |
| 17:12.17 | Carlos_PHX | It's actually not bad after 30 days or so. |
| 17:12.27 | Carlos_PHX | Quite intense during install phase and first month. |
| 17:12.27 | [TK]D-Fender | Carlos_PHX: If you wanted godly attendant buttons, you could ahve gone for the Aastra... |
| 17:12.34 | *** join/#asterisk |Magrao| (n=eusei@unaffiliated/magrao/x-2903) |
| 17:13.40 | *** join/#asterisk seba0606 (n=smelgin@adsl190-28-129-120.epm.net.co) |
| 17:13.51 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 17:14.40 | Carlos_PHX | I might have to look at that for receptionists. I'm just getting my first feedback on the 962/932 from a high-volume site. |
| 17:16.47 | smelgin | (nod) |
| 17:17.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:17.14 | [TK]D-Fender | Carlos_PHX: my dislikes on SPA < 962 = too light, they slid around. tinny handset, small screen with poor pixel resolution for soft-keys. Drastically inferior call handling to Polycom (no split/join, spanning or mutiple calls per key), etc |
| 17:17.27 | [TK]D-Fender | Carlos_PHX: Speakerphone goes without saying. |
| 17:18.46 | *** part/#asterisk damnpoet (n=UCF\inf2@orion.ucf.edu.cu) |
| 17:19.16 | [TK]D-Fender | Carlos_PHX: I'm betting the 962 shares many of the same caveats as the lower models, I just haven't had the eval time on them |
| 17:21.29 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:22.38 | Carlos_PHX | Those are interesting comments, I might have to test them. I can tell you that they now do multiple calls per registration. |
| 17:23.04 | Carlos_PHX | The 962 does have some differences, I haven't put a 942 on my desk yet. |
| 17:23.10 | [TK]D-Fender | Carlos_PHX: You can't do multiple per line-key. |
| 17:23.25 | [TK]D-Fender | Carlos_PHX: So thats 4 appearances on a 94X |
| 17:23.37 | [TK]D-Fender | Carlos_PHX: And I don't believe thats even 4 reg's |
| 17:24.03 | [TK]D-Fender | Carlos_PHX: An IP 650 can juggle 24 calls easy without expansion. |
| 17:24.48 | [TK]D-Fender | Carlos_PHX: join/split for ad-hoc conferencing is a huge convenience as is being able to drop a 3-way call leaving the 2 other sides bridged. |
| 17:25.03 | Carlos_PHX | I just did two on my first line key. |
| 17:25.11 | [TK]D-Fender | Carlos_PHX: Frankly I've never seen any phone come anywhere near Polycom on handling. |
| 17:25.58 | [TK]D-Fender | Carlos_PHX: Aastr's strong suit was their soft-keys which are unmatches.. state-based, better presence support... if only it didn't crash the phone |
| 17:26.02 | Carlos_PHX | If we had someone doing 24 calls I'd have to re-think everything I suppose. |
| 17:26.51 | [TK]D-Fender | Carlos_PHX: And there is the microBrowser as well.... I use that for live queue stats, etc |
| 17:27.24 | Carlos_PHX | I'm just about to try that out. But the Polycom has one too, doesn't it? |
| 17:27.43 | *** join/#asterisk kdas (n=kdas@c-98-207-248-194.hsd1.ca.comcast.net) |
| 17:27.53 | kdas | howdy all |
| 17:29.25 | [TK]D-Fender | Carlos_PHX: Polycom has it, Linksys didn't for any model I'd ever heard of. |
| 17:30.00 | Carlos_PHX | Joining is supported in my phone. |
| 17:30.32 | Carlos_PHX | Pretty sure I saw split in the docs. |
| 17:30.34 | [TK]D-Fender | Carlos_PHX: So you can take 2 unrelated calls and join / sploit at will? |
| 17:30.41 | Qwell | and of course the audio quality on a 650... |
| 17:31.02 | jaytee | I sold my iPod and bought a 650. |
| 17:31.05 | Carlos_PHX | [TK]D-Fender: Tested join, didn't test split. |
| 17:31.35 | hesco | Here are two pastes which outline an issue I'm having with cdr_pgsql. I had this working until I ran make samples a second time. Last night I ran a make clean, then rebuilt and re-installed *. But I'm still seeing these same errors. Any clues how to resolve this issue and start collecting data in my postgreSQL backend again? http://pastebin.ca/1235438 and http://pastebin.ca/1234977 |
| 17:32.10 | jaytee | hesco, is this the duplicate calldate problem? |
| 17:32.21 | hesco | yes, same one |
| 17:32.47 | jaytee | you've been at this for days now. I would have repartitioned and started from scratch by now. |
| 17:33.10 | Carlos_PHX | [TK]D-Fender: As far as the browser, it's an RSS reader, don't know how it would compare to the Polycom. Have not used either, but am going to test this week and see what we can do with them. |
| 17:33.52 | hesco | This * server doubles as a db server. Building my base database was a five week project, that is five weeks of running perl scripts to import the plain text data, phone ringing bb soon |
| 17:34.37 | kdas | i got my * box all up and running with outgoing and incoming calls, but i can't hear anything when i answer/call out and the other party can't hear me. any ideas? i am running a handytone 286 through a asterisk box that connects to callwithus.com. i have a normal phone connected to the handytone obviously ;). any ideas ? |
| 17:35.16 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 17:35.42 | Carlos_PHX | [TK]D-Fender: You may be right on split, can't find an obvious way to do it. |
| 17:36.03 | Carlos_PHX | Once conferenced, the only obvious answer is to join the calls and leave the conf. |
| 17:36.07 | [TK]D-Fender | Carlos_PHX: Can you drop a 3-way and have the other 2 stay joined? |
| 17:36.13 | Carlos_PHX | I mean, only obvious option. |
| 17:36.14 | Carlos_PHX | Yes |
| 17:36.21 | [TK]D-Fender | Carlos_PHX: Thats good then... |
| 17:36.25 | Carlos_PHX | But the display has no obvious way to drop the first call. |
| 17:36.31 | Carlos_PHX | It's easy to drop the second. |
| 17:36.47 | Carlos_PHX | It's almost like an announced transfer. |
| 17:36.52 | [TK]D-Fender | kdas: Does the HT work 100% both ways direct to *? Then does your ITSP do the same? |
| 17:38.01 | kdas | [TK]D-Fender, the HT can call out and recieve calls 100% i hear ring an all but no audio when i answer. ITSP = ? |
| 17:38.08 | hesco | ok, back. |
| 17:38.25 | Carlos_PHX | [TK]D-Fender: Linksys has done firmware updates consistently every couple months, if you looked at them 6 months ago it's possible they're totally different. We've only been selling/installing for about 4 months, and one went on my desk Monday. |
| 17:39.05 | smelgin | ;) |
| 17:39.24 | jameswf-home | burns kubuntu 8.10 rc1 |
| 17:39.27 | Carlos_PHX | We started selling them because we took a 50-seat company from their dead * server to our service and they had the 942s everywhere. It was SO much faster to convert them than to convert Polycoms. |
| 17:39.27 | hesco | so jaytee, repartitioning this box is not a two hour project, but a five week project. |
| 17:39.37 | jaytee | ok |
| 17:40.16 | hesco | If there is any way to tear out ONLY the *, and rebuild that, I am ready. In fact I thought that was what i was doing last night with my make clean, etc. |
| 17:41.20 | hesco | I emptied my /etc/asterisk and moved back ONLY the files I had edited since the preceeding install. |
| 17:41.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:42.24 | *** join/#asterisk StephenF[W] (n=none@198.144.197.28) |
| 17:42.28 | jaytee | hesco, how about pastbining your cdr_custom.conf file? |
| 17:42.38 | hesco | coming at you |
| 17:42.46 | jaytee | and your cdr_pgsql.conf |
| 17:42.57 | hesco | ok |
| 17:43.53 | kdas | [TK]D-Fender, is there anything else you need to know ? |
| 17:45.29 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:45.34 | hesco | here you go: http://pastebin.ca/1235838 |
| 17:47.42 | kdas | i got my * box all up and running with outgoing and incoming calls, but i can't hear anything when i answer/call out and the other party can't hear me. any ideas? i am running a handytone 286 through a asterisk box that connects to callwithus.com. i have a normal phone connected to the handytone obviously ;). any ideas ? |
| 17:47.44 | Katty | growls lightly. |
| 17:48.05 | *** join/#asterisk pg1054 (n=pg1054@unaffiliated/pg1054) |
| 17:49.30 | LeddyHM | Is there a change between 1.2 and 1.4 that needs to be updated to hear "ring" sound when dialing an extension (to the calling party) |
| 17:49.30 | kdas | LeddyHM, i don't think so |
| 17:49.30 | LeddyHM | users hear dead silence |
| 17:49.39 | Qwell | how are you dialing? |
| 17:49.48 | LeddyHM | users extension |
| 17:49.59 | *** join/#asterisk Cyberpony (n=CyberPon@66-194-25-11.static.twtelecom.net) |
| 17:50.16 | LeddyHM | if I dial extension to extension it seems fine (just checked) |
| 17:50.30 | LeddyHM | if I go through the IVR I get zip |
| 17:50.32 | Qwell | okay, but how are you dialing? |
| 17:50.40 | Qwell | so you answered in the IVR |
| 17:50.43 | Qwell | don't do that |
| 17:51.18 | pg1054 | I've built * from svn .../branch/160/. No errors in the build, and launches OK. current frontend is FreePBX. I can connect local extensions OK, but if I create a trunk, specifying host=callcentric.com (my provider), I see no registration. looking at tcpdump, there is *NO* traffic to/from callcentric showing up at all. |
| 17:51.18 | pg1054 | I've clearly misconfigured something, but what? |
| 17:51.51 | jeev | hey doods, when i call from a landline to http://hahpo.pastebin.com/d19724ad6 it makes the ringing noise, when i call from a tmobile cell phone.. it doesn't have that sound of ringing on the callers side but the other side will answer the phone just fine. how can i make it make the tone ? |
| 17:51.56 | LeddyHM | qwell: what am I supposed to do instead? |
| 17:52.43 | LeddyHM | the auto attendant anwers and asks for an extension |
| 17:52.55 | Cyberpony | I have a question regarding AEL macros (using macros vs using Gosub/Return) in my configuration files. |
| 17:53.26 | Cyberpony | LeddyHM: does the IVR context know about the extensions? |
| 17:53.43 | LeddyHM | if I dial direct there is no ringtone either |
| 17:53.55 | LeddyHM | cyber: I'd imagine so |
| 17:54.05 | LeddyHM | it worked beautifully in 1.2 |
| 17:54.19 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:55.37 | Cyberpony | The documentation for version 1.6.x says that Macro() is deprecated and I should change to Gosub()/Return().... but aelparse give warnings about Gosubs and Returns...... |
| 17:55.56 | Cyberpony | which way should I be using? |
| 17:56.24 | jaytee | hesco, are you doing anything in your dialplan with CDR or are you just letting it log all calls with defaults? |
| 17:56.32 | kdas | i got my * box all up and running with outgoing and incoming calls, but i can't hear anything when i answer/call out and the other party can't hear me. any ideas? i am running a handytone 286 through a asterisk box that connects to callwithus.com. i have a normal phone connected to the handytone obviously ;). any ideas ? |
| 17:56.38 | kdas | sip.conf = http://pastebin.com/m156134f6 |
| 17:56.57 | ManxPower | kdas: See the sipnat doc |
| 17:56.59 | ManxPower | ~sipnat |
| 17:56.59 | 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 |
| 17:57.14 | Cyberpony | LeddyHM: does the IVR context include the context(s) that the extensions are in? |
| 17:57.34 | kdas | ManxPower, i am not behind a nat though |
| 17:58.24 | hesco | jaytee: log all calls by default. |
| 17:58.25 | LeddyHM | Cyber: I'll have to check with someone on that I didn't write it |
| 17:58.34 | [TK]D-Fender | kdas: http://pastebin.com/m5cff4b81 |
| 17:58.44 | LeddyHM | and there he is |
| 17:59.00 | LeddyHM | rubs his genie [TK]D-Fender |
| 17:59.04 | kdas | [TK]D-Fender, i have dynamic ip addy so what should i put there? |
| 17:59.25 | [TK]D-Fender | kdas: if your * is public you don't need localnet or externip/externhost |
| 17:59.52 | kdas | [TK]D-Fender, my asterisk is direct hooked to net |
| 18:00.03 | [TK]D-Fender | kdas: then no. |
| 18:00.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:00.31 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 18:00.33 | LeddyHM | tk: when people dial in they don't get a "ring ring" sound. Can you make sense of what Cyber is asking for? :) |
| 18:01.06 | kdas | ok let me try |
| 18:02.26 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d282318e8990babb) |
| 18:02.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:02.48 | jaytee | hesco, I don't see anything in those config files that look wrong. You're not using odbc are you? |
| 18:03.04 | hesco | no sir, no odbc |
| 18:03.05 | [TK]D-Fender | LeddyHM: I'd have to look at things... |
| 18:03.38 | jaytee | hesco, and you've verified that the cdr table structure is correct? |
| 18:03.58 | kdas | [TK]D-Fender, all seems to work but the VM :( |
| 18:04.15 | [TK]D-Fender | kdas: VM? |
| 18:04.22 | kdas | voicemail |
| 18:04.25 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:04.38 | LeddyHM | TK: if you wouldn't mind that would be awesome |
| 18:04.54 | [TK]D-Fender | kdas: How is it that wouldn't work? BTW, you are being very sloppy with codec assignment.. or lack thereof... fix this globally, and for each peer... |
| 18:04.56 | Cyberpony | so in 1.6 AEL should I still be using macros? |
| 18:04.56 | jaytee | VM! easily confused between voicemail, virtual machine or vagina monologues |
| 18:06.04 | hesco | I have not changed the table structure since this was working. I'll pastebin the \dt output though so you can see for yourself. I'm new to the schema. I found my create statement in a google search on one of the asterisk related wikis. |
| 18:06.32 | kdas | [TK]D-Fender, ok so set all peers to gsm and alaw ? (those are the 2 sound files i use) ? |
| 18:07.02 | jaytee | hesco, when it broke what was the last thing you'd done? |
| 18:08.09 | rwaite | i wonder how hard it would be to put my server under a scm |
| 18:08.44 | codefreeze-lap | Cyberpony: the only way to form a subroutine in AEL is with the "macro" keyword; How AEL generates code in the dialplan to perform that task is up to AEL. So, yes, keep using "macro" in 1.6. Try not to use "Gosub" or "Macro" apps in AEL. Use the AEL keywords instead, and you'll be better off. |
| 18:09.06 | hesco | jaytee: http://pastebin.ca/1235856 |
| 18:09.07 | Cyberpony | ok thank you very much. |
| 18:09.33 | Cyberpony | the README's are a little confusing on that point. |
| 18:10.03 | Cyberpony | putting what you just said should be stated like that in the docs. |
| 18:10.47 | kdas | all my incoming calls callerid are from the state/country rather then owner of phone. is that a * thing or callwithus ? |
| 18:11.57 | ManxPower | kdas: your provider is who is sending you the callerid name info |
| 18:12.15 | kdas | [TK]D-Fender, when i check my voice mail it says "you have one new message" and then it gives me a busy tone rather then playing the message :( |
| 18:12.26 | kdas | ManxPower, ok thankyou sir |
| 18:12.44 | kdas | ManxPower, to you happen to know how to change that with callwithus ? |
| 18:12.52 | *** join/#asterisk jdjurici (n=jdjurici@78-1-130-148.adsl.net.t-com.hr) |
| 18:12.54 | jdjurici | yo |
| 18:13.05 | jdjurici | got a question, folks from dev told me to ask here |
| 18:13.26 | jdjurici | can I use any of these function to get ip from dns name? |
| 18:13.28 | jaytee | hesco, well since the error message makes it seem like it's trying to insert a record with two calldate fields in it either your table is corrupt or the cdr_pgsql.so module is corrupt and it's most likely the latter unless you've added something in your dialplan that massages the CDR data. |
| 18:13.33 | jdjurici | ast_get_ip or ast_get_ip_or_serv? |
| 18:13.44 | jdjurici | or is it some other function? |
| 18:14.02 | jdjurici | I'm trying to change mgcp reload function a little bit... |
| 18:14.22 | hesco | yatee: sorry, had missed that question. the last thing I did was run make sample again. |
| 18:14.40 | jaytee | make sample in asterisk? |
| 18:15.40 | hesco | yes |
| 18:15.51 | hesco | no, at the bash prompt |
| 18:15.53 | *** join/#asterisk Shadow98 (n=thompsbm@cpe-76-181-153-103.columbus.res.rr.com) |
| 18:16.14 | Shadow98 | hey guys i just have a quick question about whether asterisk is able to provide this functionality or not.. |
| 18:16.19 | jaytee | hesco, yes I know at the prompt. I meant in the asterisk source directory |
| 18:16.32 | hesco | yes, in the source directory |
| 18:16.48 | jaytee | hesco, had you backed up the config files in /etc/asterisk prior to running make samples again? |
| 18:16.57 | kdas | any one have idea why my voicemail messages are busytones rather then the messages? |
| 18:17.36 | Shadow98 | im going to have a website where people enter their phone numbers and receive a call back...when called back they are going to be prompted with choice press one for this press 2 for that...and so on...it will then ask them a time from like enter 30 for 30 days or whatever...will asterisk allow calls to be placed like this and receive user input.. |
| 18:18.23 | hesco | oh yes. I now have five directories in the form /etc/asterisk_*, besides the current one. |
| 18:18.49 | jaytee | kdas, does it happen with internal calls too or just from outside callers and are they coming in SIP or over PSTN? |
| 18:18.54 | hesco | might those be read as well? |
| 18:19.37 | kdas | jaytee, umm it when i try to call my voicemail on my pstn which is connected to handytone which is connected to my * box |
| 18:19.39 | jaytee | hesco, i don't understand what you mean by /etc/asterisk_* |
| 18:20.25 | jaytee | hesco, you should only have one asterisk directory under /etc |
| 18:23.03 | jaytee | kdas, voicemail on PSTN? not voicemail on your * server? |
| 18:23.39 | kdas | jaytee, voicemail one my * server |
| 18:24.13 | *** join/#asterisk stoffell (n=stoffell@d51A4D1C0.access.telenet.be) |
| 18:24.48 | jaytee | kdas, you don't hear a prompt to leave a message? it's just hanging up on you when you call it? |
| 18:25.06 | kdas | jaytee, when i check my voicemail |
| 18:25.17 | kdas | jaytee, not when i leave a message |
| 18:25.21 | jaytee | kdas, I meant if you do a test call |
| 18:25.31 | kdas | jaytee, that works fine |
| 18:26.00 | jaytee | kdas, what happens if you make a test call and then hangup at the prompt without leaving a message? do you get a busy tone message? |
| 18:26.59 | kdas | jaytee, i think i found the problem... i was missing vm-onefor.* ahha |
| 18:33.29 | Katty | anyone know anything about displaying Flickr on a polycom? |
| 18:33.48 | Katty | i have a url that goes into the idlebrowser |
| 18:33.52 | *** join/#asterisk pimpwell (n=portches@ool-457e6e03.dyn.optonline.net) |
| 18:34.08 | pimpwell | do the fonality guys hang out here? |
| 18:34.10 | *** join/#asterisk Anon472 (n=Anon472@86.99.127.196) |
| 18:34.17 | pimpwell | considering its powered by * |
| 18:34.24 | Shadow98 | anybody able to answer my question above.. |
| 18:34.28 | Anon472 | any wayt to make IVR by text to voice instead of using pre packaged selections? |
| 18:34.40 | Anon472 | any text to voice reader for asterisk? |
| 18:35.03 | Katty | Anon472: i usually just record the prompts myself |
| 18:35.13 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:35.18 | Anon472 | Katty: my voice is not so sexy |
| 18:35.26 | iCEBrkr | Ok, So, my G1 won't play .wav49 |
| 18:35.28 | iCEBrkr | :( |
| 18:35.28 | Katty | Anon472: i'm sorry to hear that. |
| 18:35.31 | Carlos_PHX | pimpwell: Probably not, there are some political issues with Fonality and the * community |
| 18:35.35 | Katty | iCEBrkr: :< |
| 18:35.41 | Katty | [TK]D-Fender: ping? |
| 18:35.45 | iCEBrkr | and I think I've tried asterisk-addon's and doing format=mp3 in voicemail.conf |
| 18:35.48 | iCEBrkr | I don't remember it working |
| 18:36.02 | Anon472 | FreeSWITCH has that built-in.. text to voice converter.. so the dialplan has text, and FS converts it to voice |
| 18:36.06 | pimpwell | Im in NY, I need to hire someone to setup fonality for a few days |
| 18:36.14 | Anon472 | i need same thing for Asterisk |
| 18:36.27 | iCEBrkr | Anon472: Don't so those dirty words in here.. |
| 18:36.29 | iCEBrkr | :P |
| 18:36.37 | |Magrao| | is away: ninguem merece atender OS de siemens |
| 18:36.54 | Anon472 | iCEBrkr: which words? |
| 18:37.25 | pimpwell | actually im in Westchester, NY |
| 18:37.25 | Anon472 | freeswitch? i love it but can't manage to use it so far hehehehe |
| 18:37.36 | iCEBrkr | Anon472: hehe |
| 18:37.36 | Anon472 | xml configs are too complicating |
| 18:37.43 | iCEBrkr | Yeah.. |
| 18:37.51 | iCEBrkr | IT's just another learning curve. |
| 18:37.57 | iCEBrkr | I felt the same way when I started with Asterisk |
| 18:38.00 | stoffell | Katty, that'll probably only work on the higher end of polycom phones, right? |
| 18:38.01 | iCEBrkr | Really overwhelmed. |
| 18:38.13 | iCEBrkr | Now that you're used to Asterisk, it's hard to deal with FS XML's |
| 18:38.15 | Anon472 | iCEBrkr: asterisk is the windows of telephony :P |
| 18:38.25 | [TK]D-Fender | Katty: Mew? |
| 18:38.25 | iCEBrkr | :-X |
| 18:38.53 | Katty | [TK]D-Fender: i want to put this: http://plcmapps.zipdx.com/plcmapps/IdleFlickr.aspx?name=veladeptus into the idle browser of sip.cfg |
| 18:38.57 | Katty | [TK]D-Fender: but not the main sip.cfg |
| 18:39.12 | Katty | [TK]D-Fender: i want to put it into MAC.cfg, which i guess technically would be phone-MAC.cfg |
| 18:39.28 | jaytee | Anon472, check out Digium's site for Cepestral text to speech addon |
| 18:39.35 | Katty | [TK]D-Fender: but i can't seem to find much info on how to dump that in there. have any documents? |
| 18:39.55 | [TK]D-Fender | Katty: dump it on a web-server, point your idle page to it. End of story. |
| 18:40.23 | [TK]D-Fender | Katty: Katty Keeping in mind that is so dark you are FUBAR'd on anything but an IP 670 |
| 18:40.33 | Katty | [TK]D-Fender: still be neat. |
| 18:40.45 | Katty | [TK]D-Fender: i just can't figure out what to put into phone-MAC.cfg to get it to display |
| 18:40.46 | [TK]D-Fender | Katty: If your idea of neat is a solid grey block... |
| 18:40.52 | Katty | [TK]D-Fender: i'm guessing some sort of <microbrowser> tag or something |
| 18:41.08 | [TK]D-Fender | Katty: the mb idle tags are blatantly obvious. you shove a URL there |
| 18:41.24 | Katty | [TK]D-Fender: my phone1.cfg file doesn't have a sample of what the line looks like. |
| 18:41.30 | Katty | [TK]D-Fender: could you pastebin me yours? |
| 18:41.43 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 18:42.00 | [TK]D-Fender | Katty: Its in the main. Copy that block off sip.cfg and paste it into your phoneXXX.cfg |
| 18:42.05 | Katty | kk |
| 18:42.12 | [TK]D-Fender | Katty: then you can personalize it per phone |
| 18:43.56 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 18:44.10 | Katty | [TK]D-Fender: http://pastebin.ca/1235865 <- is that the correct bit from sip.cfg? |
| 18:45.04 | Katty | i'd think there'd be a timeout or something for each image |
| 18:45.10 | Katty | err, refresh |
| 18:45.49 | [TK]D-Fender | Katty: right section, but you are missing the IDLE tag |
| 18:45.58 | [TK]D-Fender | Katty: that would fall under the services key |
| 18:46.14 | [TK]D-Fender | Katty: Go check your admin guide |
| 18:47.52 | *** join/#asterisk nido (i=nido@5ED105AD.cable.ziggo.nl) |
| 18:52.35 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:54.09 | onats | ~stun |
| 18:54.10 | jbot | rumour has it, stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs, or a client side method to cater to crappy sip servers, or a phaser setting |
| 18:54.40 | jameswf | I truely under estimated the pain of a clean OS install on my laptop |
| 18:54.43 | jeev | stun is cool, i want to learn it |
| 18:54.49 | jeev | jameswf, what laptop |
| 18:55.15 | jameswf | my work laptop was getting old and moldy so I wiped it and updated |
| 18:55.48 | jeev | model? |
| 18:55.50 | jeev | dell rules all |
| 18:56.10 | nido | I've set up an asterisk 1.4 server on OpenBSD; I can connect and register but when I connect I get a strange error: [Oct 25 20:23:37] NOTICE[27313]: chan_sip.c:13879 handle_request_invite: Call from '' to extension '10.1.1.241' rejected because extension not found. |
| 18:56.15 | jameswf | vostro 1000 |
| 18:56.19 | nido | 10.1.1.241 is the ip of the asterisk box |
| 18:56.22 | nido | any ideas? |
| 18:56.38 | De_Mon | how are you connecting? |
| 18:56.56 | nido | using ekiga on a linux box in the same network |
| 18:57.01 | jameswf | ~bsd |
| 18:57.02 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 18:57.13 | kerx | hey Katty & [TK]D-Fender |
| 18:57.30 | Anon472 | why my asterisk is unable to play wav files? is there any module to look at? |
| 18:57.36 | Anon472 | what module is responsible for playing wav? |
| 18:57.42 | kerx | make sure your wav file is sampled correctly |
| 18:57.57 | kerx | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 18:58.12 | Anon472 | kerx: they are, actually this is the moh wav file |
| 18:58.34 | De_Mon | nido are you dialing a number? what does your sip.conf and extensions.conf look like |
| 18:58.49 | nido | De_Mon: should I pm them to you? |
| 18:58.54 | De_Mon | ~pastebin |
| 18:58.55 | 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 |
| 18:58.55 | *** join/#asterisk wierdo (i=wierdo@77.78.3.107) |
| 18:59.38 | De_Mon | it's trying to connect to an extension '10.1.1.241', and I'm pretty sure you don't have an exten => 10.1.1.241,1 line |
| 18:59.54 | nido | that\s true |
| 18:59.55 | De_Mon | so, try dialing something that does exist? |
| 18:59.59 | jaytee | jameswf, vostros are cheap but pretty decent machines. they seem to hold up pretty well under the punishing conditions we use them for. |
| 19:00.34 | jameswf | I like mine does okay |
| 19:00.37 | De_Mon | returns to his OCS integration |
| 19:00.50 | jeev | jameswf, dells are the shit man. they're so easy to clean and set up |
| 19:00.50 | jaytee | made in Malaysia, supported in the Phillipines for a company headquartered in the US. |
| 19:00.56 | jameswf | is selectively restoring from backup |
| 19:01.13 | Anon472 | I get this error: [Oct 24 23:00:16] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-calm-river does not exist in any format |
| 19:01.17 | jeev | i got lenovo's now because i got a great deal on them.. but if i were in the market for a dell, i'd get the refurb XPS they have.. 2.5ghz, 4 gb ram.. for ~850 |
| 19:01.21 | jameswf | is running kubuntu 8.10 |
| 19:01.25 | jeev | in the market for a LAPTOP i meant |
| 19:01.51 | jameswf | (rc) |
| 19:02.02 | jaytee | dell lappys are nice. I'm not that fond of the small form factor Optiplex desktop though. Of the 24 SX-280's we purchased back in 2005 all of them have had the system boards fail. |
| 19:02.25 | Katty | [TK]D-Fender: think i got it--rebootin ma phone |
| 19:02.26 | nido | ah. I'm getting it now |
| 19:02.35 | jaytee | we're now buying all our desktops with the 3 year warranty plan to ensure that we get at least 3 years of of them. |
| 19:04.21 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 19:05.38 | jaytee | so the release candidate for Ibex is out. When's the actual final due out? next week? |
| 19:06.23 | [TK]D-Fender | jaytee: Oct 31 |
| 19:06.24 | jameswf | next Fri |
| 19:06.35 | jameswf | im inpatient |
| 19:06.51 | jaytee | I'm going to wipe my drive on my lappy and try it |
| 19:07.24 | jameswf | it has KDE 4.1.2 so you may have to hold your nose a little on kubuntu |
| 19:08.02 | Anon472 | jaytee: how long does in_background: ast_streamfile failed on IAX2/sameer-10790 for moh/fpm-sunshine |
| 19:08.05 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-sunshine does not exist in any format |
| 19:08.06 | Carlos_PHX | < Wonders if he should mention using a Mac laptop after the Polycom thread... |
| 19:08.09 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:912 ast_streamfile: Unable to open moh/fpm-sunshine (format 0x2 (gsm)): No such file or directory |
| 19:08.11 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on IAX2/sameer-10790 for moh/fpm-sunshine |
| 19:08.14 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-sunshine does not exist in any format |
| 19:08.17 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:912 ast_streamfile: Unable to open moh/fpm-sunshine (format 0x2 (gsm)): No such file or directory |
| 19:08.18 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 19:08.20 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on IAX2/sameer-10790 for moh/fpm-sunshine |
| 19:08.23 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-sunshine does not exist in any format |
| 19:08.26 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:912 ast_streamfile: Unable to open moh/fpm-sunshine (format 0x2 (gsm)): No such file or directory |
| 19:08.28 | jameswf | holy crap |
| 19:08.29 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on IAX2/sameer-10790 for moh/fpm-sunshine |
| 19:08.32 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-sunshine does not exist in any format |
| 19:08.33 | jameswf | ~pb |
| 19:08.34 | 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 |
| 19:08.35 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:912 ast_streamfile: Unable to open moh/fpm-sunshine (format 0x2 (gsm)): No such file or directory |
| 19:08.38 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on IAX2/sameer-10790 for moh/fpm-sunshine |
| 19:08.41 | jeev | owowow |
| 19:08.41 | Anon472 | [Oct 24 23:02:05] WARNING[13656]: file.c:602 ast_openstream_full: File moh/fpm-sunshine does not exist in any format |
| 19:08.41 | Carlos_PHX | Anon472: Hope you have your Nomex underwear. |
| 19:08.42 | jblack | wonders how many people just /ignored anon472 |
| 19:08.43 | jeev | anonn, stop |
| 19:08.45 | Anon472 | oops |
| 19:08.48 | Anon472 | <PROTECTED> |
| 19:08.58 | Anon472 | sorry, was a mistake |
| 19:09.04 | jeev | Anon472, can you first acknowledge pastebin.com? |
| 19:09.05 | jaytee | I did a clean install of Gutsy on my laptop and it worked great, even with NDISwrapper. I did the dist-upgrade to Hardy and it was like my CPU lost half it's speed |
| 19:09.16 | jeev | oh, linux.. no wodner |
| 19:09.17 | Anon472 | jeev: it was a mistake... |
| 19:09.19 | jeev | ok |
| 19:09.24 | jaytee | Anon472, what demo? |
| 19:09.44 | Anon472 | jaytee: for swift |
| 19:09.49 | jameswf | jaytee: I have been through 2 dist upgrades on this laptop thats why this time i opted for clean reinstall |
| 19:09.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:10.31 | jaytee | Anon472, don't know. is swift for streaming media? I don't use it whatever it is |
| 19:10.43 | jameswf | 2 X dist-upgrade = holy crap my system is borked |
| 19:10.48 | jaytee | thinks he's confusing me with someone else |
| 19:13.35 | jameswf | yay i can watch youtube |
| 19:15.05 | Qwell | youtube watches you. |
| 19:15.11 | Katty | next they'll be putting youtube on polyucom phones |
| 19:16.14 | De_Mon | mine already has that |
| 19:16.42 | tzafrir_laptop | jameswf, so fix it |
| 19:16.44 | hesco | jaytee: sorry, pulled away by the phone for too long. I hear you to say that these backups might be the issue? My plethora of /etc/asterisk_thisbackup, /etc/asterisk_thatbackup ??? |
| 19:16.53 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:17.08 | jameswf | break youtube? |
| 19:18.10 | hesco | Is it reading and rereading these configuration files on top of each other? If so its a wonder that this db issue is the only thing that is broken. |
| 19:18.31 | hesco | I'll move those elsewhere reload and test again |
| 19:19.29 | jameswf | http://www.youtube.com/videoyourvote << upload your vote to youtube.... |
| 19:19.32 | jameswf | wow |
| 19:21.14 | jeev | lets go take over youtube.com with the old network solutions spoof email |
| 19:24.26 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:25.17 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:25.55 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 19:27.52 | Qwell | jameswf: you did not just use <marquee>... |
| 19:28.46 | jameswf | no |
| 19:28.52 | jameswf | [move] |
| 19:28.58 | jameswf | :) |
| 19:29.20 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.68) |
| 19:30.15 | Daejeo | how many a big cellphone carriers are in US? can anyone tell me the names? |
| 19:30.27 | Daejeo | how many big cellphone carriers are in US? can anyone tell me the names? |
| 19:30.36 | jameswf | ummm |
| 19:30.39 | jameswf | ummm |
| 19:31.00 | [TK]D-Fender | Daejeo: 1. AT&T |
| 19:32.31 | Daejeo | verizon , t-mobile |
| 19:32.45 | jameswf | Daejeo: they arent big |
| 19:32.52 | Daejeo | are the part of AT&T? |
| 19:33.01 | [TK]D-Fender | Daejeo: Ask again next week ;) |
| 19:33.54 | Carlos_PHX | Why would you want a big cellphone? |
| 19:34.22 | jameswf | i has a brick |
| 19:34.31 | Daejeo | why i wanted to know, because i want to do reverse number lookup |
| 19:34.51 | jameswf | its been done |
| 19:35.23 | Daejeo | i see so many listings |
| 19:35.37 | Shadow98 | exit |
| 19:35.38 | Shadow98 | exit |
| 19:35.40 | Shadow98 | quit |
| 19:35.58 | jeev | lol |
| 19:36.06 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 19:36.11 | Daejeo | but it is hard to get the information abt celphone number |
| 19:36.53 | Daejeo | jameswf: smarty tell the URL who is providing free listing |
| 19:36.56 | jameswf | Shadow == elite enough for BITCH X not elite enough to use a / |
| 19:37.07 | jameswf | http://www.google.com |
| 19:37.26 | Katty | bummer. |
| 19:37.32 | Katty | polycom doesn't make any ip phones with video support |
| 19:37.40 | Katty | how sasd. |
| 19:37.41 | Katty | sad. |
| 19:37.49 | jameswf | seriously though http://www.fonefinder.net/ |
| 19:38.07 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:38.14 | jeev | has anyone here used heartbeat ? |
| 19:38.31 | jameswf | i use many heartbeats every minute |
| 19:38.44 | jeev | bahhhhhhhhhh |
| 19:39.51 | jaytee | jeev, just google Linux HA |
| 19:40.20 | jaytee | or if you prefer you could use http://goosh.org :-) |
| 19:40.32 | jameswf | has failover switches that heartbeat from asterisk |
| 19:40.39 | thehar | wat is heartbeat/high avail |
| 19:40.39 | jeev | i know about googling, i'm asking if anyone uses it. |
| 19:40.41 | thehar | precious |
| 19:41.04 | jeev | jameswf, i have dual wan.. everyone knows but i dont have a good failover method on the default gateway. asterisk is now set up to be redundant, incoming and outgoing calls are sent out two at a time. |
| 19:41.06 | jaytee | jeev, I plan to but I'm waiting on the hardware |
| 19:41.14 | jameswf | failover can even hit the reset pins on a panic |
| 19:41.27 | jeev | heh |
| 19:41.33 | jaytee | right now I'm just using heartbeat with Nagios for monitoring |
| 19:41.49 | jaytee | not for failover or high availabilty |
| 19:42.17 | jeev | ah |
| 19:42.23 | jeev | man i used to be 31337 with nagios. |
| 19:42.43 | jameswf | http://www.rhinoequipment.com/1portfail.html |
| 19:42.51 | Katty | does anyone have an opinion on the GXV3000 video phone? |
| 19:43.06 | jaytee | and then you had the accident and the memory loss, all the hair fell out and things just went downhill from there? |
| 19:43.09 | jeev | jameswf, i need a software method! |
| 19:43.35 | jameswf | Katty: besides it being made by grandstream... |
| 19:43.43 | Kobaz | hmmm |
| 19:43.47 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:43.51 | jaytee | rhino has some nice failover solutions in their hardware. Hey Digium!!! hint, hint!!! |
| 19:43.56 | Kobaz | what's the easyest way to see if a phone has a call on it |
| 19:43.59 | jeev | hm |
| 19:44.10 | Kobaz | i'm using ChanIsAvail, and using the 's' option, but it's still saying it's available |
| 19:44.13 | Katty | jameswf: sorry i don't have any experience with grandstream |
| 19:44.16 | Katty | jameswf: can you elaborate? |
| 19:44.28 | jaytee | Kobaz, SIP phone? |
| 19:44.32 | Kobaz | jaytee: yeah |
| 19:44.39 | Kobaz | <PROTECTED> |
| 19:44.47 | jameswf | ~grandstream |
| 19:44.48 | jbot | somebody said grandstream was the Yugo of VoIP hardware. Run. Run away now. |
| 19:44.57 | jaytee | sip show inuse? |
| 19:45.09 | Kobaz | jaytee: it needs to be dialplan level |
| 19:45.34 | jaytee | Kobaz, not sure |
| 19:45.40 | Katty | jameswf: kthx. |
| 19:45.43 | jameswf | maybe i will shoot over an email and have em send me a demo to see if they are any good... kinda need 2 people with video phones though |
| 19:45.49 | Carlos_PHX | I haven't used the high end Grandstreams, but the cheap and mid level ones were pretty crappy compared to others in the same range. |
| 19:45.58 | Katty | can anyone recommend a good video phone? |
| 19:46.02 | Katty | polycom doesn't make any |
| 19:46.13 | jameswf | try aastra? |
| 19:47.12 | Kobaz | jaytee: the problem with a failover thing like what rhino has, is you need a functioning computer to use it |
| 19:47.32 | Kobaz | jaytee: if the computer that the usb plug is connected to is down, you can't exactly switch anything |
| 19:48.13 | jaytee | Kobaz, yeah their solution is more for switching spans if one goes down to an alternate span in T1 or E1 |
| 19:49.33 | jameswf | no you need a power source no need for a pc |
| 19:50.00 | Katty | jameswf: looks like aastra doesn't have one |
| 19:50.06 | jameswf | any standard "usb" power supply works |
| 19:50.33 | jameswf | Katty: I like aastra and snom, I dont do video |
| 19:51.13 | Katty | that's fine |
| 19:51.16 | Katty | i just wanna tinker |
| 19:51.26 | *** part/#asterisk nido (i=nido@5ED105AD.cable.ziggo.nl) |
| 19:51.34 | jameswf | the failover has a power only option so it will flip on power failure |
| 19:51.48 | jeev | i guess i'm just gonna have to script something. |
| 19:51.53 | jeev | or i can nagios it! |
| 19:51.56 | jeev | if ping to gateway fails... |
| 19:52.05 | jeev | but how will nagios know what the gateway is? ahh i can set it as a host in hosts. |
| 20:00.22 | jtodd | quick survey: what ITSPs do people use here? I'm just looking for company names, not reviews. :-) Trying to see what companies are the most common. |
| 20:01.24 | russellb | ~itsplist |
| 20:01.37 | russellb | hrm ... I thought that was a factoid used frequently in here |
| 20:01.44 | russellb | fails |
| 20:02.09 | jaytee | ~itsplist-us |
| 20:02.10 | 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 |
| 20:02.17 | jaytee | ~itsplist-ca |
| 20:02.18 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 20:02.25 | jtodd | That's not what I asked, though. My question is trying to determine a coma common list of carriers used by participants who happen to be on the channel right now, not all possible ITSPs. |
| 20:02.59 | jtodd | I have no idea why the word "coma" appeared in there. My key. My keystrokes are about 10 seconds ahead of what this IRC client is printing out. |
| 20:03.34 | jeev | i just heard a girl from over there complaining to the business owner |
| 20:03.36 | jeev | that DTMF doesn't work |
| 20:03.37 | jeev | but it does |
| 20:03.41 | jeev | she just likes using the nortel phones |
| 20:03.42 | jeev | iT PSISES ME OFF |
| 20:03.49 | jaytee | jtodd, lotta people use viatalk and broadvoice. teliax too. haven't heard much about the others |
| 20:05.23 | [TK]D-Fender | jtodd: I hear mention of all of them |
| 20:05.39 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 20:05.55 | jtodd | ok, thanks. I have teliax on the list and broadvoice. I use binfone, and of course there's nufone. |
| 20:05.58 | jaytee | of the 4 people I've moved from a Nortel 3903 to a Polycom 550 I haven't had a single complaint |
| 20:06.08 | [TK]D-Fender | jtodd: and I maintain those jbot info-lets |
| 20:06.14 | jeev | jaytee, the people here bitch like crazy, they dont like change |
| 20:06.17 | jtodd | OK, so who do _you_ use as your ITSP? :-) |
| 20:06.41 | [TK]D-Fender | jtodd: Personally i use my office which doesn't count. :) |
| 20:06.47 | [TK]D-Fender | (ab) |
| 20:07.15 | [TK]D-Fender | jtodd: Almost all the listed ones there are used by quite a lot of people who come through here... |
| 20:07.28 | Carlos_PHX | Heh, there are ITSPs in here too...like, er, me. |
| 20:07.28 | jaytee | considering our "value added reseller, AT&T" (try to contain your laughter) would charge us over 400 bucks a phone for the 3903 and I can get 550's for 220 bucks I say if someone doesn't like their 550 I'll give them their Nortel back minus the dialtone :-) |
| 20:07.52 | [TK]D-Fender | jtodd: I can tell you that the majority of Canadian requests go towards les.net, the better part of leftovers to unlimitel. |
| 20:08.00 | stencil | jtodd: I really like les.net he is always adding new features without increasing the price |
| 20:08.56 | jtodd | OK, that's good data... |
| 20:09.09 | jeev | shit man, i tested dtmf to the number for her 3 times in front of her and it works |
| 20:09.14 | jeev | she says at least ONCE a day, it ignores it. |
| 20:09.18 | jeev | WTF MAN, youc all it a thousand times |
| 20:09.19 | jeev | jesus christ |
| 20:09.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:10.48 | [TK]D-Fender | jtodd: Its as solid a feeling as can be offered for any market. Canadian DID's are a little harder come-by so the playing fienld is noticably reduced |
| 20:11.06 | jtodd | mail.app has lost its mind and is at 140% cpu, so pardon my slow responses. |
| 20:14.03 | *** join/#asterisk neurosys (n=neurosys@166.193.136.13) |
| 20:14.44 | neurosys | [TK]D-Fender: what's your bot keyword for providers again? |
| 20:15.01 | [TK]D-Fender | ~itsplist-us |
| 20:15.02 | 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 |
| 20:15.03 | [TK]D-Fender | ~itsplist-ca |
| 20:15.04 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 20:15.05 | [TK]D-Fender | ~itsplist-uk |
| 20:15.05 | jbot | methinks itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 20:15.11 | neurosys | [TK]D-Fender: Thx :) |
| 20:16.33 | Carlos_PHX | So speaking of providers, anyone have an opinion on Broadvox vs. Vitelity as a wholesale provider? |
| 20:17.05 | Carlos_PHX | We're looking to switch from our current one, and can't decide. |
| 20:18.43 | *** join/#asterisk telcohitman (n=telcohit@tn-76-5-147-175.dhcp.embarqhsd.net) |
| 20:21.34 | [TK]D-Fender | checkout time... later all |
| 20:28.48 | smelgin | (t) |
| 20:33.43 | pimpwell | I know this is against the rules, but we've implemented fonality and the original guy left the company who set it up. We are looking to just hire someone to setup menu's etc remotely. Please let me know, we are located in NY. |
| 20:36.23 | Qwell | ... |
| 20:36.45 | Qwell | If you know it's against the rules, why ask? |
| 20:37.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:37.28 | jeev | hey, so i'm trying to use monast, 5038 aint listening, it's set to listen, i'd have to restart asterisk? doesn't manager run by default ? |
| 20:37.42 | telcohitman | So on a different note :) Does anyone know where I can find sizing specifications for Asterisk like how many subscribers can be held on a system based on the size of the server used? |
| 20:38.24 | Kobaz | what's the easyest way to see if a phone has a call on it |
| 20:38.29 | Qwell | Kobaz: look at it |
| 20:38.30 | Kobaz | i'm using ChanIsAvail, and using the 's' option, but it's still saying it's available |
| 20:38.34 | Kobaz | heh |
| 20:38.54 | Kobaz | ChanIsAvail(SIP/${EXTEN}) |
| 20:39.03 | ajohnson | telcohitman: This may help: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 20:39.11 | telcohitman | awesome thank you |
| 20:39.31 | jeev | ahh, wasn't enabled |
| 20:40.09 | Kobaz | Qwell: it always returns 0 for that |
| 20:42.19 | telcohitman | that article gives me everything I needed....it is much appreciated |
| 20:44.45 | pimpwell | Qwell: without a phone system at my business , we dont make money and I dont get paid |
| 20:45.16 | pimpwell | sure I can go to craigslist and post jobs... but I thought it would be nice to give the current people who are interested in the subject to do some work. That's all./ |
| 20:45.18 | Qwell | pimpwell: Hate to say it, but that's not our problem. ;/ |
| 20:45.26 | Kobaz | heh |
| 20:45.33 | Qwell | We get countless people coming in here "omg, I'm gonna lose my job if this doesn't get fixed. HELP ME NOW!" |
| 20:45.36 | Qwell | no, it doesn't fly. |
| 20:45.46 | Kobaz | hehe |
| 20:45.46 | pimpwell | not me, Im helping an employee who is stuck in the position |
| 20:46.57 | pimpwell | please respect the fact at least that I tried to bring some business here even though not wanted |
| 20:47.05 | Carlos_PHX | Stuck in this position? http://aslowerpace.com/serendipity/uploads/cow3.jpg |
| 20:47.08 | pimpwell | I forgot you all do this for personal satisfaction and fame |
| 20:47.14 | pimpwell | not $ |
| 20:47.47 | telcohitman | I do it because I like the work.... but I get cash for it too |
| 20:47.50 | telcohitman | :) |
| 20:48.43 | Carlos_PHX | Most of us work with Asterisk itself, not a GUI, so it would be like learning something new. If you had a regular Asterisk server I'm sure plenty of us would be happy to take your money. |
| 20:49.07 | Carlos_PHX | The Fonality box just has too many...um...quirks to put it nicely. I've removed all the ones I had used/sold. |
| 20:58.52 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:01.33 | jaytee | Long live Engineer Tim!!! Down with Fonality!!! |
| 21:02.33 | telcohitman | :) |
| 21:03.24 | *** join/#asterisk cathya (n=cathy@unaffiliated/cathyal) |
| 21:03.27 | *** part/#asterisk cathya (n=cathy@unaffiliated/cathyal) |
| 21:03.30 | jaytee | yay! quittin time. be back later for more fun |
| 21:06.02 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
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| 21:07.06 | orkid | . |
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| 21:14.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 21:16.07 | *** part/#asterisk smelgin (n=smelgin@adsl190-28-129-120.epm.net.co) |
| 21:18.29 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
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| 21:22.58 | StephenF[W] | Does anyone else feel like Bandwidth.com is overpriced when compared to VoicePulse? |
| 21:23.21 | StephenF[W] | I dont understand where the extra value is, is their support better? |
| 21:23.35 | StephenF[W] | anyone use either Bandwidth.com or VoicePulse for a small Business? |
| 21:25.02 | *** join/#asterisk kisu (n=kkang@ip70-179-88-179.dc.dc.cox.net) |
| 21:25.35 | [TK]D-Fender | StephenF[W]: I've had clients who were happy with VoicePulse. |
| 21:26.01 | *** join/#asterisk Greek-Boy (n=email@41.222.89.114) |
| 21:26.31 | StephenF[W] | [TK]D-Fender: im using them at home and the quality seems great, Im thinking i will try them out first... |
| 21:26.50 | StephenF[W] | plus their server is right here in the same town as us |
| 21:27.25 | *** join/#asterisk FarrisG (n=FarrisG@pool-71-164-195-61.dllstx.fios.verizon.net) |
| 21:28.39 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
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| 21:29.27 | *** join/#asterisk chazz (n=chazz@nat/digium/x-3965495c6bd67f9b) |
| 21:29.39 | jameswf | computer is so bleading edge an emo kid may try to steal it and cut them selves |
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| 21:34.11 | Carlos_PHX | StephenF[W]: We tried Bandwidth. They seem to waste a lot of money on sales people and paperwork. The paperwork is huge and what drove us away. |
| 21:34.24 | Carlos_PHX | I don't believe, my opinion, that they offer a good value. |
| 21:34.38 | Carlos_PHX | I have no experience with VoicePulse. |
| 21:34.46 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 21:36.04 | Carlos_PHX | StephenF[W]: What kind of volume? |
| 21:36.46 | Carlos_PHX | Man, I'm having fun reading the old documentation talking about how many channels you might be able to run on a Pentium 233. |
| 21:36.50 | Carlos_PHX | Ah, the bad old days. |
| 21:38.17 | *** join/#asterisk JonScarlet (n=johnnyS@user-12hdna5.cable.mindspring.com) |
| 21:39.34 | JonScarlet | Is this the right channel to ask some newbie type questions? I'm trying to build a small 4 phone system here in my office. |
| 21:40.08 | JonScarlet | All IP phones btw, aastra 57i, and i have a lingo, a vonage, and two broadcom numbers to control |
| 21:41.57 | Carlos_PHX | It would help to know your question. Don't worry, if it's not interesting, everyone will ignore it. |
| 21:43.40 | JonScarlet | haha. ok. Well I got the ubuntu box up and running, did the basic asterisk install, but I need to start entering the SIP information and stuff... is there a web interface or anything? Or is it JUST modifying various conf files? |
| 21:44.20 | encode | JonScarlet: http://www.voip-info.org/wiki-Asterisk+GUI |
| 21:44.23 | JonScarlet | trying to get just one line up and running, and have it connect to the aastra 57i, then i'll do the rest |
| 21:44.28 | JonScarlet | ahhh, thank you encode |
| 21:44.38 | encode | take a look at the configuration managers section |
| 21:44.52 | encode | the most common is freepbx |
| 21:45.04 | JonScarlet | encode, thanks, very cool of you. |
| 21:45.16 | Carlos_PHX | You need to decide whether you want to learn Asterisk, or just run a box in the most simple way possible. |
| 21:45.30 | Carlos_PHX | If you're looking for simple, also take a look at Switchvox free. |
| 21:46.03 | JonScarlet | Oh, also how involved is it to setup the webvbox that is mentioned? Need to isntall databases and such? i'm already runnign mysql for a compeltely seperate ticketing system |
| 21:46.41 | JonScarlet | Carlos, for the time being, i'm looking to run a box the simplest way possible, and add on as time goes by |
| 21:49.14 | *** join/#asterisk rycar (n=rycar@206-15-91-103.static.twtelecom.net) |
| 21:49.14 | Carlos_PHX | The Switchvox ISO method is super fast/easy/stable. I like the GUI a lot. |
| 21:49.41 | Carlos_PHX | I don't know much about the other GUIs as far as setup, have never set one up. |
| 21:50.07 | [TK]D-Fender | JonScarlet: From a practical standpoint I would start with FreePBX if you really want to go the GUI route at all. SwitchVox may be better in a few ways, but people available to support you isn't one of them |
| 21:50.15 | JonScarlet | that wont replace my ubuntu distro right? just add in to my existing setup of asterisk? |
| 21:50.17 | [TK]D-Fender | JonScarlet: And NO GUI's are supported in this channel. |
| 21:50.50 | [TK]D-Fender | JonScarlet: And you can bet that 99% of whats on the GUI list that you linked noone here has ever heard of or used. |
| 21:52.16 | JonScarlet | [TK]D-Fender , Thanks for the heads up. The Webvbox item i mentioned was mentioned in the Asterisk O'rielly book(skimmed through it during my lunch break). |
| 21:53.23 | JonScarlet | Ok, here is a non-gui related questions but also simple. Where do I find the SIP account info I need to confiugre the asterisk server? I logged into my broadvoice account, but cannot find such info, not in thier support forms either. |
| 21:54.06 | [TK]D-Fender | JonScarlet: plenty of samples on configuring BV for * out there. Get googling |
| 21:54.38 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 21:55.17 | [TK]D-Fender | JonScarlet: http://www.broadvoice.com/support_install_byod_asterisk.html |
| 21:55.23 | [TK]D-Fender | JonScarlet: You clearly looked very hard |
| 21:55.47 | JonScarlet | hehe, gotcha. I was jsut kinda hopign it was as simple as gettign the info by logging into my broadvoice account on thier website |
| 21:56.21 | [TK]D-Fender | JonScarlet: Funny I don't use them and found it in under 1 minute. |
| 21:56.22 | Carlos_PHX | Some providers do have samples, some don't, some hide them, some are wrong.... Welcome to open source telephony. |
| 21:56.30 | JonScarlet | thanks [TK}D-Fender i'll spend more time reading and less asking. I'll be back with something more challenging. :) |
| 21:56.51 | Carlos_PHX | JonScarlet: Just ask him about Polycom vs. Linksys phones. |
| 21:57.12 | JonScarlet | what about Lingo accounts? from what I googled, a lot of the psot i come across are from 2006, can we do lingo accoutns through *? |
| 21:57.13 | [TK]D-Fender | JonScarlet: I would personally forego GUI's altogether and learn from scratch. |
| 21:57.19 | hesco | what files are installed in /etc/asterisk if I don't make samples. Is it a clean slate? Or is there a skeleton to start with? |
| 21:57.42 | [TK]D-Fender | JonScarlet: as you've got the book, thats a must-have starting item. Here is another quick guide for inspiration which could help get you up and running really fast : |
| 21:57.46 | [TK]D-Fender | ~jerjerguide |
| 21:57.47 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:57.54 | [TK]D-Fender | hesco: none |
| 21:58.00 | hesco | thanks |
| 21:58.43 | *** join/#asterisk DarkRift (n=dark@206.167.240.40) |
| 21:58.45 | hesco | I noticed that my rebuild of asterisk is providing me far fewer options on the help menu this time. |
| 22:00.27 | Carlos_PHX | [TK]D-Fender: So there is no split on the Linksys phones but it's in development. The browser is simple but useful, RSS feeds scrolling with the ability to drill down a level on specific items. Nothing fancy but good for company info and news and such. |
| 22:02.07 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:02.17 | JonScarlet | That jerjerguide is pretty awesome. gonna print that out and read it on the way home |
| 22:02.21 | JonScarlet | thanks |
| 22:02.41 | Carlos_PHX | < hopes JonScarlet doesn't drive to work. |
| 22:02.46 | JonScarlet | Did anyoen know if Lingo and * are compatible btw? I never got a definite answer on that. |
| 22:02.49 | [TK]D-Fender | JonScarlet: really is for the best that you get to understand * at the lower levels for the little bits it takes to be useful |
| 22:03.21 | JonScarlet | yeah, it's all the new jargon and some new communication protocols i never had to deal with. It's my first venture into telephony |
| 22:03.33 | Carlos_PHX | The learning curve is steep but rewarding. |
| 22:03.54 | Carlos_PHX | I agree with learning the hard way, unless you plan to just make a simple one-off system and have no interest in doing more. |
| 22:03.56 | [TK]D-Fender | Carlos_PHX: if they've really picked up on their firmware they might be worth looking at again |
| 22:04.16 | Carlos_PHX | You might grab one and see. |
| 22:04.26 | Carlos_PHX | No way to know without trying both. |
| 22:04.51 | [TK]D-Fender | Carlos_PHX: Only the 962 could be worth it... |
| 22:05.07 | hesco | I finally figured out that a reload was not the same as a restart. Now things that were working are not. And I have a new set of issues to sort out. |
| 22:05.15 | Carlos_PHX | Personally I agree, but my customers rave about the 942 also over their existing 501s. |
| 22:05.41 | Carlos_PHX | The firmware is the same. |
| 22:05.46 | Carlos_PHX | But I think there's no browser on the 942. |
| 22:07.38 | [TK]D-Fender | Carlos_PHX: I use a 501 at home... very nice. |
| 22:08.03 | [TK]D-Fender | Carlos_PHX: I couldn't imagine swapping for any sub 962 model |
| 22:08.26 | Carlos_PHX | I've never had either on my own desk, so hard for me to say. |
| 22:08.41 | Carlos_PHX | But here's a customer quote... |
| 22:09.04 | Carlos_PHX | Got the new phone model, you bastard, now everyone wants these. How much are the old one's worth. |
| 22:11.54 | JonScarlet | thanks for everything guys. I have to go for now. thanks again |
| 22:11.54 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 22:12.10 | RMod | hey guys, having some issues with dtmf relay to a sonus |
| 22:14.39 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:15.30 | Carlos_PHX | What DTMF mode are you using? |
| 22:15.49 | RMod | rfc2833 |
| 22:16.08 | Carlos_PHX | All the way through? From the devices to the carrier? |
| 22:16.15 | Carlos_PHX | Have they verified they support that? |
| 22:19.21 | jeev | anyone use monast ? |
| 22:19.24 | RMod | yea |
| 22:19.51 | jeev | i think i did the installation the right way.. authentication and whatnot but when i point the browser, it just sits there.. connecting |
| 22:20.26 | RMod | if i set the devices to canreinvite it works, but thats not a good solution if they are behind nat |
| 22:22.11 | jeev | if you dont set them, whats it do? |
| 22:22.17 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 22:23.13 | RMod | dont set the dtmfmode= ? |
| 22:23.38 | jeev | are you talking to me ? |
| 22:25.07 | RMod | nope |
| 22:30.55 | jblack | Would anyone happen to know what's up with freechess.org these days? |
| 22:33.33 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
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| 22:42.01 | dennisharrison | trying to get followme to work : when it goes to dial an external number, I get this error |
| 22:42.02 | dennisharrison | [Oct 24 17:38:34] NOTICE[27976]: chan_local.c:617 local_alloc: No such extension/context 19853732317@DLPN_Default_DialPlan creating local channel |
| 22:42.02 | dennisharrison | [Oct 24 17:38:34] WARNING[27976]: app_followme.c:871 findmeexec: Unable to allocate a channel for Local/19853732317@DLPN_Default_DialPlan cause: Unknown |
| 22:44.04 | [TK]D-Fender | dennisharrison: Pastebvin your context in its entirety |
| 22:45.55 | dennisharrison | [TK]D-Fender, http://rafb.net/p/1EQBEw86.html is what I get from interactive while the call happens |
| 22:46.00 | StephenF[W] | my phone stopped being able to register over NAT, and im getting this: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission |
| 22:46.02 | dennisharrison | let me go and grab the context and paste it as well |
| 22:46.14 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 22:46.18 | *** join/#asterisk robevans (n=robevans@195-78-16-190.fibertel.com.ar) |
| 22:49.03 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 22:49.15 | dennisharrison | [TK]D-Fender, http://rafb.net/p/PBYmt224.html is my extensions.conf |
| 22:49.18 | dennisharrison | and last file coming up |
| 22:51.01 | [TK]D-Fender | dennisharrison: I only asked for 1 thing |
| 22:51.04 | [TK]D-Fender | dennisharrison: [Oct 24 17:38:34] NOTICE[27976]: chan_local.c:617 local_alloc: No such extension/context 19853732317@DLPN_Default_DialPlan creating local channel |
| 22:51.35 | [TK]D-Fender | dennisharrison: Now what line in http://rafb.net/p/PBYmt224.html Do you imagine this was supposed to match against? |
| 22:52.12 | dennisharrison | [TK]D-Fender, not there yet, sorry about that, putting up the config files for someone else as well, getting to where that context should be right now with the next url |
| 22:52.31 | [TK]D-Fender | dennisharrison: Everything we needed to know was in the dialplan... |
| 22:52.34 | dennisharrison | I suck at debugging asterisk, so I am shooting in the dark :) |
| 22:52.40 | [TK]D-Fender | dennisharrison: So tell me where you think it should ahve matched |
| 22:53.13 | dennisharrison | [TK]D-Fender, no idea |
| 22:53.14 | dennisharrison | sorry |
| 22:54.25 | [TK]D-Fender | dennisharrison: If you can't tell me what dialplan pattern in there should match that number then seem to lack the most important and basic part of * |
| 22:54.47 | [TK]D-Fender | dennisharrison: like a driver failing to understand the gas pedal. |
| 22:55.13 | [TK]D-Fender | dennisharrison: go break out the book and go learn how dialplan patterns work. |
| 22:55.25 | dennisharrison | [TK]D-Fender, tell me about it, I am using ast-gui to configure this, everything else seems to work. There are a lot of config files here, does the first part of the dialplan need to match as well, or just the part after the @ ? |
| 22:56.58 | [TK]D-Fender | dennisharrison: its looking for a match for 19853732317 in [DLPN_Default_DialPlan]. |
| 22:57.20 | *** join/#asterisk Penggu (n=me@156.39.233.220.exetel.com.au) |
| 22:57.32 | hardwire | anybody seen a USB ringdown phone? |
| 22:57.37 | hardwire | or a sip one? |
| 22:57.44 | dennisharrison | [TK]D-Fender, ahh, that is strange to me, can't I just tell it to 'dial' a number using a dialplan, and have that use the specific outbound calling rule for it? |
| 22:57.45 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d282318e8990babb) |
| 22:57.46 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 22:57.49 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 22:58.13 | [TK]D-Fender | dennisharrison: that is what its doing. |
| 22:58.21 | Penggu | hi all. anyone know of any simple 1xISDN-PRI<->SIP boxes "gateways" ? just so someone can connect ip phones to ISDN channels? |
| 22:58.32 | Penggu | cant find anything like it |
| 22:58.34 | [TK]D-Fender | dennisharrison: how do you normally dial an outbound number from your phone? |
| 22:58.35 | Penggu | simple ATA style |
| 22:58.41 | dennisharrison | [TK]D-Fender, ok, i'll go back and grep some more to see if I figure out wtf I need to look at |
| 23:00.03 | dennisharrison | [TK]D-Fender, it usually would look like this : Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/fBB14mgN23-1c29d3a0", "SIP/fBB14mgN23/19853732317") in new stack |
| 23:00.29 | Penggu | i could build an * machine with a pri+ethernet interface. i would think though that a little black box to do that would be more economical. |
| 23:01.14 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) [NETSPLIT VICTIM] |
| 23:01.14 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) [NETSPLIT VICTIM] |
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| 23:01.14 | *** join/#asterisk v4mp (n=Gary@82.118.111.250) [NETSPLIT VICTIM] |
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| 23:01.26 | [TK]D-Fender | dennisharrison: that is not what I asked you |
| 23:01.39 | *** join/#asterisk jameswf (n=james@166.128.221.174) |
| 23:01.56 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:04.06 | dennisharrison | [TK]D-Fender, I don't know how to answer you. Other then, there are no phones hooked up to this system what so ever. My guess would be that they send the number dialed to the dialplan associated with the extension, then the calling rule that best matches the input is executed, and that goes to whatever trunk is available on that dialplan... ? |
| 23:04.22 | [TK]D-Fender | dennisharrison: how do you normally dial an outbound number from your phone? <------- |
| 23:04.58 | dennisharrison | by stringing together enough digits to make something happen? :) |
| 23:05.21 | [TK]D-Fender | dennisharrison: ... give an example.. |
| 23:05.40 | dennisharrison | 19853732317 (press talk/enter) |
| 23:05.53 | dennisharrison | or sometimes 10 digit as well |
| 23:06.54 | [TK]D-Fender | dennisharrison: really... pastebin a normal call |
| 23:07.15 | dennisharrison | [TK]D-Fender, here comes |
| 23:07.27 | jeev | Fender, not all ITSP's are connected to PSTN, right ? some ITSP's just resell ITSP.. |
| 23:08.00 | dennisharrison | [TK]D-Fender, http://rafb.net/p/nNZZcl81.html |
| 23:08.28 | dennisharrison | the only other way I have of dialing out is with a voice menu |
| 23:09.02 | dennisharrison | and I don't see anything about default_dialplan there |
| 23:10.29 | [TK]D-Fender | dennisharrison: That doesn't look like you calling OUT |
| 23:10.34 | dennisharrison | it is though |
| 23:10.54 | dennisharrison | it is dialing the 985373 number after it answers the incoming |
| 23:10.57 | dennisharrison | and connects the call |
| 23:11.13 | dennisharrison | in that example, I guess you would consider it a forward |
| 23:11.26 | dennisharrison | wasting two channels, but the simplest example I could find. |
| 23:11.47 | [TK]D-Fender | dennisharrison: dial a REAL NUMBER |
| 23:12.00 | dennisharrison | it is a real number, 19853732317 is a real number |
| 23:12.37 | [TK]D-Fender | dennisharrison: not out to the real world it isn't |
| 23:12.37 | dennisharrison | I am really not trying to be thick man |
| 23:20.55 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:22.49 | hesco | my cdr_pgsql issue has been resolved. Thank you for your patience helping me to sort this out. |
| 23:24.01 | jaytee | hesco, was it the extra asterisk_* directories? |
| 23:24.11 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 23:24.28 | unpaidbill | g729 is effing sexy |
| 23:24.29 | unpaidbill | that is all |
| 23:24.45 | jaytee | unpaidbill, g729a or g729b? |
| 23:24.49 | unpaidbill | a |
| 23:24.56 | hesco | that may be a piece of it. I also think that actually restarting *, instead of only reloading its config files, after the make clean, re-install may have likely been of help. |
| 23:24.59 | jaytee | so b is a bow-wow? |
| 23:25.05 | unpaidbill | i havent used b |
| 23:25.20 | unpaidbill | i can't comment on it's attraction level |
| 23:26.04 | jaytee | hesco, yeah most times you're better off doing a complete restart of asterisk and when sql is concerned just doing a reboot is even better to start clean and restart all the services. |
| 23:26.22 | unpaidbill | this is pretty awesome though.. i have a wireless network connection |
| 23:26.34 | unpaidbill | with 4 lines running over it using g729, crystal clear audio |
| 23:26.51 | jaytee | unpaidbill, I only use ulaw cuz I likes my codecs the way I likes my women, phat and thick. |
| 23:26.55 | jtodd | G.729 is only sexy when dressed in IAX to keep it slim and overhead-free. Otherwise, it's bloated and not that much more attractive than it's smoother-sounding sister, G.711. |
| 23:27.10 | unpaidbill | if i had more than 30KB/s upstream i'd use it |
| 23:27.13 | hesco | I woke up this morning to find we had lost power here. When I fired the machine back up kiax had stopped making useful connections to the * at localhost. That will be the next issue to sort out. But likely not until after some food. |
| 23:27.19 | unpaidbill | ulaw that is |
| 23:27.48 | jaytee | 30KB/s? damn, man. you need to find a real ISP |
| 23:28.05 | unpaidbill | well the place is up on the side of a mountian in the middle of nowhere |
| 23:28.20 | jaytee | wow, what are you? some kind of Ranger Rick? |
| 23:28.36 | unpaidbill | nope, just someone in a crappy area |
| 23:29.01 | jaytee | Middle of Nowhere? is that in Alberta? Saskatchewan? |
| 23:29.12 | unpaidbill | santa claus is my neighbor |
| 23:29.18 | unpaidbill | his wife is very naughty. |
| 23:29.25 | jaytee | can you see Russia from your house? |
| 23:29.37 | unpaidbill | no but i see this head rearing in the sunset |
| 23:29.41 | unpaidbill | it's freaking me out |
| 23:29.54 | jaytee | hehe |
| 23:29.59 | unpaidbill | have any of you tried out the new asterisknow beta |
| 23:30.03 | unpaidbill | it looks pretty bitchin |
| 23:30.10 | unpaidbill | i'm so glad they changed the web interface |
| 23:30.16 | jaytee | I just loaded it last nite but haven't messed with it too much yet |
| 23:30.22 | unpaidbill | i'm about to burn the CD |
| 23:30.24 | *** join/#asterisk jstocks (n=jstocks@70.103.136.2) |
| 23:31.13 | jaytee | it loaded * 1.4 with DAHDI. I was under the impression it would give me a choice between 1.4 or 1.6 during the install but it didn't. |
| 23:32.01 | unpaidbill | i'll care about 1.6 more when it has t38 gateway support |
| 23:32.37 | unpaidbill | app_sendfax/app_receivefax do work well though |
| 23:32.48 | jaytee | i'll probably care more about AsteriskNOW period when it's support channel isn't like visiting the county morgue. |
| 23:33.29 | unpaidbill | hah |
| 23:33.36 | jstocks | Question: I have a asterisk system up and running that I have been using to play with. I have setup both confrence, and dictate. Both have worked fine, have been able to detect my DTMF digits and such. But when I have been testting with AGI and perl I can't seem to get the DTMF digits. What would I be missing? |
| 23:33.38 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 23:33.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:33.47 | *** join/#asterisk ManxPower (n=manxpowe@224.sub-70-223-66.myvzw.com) |
| 23:34.44 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:37.18 | riddlebox | hrmm I have setup an extension to work with my zap/2 channel, but when I call the wakeup.php app it says the extension is -1? |
| 23:37.38 | jaytee | http://www.dayofthejedi.com/articles/2008/04/top_10_sexually_suggestive.html |
| 23:37.40 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 23:38.14 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-20.lsanca.dsl-w.verizon.net) |
| 23:41.06 | riddlebox | hey jaytee |
| 23:41.15 | jaytee | hey |
| 23:41.29 | unpaidbill | i just installed sqlgrey on my postfix mail proxy.. i need to make an asterisk port |
| 23:42.20 | unpaidbill | this thing is great except for the 30m delays |
| 23:42.43 | jaytee | 30 minutes? |
| 23:42.53 | unpaidbill | yeah |
| 23:43.12 | jaytee | is sqlgrey like an alternative to spamassassin? |
| 23:43.18 | unpaidbill | no |
| 23:43.26 | unpaidbill | it would work in conjunction with it |
| 23:43.47 | unpaidbill | sqlgrey defers the first attempt a remote mail server makes to deliver a message forcing it to requeue it |
| 23:44.12 | unpaidbill | the theory is that most spam is sent from MTAs that dont conform to mail standards and will only try to deliver the message once |
| 23:44.36 | unpaidbill | or some such nonsense |
| 23:45.05 | unpaidbill | i bet it would work great for telemarketers |
| 23:45.20 | unpaidbill | 'This number is no longer in service' message the first time they call |
| 23:45.22 | unpaidbill | haha |
| 23:45.34 | unpaidbill | ahhh im totally doing this. i wonder how long till i get fired |
| 23:45.57 | jaytee | at least 30 minutes :-) |
| 23:46.58 | riddlebox | not until a boss tries to call in |
| 23:47.45 | unpaidbill | i shook my bosses hand earlier today and i think i squeezed it too hard.. i heard and felt his bones cracking |
| 23:48.03 | unpaidbill | probably get canned for that too |
| 23:48.35 | riddlebox | then do it, make it fun anyway |
| 23:48.52 | jaytee | I'd rather run my boss down in the parking lot and then back over him :-) more satisfying |
| 23:49.18 | unpaidbill | you must really hate work |
| 23:49.37 | riddlebox | wow, want to work for us haha |
| 23:50.27 | unpaidbill | so far sqlgrey has prevented 485/527 messages from being delivered |
| 23:50.41 | unpaidbill | then spamassassin picked up 35 more |
| 23:50.45 | jaytee | no, actually I love what I do and the organization is cool but the pay sucks cuz we're a non-profit and my boss is both an idiot and an arrogant ass. very, very difficult to work with. He knows next to nothing but he's ALWAYS right. If you make him look stupid in front of someone else you'll never live it down. |
| 23:50.50 | unpaidbill | and 7 were actually not spam and delivered |
| 23:50.50 | unpaidbill | wow |
| 23:51.18 | unpaidbill | hah, sounds like a fun guy |
| 23:52.40 | jaytee | yeah, I have 92 hours of paid time off left I have to use by the end of the year. I called in a week ago feeling like crap and called me back and lectured me with bullshit threats about having to note it in my review. I've been out sick 2 days this year. |
| 23:53.20 | unpaidbill | haha |
| 23:54.02 | jaytee | he sets a higher standard for everyone in his department than the rest of the company sets and yet he then goes and does the same crap that he'd count against us. |
| 23:54.16 | Qwell | jaytee: you just need to make him look stupid in front of the right people |
| 23:55.09 | jaytee | Qwell, I've thought of that but I need an exit strategy that doesn't leave me unemployed and living in a cardboard box under a highway overpass. |
| 23:55.18 | hesco | if I invoke the console at localhost as: sudo asterisk -vvvvvvvvvvgcr and attempt to run a call through that server from kiax, and I see no evidence of it on the console, that's a pretty safe bet my call isn't reaching *, right? |
| 23:55.26 | unpaidbill | jaytee: sugar momma. EOT |
| 23:55.27 | jaytee | cuz he's a fundamentalist and he's not the forgiving kind |
| 23:56.02 | jaytee | unpaidbill, thanks for the suggestion. I'll put an ad in Craigslist |
| 23:56.38 | unpaidbill | no problem, i'm sure you'll have really good luck |
| 23:57.10 | jaytee | yeah, I'll get all these photos from fat guys saying "Would you consider a sugar daddy instead?" |
| 23:57.50 | *** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod) |
| 23:57.54 | unpaidbill | 'sugar daddy needs JO buddy no gay stuff, also have TE110P for pics' |
| 23:58.02 | jaytee | hahaha |
| 23:59.05 | unpaidbill | opal/t38modem is the worst thing in the world |
| 23:59.17 | russellb | OH |
| 23:59.20 | unpaidbill | i wish asterisk supported t38 h323 passthrough |
| 23:59.29 | unpaidbill | is it in the works |
| 23:59.37 | unpaidbill | i'll send you a pallet of dr pepper |
| 23:59.41 | jaytee | I remember the first time I was playing with NetMeeting and a webcam back in the late 90's and clicked on a link for Hot Sheila Nekkid. 300 lb fat man masturbating. took months of therapy and heavy drinking to rid myself of that haunting image. |
