| 00:04.57 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:13.49 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 00:16.06 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 00:22.38 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:26.51 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 00:29.18 | redondos | Any word on my Music on hold problem? |
| 00:29.29 | redondos | Does asterisk cache the audio streams of the files in the MoH dir? |
| 00:32.28 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 00:38.29 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
| 00:42.07 | *** join/#asterisk reth (n=reth@hjaerterdam.csbnet.se) |
| 00:56.37 | De_Mon | redondos linux does |
| 00:57.05 | redondos | Ok... Do you have any tips for fixing my problema> |
| 00:57.11 | redondos | s/problema>/problem?/ |
| 00:57.55 | De_Mon | what is playing the moh? |
| 00:57.58 | redondos | Music on hold plays for half a second, then it stops. No error log, no nothing. |
| 00:58.19 | redondos | <PROTECTED> |
| 00:58.19 | redondos | <PROTECTED> |
| 00:58.21 | De_Mon | (if you say asterisk i'm going to smack you) |
| 00:58.28 | redondos | nope, mpg123 |
| 00:58.37 | redondos | BTW, MP3Player() works. |
| 00:58.45 | De_Mon | does mpg123 work ok by itself? |
| 00:58.49 | dlynes_home | De_Mon: what's wrong with native mode? |
| 00:58.53 | redondos | yes |
| 01:00.09 | *** join/#asterisk shido6 (n=shido7@d38-45-81.commercial1.cgocable.net) |
| 01:01.07 | [TK]D-Fender | redondos : What happened to priority #2 ? |
| 01:01.33 | [TK]D-Fender | redondos : I see #1 and #3.... no #2 |
| 01:05.17 | De_Mon | dlynes_home nothing, but.. nobody uses it? |
| 01:06.06 | *** join/#asterisk bigmac4444 (n=mtur2848@CPE-60-231-112-137.qld.bigpond.net.au) |
| 01:06.14 | bigmac4444 | h iguys |
| 01:06.31 | dlynes_home | De_Mon: I do |
| 01:06.48 | dlynes_home | De_Mon: why not reduce the load on the server as much as possible, if you can? |
| 01:07.26 | [TK]D-Fender | De_Mon : I do, exclusively. |
| 01:07.33 | bigmac4444 | question... |
| 01:07.40 | [TK]D-Fender | answer... |
| 01:07.44 | bigmac4444 | May 15 10:46:44 NOTICE[15976]: chan_sip.c:10319 handle_request_invite: Failed to authenticate user |
| 01:07.57 | [TK]D-Fender | bigmac4444 : bad user/pass |
| 01:08.01 | dlynes_home | bigmac4444: it means it couldn't authenticate |
| 01:08.35 | bigmac4444 | cant be, spent 2 days, password is correct, i can log it on from other machines on a different ISP. |
| 01:09.06 | bigmac4444 | The person gets the same error when on any other account too.. registers fine, just wont dial out |
| 01:09.27 | bigmac4444 | 403 forbidden. its very wierd |
| 01:09.54 | Flauto | bigmac, what service is that |
| 01:09.58 | bigmac4444 | secret correct on server and user spc |
| 01:10.07 | bigmac4444 | service? |
| 01:10.18 | De_Mon | i'll stop digging my hole deeper now :x |
| 01:12.47 | bigmac4444 | this has got me stumped. Absolutely nothing wrong with secret on server or end users PC. Tested his account my end, works fine. |
| 01:12.51 | *** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net) |
| 01:13.00 | *** part/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
| 01:13.19 | NoRemorse | hello all, do I need any extra modules to set up a conference extension on asterisk please? or is it just a straight extensions.conf mod? |
| 01:13.35 | bigmac4444 | zaptel? |
| 01:13.39 | NoRemorse | nope |
| 01:13.48 | NoRemorse | all SIP and IAX trun king |
| 01:14.31 | De_Mon | you need zaptel to setup a meetme conference |
| 01:14.39 | NoRemorse | just the module or a card? |
| 01:14.45 | De_Mon | there are alternatives, voip-info.org |
| 01:15.06 | NoRemorse | yeha looked tehre looks messy, just wanted to check first, if I do have zaptel is it just a config? |
| 01:15.30 | bigmac4444 | meetme.conf and extension |
| 01:16.01 | NoRemorse | ah cool, I do have aptel and ztdummy modules laoded, i was playing with an E1 card awhile abck I just dont use it |
| 01:16.06 | redondos | [TK]D-Fender: Sorry, that was just an example. Here's how it really looks like: |
| 01:16.06 | NoRemorse | thanks |
| 01:16.11 | redondos | <PROTECTED> |
| 01:16.14 | redondos | <PROTECTED> |
| 01:16.16 | redondos | <PROTECTED> |
| 01:16.20 | redondos | (is a 3-line paste allowed here?) |
| 01:16.32 | redondos | Sorry if it isn't. |
| 01:16.41 | NoRemorse | pastebin.ca is your friend |
| 01:16.51 | [TK]D-Fender | redondos : pastebin your musiconhold.conf and apstebin the call attempt |
| 01:17.12 | redondos | great. |
| 01:18.11 | redondos | NoRemorse: Unfortunately, some router in between me and pastebin.ca's has banned my ISP since December. I like .ca's template, but well. I use it with tor, but that's slower. :) |
| 01:19.21 | dlynes_home | redondos: you can use pastebin.ca as well |
| 01:19.49 | dlynes_home | redondos: or try: http://erxz.com/pb/ or http://dragon.cbi.tamucc.edu:8080 |
| 01:20.05 | NoRemorse | pastebin.com? |
| 01:20.13 | dlynes_home | erm pastebin.com yeah |
| 01:20.13 | NoRemorse | or is it the same crowd |
| 01:20.32 | NoRemorse | anyways, yeah meetme.conf isn't being loaded is there anywhere I should tell * to load it? |
| 01:20.46 | redondos | [TK]D-Fender: http://pastebin.com/717918 |
| 01:21.05 | NoRemorse | ah ignore that |
| 01:21.20 | redondos | That's what I said, pastebin.com is what I use, but I like .ca's theme better. |
| 01:21.47 | Qwell | redondos: Do you use ipv6? |
| 01:21.55 | Qwell | redondos: If so, try using ipv4.pastebin.ca |
| 01:22.19 | Qwell | if it's flakey (on either end), it won't resolve, so, I had him add the ipv4. one :p |
| 01:22.33 | redondos | Qwell: Right, no, it's router banning my IP range. It's a known problem. |
| 01:23.10 | NoRemorse | thanks guys worked a treat |
| 01:23.14 | redondos | This IP subnet my address is in, it appears my ISP bought it cheap because it belonged to an ISP full of spammers. |
| 01:23.21 | Qwell | heh |
| 01:23.32 | redondos | At first, I couldn't even access any wikimedia sites, including wikipedia. :( |
| 01:23.43 | redondos | Good thing things like Tor exist. |
| 01:23.59 | [TK]D-Fender | redondos : and the call attempt? |
| 01:24.14 | redondos | Oh, yes, just a second. |
| 01:26.13 | redondos | [TK]D-Fender: http://pastebin.com/717922 |
| 01:27.56 | [TK]D-Fender | redondos : And that part where it fails? |
| 01:28.09 | [TK]D-Fender | redondos : I'd like to see the error. |
| 01:28.30 | redondos | The first time I stated the problem I said that I saw no errors. But the music just doesn't play. |
| 01:28.55 | [TK]D-Fender | redondos : make sure they have no ID3 tags <- |
| 01:28.55 | redondos | It's like it started playing, but before you know it (400ms?) it stops. |
| 01:29.00 | redondos | Oh, ok. |
| 01:29.47 | redondos | [TK]D-Fender: An uncompressed (and of course untagged) PCM stream presents the same problem. |
| 01:30.35 | [TK]D-Fender | Sure all you have in there are good MP3's? no extra garbage? |
| 01:30.58 | redondos | Let me recheck. |
| 01:31.47 | redondos | Yes, in fact it's empty with just one mp3 or one wav file, as I am testing it. |
| 01:33.02 | redondos | Has it got anything to do with the `mode' option? |
| 01:35.27 | Qwell | "rtfm nub" |
| 01:36.01 | DarthClue | and where is tfm you anal expert? |
| 01:36.24 | [TK]D-Fender | DarthClue : Sorry, you have exceeded your question quota, please call again! |
| 01:36.38 | Qwell | DarthClue: ubuntu.org? :p |
| 01:36.42 | DarthClue | yep, sounds about like what I got in #Ubuntu |
| 01:36.57 | orlok | guide to do what? |
| 01:37.03 | orlok | its all linux when you get down to it. |
| 01:37.13 | [TK]D-Fender | DarthClue : Ubuntu is basically Debian missing all the good stuff you need to compile *, and twisting the concept of having a root account. Asking for trouble if you ask me. Go straight Debian. |
| 01:37.20 | orlok | but ubuntu isnt really made for command line amdin users |
| 01:37.26 | DarthClue | no help there, i had to scour the endless masses of junk on google that simply referenced the GUI |
| 01:37.47 | orlok | DarthClue: there is a reason for that.. thats what it is designed for |
| 01:38.49 | orlok | DarthClue: i use it on my desktop |
| 01:38.53 | orlok | it "just works" |
| 01:39.08 | Nugget | Ubuntu is an ancient african word that means "I couldn't figure out how to install Debian" |
| 01:39.25 | orlok | i have never needed a manual for anything ubuntu specific, only the usual RTFM for whatever software i'm dealing with |
| 01:39.31 | DarthClue | [TK]D-Fender, twisted root eh? Debian is worth a try then. So does doze and I have more games than ubuntu could ever imagine handling. |
| 01:39.40 | orlok | so i can bang my head against all the rpm based servers i run |
| 01:39.50 | DarthClue | Nugget, i'm starting to see that. |
| 01:43.54 | *** join/#asterisk zparta (i=zparta@ipv6.hispan.se) |
| 01:46.48 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 01:56.15 | asterboy | howdy |
| 01:58.11 | [TK]D-Fender | doody |
| 02:03.34 | Az_au | what's the best way to log iax debug information to a file? |
| 02:06.02 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 02:15.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:15.57 | asterboy | trying to get this phone to work with * |
| 02:15.59 | asterboy | http://cgi.ebay.com/ANTIQUE-DEVEAU-OAK-WALL-TELEPHONE-PHONE-1899-RARE_W0QQitemZ6629900788QQcategoryZ38037QQrdZ1QQcmdZViewItem |
| 02:16.56 | asterboy | thought they supported sip |
| 02:17.19 | [TK]D-Fender | asterboy : TDM card and "immediate=yes". I'd disable the crank as not to fry the port and you're good to go :) |
| 02:17.21 | SwK | asterboy: just put it on a fxs zap set for imediate start |
| 02:17.28 | [TK]D-Fender | :D |
| 02:17.31 | SwK | haha |
| 02:17.48 | [TK]D-Fender | SwK :*I* remembered the A/C crank so pftt! |
| 02:17.55 | asterboy | lol |
| 02:18.04 | SwK | I was trying to give him a little more credit :P |
| 02:18.22 | [TK]D-Fender | SwK : I got a request to cut the card ;) |
| 02:18.37 | SwK | cut what card? |
| 02:18.56 | [TK]D-Fender | SwK : His credit card ;) |
| 02:18.56 | SwK | oh duh |
| 02:19.10 | SwK | damn i'm a little slow tonight |
| 02:19.26 | asterboy | ok, forget that....I want this on my * box: |
| 02:19.27 | [TK]D-Fender | Yeah! This is stream of consciousness, not Field & Stream! |
| 02:19.27 | asterboy | http://cgi.ebay.com/Morse-Code-Wood-Bronze-England-WWII-IDF-40s-Rare_W0QQitemZ6627506573QQcategoryZ38037QQrdZ1QQcmdZViewItem |
| 02:19.54 | [TK]D-Fender | asterboy : Gimme a few... I'll come up with something :D |
| 02:20.13 | asterboy | lol, would be funny to see in action. |
| 02:20.32 | [TK]D-Fender | asterboy : I'm sure we could make an open-loop zaptel timed interface for it :) |
| 02:20.59 | asterboy | actually, wonder if they interfaced with them when they were bringing in the bells |
| 02:21.18 | asterboy | back in the day when morse was being replaced |
| 02:22.14 | orlok | telegraph would have been seperate altogether woulent it? |
| 02:22.56 | asterboy | that's what I'm wondering...they may have ran on the same wire for a while |
| 02:24.01 | *** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net) |
| 02:24.07 | asterboy | lol |
| 02:24.28 | asterboy | well look who joined! |
| 02:24.42 | NoRemorse | hi all, does the grandstream 2000 handle multiple "lines" in the same manner as say x-lite? ie are they additional lines on the same sip account or does each "line" on the grandstream require separate accounts? |
| 02:24.57 | NoRemorse | ? |
| 02:25.40 | [TK]D-Fender | NoRemorse : I believe they can work either way |
| 02:25.44 | asterboy | gxp can have multiple lines |
| 02:25.52 | asterboy | same account |
| 02:26.13 | asterboy | all lights ring at the same time |
| 02:26.50 | [TK]D-Fender | asterboy : EW |
| 02:27.03 | asterboy | ya, it's silly |
| 02:27.22 | NoRemorse | ah cool, so say an incoming call to your account goes to line 2 if u have line 1 in use (like on x-lite) cool |
| 02:27.31 | asterboy | yep |
| 02:27.39 | NoRemorse | awesome. |
| 02:27.57 | asterboy | make sure you upgrade the firmware before you touch those phones. |
| 02:28.02 | NoRemorse | lol ok |
| 02:28.14 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 02:28.38 | [TK]D-Fender | asterboy : Make sure you have you sanity checked before buying that crap ;) |
| 02:29.00 | asterboy | ya, get ready to dye your hair black. |
| 02:29.20 | mitcheloc | did remorese buy that morse code thing? because i don't think they have released any new firmware revisions for a while now... |
| 02:29.47 | asterboy | oh, ya...there was a major release just recently. |
| 02:29.54 | mitcheloc | oh? does it do sip yet? |
| 02:29.59 | asterboy | lol |
| 02:30.32 | DarthClue | i didn't mind the gxp2000 but it couldn't compare to the warm fuzzies i got from the ip501s |
| 02:30.52 | asterboy | nothing really to compare there |
| 02:31.24 | orlok | NoRemorse: multiple lines |
| 02:31.30 | [TK]D-Fender | DarthClue : Try a 601 w/ Att module then... better goodness |
| 02:31.36 | orlok | asterboy: i would not call NoRemorse a goth :) |
| 02:31.54 | asterboy | goth? |
| 02:32.07 | DarthClue | might be able to get my hands on of those by christmas...maybe |
| 02:32.30 | orlok | NoRemorse: i got a pair of gxp's here, they handle multiple line |
| 02:32.31 | orlok | lines |
| 02:32.45 | orlok | NoRemorse: the sipuras are much nicer phones |
| 02:32.45 | mitcheloc | <--doesn't need a 601 whatchyamacallit, i'll stick to my morse code thingy =) |
| 02:32.52 | orlok | even the handsets on the gxp's are naf |
| 02:32.55 | asterboy | Check out this new handset: |
| 02:32.58 | asterboy | http://cgi.ebay.com/Royal-Army-Ordnance-Corps-R-A-O-C-MORSE-KEY-IMMACULATE_W0QQitemZ9722979761QQcategoryZ48706QQrdZ1QQcmdZViewItem |
| 02:33.33 | catch23 | does that work with asterisk? |
| 02:33.42 | mitcheloc | omg, that's perfect *buys the morse key thing so he will have someone else to "morse" with!!* |
| 02:33.52 | catch23 | i didn't see a serial port anywhere on that handset |
| 02:34.06 | mitcheloc | yes, but only the void asterisk version |
| 02:34.15 | mitcheloc | ^ voice over inputted dotts/dashes |
| 02:34.24 | asterboy | Need to write a SIP interface |
| 02:34.29 | asterboy | for it |
| 02:34.38 | mitcheloc | nah void works just fine |
| 02:35.14 | asterboy | might need a relay |
| 02:35.15 | asterboy | http://cgi.ebay.com/Morse-Code-Key-Relay_W0QQitemZ9724582466QQcategoryZ1502QQrdZ1QQcmdZViewItem |
| 02:35.17 | robl^ | no no no! must run morse over chan_sccp so we can see when the button is up or down remotely |
| 02:35.45 | asterboy | lol, that would be a cool module |
| 02:35.59 | mitcheloc | *patents it* |
| 02:37.14 | robl^ | mitcheloc: you are infringing on my patent. I have a patent for the process of patenting things. ;-) |
| 02:37.21 | asterboy | lol |
| 02:37.34 | mitcheloc | so sue me? ba-da-bing! |
| 02:37.56 | robl^ | mitcheloc: I will as soon as we finish with RIM, Apple, and Microsoft |
| 02:38.17 | DarthClue | robl^, but I have already patented the process of patenting the patent process and also hold the patent for the word patent |
| 02:38.42 | robl^ | actually that would be a trademark on the word... not a pattent.. ;-) |
| 02:39.29 | DarthClue | no, i have patented the process of the word patent, all methods of creating the word including written and electronic have been patented |
| 02:39.56 | *** join/#asterisk sphing (n=sphing@n128-227-50-235.xlate.ufl.edu) |
| 02:40.07 | mitcheloc | good, nobody is allowed to think until i say so |
| 02:40.10 | DarthClue | the patent itself is very long and boring and really just describes the motion required to write out the letters but it serves it's purpose |
| 02:40.16 | robl^ | ok.. I spend too much time at a law firm.. I am starting to sound like the attorneys that complain to me when they can't find a record in a database |
| 02:40.19 | mitcheloc | and since i'm nice, your welcome to think, but be careful!!! i can take it away at any time ;) |
| 02:40.23 | sphing | has anyone gotten asterisk and packet8 working? |
| 02:40.54 | DarthClue | robl^, we'll forgive you. i've been there myself before. |
| 02:42.23 | robl^ | *shiver* yeah. I have to go back to an office filled with them tomorrow |
| 02:43.56 | DarthClue | i work at a five letter gov agency that is so paranoid it maintains duplicate copies of the same non-secure publicly available information so that it can be accessed over multiple secure and insecure channels. |
| 02:44.25 | mitcheloc | *confused* |
| 02:45.48 | robl^ | DarthClue: hah! Sounds very musch like a law firm. they create PDFs of documents with passwords (for security), but tend to write passwords with felt tips of CD-Rs filled with PDFs.. |
| 02:46.33 | robl^ | so all you need is to read the handwritten password on the CD-R to open the docs.. |
| 02:46.50 | DarthClue | robl^, yeah, something like that. of course, the group i'm working in has something like 6 people doing the job of 1 person in the private sector |
| 02:46.51 | Olobola | So is Nufone toast? What is your opinion? |
| 02:47.36 | orlok | Ahh, you work for govco? |
| 02:47.37 | robl^ | NuFon is more like eggs over easy. Temporary hiatus.. not quite finished yet. |
| 02:48.25 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:49.09 | robl^ | I still can dial out with NuFone. Inbound.. well.. I am.. umm.. holding my breath.. glad I have back up providers ;-) |
| 02:49.52 | paolob | Hi guys! I'm trying to register a zoom 5801, but asterisk keeps giving the answer: "chan_sip.c:10886 handle_request_register: Registration from 'sip:monjas@10.152.58.1;tag=as0378706e' failed for '10.152.58.13' - Wrong password"... The password is the same in sip.conf and in the zoom ATA... How can I see the password the ATA is sending asterisk? sip debug doesn't show it |
| 02:51.03 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 02:51.33 | ManxPower | If I have a street address and zip/city/state can anyone here give me a list a NPA-NXX's for the serving CO? |
| 02:51.35 | robl^ | MP! |
| 02:52.05 | ManxPower | Rob L Carrot! |
| 02:52.18 | robl^ | MP: that's not as easy as it once was.. due to number portability |
| 02:53.03 | ManxPower | Yeah, I just need some general stuff. |
| 02:53.27 | ManxPower | Can't get DSL qualification without a phone number, can't decide if we want a phone at that location until we know of there's DSL |
| 02:53.38 | ManxPower | Chicken, meet Egg. |
| 02:53.45 | robl^ | hehehe |
| 02:53.56 | [TK]D-Fender | ManxPower : Got a neighbour to test that with? |
| 02:54.05 | ManxPower | [TK]D-Fender, not moved in yet. |
| 02:54.16 | paolob | Guys, anyone could tell me how can I see the password asterisk receives from a sip device trying to register? thank you! |
| 02:54.30 | [TK]D-Fender | ManxPower : You don't have to. if there is a neighbouring unit, get their #, odds are you'll rate whatever they do |
| 02:54.33 | ManxPower | all broadbandreports.com wants is a street, zip, and npa-nxx |
| 02:54.35 | robl^ | yeah.. had that issue too. I have "naked DSL" at home.. no dialtone. they demanded a phone number for qualification. I gave the number for the apt leasing office |
| 02:54.37 | orlok | ManxPower: heh, its a PITA isnt it :0 |
| 02:54.49 | orlok | ManxPower: we have the same problem in aus |
| 02:54.51 | ManxPower | [TK]D-Fender, I'll be sure to tell him. |
| 02:55.13 | [TK]D-Fender | robl^ : Did just about the same.. found another Apt # in my building. I'm at a nice synch of 5mbps solid |
| 02:55.46 | robl^ | I really need to up mine. I still have 1.5/384 |
| 02:55.50 | pjchilds | paolab: the 407 requires auth from asterisk contains some stuff, which is then hashed with the password -- so you won't see the password in any tracing or logs... (its MD5 hashed...) |
| 02:56.10 | ManxPower | [TK]D-Fender, I know all this. That does not answer my question. |
| 02:56.32 | redondos | [TK]D-Fender: Thank you for your suggestion: removing the ID3 tags. That solved it. :) |
| 02:56.35 | ManxPower | since he currently lives 2 hours away from where he's moving to..... |
| 02:57.03 | ManxPower | so if I can get an NPA-NXX for the serving CO it will help with the prelim search for broadband \. |
| 02:57.09 | robl^ | ManxPower: try whit pages.com. I think you can search on an address |
| 02:57.33 | robl^ | it can give ya something nearby |
| 02:57.50 | [TK]D-Fender | redondos : ywc |
| 02:58.51 | robl^ | hrmm.. 6.0/768 is $115/mo |
| 02:59.18 | [TK]D-Fender | EEK |
| 03:00.03 | robl^ | http://speakeasy.net/home/compare/ |
| 03:00.44 | ManxPower | robl^, whitepages.com wants money for reverse address |
| 03:01.02 | websae | Speakeasy needs to up their upload bandwidth! |
| 03:01.02 | ManxPower | I hate all you people that are close to the CO. I hate you. |
| 03:01.08 | robl^ | when did they start that? hmmm.. never had to pay before |
| 03:01.34 | [TK]D-Fender | robl^ : Mine set me back $40 CDN / month on Dry.... |
| 03:01.41 | ManxPower | looks like there is no broadband available at his new place. |
| 03:02.36 | robl^ | ManxPower: Well you are in the middle of nowhere.. a hidden cult...er...exclusive housing complex |
| 03:02.53 | ManxPower | robl^, this is for one of my significant others who is moving to Tuscaloosa |
| 03:03.13 | ManxPower | Ya, know, the city where the largest university in the state is located. |
| 03:03.33 | ManxPower | And I like my cult, thankyouverymuch |
| 03:03.40 | robl^ | hrmm.. I ha 7690 ft from the CO.. how fast do ya think I can get? |
| 03:04.41 | CunningPike | ~seen dlynes_ |
| 03:04.51 | jbot | dlynes_ is currently on #asterisk (2d 6h 42m 39s). Has said a total of 143 messages. Is idling for 2d 3h 37m 54s, last said: 'achandra: and if not trunk, then oej's test-this-branch'. |
| 03:04.51 | robl^ | ManxPower: hrmm.. I wonder if he could tap into the university network??? I use to get dial up and later ISDN from my university.. (shows my age) |
| 03:04.51 | ManxPower | I dunno. I'm 11 miles from the CO. |
| 03:04.58 | ManxPower | oh, he can get dialup |
| 03:05.04 | CunningPike | ~seen dlynes_home |
| 03:05.06 | jbot | dlynes_home is currently on #asterisk (21h 27m 39s). Has said a total of 97 messages. Is idling for 1h 44m 53s, last said: 'erm pastebin.com yeah'. |
| 03:05.38 | robl^ | what about like sprint broadband wireless? my SO uses that.. its basically like a microwave dish and a cable modem |
| 03:05.56 | ManxPower | I thought they stopped offering that service. |
| 03:06.18 | robl^ | as long as he has clear line of site.. he's 21 miles from the tower and gets decent servoce |
| 03:06.30 | robl^ | they started offering it again.. at least in our area |
| 03:07.00 | ManxPower | robl^, URL? |
| 03:07.13 | ManxPower | sprint.com still wants a phone number |
| 03:07.14 | robl^ | http://www.sprintbroadband.com/ |
| 03:08.18 | robl^ | it has a bit of latencey.. and packets are in "bursts".. but he can do SIP/g729 no problem.. sounds crystal clear about 95% of the time |
| 03:09.17 | ManxPower | fount it. no service |
| 03:09.39 | robl^ | ugh!! |
| 03:10.22 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 03:11.13 | robl^ | what about like a cell phone card? EDGE/GPRS? |
| 03:13.08 | litage | i hooked up my analog cordless phone to a Grandstream ATA and there was a huge amount of background noise and static. when i made a call, the recipient could hear it too. is that normal when using a cordless phone with an ATA? |
| 03:13.50 | inv_Arp | litage: nope.. |
| 03:14.24 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:14.53 | brookshire | litage: it could be possible that your cordless phone is putting voltage back on the line.. i've had that problem before |
| 03:14.54 | ManxPower | robl^, the term you should think of is "flat broke" when thinking of options for him. |
| 03:15.17 | ManxPower | litage, only if you have wifi |
| 03:15.45 | brookshire | manx: and it's a 2.4 ghz phone |
| 03:15.46 | robl^ | ManxPower: ahhh! how about a large pringles can and some coax cable? |
| 03:16.18 | litage | brookshire: how can i determine if that's the case? |
| 03:16.27 | ManxPower | robl^, I was thinking of the spare 18" dish I had and steal someone's wifi 8-) |
| 03:16.36 | litage | ManxPower: the wireless capabilities on my router are disabled |
| 03:16.44 | brookshire | try a different phone |
| 03:16.54 | ManxPower | litage, that does not mean the radio is disabled. |
| 03:17.18 | robl^ | the problem is that is if he is so far from any CO, he won't be able to find a hot spot close enuff |
| 03:17.21 | litage | brookshire: i only have the one cordless phone. on my corded phone, it's crystal clear |
| 03:17.36 | litage | ManxPower: i've unscrewed and removed the wireless antennae from my router, too |
| 03:17.45 | ManxPower | robl^, which is WHY I wanted info on his serving co |
| 03:18.30 | robl^ | ManxPower: I understand the why.. just need to figure out the HOW ;-) |
| 03:18.55 | ManxPower | robl^, *nod* I'll have him try to scam a phone number from a neighbor when he's down there tomorrow |
| 03:19.25 | ManxPower | There's no way in hell I would give some random guy my phone number who claims he's moving next door |
| 03:19.35 | robl^ | try doing a search for like.. a McDonalds in his area.. |
| 03:20.04 | robl^ | maybe you can find one VERY close to his address |
| 03:21.15 | robl^ | also.. what about cable Internet? |
| 03:25.04 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 03:27.14 | *** join/#asterisk Altair256 (n=Altair25@mail.clccorp.com) |
| 03:32.47 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:43.59 | chiaBORG | can someone explain the purpose/context of phone.conf for me.. I understand it's a channel config file.. but it what context is it used? |
| 03:47.53 | chiaBORG | or asked another way.. why would I use phone.conf |
| 03:49.28 | robl^ | chiaBORG: if you have to ask, you don't need it. its for use with "Quicknet PhoneJack" cards. |
| 03:49.49 | robl^ | if you don't have one of those cards, just ignore the file |
| 03:50.48 | chiaBORG | robl^ : that what I though (I didnt need it).. Im trying to gain understanding of the different config files.. how did you learn what it was for? |
| 03:51.13 | robl^ | chiaBORG: I asked the same question 2 years ago -) |
| 03:51.34 | chiaBORG | Im always late to the show ;) |
| 03:52.32 | robl^ | hehe. that's not a problem. ;-) |
| 03:52.46 | chiaBORG | robl^ : thanks for the input! |
| 03:52.52 | orlok | Can somebody look at this pastebin for me and tell me why incoming calls might not be working? |
| 03:52.57 | orlok | http://pastebin.ca/55908 |
| 03:53.37 | *** join/#asterisk bmg505 (n=leon@c1-135-2.rndf.isadsl.co.za) |
| 03:54.42 | robl^ | chiaBORG: several of the conf files are rarely used and can safely be ignored by 99% of the asterisk installs |
| 03:55.25 | chiaBORG | robl^ : I'm starting to notice this.. |
| 03:55.56 | [TK]D-Fender | orlok : SIP/2.0 407 Proxy Authentication Required |
| 03:56.12 | [TK]D-Fender | orlok : Probably doesn't like your use of "asterisk" as a realm. |
| 03:56.18 | [TK]D-Fender | orlok : check with your ITSP |
| 03:58.35 | orlok | [TK]D-Fender: asterisk as a realm? wheres that? |
| 03:58.49 | orlok | ahh |
| 03:58.52 | orlok | digest-realm |
| 03:58.59 | [TK]D-Fender | orlok : indeed |
| 03:59.40 | orlok | Where is that set? |
| 03:59.59 | orlok | ahh |
| 04:00.00 | orlok | realm= |
| 04:01.05 | orlok | hmm, does the verbosity and debug states get lost after a reload? |
| 04:01.48 | orlok | [TK]D-Fender: doesnt the 407 happen every time as part of the handshaking? |
| 04:02.07 | [TK]D-Fender | orlok : if it stops there I think thats where it implies that it cares. |
| 04:03.00 | *** join/#asterisk [bsd] (n=bsd@59.144.245.188) |
| 04:03.10 | robl^ | nah.. they really don't care.. they are just playing a practical joke. ;-) |
| 04:04.02 | orlok | [TK]D-Fender: it doesnt stop there though |
| 04:04.49 | [bsd] | hi, friends! |
| 04:05.38 | [bsd] | How do I compile h323 chan/support on a debian based distro? |
| 04:05.46 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:06.05 | [TK]D-Fender | [bsd] : get the Asterisk-Addons package from digium's FTP |
| 04:06.34 | orlok | hmm, is there a way i can see if the inbound call is actually reaching a context? |
| 04:07.29 | [TK]D-Fender | orlok : no-op it otherwise you'll just see the error if there's no match |
| 04:07.51 | [bsd] | [TK]D-Fender: I do have the latest from SVN but none is compiling i mean not even NuPhone included in Asterisk |
| 04:08.05 | orlok | ahh i can see the contaxt and whats being displayed when i dial the echo test |
| 04:09.27 | [bsd] | [TK]D-Fender: I also attempted to get the InAccessNetworks version |
| 04:09.49 | [TK]D-Fender | [bsd] .... huh?! |
| 04:10.07 | orlok | [TK]D-Fender>: no-op? |
| 04:10.41 | [TK]D-Fender | orlok : exten => _x.,1,NoOp(I gues something caming in on ${EXTEN}) |
| 04:13.11 | [TK]D-Fender | ok, well thats it for me tonight, good luck, and later all.... |
| 04:19.22 | *** join/#asterisk EastWolf (i=eastwolf@61.48.130.223) |
| 04:19.49 | *** part/#asterisk [bsd] (n=bsd@59.144.245.188) |
| 04:20.02 | EastWolf | hi baby |
| 04:25.48 | catch23 | i'm not a baby! I'm a fully grown dinosaur! |
| 04:33.30 | websae | sleep |
| 04:33.35 | websae | everyone is asleep |
| 04:34.40 | enkrypted | would someone care to call my asterisk/sjphone client? |
| 04:34.56 | enkrypted | I was having trouble with inbound audio that I think I fixed. |
| 04:35.28 | enkrypted | ie. call my sip:url, say a few words, hang up |
| 04:40.56 | orlok | Hmm... |
| 04:45.58 | *** join/#asterisk lately (n=doug@ppp167-252-31.static.internode.on.net) |
| 04:49.40 | Faithful | what would be the best way to do firewall / QoS on the linux box * is on? |
| 04:50.32 | Faithful | Bearing in mind QoS on the gateway doesn't overcome network switch congestion on the LAN |
| 04:51.32 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:57.20 | *** join/#asterisk Meeptacular (n=Mage@unaffiliated/mageofchrisz) |
| 04:58.36 | *** join/#asterisk robbie2 (n=rob@howzat.dsl.onthenet.net) |
| 04:58.54 | sevard | anyone looking to sell any used sipuras? |
| 04:59.23 | robbie2 | anyone able to give me 5 mins for a hand with agents, queues and a problem i have |
| 04:59.40 | Meeptacular | So, Asterisk can do anything normal telecom companies can do and much more, right? |
| 05:00.00 | robbie2 | im using agent callback login, if they login, and then switch the phone off without logging out, they system will still try pass them calls, of course it signals busy and sends to voicemail |
| 05:00.25 | mitcheloc | why would they switch off the phone? |
| 05:00.47 | robbie2 | softphone |
| 05:00.48 | sevard | I just have it so the agents can log in but i also have a line below the agent SIP/exten# |
| 05:00.52 | robbie2 | they go home and forget to logout |
| 05:01.08 | robbie2 | they just turn off the computer |
| 05:01.15 | mitcheloc | ah yes |
| 05:01.29 | robbie2 | need tech fix to a people problem |
| 05:01.45 | sevard | I thought logging out meant you disconnected your phone. eg you dial the exten to log you in and it says agent loged in, but when you disconnect from that call you've logged out |
| 05:02.02 | robbie2 | im using agent callback login |
| 05:02.13 | robbie2 | it hangs up and calls them back when a call is coming in |
| 05:02.24 | sevard | do you have a need for that feature? |
| 05:02.39 | robbie2 | the way we run this place, its advantageous |
| 05:02.44 | robbie2 | as agents also make outgoing calls |
| 05:02.51 | Meeptacular | Actually I have a less stupid question. I have a setup of a gentoo box w/ asterisk, sipura, and a normal phone |
| 05:02.55 | mitcheloc | even if he doesn't have a need, this should be fixable no? |
| 05:02.55 | robbie2 | so they can log out and not be bothered by incoming ones |
| 05:03.13 | Meeptacular | The gentoo box only has no rj11 line, but it has lan so I had the sip box on the router with the linux box |
| 05:03.14 | sevard | I've never actually used this feature... i didn't know it existed :) |
| 05:03.16 | robbie2 | and since we use IAX2, there is no no calllimit that can be set in iax.conf |
| 05:03.18 | Meeptacular | Would that actually work? |
| 05:03.33 | sevard | I have my agents set up with SIP/exten instead of AGENT/number |
| 05:04.00 | robbie2 | sevard, no good when they are busy and cant take incoming calls |
| 05:04.12 | robbie2 | we are a place that also does quite a bit of calling back |
| 05:04.32 | robbie2 | i ahvea dirty fix, drops them into another queue that does a call everyone |
| 05:04.32 | sevard | what do you mean? they are busy and can't be on the phone or they are busy calling out? |
| 05:04.40 | robbie2 | i hsve a dirty fix, drops them into another queue that does a call everyone |
| 05:04.57 | robbie2 | sevard, either or |
| 05:05.23 | sevard | i'm inclined to bitch at my agents and tell them if they don't log off the system they are fired |
| 05:05.24 | sevard | or or |
| 05:05.38 | sevard | somehow script it to forcefully log them off after X amount of hours |
| 05:06.12 | robbie2 | yeah there is that |
| 05:06.17 | sevard | can you force log them off? |
| 05:06.22 | robbie2 | i could |
| 05:06.36 | sevard | how do you go about that? i'm still learning :) |
| 05:06.56 | robbie2 | System(asterisk -rx "agent logoff Agent/${SOMEVARIABLE}") |
| 05:07.12 | sevard | neat |
| 05:07.28 | *** join/#asterisk dasmoot (n=dasmoot@S0106001195567012.no.shawcable.net) |
| 05:07.40 | robbie2 | be neater a few more months down the track when i know more what i'm doing |
| 05:07.47 | sevard | heh |
| 05:07.48 | robbie2 | still a bit green around the gills |
| 05:08.05 | sevard | we're in the same pocket |
| 05:09.17 | robbie2 | on another note |
| 05:09.31 | dasmoot | wondering if anyone can help me out... is there any reason why Asterisk wouldn't be able to resolve a hostname even though DNS is properly configured? I'm getting: |
| 05:09.31 | dasmoot | WARNING[8508] chan_sip.c: No such host: sip.domain.com |
| 05:09.31 | dasmoot | WARNING[8508] chan_sip.c: Probably a DNS error for registration to username@sip.domain.com, trying REGISTER again (after 20 seconds) |
| 05:09.58 | Qwell | dasmoot: Did you ask this yesterday too? |
| 05:10.07 | Qwell | because if not, you're the second person |
| 05:10.10 | dasmoot | no, first time here... is there an archive I can look at somewhere? :) |
| 05:10.13 | Qwell | hmm |
| 05:10.21 | Qwell | I don't think we solved his problem either. What version of *? |
| 05:10.26 | robbie2 | anyone getting dropouts on a TE110P |
| 05:10.49 | dasmoot | 1.2.1 installed from Debian apt repository |
| 05:10.58 | Qwell | upgrade...and don't use the package |
| 05:11.07 | robbie2 | use the sauce |
| 05:11.18 | litage | what's ACD? |
| 05:11.52 | robbie2 | dasmoot, so you can ping the sip server from console ? |
| 05:12.42 | dasmoot | yeah, ping sip.domain.com works fine... I even tried replacing all instances of sip.domain.com with the actual IP of my VoIP provider's sip server and it still complained that it couldn't resolve it |
| 05:14.03 | dasmoot | I'll try compiling it from source... maybe there's something awry in the Debian packages |
| 05:16.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:39.59 | *** join/#asterisk pjchilds-at-home (n=pjchilds@pdpc/supporter/student/pjchilds) |
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| 05:59.22 | Faithful | this is a big nusance... my adsl drops momentarily but I get the same IP because it is static but * stops working |
| 05:59.26 | *** part/#asterisk dasmoot (n=dasmoot@S0106001195567012.no.shawcable.net) |
| 06:00.04 | Faithful | unless I reboot |
| 06:02.10 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:06.18 | stoffell | Faithful, could it have to do something with dns ? |
| 06:07.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:14.46 | *** part/#asterisk Meeptacular (n=Mage@unaffiliated/mageofchrisz) |
| 06:18.40 | *** join/#asterisk Vhata (i=vhata@shell.rucus.ru.ac.za) |
| 06:19.37 | Vhata | how do I "route a number in E.164 format" to an IP address? What does that mean in terms of SIP URIs? |
| 06:19.44 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 06:20.59 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
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| 07:13.23 | stoffell | is it possible to have a specific RTP port range for some SIP clients? (softphones in a VPN for example..) |
| 07:14.10 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:14.39 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:15.48 | tzafrir | stoffell, sure. define it in the client, right? |
| 07:15.56 | tzafrir | Or am I missing something? |
| 07:16.22 | stoffell | tzafrir, uh, okay, so specify it in x-lite then? (but it has to in the range, specified in rtp.conf then?) |
| 07:17.01 | Qwell | stoffell: correct |
| 07:17.17 | stoffell | okay, thanks Qwell |
| 07:17.22 | stoffell | and tzafrir |
| 07:30.21 | azeteg | I have a SIP peer with status = UNREACHABLE |
| 07:30.25 | azeteg | what would cause that? |
| 07:30.33 | azeteg | I can call from the phone |
| 07:32.12 | dlynes_home | zwelch: there's a telemarketer torture script on voip-info already |
| 07:32.48 | dlynes_home | azeteg: it hasn't qualified recently |
| 07:33.09 | dlynes_home | azeteg: and it's probably behind a firewall |
| 07:33.15 | azeteg | it is not |
| 07:33.25 | azeteg | I need qualify=yes in sip.conf? |
| 07:33.51 | tzafrir | zwelch, there is also a nice little "telemarketers not welcome" sound file in the asterisk-sounds tarball |
| 07:34.22 | dlynes_home | azeteg: or qualify=xxxxx, where xxxxx is the size of period after which it needs to qualify, xxxxx being in seconds |
| 07:34.36 | azeteg | dlynes_home: when I startup asterisk, I get this: *CLI> May 15 07:33:38 NOTICE[7166]: chan_sip.c:11396 sip_poke_noanswer: Peer '031834290' is now UNREACHABLE! Last qualify: 0 |
| 07:34.51 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 07:35.06 | dlynes_home | ah...weird |
| 07:35.12 | dlynes_home | I don't know offhand, then |
| 07:35.37 | azeteg | hmm |
| 07:36.16 | dlynes_home | but then again, i don't have anywhere near the amount of experience a lot of other people on here, do with sip |
| 07:36.21 | azeteg | on a "sip show peer xxxxxx" command, I see it says "Nat: RFC3581" |
| 07:36.27 | azeteg | I am not running NAT |
| 07:36.41 | azeteg | and I have nat=no in the sip.conf |
| 07:36.53 | dlynes_home | azeteg: you need to add nat=no, and canreinvite=yes |
| 07:37.07 | azeteg | why canreinvite? |
| 07:37.30 | dlynes_home | It means that asterisk will only stay in the media path if it needs to |
| 07:37.37 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 07:37.39 | dlynes_home | otherwise, it puts itself out of the media path |
| 07:37.52 | azeteg | I am using asterisk to talk to h323 gatekeeper |
| 07:37.59 | azeteg | with asterisk-ph323 chan driver |
| 07:38.07 | azeteg | asterisk-oh323 that is |
| 07:38.20 | dlynes_home | Yeah...I have no idea about that scenario,,,but that has nothing to do with sip |
| 07:38.25 | *** join/#asterisk motu (n=motu@192.165.166.183) |
| 07:38.25 | azeteg | someone told me to put canreinvite=no |
| 07:38.46 | dlynes_home | azeteg: Did they know whether you were behind a firewall, or not? |
| 07:39.00 | azeteg | the main problem I'm looking for - is that I get no incoming sound on the phone |
| 07:39.15 | dlynes_home | azeteg: and you're not behind a firewall? |
| 07:39.24 | azeteg | I have done a tcpdump in front of the phone - and it shows the rtp stream coming in |
| 07:39.33 | azeteg | and I save it to file, audio IS indeed coming in |
| 07:39.35 | dlynes_home | azeteg: that's usually a good indicator the sip device is on the other side of the firewall from the asterisk machine |
| 07:39.39 | zwelch | dlynes_home: thanks for that tip; i found the torture.extensions script, which seems like it could be a lot of fun. :) |
| 07:39.44 | azeteg | the net is NOT firewalled |
| 07:39.59 | azeteg | asterisk is in same switch as phone |
| 07:40.06 | dlynes_home | zwelch: yeah...it looks pretty damned funny :) |
| 07:40.21 | dlynes_home | zwelch: with a bit of work, it could be even funnier :0 |
| 07:42.01 | dlynes_home | azeteg: have you tried replacing the sip phone with some other sip phone? |
| 07:43.26 | azeteg | I have tried cisco 7960 and cisco 7940 |
| 07:43.34 | azeteg | those are the ones I have |
| 07:44.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:44.15 | zwelch | dlynes_home: well, to make it work, you have to do your own sounds; i regularly perform music as the frontman in a six piece bluegrass band, so i can improvise well. |
| 07:44.16 | azeteg | I have swissvoice ip10s and ATA186 as well |
| 07:44.23 | azeteg | but haven't configured those for asterisk yet |
| 07:45.47 | dlynes_home | azeteg: well, like i said...i'm not a sip expert, but i would try another phone, to see if that works |
| 07:46.00 | dlynes_home | azeteg: then you'll know if it's a phone configuration issue, or not |
| 07:46.04 | dlynes_home | zwelch: cool |
| 07:46.41 | zwelch | dlynes_home: i have a digium dev kit on the way, so i am really excited to go nuts with these kind of things. |
| 07:47.11 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 07:47.19 | dlynes_home | zwelch: damn...i've seen you on here for a while...figured you'd been working with asterisk for a while |
| 07:47.46 | zwelch | well, i have been using it for minisip work since december |
| 07:48.01 | zwelch | but now i have business needs to interface with the PSTN |
| 07:48.07 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 07:48.49 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 07:48.57 | zwelch | i bought direct, 'cause i really am thrilled with * |
| 07:49.44 | dlynes_home | heh |
| 07:50.19 | zwelch | on a completely unrelated matter, i haven't heard a peep since i sent off my inquery about the embedded position ;D |
| 07:50.29 | dlynes_home | Well, as long as Bush keeps doing such a great job of killing the US economy |
| 07:51.05 | dlynes_home | The digium cards will be a decent price for me by the end of the summer :0 |
| 07:51.44 | dlynes_home | It'll be nice to have the US dollar be the same value as the Canadian dollar :) |
| 07:52.19 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 07:52.34 | dlynes_home | Is dibbler something like diddler? |
| 07:52.59 | Dibbler | No, Dibbler is a char from the Terry Pratchet books |
| 07:53.08 | Dibbler | "cut me own throat" Dibbler |
| 07:53.18 | dlynes_home | ah...whoever Terry Pratchet is :) |
| 07:53.31 | Vhata | Disembowel Myself Honorably Dibblah |
| 07:53.32 | Dibbler | An authour :) |
| 07:54.20 | dlynes_home | ru.ac? Where's that in South Africa? |
| 07:54.53 | Vhata | Rhodes University, Grahamstown, Eastern Cape |
| 07:54.59 | dlynes_home | ah |
| 07:55.31 | dlynes_home | I guess ac is short for an African or Afrikaaner word for Grahamstown, Eastern Cape? |
| 07:55.40 | Vhata | "academic circuit" |
| 07:55.43 | dlynes_home | ah |
| 07:55.57 | Vhata | all the universities |
| 07:56.01 | dlynes_home | Just knew za was Zuid Afrika |
| 07:56.02 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:56.04 | Vhata | like .edu |
| 07:56.08 | Vhata | only, not. |
| 07:56.10 | RoyK | morning |
| 07:56.31 | Vhata | dlynes_home: you know more than most Americans, it seems ;-) |
| 07:56.39 | dlynes_home | I'm not american, either :) |
| 07:56.57 | Vhata | ah well. that explains the fact, but doesn't change it |
| 07:57.25 | RoyK | :) |
| 07:57.41 | catch23 | not all americans have the IQ of president bush either :P |
| 07:57.49 | dlynes_home | lol |
| 07:57.58 | dlynes_home | I love President Bush |
| 07:58.05 | zwelch | catch23: nope, just most of them... |
| 07:58.11 | dlynes_home | He's making the Canadian dollar worth a lot more than the American dollar :) |
| 07:58.26 | Vhata | George Bush owes me R200. |
| 07:58.30 | dlynes_home | Originally when he got voted in, I hated him :0 |
| 07:58.42 | Vhata | (that's 200 rand, the South Africa currency) |
| 07:58.53 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 07:59.17 | dlynes_home | oh....thought you mean 200 renminbi :) |
| 07:59.17 | enkrypted | why's that Vhata? |
| 07:59.21 | RoyK | the USA with its 20% of the population with functional illiteracy |
| 07:59.29 | catch23 | at least we'll never have him as president again... unlike those countries that are stuck with worse rulers :P |
| 07:59.49 | catch23 | though, it's tough to be worse than the bush man |
| 07:59.49 | Vhata | enkrypted: screwing with the exchange rate, when I went to California last year, I lost out ;-) |
| 08:00.00 | RoyK | the US is certainly in the lead in illiteracy in the west... |
| 08:00.16 | Vhata | RoyK: but the US is also in the lead population-wise in the west |
| 08:00.57 | dlynes_home | That's only because we haven't made all of our moose and polar bears honorary citizens yet! |
| 08:01.07 | zwelch | catch23: if there was another 9/11 style attach near the end of his term, i bet he'd try to grab a third term. The war on terror would be grossly miscompared to WWII, though GWB definitely is a bigger fascist than FDR was. ;) |
| 08:01.26 | Vhata | err, he's not allowed a third term |
| 08:01.38 | zwelch | neither was FDR, and yet.... |
| 08:01.47 | RoyK | Vhata: it doesn't really matter. compare Europe with its ~900M inhabitants with the US |
| 08:01.48 | catch23 | zwelch: even at his current poll ratings? I doubt it... he's more likely to be impeached than re-elected |
| 08:01.50 | enkrypted | http://en.wikipedia.org/wiki/List_of_countries_by_literacy_rate |
| 08:01.55 | Vhata | oooooh look at you with your "facts" and "historical comparisons" :P |
| 08:01.56 | enkrypted | Check your facts? |
| 08:02.00 | zwelch | catch23: that's the *only* blessing i count at the moment ;) |
| 08:02.29 | dlynes_home | Yeah!!!! |
| 08:02.34 | dlynes_home | Canada's at 99.9% |
| 08:02.44 | dlynes_home | But I think that chart's totally whack |
| 08:02.44 | RoyK | enkrypted: heh. 99.9% literacy in the US is BULLSHIT |
| 08:02.55 | dlynes_home | 99.9% literacy in Canada is not correct |
| 08:03.07 | dlynes_home | Our literacy rate is lower than that |
| 08:03.07 | catch23 | wow sierra leone is low.... 30% :P |
| 08:03.24 | Vhata | wikipedia's great - the article on quantum mechanics was written by a real quantum physicist. ... AND by a plumber and a street sweeper. |
| 08:03.27 | dlynes_home | And Iceland's literacy rate is 100% |
| 08:03.29 | RoyK | enkrypted: see the footnote about the US: "According to the National Institute of Literacy between 21 and 23 percent of the adult population, or approximately 44 million people scored in Level 1 which means that they "can read a little but not well enough to fill out an application, read a food label, or read a simple story to a child"." |
| 08:03.31 | dlynes_home | not 99.9% |
| 08:04.02 | enkrypted | Its still not the lowest in the West as you stated. |
| 08:04.11 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 08:04.11 | RoyK | dlynes_home: there are 300k people up there, and I doubt there aren't 300 that can't read or write |
| 08:04.11 | Vhata | also, most of the countries below the US in that chart are non-West |
| 08:04.23 | catch23 | i guess you need to define literacy before you define a percentage of literacy |
| 08:04.52 | catch23 | we have crappy schools, but our homeless can read food labels at least :P |
| 08:05.03 | Vhata | or identify the can with the picture of the beans on it? |
| 08:05.31 | dlynes_home | RoyK: That was a news story that they had on BBC, so the chances of it being accurate is pretty good |
| 08:05.58 | *** join/#asterisk Freman (n=twitsrus@ppp120-236.lns1.bne3.internode.on.net) |
| 08:06.18 | Freman | Extension B is ringing... how can I pick it up from extension A? |
| 08:06.22 | dlynes_home | RoyK: Besides...if 21-23% of the adult population in the US is functionally illiterate, how does that equate to a 97% literacy rate, overall? |
| 08:06.38 | enkrypted | Also that survey was in 1992. |
| 08:06.41 | brookshire | royk: i dunno, i've only met one person in my life who couldn't read.. and that was because of birth defect |
| 08:06.46 | dlynes_home | Freman: you don't; asterisk doesn't support shared line appearances yet |
| 08:06.59 | dlynes_home | Freman: you'll have to wait for asterisk 1.4 (that feature is in trunk) |
| 08:07.18 | Freman | doh, that's a feature we use all the time on our analog one |
| 08:07.24 | Freman | okay, thanks (c: |
| 08:07.29 | dlynes_home | Freman: you mean digital, right? |
| 08:07.42 | dlynes_home | Freman: it's a panasonic or nortel system, or something? |
| 08:07.44 | Freman | yeh, 'non ip pbx' |
| 08:08.02 | dlynes_home | Yeah...that'd be digital...analog systems dont' have 'extensions' |
| 08:08.22 | stoffell | i setup x-ten with RTP Listen port to 16384, but still conversations go through other ports (like 11584 for example), is that normal |
| 08:08.24 | stoffell | ? |
| 08:08.33 | Freman | telstra commander... there's a call on line y which rings extensions 2 and 3 but someone at the other end of the building can pick up and push the button for line y |
| 08:09.16 | dlynes_home | Freman: think of that as accounts |
| 08:09.33 | dlynes_home | Freman: and you can different 'lines' light up on your phone for different accounts |
| 08:09.51 | Freman | all I have is a sip ata with one line per hone |
| 08:09.55 | dlynes_home | Freman: but, after one person picks it up, the light will go out on the other phones |
| 08:10.22 | Freman | spoze once I upgrade to voip phones instead of sip ata's |
| 08:13.43 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 08:18.06 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:19.36 | Freman | any way I can check to see what extensions are off hook before initiating a dial? perhaps an agi? |
| 08:19.48 | Vhata | dlynes_home: "A-Tuin" is also a character from a Terry Pratchett book, btw ;-) |
| 08:20.39 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 08:20.53 | Freman | ie I want to dial all extesions, but I don't want to do this if one of them is off hook |
| 08:22.38 | *** join/#asterisk motu (n=motu@192.165.166.183) |
| 08:29.48 | litage | i hooked up my analog cordless phone to a Grandstream ATA and there was a huge amount of background noise and static. when i made a call, the recipient could hear it too. is that normal when using a cordless phone with an ATA? |
| 08:29.53 | *** join/#asterisk NoRemorse (n=bah@210-84-1-102.dyn.iinet.net.au) |
| 08:30.08 | *** join/#asterisk hensema (n=erik@scrat.hensema.net) |
| 08:30.14 | NoRemorse | Hi all, does anyone know of a presence client that can query asterisk and show what users are registered? |
| 08:30.47 | *** join/#asterisk robby (n=robby@host23-229.pool8252.interbusiness.it) |
| 08:30.49 | mitcheloc | afaik the asterisk management api can't do that, you'd probably need a script on the box |
| 08:31.40 | *** join/#asterisk kay2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 08:31.47 | NoRemorse | yeah thought as much |
| 08:32.35 | robby | hi there, a question: I know how to redirect an active channel to a defined extension of the dialplan, via the AMI command Redirect. the problem is that I'd like to redirect the active channel A (say, SIP/201-xxxx) to another *active* channel B (say, SIP/202-yyyy) |
| 08:32.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:33.40 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 08:35.40 | kay2 | how can I get the queue_log in real time |
| 08:35.45 | kay2 | and not everytime asterisk has restarted |
| 08:35.56 | *** join/#asterisk chapeaurouge (n=chap@80.92.83.2) |
| 08:36.42 | hensema | bah, I've got a reproducable segfault in asterisk when answering a call from chan_capi :-( |
| 08:38.23 | robby | any suggestion? thx in advance. |
| 08:39.42 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 08:40.12 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 08:42.04 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 08:42.07 | ramtha | hey |
| 08:42.16 | ramtha | something really strange happens to my * :) |
| 08:42.58 | ramtha | i have users with a strange content in my mysqldb (sip_buddies-> filed username) |
| 08:43.30 | ramtha | there should be the telefonnumber in the username filed but no there is the customers name in the filed |
| 08:43.32 | satlan32 | hi |
| 08:43.35 | ramtha | how can this happen? |
| 08:43.47 | satlan32 | anyone know about a good and free sip testing software? |
| 08:43.50 | ramtha | how can a sip client change the db filed of the asterisk server? |
| 08:43.53 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-252-202.telkomadsl.co.za) |
| 08:45.49 | ramtha | satlan32: what do you want to test? |
| 08:46.32 | satlan32 | sip VM system we develope |
| 08:47.06 | satlan32 | we have abacus for the PRI and SS7 testing, but the systems for sip are way too much expensive |
| 08:48.52 | ramtha | you know sipsak? |
| 08:51.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:52.42 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 08:52.44 | kernel20 | hi there |
| 08:53.05 | Vhata | you're somewhat out of date |
| 08:53.17 | kernel20 | is there any softphone that is free and it has a functionality that can do voice conference? |
| 08:53.33 | satlan32 | ramtha: i don't know this |
| 08:53.57 | ramtha | test it |
| 08:54.00 | kernel20 | just like x-lite but unfortunately x-lite doesnt have conferencing |
| 08:54.08 | kernel20 | any ideas? |
| 08:54.40 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 08:54.52 | dyn | hi |
| 08:55.15 | Qwell | kernel20: app_meetme |
| 08:55.16 | kernel20 | ? |
| 08:55.26 | Qwell | any softphone can do conferencing |
| 08:55.27 | kernel20 | QwellL: what is that? |
| 08:55.29 | dyn | (noob) when trying to access asterisk from a configured softphone, I get "Security check failed" - what's this about? |
| 08:55.31 | Qwell | google it |
| 08:55.39 | kernel20 | not with x-lite |
| 08:55.45 | Qwell | and no, you can't escape RTFMing on freenode either |
| 08:55.59 | satlan32 | ok i will |
| 08:56.00 | dyn | Qwell: i did but no match (if you meant it for me) |
| 08:56.00 | satlan32 | thanks |
| 08:56.06 | Qwell | dyn: no |
| 08:56.13 | *** part/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 08:56.17 | Qwell | dyn: and..you probably have the wrong user/pass somewhere |
| 08:56.38 | Qwell | bed |
| 08:56.39 | kernel20 | Qwell: u can check x-lite if u want |
| 08:56.48 | kernel20 | and read ur damn RTFM |
| 08:56.50 | dyn | Qwell: yes I guessed so but I wonder where I managed to set user/pass in my asterisk config.. |
| 08:56.51 | kernel20 | on it |
| 08:57.57 | kernel20 | what I am refering is a softphone |
| 08:58.03 | kernel20 | cant u understand it? |
| 08:59.14 | dyn | kernel20: i'm not sure what're you on but better calm a little bit down, nah? |
| 08:59.37 | kernel20 | Qwell: sucks |
| 08:59.51 | kernel20 | very hard headed and keep on insulting |
| 09:00.16 | kernel20 | considering the he himself doesnt know how to do it |
| 09:00.58 | Vhata | kernel20: a softphone is just some software that allows you to talk to the Asterisk box. The asterisk box is the thing that does conferencing, you need to set the asterisk box up, using the asterisk application called "app_meetme" |
| 09:01.07 | Vhata | as Qwell said. |
| 09:01.23 | ramtha | aeh |
| 09:01.31 | ramtha | i dont think that right Vhata |
| 09:01.33 | kernel20 | like skype it has an option for conferencing |
| 09:01.40 | kernel20 | where I just click on |
| 09:01.46 | ramtha | it is one think to do conference with asterisk |
| 09:01.46 | Vhata | don't bring skype into this. |
| 09:01.48 | kernel20 | i have set up asterisk box myself |
| 09:02.00 | kernel20 | i am refering to a damn app Vhata |
| 09:02.02 | ramtha | but you can make several calls(conference) with a softphone too |
| 09:02.08 | mitcheloc | heh Vhata, good call |
| 09:02.21 | kernel20 | how? |
| 09:02.31 | kernel20 | in x-lite how can i do it? |
| 09:02.56 | kernel20 | say 5 people talking at a time |
| 09:03.05 | kernel20 | where can i set it in x-lite |
| 09:03.30 | dyn | kernel20: I *guess* (but i'm much of a noob) you set up a conference room in asterisk and you call that extension from your softphone |
| 09:03.36 | Vhata | kernel20: you just lost any sympathy you were going to get, and therefore any help I was going to give. do stop swearing at us. thanks. |
| 09:03.40 | dyn | so it's not related to your x-lite |
| 09:04.09 | kernel20 | hmmm |
| 09:04.19 | kernel20 | thanks dyns |
| 09:04.29 | dyn | welcome.. |
| 09:04.37 | satlan32 | ramtha: this sipsak is for single port each time... i need something that can produce 30-120 concurent calls |
| 09:06.09 | dyn | incoming calls (eg. from softphones) arrive in the [default] context? |
| 09:06.22 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:07.46 | dyn | ahh ok I see it in sip.conf, context=default |
| 09:09.46 | {zombie} | dyn: the idea is that you add your phones to sip.conf, and put them in contexts other than [default] |
| 09:10.05 | {zombie} | so you can give them more privileges (eg being able to ring the outside world) than unauthenticated users |
| 09:10.26 | {zombie} | because putting stuff like that in [default] is asking for trouble |
| 09:10.58 | dyn | i see your point.. so I better set up softphone contexts like 'staff', 'bosses', etc to be able to route accordingly.. I guess |
| 09:11.05 | {zombie} | right |
| 09:11.19 | {zombie} | I know of one large company in Brisbane, Australia that ended up with a hefty phone bill |
| 09:11.27 | {zombie} | because their VoIP wasn't locked down enough |
| 09:11.29 | dyn | *snickers* |
| 09:12.04 | dyn | and the callerid in sip.conf should match the extension of the owner of that softphone? |
| 09:12.27 | {zombie} | that would be useful if you want people to be able to ring each other back easily, yes :) |
| 09:13.11 | dyn | i asked that end of last week but i wish there was a shorten way to configure ~20 softphones in sip.conf (basically the username/secret/defaultip seems to be changing but each entry is now about 15 lines) |
| 09:13.34 | dyn | you might know any better solution than scripting it in bash/perl/anything? |
| 09:14.08 | {zombie} | I use vim macros :) |
| 09:14.25 | dyn | hehe fair enough |
| 09:15.01 | Vhata | dyn: use extconfig |
| 09:15.47 | dyn | extconfig.conf? ok i dig up some docs on it |
| 09:15.47 | *** join/#asterisk magic_1 (n=quinton@wbs-196-2-123-139.wbs.co.za) |
| 09:16.05 | Vhata | dyn: it lets you store your SIP peers/etc in a database, isntead of in the flat file |
| 09:17.45 | dyn | that sounds handy, even if we have a few peers only (at least not thousands) |
| 09:18.35 | Vhata | voip-talk.org is your best friend |
| 09:19.01 | Vhata | (unless you have a dog) |
| 09:19.22 | dyn | I aint.. and i'm getting already in touch with voip-info.org :) |
| 09:20.47 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 09:20.51 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 09:21.30 | Freman | I'm trying to call asterisk_chanlist in perl but it want's (fd), I can't find any documentation for 'fd' anywhere |
| 09:22.16 | Vhata | file descriptor. it'll be the handle to the current open session with asterisk? |
| 09:23.20 | Freman | that may be right... but where do I get the FD from? |
| 09:23.50 | Vhata | is this in res_perl, or, erm, where? |
| 09:23.57 | Vhata | I mean, how does your perl get called? |
| 09:23.57 | Vhata | beca |
| 09:24.06 | Vhata | because the file descriptor should get passed in to you |
| 09:24.25 | Vhata | it's two years since I wrote an asterisk application, and that was in C |
| 09:25.29 | Freman | yes res_perl |
| 09:27.02 | Freman | I make use of perl(Loadfile) |
| 09:33.18 | hensema | how can I set the default language of all channels? |
| 09:33.25 | hensema | (for playback of sounds etc) |
| 09:35.22 | wasim | Set(Language=xx) |
| 09:36.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:38.01 | RoyK | methinks language should be xx_XX, not xx. |
| 09:38.30 | ramtha | i have users with a strange content in my mysqldb (sip_buddies-> filed username) |
| 09:38.34 | ramtha | there should be the telefonnumber in the username filed but no there is the customers name in the filed |
| 09:38.44 | ramtha | how can a sip client change the db filed of the asterisk server? |
| 09:44.16 | *** join/#asterisk my007 (n=noor@217.139.224.194) |
| 09:45.55 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 09:48.54 | Freman | I'll be buggered if I can find my file discriptor |
| 09:49.42 | SheriF_WorK | any one have any experinc with Multi Tech VOIP Devices analog + asterisk ? |
| 09:55.34 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 09:55.50 | hwt | hi. my snom 360 keeps getting UNREACHABLE: |
| 09:55.50 | hwt | May 15 10:55:57 NOTICE[23467]: chan_sip.c:11389 sip_poke_noanswer: Peer '5005' is now UNREACHABLE! Last qualify: 78 |
| 09:55.58 | hwt | is there a way to avoid this? |
| 09:56.02 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 09:56.15 | hwt | that does not happen on any of my softphones, or on my sipura ATA. |
| 10:07.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 10:10.11 | mut | mornin all |
| 10:11.12 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 10:20.43 | *** join/#asterisk saftsack (n=saftsack@p54A7E35D.dip.t-dialin.net) |
| 10:22.10 | ghenry | morning all |
| 10:22.41 | ghenry | I am trying to get round a bad nat issue with sip |
| 10:22.45 | ghenry | we have 2 offices |
| 10:22.53 | ghenry | I want to check if this idea is sane |
| 10:23.01 | ghenry | we have 2 asterisk instances |
| 10:23.15 | ghenry | users connect inside lan and each asterisk box is a sip proxy |
| 10:23.19 | ghenry | at each office |
| 10:23.56 | ghenry | going to connect them (each site) via iax, and have 2 sets of exten. 5xxx and 6xxx, 5xxx is one office, 6xxx is the other office |
| 10:24.13 | ghenry | asterisk routes the calls via each box dependant on exten over iax. sound ok? |
| 10:24.33 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 10:24.59 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 10:25.00 | qdk | ghenry: yes, sounds just about right |
| 10:25.09 | ghenry | cool |
| 10:25.30 | ghenry | follow iax.conf sample and sip nat howtos on voip-info for guidance, aye? |
| 10:25.37 | qdk | ghenry: you do need something else if you want them do dial "out". |
| 10:25.46 | ghenry | out? |
| 10:25.54 | ghenry | over sip or zap? |
| 10:25.58 | qdk | ghenry: out = the rest of the world. |
| 10:26.01 | ghenry | ah |
| 10:26.15 | ghenry | we would do 01xxxx blah out over zap channel |
| 10:26.21 | qdk | ghenry: yes, that a good place to start. |
| 10:26.24 | ghenry | that's the plan |
| 10:26.30 | Faithful | Hey has PennyTel in Au gone bust or something? |
| 10:26.33 | qdk | ghenry: ok. |
| 10:26.41 | ghenry | only offices to use sip for saving money qdk ;-) |
| 10:27.10 | ghenry | qdk: As getting sip hard/soft phones to register from behind nat to a server behind nat is proving very hard :-( |
| 10:27.41 | ghenry | qdk: They could even be VPNs and not a iax connect if we could be arsed ;-) |
| 10:29.20 | qdk | ghenry: only the IAX is "behind" nat in your setup. |
| 10:29.37 | ghenry | qdk: aye. each box is behind nat |
| 10:29.47 | qdk | ghenry: and you just portforward the IAX port, and then NAT isnt an issue at all. |
| 10:29.54 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 10:30.30 | qdk | ghenry: and? both * and SIPs a behind NAT = non of them is behind NAT. |
| 10:30.39 | ghenry | qdk: That's the plan. Bought 2 of these: http://www.voipon.co.uk/product_info.php?cPath=1_50&products_id=124 So hopefully they won't be a waste with this setup. |
| 10:30.41 | AsteriskAlbania | smth wrong with my configuration if I dial 00 in front of the number it fails if i do not dial 00 it make the call, any idea please ? |
| 10:30.49 | qdk | a = are |
| 10:30.58 | SheriF_WorK | is there a ring application i nasterisk ? |
| 10:31.17 | qdk | SheriF_WorK: you mean like playtones(ring)? |
| 10:31.23 | SheriF_WorK | qdk: yes |
| 10:31.38 | SheriF_WorK | qdk: when a personal dial 100 " which is not a real phone " he got 2 rings 1st |
| 10:32.05 | ghenry | qdk: both sip phones and * are behind nat, but like you say, with the IAX link, should be ok? Or am I not coming across clearly? |
| 10:32.14 | file | ooh regulated prices on gas take effective on July 1st here... most excellent |
| 10:32.15 | ghenry | sorry ;-) |
| 10:32.37 | qdk | SheriF_WorK: hmm.. did you answer the channel? have you tried using ringing() and playtones(ring)? |
| 10:33.39 | SheriF_WorK | qdk: nop |
| 10:33.40 | SheriF_WorK | let me try |
| 10:33.42 | qdk | ghenry: as i said... you only have a NAT issue when connection the * together with IAX, but thats just a simple portforward for 1 port. |
| 10:34.10 | ghenry | qdk: Cool. As I thought. Will test out later today. Thanks for the confirmation. |
| 10:34.14 | *** join/#asterisk eds0n (n=edson@pc36-73.vlab.iu.hio.no) |
| 10:34.50 | qdk | ghenry: np |
| 10:35.20 | ghenry | qdk: I was going to similar to: http://www.voip-info.org/wiki-Asterisk+avoid+SIP+NAT+Traversal |
| 10:35.49 | SheriF_WorK | qdk: perfect ;-) thx alot |
| 10:36.00 | qdk | SheriF_WorK: great |
| 10:36.32 | *** join/#asterisk core-ix (n=b3l_l3b@pirus.securax.be) |
| 10:36.34 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 10:37.11 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 10:37.21 | eds0n | hey, has anyone tried astertest here? Does anyone know how to get or make the ast-test-rtp file? |
| 10:38.16 | qdk | ghenry: yes, thats is just about the idea.... SIP <-> Asterisk <-> NAT <-> IAX <-> NAT <-> Asterisk <-> SIP |
| 10:38.30 | ghenry | qdk: ta. |
| 10:38.45 | Faithful | qdk: that is not hard to do |
| 10:38.46 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 10:39.22 | qdk | Faithful: i know, but ghenry wasnt sure. |
| 10:39.44 | ghenry | Faithful: Yeah, was just sounding out the idea ;-) |
| 10:40.03 | Faithful | Most of us basically do that. |
| 10:40.31 | ghenry | Faithful: Good to know. |
| 10:41.15 | Faithful | unless you make your * box your gateway (there are some advantages to that). |
| 10:41.25 | ghenry | qdk, Faithful I think this is the exact link we need: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 10:42.51 | ghenry | I hate all the Perl programs on voip-info.org for *. I have been cleaning them all up. NONE of them have "use strict;" or "use warnings;" bad ;-) |
| 10:44.05 | azeteg | I'm having this weird problem where I cannot hear what the other end says |
| 10:44.18 | azeteg | I can call between phones internally without problem |
| 10:44.39 | azeteg | but when I go over h323 gk using asterisk-oh323, I cannot hear the incoming audio |
| 10:44.48 | azeteg | analyzing the traffic shows that it indeed is there |
| 10:45.15 | azeteg | I hear first 0.3-0.5 secs of incoming audio sometimes |
| 10:45.21 | azeteg | then it goes silent |
| 10:45.29 | azeteg | anyone that might know what that could be? |
| 10:45.33 | ghenry | qdk: WOuld you be so kind as to point me to a link or explain "switch"of extensions.conf? |
| 10:49.03 | dyn | May 15 12:47:35 NOTICE[5218]: chan_iax2.c:2447 iax2_read: I should never be called! |
| 10:49.04 | dyn | <PROTECTED> |
| 10:49.13 | dyn | anyone can tell me what should that be and how to fix it? |
| 10:49.47 | *** join/#asterisk etel (n=khalid@mbl-82-49-5.dsl.net.pk) |
| 10:50.22 | etel | Hi everyone |
| 10:50.25 | ghenry | hi |
| 10:50.32 | etel | has anyone used chan_ss7 |
| 10:50.44 | etel | Hi ghenry |
| 10:50.47 | etel | ?? |
| 10:50.56 | *** join/#asterisk michael_tcf (n=michael@de1-b11358.alshamil.net.ae) |
| 10:50.57 | eds0n | hey, has anyone tried astertest here? Does anyone know how to get or make the ast-test-rtp file? |
| 10:51.27 | etel | Do you need it for AsterTest |
| 10:51.30 | etel | ?? |
| 10:52.01 | *** join/#asterisk Ahrimanes (n=michael@62.61.133.90.generic-hostname.arrownet.dk) |
| 10:52.46 | michael_tcf | Hi, is there anyone who got completely rid of the echo issue when using voip <-> pstn bridge through a Digium tdm card? |
| 10:52.47 | *** join/#asterisk burizaa (n=freeee@cm228.omega106.maxonline.com.sg) |
| 10:53.11 | michael_tcf | Tried many things, echo cancellation works quite well, but it is now reduced to a crackle |
| 10:53.55 | michael_tcf | how important is correct configuration of asterisk for your particular pstn network? |
| 10:54.23 | michael_tcf | Because my country is not in the zones list, i'm guessing on uk zone mapping |
| 10:54.35 | michael_tcf | but can this affect echo problems? |
| 10:56.01 | michael_tcf | anyone? :) |
| 10:56.18 | Zeeek | don't think so. |
| 10:57.14 | stoffell | nope |
| 10:57.23 | stoffell | what country you are in, michael_tcf |
| 10:57.30 | Vhata | United Arab Emirates |
| 10:57.30 | michael_tcf | united arab emirates |
| 10:57.33 | michael_tcf | lol |
| 10:58.17 | michael_tcf | i've managed to pass the country option to the wctdm kernel module |
| 10:58.18 | AsteriskAlbania | how can asterisk perform call back |
| 10:58.23 | michael_tcf | it's in UAE mode |
| 10:58.49 | michael_tcf | but the timings zones defined in zonedata.c of the zaptel drivers doesn't include uae |
| 10:58.54 | michael_tcf | so I'm using uk |
| 10:59.08 | michael_tcf | But I'm not sure this really matters for the echo issue |
| 10:59.16 | Zeeek | prolly not |
| 10:59.18 | michael_tcf | It's like trying to find a needle in a haystack |
| 10:59.33 | michael_tcf | Who knows, maybe it's the motherboard that's causing problems :) |
| 10:59.41 | Zeeek | make sure the mic volume isn't too loud, that will always ensure plenty of echo |
| 11:00.08 | michael_tcf | yes, I've been tweaking the tx and rx gains as well and I got some positive results from that |
| 11:00.43 | michael_tcf | but what happens as well is that if the gains or mic volumes go too low, people on the other end don't hear us anymore :) |
| 11:00.48 | qdk | <PROTECTED> |
| 11:00.54 | michael_tcf | solves the echo problem though |
| 11:01.00 | ghenry | qdk: Cheers though. |
| 11:01.33 | Zeeek | michael_tcf the mute button does wonders with echo too |
| 11:01.40 | stoffell | michael_tcf, do you use mg2 or other echo canceller? |
| 11:01.41 | michael_tcf | lol, I bet |
| 11:01.43 | michael_tcf | mg2 |
| 11:01.48 | michael_tcf | works very well |
| 11:01.56 | michael_tcf | I compared it with the others |
| 11:02.19 | michael_tcf | I can no longer distinguish my own voice in the echo, but is becomes a crackle everytime I speak |
| 11:02.34 | michael_tcf | tried idefisk, some sip hardware phones |
| 11:02.56 | stoffell | are you sure the crackle is because of echo cancellation or something related? |
| 11:03.14 | michael_tcf | Well, it happens everytime I say something |
| 11:03.49 | michael_tcf | and if I change algorithms, it becomes more of a crackle or more distinguishable echo depending on the quality of the algorithm |
| 11:04.09 | stoffell | hm, you are doing transcoding alot? |
| 11:04.21 | michael_tcf | can you explain? |
| 11:04.34 | stoffell | what codecs do you use on the phones? |
| 11:04.38 | michael_tcf | ulaw |
| 11:04.45 | stoffell | oh, okay, no transcoding then :) |
| 11:04.45 | michael_tcf | it's on a lan |
| 11:04.49 | michael_tcf | I see :) |
| 11:05.02 | michael_tcf | so ulaw is what is used on the pstn then? |
| 11:05.07 | michael_tcf | or related to the tdm? |
| 11:05.49 | michael_tcf | interesting to know. No I use ulaw because as far as I've read it seems to give the best quality and bandwith doesn't really matter because we're on a 100mbit lan |
| 11:06.03 | stoffell | it's not ulaw, but it doesn't need transcoding either.. (your cpu doesn't need to convert them) |
| 11:06.08 | michael_tcf | I see |
| 11:06.13 | dyn | <PROTECTED> |
| 11:06.50 | michael_tcf | which message? |
| 11:06.58 | etel | Has anyone tested chan_ss7 with asterisk |
| 11:07.00 | etel | ?? |
| 11:07.33 | stoffell | michael_tcf, it's not that your server is having disk i/o when that crackling occurs? |
| 11:08.08 | michael_tcf | No, I think the crackling is related to my voice, because it only happens when I speak. The machine in question is pretty idle |
| 11:08.21 | michael_tcf | it has 1 gb of ram, 3.2 ghz processor |
| 11:08.29 | michael_tcf | load is extremely low |
| 11:08.37 | core-ix | where is chan_ss7 ? |
| 11:08.37 | stoffell | and ide/scsi ? |
| 11:08.41 | michael_tcf | sata |
| 11:08.48 | stoffell | no irq sharing ? |
| 11:08.57 | michael_tcf | no, I checked |
| 11:09.03 | michael_tcf | tdm uses irq 17 solely |
| 11:09.23 | dyn | michael-i: May 15 12:47:35 NOTICE[5218]: chan_iax2.c:2447 iax2_read: I should never be called! |
| 11:09.28 | dyn | michael-i: that's the error |
| 11:09.36 | michael_tcf | hehe |
| 11:09.41 | michael_tcf | haven't seen that one yet |
| 11:09.43 | dyn | well, it's a notice but looks quite bad when you get 10/sec :) |
| 11:10.07 | etel | what do you mean by where is chan_ss7?? |
| 11:10.21 | etel | core-ix: |
| 11:10.46 | dyn | michael-i: it's between an asterisk v1.0.9 and v1.0.8 (i dont think that matters much) |
| 11:10.59 | michael_tcf | hmm me neither |
| 11:11.00 | core-ix | etel, i have svn-ed asterisk code from friday ... and i can't find chan_ss7 in it |
| 11:11.22 | etel | its an addon sort of thing |
| 11:11.25 | etel | equivlant to LibISUP |
| 11:11.32 | etel | but open source and free |
| 11:11.50 | michael_tcf | stoffel: pbx ~ # cat /proc/interrupts |
| 11:11.50 | michael_tcf | <PROTECTED> |
| 11:11.50 | michael_tcf | <PROTECTED> |
| 11:11.50 | michael_tcf | <PROTECTED> |
| 11:11.50 | michael_tcf | <PROTECTED> |
| 11:11.51 | michael_tcf | <PROTECTED> |
| 11:11.53 | michael_tcf | <PROTECTED> |
| 11:11.55 | michael_tcf | <PROTECTED> |
| 11:11.55 | etel | its forr ss7 signalling support on PSTN end |
| 11:11.57 | michael_tcf | <PROTECTED> |
| 11:11.59 | michael_tcf | <PROTECTED> |
| 11:12.01 | michael_tcf | <PROTECTED> |
| 11:12.03 | michael_tcf | NMI: 0 0 |
| 11:12.03 | stoffell | michael_tcf, use pastebin.ca ;) |
| 11:12.05 | michael_tcf | LOC: 2560934 2560809 |
| 11:12.07 | michael_tcf | ERR: 0 |
| 11:12.09 | michael_tcf | MIS: 0 |
| 11:12.11 | michael_tcf | sorry? |
| 11:12.24 | stoffell | to paste, use http://pastebin.ca |
| 11:12.33 | core-ix | i know what ss7 stands for ... i'm asking is there chan_ss7.c in asterisk ? |
| 11:12.38 | michael_tcf | sorry, it's been many many years since I used irc :) |
| 11:12.46 | stoffell | but I get the picture, seems okay.. have you done zttest , and are your results good? |
| 11:12.50 | etel | no you have to download and patch it |
| 11:12.51 | burizaa | any one got idea settin up SPA 3000 ? |
| 11:12.57 | core-ix | etel, where from |
| 11:13.06 | dyn | how do I specify the username/secret on a softphone when trying to place a call? is it something like sip:??exten:host ? |
| 11:13.16 | dyn | or i need to set that somewhere else, not in the Sip url? |
| 11:13.34 | michael_tcf | stoffel: yes, average of 100% |
| 11:13.35 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 11:13.37 | etel | http://www.sifira.dk/chan-ss7/ |
| 11:13.44 | michael_tcf | so I guess that's good |
| 11:13.53 | core-ix | etel, thanks |
| 11:13.54 | core-ix | :) |
| 11:14.22 | etel | Do you know anyone who has any experience with it |
| 11:14.22 | etel | ?? |
| 11:14.36 | stoffell | yeah, that's good |
| 11:15.28 | michael_tcf | maybe it would take some hardware hacking of the phone line as in putting a condenser on the lines or some of that diy stuff |
| 11:15.35 | michael_tcf | capacitor I mean |
| 11:16.31 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:17.28 | *** join/#asterisk adorah (n=FreePBX9@87.68.145.182.cable.012.net.il) |
| 11:17.29 | stoffell | no idea, i''m out of ideas :) |
| 11:19.01 | michael_tcf | haha, thanks anyway |
| 11:19.35 | michael_tcf | I'm looking for people who actually got it working as well as a legacy pbx would |
| 11:19.49 | michael_tcf | or a normal phone connected to a phone line for that matter |
| 11:21.09 | michael_tcf | And I'm trying to find out what are the main causes of echo. Is it the phones, is it the hardware bridge, is it the cpu / motherboard, whatever of the box |
| 11:21.51 | michael_tcf | or is it the network? Who knows. I have been able to eliminate some speculations because voip to voip calls over the same network work fine |
| 11:21.59 | Zeeek | michael_tcf I think most people using TDM have echo |
| 11:22.11 | michael_tcf | that's what I was afraid of :D |
| 11:22.30 | michael_tcf | so what's the solution? isdn? :-/ |
| 11:22.33 | Zeeek | the new 2400 has hardware EC built in |
| 11:22.41 | stoffell | i also have sporadic echo issues (using BRI), but it's mostly because of the other end using bad phones/bad volume/.. |
| 11:22.59 | michael_tcf | Zeek: so has the TDM400P, no? |
| 11:23.08 | Zeeek | stoffell maybe if two people with echo connect to each other it'll be cancelled out ? |
| 11:23.19 | michael_tcf | lol |
| 11:23.23 | Zeeek | I don't think the TDM400 has hardxware echo cancel |
| 11:23.28 | stoffell | lol |
| 11:23.33 | Zeeek | not that I know of anyway |
| 11:23.52 | nettie | Hi guys, I'm still having call progressing issues with my voip carrier, I have a sip debug output but I can't see anything strange there.. it seems normal. Does anyone can give it a look please? Just wanted to doublecheck with someone which has more expertise on the matter :) Thanx in advance. |
| 11:24.05 | Zeeek | what carrier? |
| 11:24.14 | nettie | Zeeek skypho |
| 11:24.33 | nettie | seems they're using cisco sip gw |
| 11:24.50 | nettie | works great using grandstream ata286 |
| 11:25.00 | nettie | have issues with asterisk |
| 11:25.43 | Zeeek | speaking of providers, anyone heard anything about nufone? I'm beginning to wonder what's really up? |
| 11:26.50 | nettie | from what I Can see |
| 11:27.11 | nettie | in their news .. I'm not sure I would reccomend them eheh.. |
| 11:27.19 | michael_tcf | Zeeek: you seem to be right, i've checked the fact sheets and the tdm400p doesn't seem to have hardware echo cancellation |
| 11:27.30 | Zeeek | I've been with them for a couple of years, they're wgreat |
| 11:27.42 | michael_tcf | Zeeek: tdm2400 does mention hardware echo cancellation |
| 11:27.43 | nettie | at least they say the truth |
| 11:27.44 | Zeeek | but the only news I find |
| 11:28.02 | Zeeek | is regarding an investigation about callerid spoofing from Febrary |
| 11:28.11 | nettie | it's strange they're having such issues |
| 11:28.15 | nettie | ah |
| 11:28.25 | nettie | someone took them down? |
| 11:28.36 | Zeeek | if that is serious and the reson why, we customers are up the creek too |
| 11:29.03 | Zeeek | I felt good about what they were saying last time I had direct news |
| 11:29.06 | michael_tcf | Stoffel: interesting, hardware compatibility list on the tdm400p mentions issues with the e1000 onboard ethernet controller |
| 11:29.17 | Zeeek | the May 5th announcement is getting old though |
| 11:29.35 | michael_tcf | Stoffel: I'm using that one, so gonna try replacing it with an external pci one |
| 11:32.18 | michael_tcf | I'll keep you updated on my progress, off to the computer shop |
| 11:32.22 | michael_tcf | see you guys later |
| 11:32.44 | Zeeek | good luck |
| 11:33.06 | adorah | Sangoma has a PCI analog ports card that has Echo Cancellation on board I think.. |
| 11:33.19 | Zeeek | actually, to be fair to nufone, outgoing calls were still working fine the last I tried one, so the callerid thing can't be the problem |
| 11:35.41 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 11:35.47 | znoG | hi, quick question... do I need to enable anything else for the TDM2400 FXS modules to be able to pass on caller ID info being sent in? |
| 11:36.21 | znoG | i have a lucent PBX which dials via a FXS port on the TDM2400 and no caller ID info is being read' in |
| 11:41.18 | nettie | Zeeek do you have a sec to give a look at my sip debug please? |
| 11:43.10 | ghenry | Faithful or qdk: One last question if you don't mind. Is it ok to use type=friend to the iax settings, as each will be making and receiving calls to each other? Just asking as some docs on the wiki say best to keep them as user and peer with matching statments on each * box. |
| 11:43.27 | ghenry | thanks |
| 11:44.06 | *** join/#asterisk burizaa (n=freeee@cm228.omega106.maxonline.com.sg) |
| 11:44.37 | qdk | ghenry: well I use friends because I controll both ends of all my IAX trunks. |
| 11:45.00 | ghenry | qdk: as do we, so I guess that's fine. Thanks. |
| 11:45.15 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 11:45.29 | qdk | ghenry: and context is only valid for incoming calls, so very little reason to make both a user and a peer. |
| 11:45.42 | boddy | is there anybody who installed asterisk on suse via yast2 |
| 11:45.58 | ghenry | qdk: Cool. Is that who to refer to this setup then? IAX Trunking/Trunks? |
| 11:46.15 | Zeeek | nettie, I don't think I can help, sorry |
| 11:46.22 | qdk | boddy: nope... debian or gentoo using the force^Wsource |
| 11:46.25 | ghenry | qdk: i.e. "Ours calls go out over a IAX Trunk. |
| 11:46.42 | qdk | ghenry: who = how? |
| 11:47.04 | qdk | ghenry: yes, that is correct. |
| 11:47.04 | ghenry | qdk: Well, I just want to get the * lingo right |
| 11:47.11 | boddy | qdk I have already installed asterisk on gentoo but intel g729 codec dosent working properly on gentoo |
| 11:47.20 | ghenry | qdk: coolio. Thanks. |
| 11:47.34 | boddy | so I have decided to try suse |
| 11:47.54 | boddy | what a stupid distro |
| 11:47.57 | Zeeek | use slackware |
| 11:47.59 | qdk | boddy: i dont use g729, but I really dont think that gentoo is to blame for a br0ken g729. |
| 11:48.11 | ghenry | boddy: use fedora. |
| 11:48.15 | ghenry | right, bye. |
| 11:48.17 | Zeeek | use windows |
| 11:48.19 | qdk | ghenry: NO! |
| 11:48.30 | Zeeek | FreeBSD |
| 11:48.34 | Zeeek | solaris |
| 11:48.41 | qdk | RPM-based distros are pr. definition br0ken... |
| 11:48.44 | Zeeek | TRS-DOS |
| 11:48.45 | boddy | I dont want worry about depecies |
| 11:48.54 | boddy | I love gentoo |
| 11:48.57 | boddy | :) |
| 11:49.00 | qdk | (in my world, anyway :-D) |
| 11:49.18 | michael_tcf | gentoo rocks |
| 11:49.47 | michael_tcf | but I assume it's better not to use the portage asterisk packages, right? |
| 11:49.51 | qdk | boddy: maybe you should try and solve the g729 issue on gentoo instead of getting new problems on suse. |
| 11:50.19 | boddy | qdk I have tried |
| 11:50.23 | boddy | bur didnt |
| 11:50.34 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:50.35 | boddy | on intel web page they said |
| 11:50.51 | boddy | working properly on redhat suse ... |
| 11:50.54 | qdk | michael-i: Asterisk in gentoo is often MASKED, if you dont care about that then portage would probably be the best choise... I use pure source for it. |
| 11:51.13 | michael_tcf | qdk: yes me too, that's what I meant |
| 11:51.32 | qdk | boddy: ok, then g729 devs ppl just suck. hehe |
| 11:51.34 | michael_tcf | qdk: even though it's available in portage, I assume it's better to compile from source directly |
| 11:52.06 | qdk | michael-i: i feel more confident with the source for * than the packages-system. |
| 11:52.33 | michael_tcf | qdk: Yes, I think it's a must with this bleeding edge technology |
| 11:52.50 | RoyK | ~lart himself |
| 11:56.12 | dyn | michael-i: i'm using the gentoo packages, no problems here |
| 11:56.59 | azeteg | and I'm using UPS packages |
| 11:57.12 | dyn | azeteg: they're usually a bit faster, lol |
| 11:57.22 | dyn | same goes for dhl ;) |
| 11:57.26 | azeteg | ;) |
| 11:57.38 | azeteg | nah - centos here |
| 11:57.43 | michael_tcf | dyn: yes, but they are quite behind, no? |
| 11:57.44 | *** join/#asterisk lately (n=doug@15.5.233.220.exetel.com.au) |
| 11:57.48 | azeteg | seems to work just fine |
| 11:57.57 | azeteg | except I have a problem with oh323-sip |
| 11:58.07 | azeteg | no incoming audio to phone |
| 11:58.10 | azeteg | which bugs me |
| 11:58.29 | dyn | michael-i: not if you use /etc/portage/* and unmask the dev packages |
| 11:58.45 | dyn | michael-i: also you can use the asterisk from ~arch (not from the default arch) |
| 11:58.56 | michael_tcf | dyn: ah ok, I see |
| 11:59.05 | azeteg | anyone here into video editing etc? |
| 11:59.23 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 12:02.25 | *** join/#asterisk tdi (n=tdi@80.48.205.2) |
| 12:02.29 | tdi | re |
| 12:02.54 | tdi | howto switch off silence suppression in asterisk ? |
| 12:04.26 | robl^ | tdi: asterisk always has it turned off. it doesn't support it |
| 12:05.24 | dyn | can I specify a single extension for accessing the user's voicemail without having them enter their extension number? |
| 12:05.30 | dyn | eg. by using their caller id? |
| 12:05.54 | tdi | robl^: so why it marks packets separated by silence ?? |
| 12:06.05 | azeteg | dyn: good question |
| 12:06.16 | azeteg | dyn: can you also make it authenticate without passowrd? |
| 12:06.27 | dyn | azeteg: as I see in the docs, Voicemail(number) would bring the caller into recording a voicemail message |
| 12:06.38 | dyn | azeteg: i think that's possible but i'd keep the password |
| 12:06.56 | dyn | azeteg: it'd be only more comfortable to skip the question regarding their extension |
| 12:06.57 | azeteg | I mean - it should verify to the account name directly |
| 12:06.58 | robl^ | tdi: it must be a phone or the provider using silence suppression. |
| 12:07.18 | azeteg | just pressing a button to listen to the voicemail would be neater |
| 12:07.55 | tdi | robl^: it is asterisk->asterisk comuunication (sangoma cards) |
| 12:08.21 | tdi | PSTN->sangoma->asterisk->sipchan->asterisk |
| 12:08.24 | tdi | like that |
| 12:08.26 | SheriF_WorK | any one have any experinc with Multi Tech VOIP Devices analog + asterisk ? |
| 12:08.40 | *** join/#asterisk delmar (n=delmar@203-114-178-231.static.inspire.net.nz) |
| 12:08.56 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 12:09.30 | uwe | hello, does any one know a clean way to name the recorded files in monitor by extension? |
| 12:10.55 | delmar | Can anyone tell me what.. specifically .. sux.. about IAX2 ? other than a vague statement such as "poor design and implimentation" hmm? |
| 12:11.33 | boddy | is there anybody using intel g729 codec on gentoo |
| 12:11.34 | boddy | :) |
| 12:11.35 | Malthus | I don't think you'll find many people in here bashing iax2 |
| 12:11.56 | delmar | Malthus, exactly... then what the hell is Asterlink's problem. |
| 12:11.57 | Zeeek | yes, sometimes you will |
| 12:12.08 | delmar | I'm so sick of Asterlink and their bull sh1t |
| 12:12.10 | Malthus | but |
| 12:12.18 | Malthus | since IAX2 uses one port |
| 12:12.50 | Malthus | you can't easily separate the signaling from the actual media paths |
| 12:13.07 | delmar | Malthus, but it works... |
| 12:13.21 | Malthus | so you can't have a softswitch + separate media gateways |
| 12:13.35 | burizaa | anyone have try spa 3000 as fxo ? |
| 12:13.36 | delmar | Malthus, Asterisk/SIP to Asterisk/SIP .. has no tolerance for any network jitter at all.. there is no jitter handling in that scenario. |
| 12:13.39 | lately | azeteg: I just did a general voicemail macro. exten => s,1,VoiceMailMain(s${CALLERIDNUM}@default) |
| 12:13.43 | Malthus | it works and it has advatages in several specific use cases |
| 12:14.12 | Malthus | chan_sip has no jitter buffer? |
| 12:14.31 | lately | azeteg: I now only have to press the voicemail button and I get the list of messages. |
| 12:14.33 | Malthus | anyway, I have to go do some analog Asterisking |
| 12:14.35 | Malthus | later |
| 12:15.04 | file | delmar: get my msgs? |
| 12:15.22 | delmar | Malthus, u cant have a softswitch with seperate media gateways? am I to take your meaning .. as in.. lets say you had one route via one upstream IAX2 carrier.. and another route via a seperate and different IAX2 carrier... are u saying there would be issues running calls to both at the same time? |
| 12:15.34 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 12:15.50 | delmar | file, wish my client would beep when i get a priv message :P |
| 12:17.25 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 12:17.36 | dyn | how do I enable the MeetMe application? |
| 12:17.39 | dyn | in the default install? |
| 12:18.44 | dyn | or MeetMe exists only in asterisk 1.2.x, not 1.0.x ? |
| 12:18.49 | dyn | (i see a meetme.conf though) |
| 12:19.17 | lately | dyn: read the handbook. I read that section 30 minutes ago and seemed quite easy. |
| 12:19.45 | dyn | lately: i'm trying to figure my way based on the voip-info.org entry about meetme |
| 12:19.56 | dyn | lately: the handbook doesn't seem to cover this question (checked a few mins ago) |
| 12:20.24 | Zeeek | meetme just works. It's been included since 0.8 or something |
| 12:20.42 | Zeeek | show application meetme |
| 12:21.02 | lately | dyn: See page 117 on the handbook |
| 12:21.46 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:24.40 | *** part/#asterisk chiaBORG (i=chiaborg@concorde.hayburn.net) |
| 12:26.27 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 12:26.33 | nettie | Zeeek thanx anyway.. |
| 12:26.41 | nettie | (got disconnected.. ) |
| 12:27.01 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:27.06 | Zeeek | most decent providers actually have either a config example or a generator |
| 12:27.51 | nettie | yeah they have some examples but it didnt work |
| 12:28.57 | nettie | I mean it's strage doesnt work because I'm sure they've other customers actually using the service with asterisk, and at the same time my setup works perfectly with other 5 carriers. |
| 12:29.14 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 12:29.21 | Zeeek | have you used a SIP debug? |
| 12:29.26 | nettie | sure |
| 12:29.32 | Zeeek | and? |
| 12:30.08 | ghento | Hi there. Was wondering if someone could help suggest a call-limit number for me. I will be on a 1000mbps server, in my sip.conf i have bandwidth=high, allow=all |
| 12:30.20 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 12:30.27 | nettie | and it's pretty strange |
| 12:30.35 | nettie | I mean |
| 12:30.47 | nettie | seems normal but I get a Noisy feedback message |
| 12:31.11 | nettie | I just wanted to doublecheck it with one of the asterisk gurus here :) |
| 12:31.22 | Zeeek | and googling for that phrase has shown what? |
| 12:31.59 | nettie | I did that |
| 12:32.44 | michael_tcf | Stoffel & Zeeek: Great, changing network cards didn't do sh*t :) |
| 12:32.45 | nettie | but everybody seems to say I shouldnt pay attention to it |
| 12:33.56 | Zeeek | in short it means nothing to the end user, and has NO bearing upon quality, problems etc...its just for the techies. |
| 12:34.23 | Zeeek | that's a quote from http://www.voip.org.uk/forums/index.php?action=vthread&topic=1491&forum=6&page=-1 |
| 12:34.38 | nettie | ok |
| 12:34.40 | ghento | Any thoughts? I'm only going to be accepting incoming calls, and would like to allow as many as possible while keeping quality up |
| 12:34.40 | nettie | I just figure out |
| 12:34.43 | nettie | damn |
| 12:34.53 | nettie | firewall issue |
| 12:34.57 | nettie | cant bealive that |
| 12:35.33 | mut | ghento: what are the pc specs |
| 12:35.41 | mut | and is it a LAN? |
| 12:35.46 | *** join/#asterisk papo (n=mathias@84-72-33-221.dclient.hispeed.ch) |
| 12:36.27 | ghento | mut: it's at a big hosting center. pentium 920, 2.8ghz, 1GB ram. |
| 12:36.36 | ghento | this is also going to be shared with an httpd server running |
| 12:37.01 | mut | so all calls are going to be on the lan then |
| 12:37.07 | mut | or are they over the internet? |
| 12:37.23 | ghento | over the net |
| 12:37.43 | ghento | from broadvoice -> my server |
| 12:38.31 | papo | Hi. I have a sip client behind NAT connecting to my asterisk server outside nat over the internet. It's only working with canreinvite=no for outgoing calls, otherwise there is no sound on the calling client. I'd like to stay out of the media path and wonder whther there is an trick to make it work without canreinvite=no, and I'd like to know why exactly it's not working |
| 12:38.33 | mut | depends on a few things then, what kinda load the http server gets and in what bursts, cause that can cause a call to flake if you get a big request fast |
| 12:38.44 | mut | your latency to their server |
| 12:38.56 | papo | I understand some RTP packages are dropped on the box doing NAT, but I don't understand why incoming calls are working |
| 12:38.58 | mut | and if you're going to transcode any calls or stay ulaw the entire path |
| 12:39.11 | mut | latency shouldn't be a problem i wouldn't imagine |
| 12:39.18 | ghento | nope |
| 12:39.25 | mut | you'de probly not need to transcode anything either |
| 12:39.33 | ghento | as for server load, i'm not sure at this point, but i'm using lighttpd to keep memory use, etc as low as possible |
| 12:39.49 | mut | so you could hold a lot of calls |
| 12:40.04 | mut | i'de imagine 60-80 calls |
| 12:40.15 | mut | specailly if you don't do music on hold |
| 12:40.17 | ghento | thats the goal. this is more temporary, until I am able to offload asterisk to it's own server |
| 12:40.22 | ghento | nope |
| 12:40.26 | ghento | okay great, thanks! |
| 12:40.27 | mut | yea |
| 12:40.33 | mut | the server load is next to nothing then |
| 12:40.36 | mut | ought to be fine |
| 12:42.34 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 12:42.40 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 12:44.52 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:45.04 | dyn | lately: back.. well, my handbook only has 71 pages :) |
| 12:45.45 | *** join/#asterisk exonic (n=exonic@sig.triton.net) |
| 12:46.20 | exonic | I'm trying to find that program that starts a traceroute and updates it periodically, Why is remembering program names so hard =) |
| 12:46.53 | dyn | lately: we're talking about this: The Asterisk Handbook Version 2 ? |
| 12:47.55 | dyn | existx: mtr |
| 12:47.58 | dyn | existx: Matt's TraceRoute |
| 12:48.15 | *** join/#asterisk zeppelin_ (n=zeppelin@200-186-48-74.dns.impsat.net.br) |
| 12:49.55 | exonic | dyn, sweet! |
| 12:50.00 | exonic | now I have an acronym |
| 12:50.01 | dyn | existx: :) |
| 12:52.32 | [TK]D-Fender | dyn : you referring to that PDF out a while back? |
| 12:52.49 | dyn | [TK]D-Fender: yep, this one: http://www.digium.com/handbook-draft.pdf |
| 12:52.58 | [TK]D-Fender | dyn : Worthless today |
| 12:53.00 | tzanger | [TK]D-Fender: got a moment for a ip501 question? |
| 12:53.01 | dyn | [TK]D-Fender: but lately's version has about 30 pages more :) |
| 12:53.03 | [TK]D-Fender | dyn : get THEBOOK |
| 12:53.10 | [TK]D-Fender | ~thebook |
| 12:53.11 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 12:53.13 | dyn | [TK]D-Fender: which one are you referring to? |
| 12:53.13 | dyn | ah |
| 12:53.15 | dyn | wow |
| 12:53.17 | dyn | thanks :) |
| 12:54.59 | *** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:58.10 | SheriF_WorK | god damn this MultiVOIP !! |
| 12:58.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:58.12 | SheriF_WorK | ah:-s |
| 12:59.42 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 13:00.29 | uwe | argh, how do i know whats the file name of the recording of some fiile?!! i cant find it in asteriskcdr ! or should i ask in #freepbx ? |
| 13:01.14 | tzafrir | uwe, set verbose 3 |
| 13:01.23 | wasim | MONFILE |
| 13:01.31 | tzafrir | then watch the CLI trace where it records... |
| 13:01.38 | Zeeek | wasim, my man, how are you? |
| 13:01.39 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:01.45 | uwe | thanks all :) |
| 13:01.47 | [TK]D-Fender | tzanger : shoot |
| 13:01.54 | wasim | Zeeek: lately it seems there is no sleep left in this world |
| 13:02.02 | tzanger | [TK]D-Fender: ok, it's about the alert-info stuff |
| 13:02.07 | tzanger | I have it working, but I just want a better understanding |
| 13:02.09 | tzanger | let me pull up the file |
| 13:02.16 | [TK]D-Fender | tzanger : Never really tried it myself. |
| 13:02.34 | [TK]D-Fender | tzanger : Show me and I'll see what I can interpret. That may suffice |
| 13:02.38 | tzanger | [TK]D-Fender: ok |
| 13:02.59 | tzanger | voIpProt.SIP.alertInfo.2.value="VISUAL_ONLY" voIpProt.SIP.alertInfo.2.class="2" |
| 13:03.03 | tzanger | I understand that |
| 13:03.04 | dyn | [TK]D-Fender: i followed the book's instructions closely but still get this: No application 'MeetMe' for extension (office, 501, 1) |
| 13:03.21 | tzanger | if an Alert-Info: VISUAL_ONLY comes in, it goes to see what ringing class 2 says to do |
| 13:03.24 | hensema | can anybody help me with setting up echo cancellation on chan_capi? |
| 13:03.24 | tzanger | that's easy |
| 13:03.54 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 13:04.11 | tzanger | not ring class 2, ring type 2 |
| 13:04.31 | tzanger | but now in <ringType> |
| 13:04.32 | tzanger | <VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/> |
| 13:04.54 | tzanger | it's looking up the '2' there, so why the hell do I need to (or do I?) set VISUAL_ONLY there? |
| 13:05.02 | tzanger | it does not look like it's being used a tall and is only an identifier for us mere humans |
| 13:05.41 | tzanger | if I set the alertInfo config up above to class "7" but leave everything else the same, it looks up class 7, regardless that it's called OOGA_BOOGA |
| 13:06.02 | tzanger | so what is the purpose of naming each ringType? |
| 13:06.14 | tzanger | Polycom doesn't seem to require or need it for anything else |
| 13:06.21 | tzanger | which suggests to me that it must be there for a reason |
| 13:06.46 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-44-211.cust-adsl.tiscali.it) |
| 13:07.26 | [TK]D-Fender | dyn: ... huh? |
| 13:07.35 | *** part/#asterisk tdi (n=tdi@80.48.205.2) |
| 13:07.56 | dyn | [TK]D-Fender: is MeetMe supported in asterisk 1.0.x? |
| 13:08.18 | [TK]D-Fender | dyn : sure |
| 13:08.59 | dyn | [TK]D-Fender: i enabled 2 conference rooms in meetme.conf, hooked them to two extensions, reloaded the config and asterisk claims 'No application 'MeetMe' for extension' |
| 13:10.11 | file | do you have zaptel installed? |
| 13:10.28 | dyn | file: zaptel is for the isdn lines, right? |
| 13:10.41 | dyn | file: (plz forgive me i'm a real noob here) |
| 13:10.47 | file | well more importantly, it has to be installed and a zaptel timing device has to be available... |
| 13:11.01 | file | if it's not installed, meetme will not be built - and you see the message above you gave |
| 13:11.18 | dyn | i have not yet configured the physical lines. i was trying to get the softphones and dialplan set up first |
| 13:11.33 | dyn | file: okay i see your point. most likely that caused my error.. thanks! |
| 13:11.35 | Hmmhesays | wiki wild wild |
| 13:11.53 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 13:12.35 | dyn | (however i'm on ubuntu with that server so no build-time problems can happen here.. but i'll install zaptel anyway and see) |
| 13:13.14 | file | you'll need to recompile asterisk afterwards, so it picks up zaptel is installed and compiles meetme... |
| 13:13.48 | dyn | file: do I get you right you suggest compiling asterisk manually and not to use the ubuntu packages? |
| 13:14.02 | dyn | file: i'm used to gentoo so I think it's no problem, just to get you right :) |
| 13:14.03 | file | if you're using packages, I can't help you... |
| 13:14.17 | dyn | okay |
| 13:14.59 | Sonderblade | i have a problem with my grandstream phone, it seems like when you set it to use dhcp, it sometimes loses configuration such as subnet mask and default gateway |
| 13:15.06 | Sonderblade | have anyone experienced that problem? |
| 13:15.23 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 13:16.01 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 13:18.05 | dyn | file: i guess `find /usr/lib/asterisk/ -iname meetme` should show some results, right? |
| 13:18.43 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 13:19.47 | uwe | wasim, i cant find something related to MONFILE ! |
| 13:20.06 | tzanger | dyn: nope |
| 13:20.21 | tzanger | you need to learn how to use the 'find' command |
| 13:20.43 | tzanger | -iname meetme will find a file named meetme in any case. not meetme.c not areyougonnameetme, just meetme |
| 13:21.44 | wasim | uwe: egrep MON /usr/src/cvs-src/asterisk/doc/README.variables |
| 13:22.00 | [TK]D-Fender | dyn : if you have no "Meetme" then its not compiled in. |
| 13:22.16 | [TK]D-Fender | dyn : Or no timing source to run it on maybe |
| 13:22.41 | wasim | uwe: orlook at the options or the app you're using to record the file |
| 13:22.44 | dyn | tzafrir: i meant meetme* |
| 13:22.47 | dyn | tzafrir: ofc |
| 13:22.58 | [TK]D-Fender | tzanger : Purpose for naming ech ring type is so taht you can have 3 identifiers that maybe all point to the same action calss for NOW, but can be split up later. |
| 13:23.10 | dyn | tzafrir: i'm no find expert for sure (nor asterisk), i've been admin-ing linux only since 97 or so :) |
| 13:23.32 | tzanger | [TK]D-Fender: yes but hte names aren't used for anything |
| 13:23.43 | uwe | thank you wasim |
| 13:23.46 | tzanger | [TK]D-Fender: could you just not have class 2 visual only and class 3 visual only? the name isn't used for anything, not matching, nothing |
| 13:23.53 | *** join/#asterisk dwclarkNU (n=dwclark@129.10.202.97) |
| 13:24.00 | [TK]D-Fender | tzafrir : The name is just the ID for the entry, the ring-type does the rest from what I can see |
| 13:24.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:24.27 | [TK]D-Fender | tzanger : Could be doubled on both ends... |
| 13:24.55 | tzanger | [TK]D-Fender: ? |
| 13:24.58 | tzanger | "doubled on both ends" ? |
| 13:25.02 | dwclarkNU | is there a way to require a caller to enter an access code / password in they try to dial an international number? |
| 13:25.29 | tzanger | dwclarkNU: you can do anything in the dialplan |
| 13:25.38 | tzanger | launch a tactical nuke if you have the equipment to hook up |
| 13:26.14 | tzanger | just have your international dialplan Read() digits and compare them, or launch DISA, dropping into another dialplan that picks up the extension... think about what you want it to do, then codify it |
| 13:26.32 | dwclarkNU | tzanger: thanks |
| 13:27.15 | scanna | hi all |
| 13:27.41 | scanna | does iax2 protocol support fromuser/fromdomain like sip? |
| 13:27.55 | tzanger | scanna: there is no concept of that in iax2 |
| 13:28.29 | scanna | thanks! |
| 13:31.46 | *** join/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 13:34.23 | *** part/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 13:34.26 | *** join/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 13:34.34 | Shark_y | Can anybody help me for a basic dialplan with misdn? THX |
| 13:39.02 | sevard | Does anyone have any used Sipuras they are willing to sell? |
| 13:39.30 | websae | we have lots of sipuras |
| 13:39.35 | websae | but they are new in box |
| 13:40.10 | websae | we actually have a lot of Sipuras, Grandstreams, and Polycoms.. |
| 13:40.26 | websae | in case anyone was looking for a good deal on new ones hehehe |
| 13:41.09 | ghento | hmm, when I installed asterisk, shouldn't it have placed a file in /etc/init.d ? |
| 13:41.24 | Hmmhesays | no |
| 13:41.49 | Hmmhesays | if you used some type of package system to install it *might* have |
| 13:42.05 | ghento | ah, okay. |
| 13:47.47 | sevard | websae: what model sipura / what's your price? |
| 13:50.03 | *** join/#asterisk bufh (n=user@ip216-239-71-75.vif.net) |
| 13:50.07 | bufh | hello |
| 13:52.19 | blitzrage | websae: how much for a polycom? |
| 13:53.06 | blitzrage | ghento: no -- if you're expecting that, you have to run "make config" |
| 13:53.21 | SheriF_WorK | ghenry: how did u install asterisk ? and what is ur distro. ? |
| 13:53.29 | blitzrage | assuming you're running a redhat based distro |
| 13:53.39 | blitzrage | if you're running debian, I'm not sure there is an init script for that |
| 13:53.54 | SheriF_WorK | blitzrage: there is init for debian |
| 13:53.56 | ghenry | Shark_y: FC4 and FC5. From source. |
| 13:54.30 | Splas | websae: yea, whatcha got in the polycom realm? |
| 13:54.35 | SheriF_WorK | ghenry: i don't know about the make config but anyway it should be in the source @ contributes/init/ |
| 13:55.05 | blitzrage | ghento: yah -- just run 'make config' and it'll run chkconfig for you too so asterisk starts automatically on reboot |
| 13:55.08 | ghenry | SheriF_WorK: You should be speaking to ghento ;-) |
| 13:56.29 | ghento | thanks guys! |
| 13:57.25 | SheriF_WorK | ghenry: oh sorry i'm soo tired really :-) sorry to bother |
| 13:57.36 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 13:57.39 | ghenry | ha, it's ok |
| 13:57.58 | SheriF_WorK | ghento: it's on the source and called rc.debian or rc.redhat.asterisk and there is one for zaptel. |
| 13:58.21 | SheriF_WorK | ghenry: trying to get multi tech voip device and asterisk to play along but it's they are not :-) |
| 14:00.21 | Shark_y | ghenry: what do you mean by "FC4 and FC5"? |
| 14:00.46 | ghenry | Shark_y: SheriF_WorK asked about distros. Fedora Core 4 and 5 ;-) |
| 14:01.09 | bufh | hum, i'd like to have a "per call" variable to track some informations about the caller (the language he choose, etc), what should i use ? Set(var=value) ? should i prefix the "var" with "_" ? |
| 14:01.09 | Shark_y | gentoo |
| 14:01.36 | ghento | Hmm, i'm trying to play a wav using Read(). I am seeing the error "format_wav.c:169 check_header: Unexpected freqency 22050" |
| 14:01.37 | Shark_y | ghenry: I'm on gentoo |
| 14:02.00 | ghento | and "ast_filehelper: Unable to open file on /home/supcast/audio/welcome.wav" |
| 14:02.18 | ghenry | Shark_y: And? ;-) Sorry, couldn't resist ;-) |
| 14:03.27 | bkw_ | you need 8000 not 22050 |
| 14:03.27 | bkw_ | duh |
| 14:03.40 | ghenry | can your voicemail setup be on another server? i.e. in extensions.conf exten => _9XXX,2,Voicemail(IAX2/servername/u${EXTEN}@default) |
| 14:04.19 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:04.23 | Shark_y | ghenry: I have an hfc-pci card with mISDN and I cannot place a basic call :( |
| 14:04.27 | MikeJ__ | ghenry, not like that, but yes |
| 14:04.39 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:05.12 | ghenry | MikeJ__: Ah, ok. so I shoul read that application docs more? ;-) |
| 14:05.14 | MikeJ__ | you can just make your "voicmail"server dialplan just be all Voicemail in ext |
| 14:05.42 | MikeJ__ | then just dial over iax or sip to that box instead of calling Voicemail() |
| 14:06.09 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:06.16 | MikeJ__ | MWI will be a littler more interesting in that situation.. |
| 14:06.33 | C4T3l | greetings all |
| 14:06.38 | *** join/#asterisk esandeen (n=sandeen@sandeen.net) |
| 14:06.39 | *** part/#asterisk Freman (n=twitsrus@ppp120-236.lns1.bne3.internode.on.net) |
| 14:07.03 | esandeen | hey al, maybe a bit OT, but can anyone tell me what the linksys "PAP2T" is vs. the "PAP2?" |
| 14:07.21 | esandeen | s/al/all :) |
| 14:07.24 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 14:07.53 | ghenry | MikeJ__: Ah, yes. THe voicemail is handled on the otehr box you are dialing. Silly me. Thanks. |
| 14:08.38 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.140.Dial1.SanJose1.Level3.net) |
| 14:08.57 | *** join/#asterisk timgoh0 (n=timgoh0@unaffiliated/timgoh0) |
| 14:10.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:10.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:11.44 | *** join/#asterisk moruziel (n=moruziel@62.90.121.238) |
| 14:11.52 | moruziel | hi |
| 14:11.56 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.140.Dial1.SanJose1.Level3.net) |
| 14:15.15 | *** part/#asterisk timgoh0 (n=timgoh0@unaffiliated/timgoh0) |
| 14:15.48 | bufh | (ok, forget my question, i found the SetLanguage() method) |
| 14:17.28 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 14:20.47 | ManxPower | "show applications" in the Asterisk CLI |
| 14:22.06 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 14:22.52 | ManxPower | I HATE WINDOWS XP!!!!!!!!!! |
| 14:23.28 | brettnem | oh I can't imagine why |
| 14:23.31 | ManxPower | They have trouble connecting to Samba shares, they don't release their IP address when they move between networks, they have trouble with Apache WebDAV servers |
| 14:23.32 | cytrak | is a dell optiplex with 2.8ghz and 1.5gb of ram ok for running asterisk with IAX2 peers and a TE205P & TE110P card ? |
| 14:24.14 | brettnem | windows+webdav really kinda sucks |
| 14:24.34 | brettnem | cytrak: all depends on what you are going to do with it. |
| 14:24.35 | bufh | manxpower < maybe that's why asterisk has been developed for unixes ? |
| 14:24.48 | sevard | Does anyone have any used Sipuras they want to get rid of? |
| 14:24.52 | sevard | used + working :) |
| 14:24.53 | brettnem | bufh: hey they got asterisk for windows ;) |
| 14:25.24 | bufh | brettnem < yep, but that was not done for it at first, and try to run a windows on a soekris :) |
| 14:25.41 | cytrak | brettnem: well I using it to pass calls from and to the PSTN to a siemens PBX, voicemail, iax2 users to make internal and outside calls |
| 14:26.08 | brettnem | cytrak: how many calls? transcoding? |
| 14:26.39 | cytrak | the thing is I'm getting a low quality on my calls , not sure if its the cheap headset or some rxgain txgain that I have to reconfigure |
| 14:27.18 | ManxPower | brettnem, our users are too stupid to use anything else. |
| 14:27.32 | brettnem | ManxPower: I want to move everything to OSX :) |
| 14:27.37 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:27.51 | ghenry | Hi again. I have type=friend in iax.conf, but keep getting "Rejected connect attempt from" when placing a call to that server? |
| 14:28.09 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:28.15 | cytrak | brettnem: not a huge load , I would say the most 5 calls simultaneously from/to pstn <->asterisk<->siemens |
| 14:28.46 | brettnem | cytrak: transcoding? |
| 14:28.48 | cytrak | brettnem: how can I check the transcoding ? |
| 14:29.08 | brettnem | you shoudl know if you are doign transcoding |
| 14:29.28 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 14:29.30 | cytrak | brettnem: sorry don't know , but would like to check on that |
| 14:29.31 | brettnem | what codecs are you using? |
| 14:29.37 | cytrak | oh ok |
| 14:29.47 | cytrak | ulaw, alaw and gsm are anabled |
| 14:30.18 | brettnem | yeah, you should be fine.. |
| 14:30.32 | brettnem | you might want to be sure that the interrupts for your pstn cards don't share with anything else. |
| 14:31.12 | cytrak | brettnem: but those codec is only for my iax clients right |
| 14:31.23 | brettnem | yep |
| 14:31.32 | ghenry | done it. wrong context |
| 14:31.51 | cytrak | asterisk shouldn;t do any transcoding when the call is just been routed from span2 to span 1 |
| 14:31.55 | cytrak | right ? |
| 14:32.09 | brettnem | most likely |
| 14:32.34 | brettnem | just be sure you arn't sharing interrupts |
| 14:32.53 | cytrak | brettnem: got google to see chow I can check on that |
| 14:33.01 | *** join/#asterisk saftsack (n=saftsack@p54A7E35D.dip.t-dialin.net) |
| 14:33.03 | brettnem | cat /proc/interrupts |
| 14:33.39 | cytrak | ztttol hasn't shown me any IRQ misses if that means anything , I know that before when I had an SATA drive I had a lot of those |
| 14:34.07 | *** join/#asterisk tangel (i=tangel@208.239.77.234) |
| 14:34.16 | tangel | is it possible to port forward an iax or sip connection over ssh? |
| 14:34.21 | tangel | i'm stuck behind a firewall at work =( |
| 14:34.21 | brettnem | well that's important, but you need to check on interrupt sharing |
| 14:34.26 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 14:34.32 | carlos-the-man | hi |
| 14:34.40 | brettnem | tangel: sure it is, but you'll probably have all sorts of crazy SDP problems |
| 14:34.49 | carlos-the-man | please help me setm up my first viop softphone :) |
| 14:34.59 | brettnem | ~gwypf |
| 14:35.03 | jbot | gwypf is, like, Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 14:35.24 | carlos-the-man | I am yanking my hair just trying to find an online instructions that actually work :S |
| 14:35.35 | brettnem | the wiki is very good |
| 14:35.41 | cytrak | brettnem: yeah they are been shared, my 205P is sharing it with some Intel ICH5 and my TE110P is sharing it with some usb modules I thinkg uhci_hcd (twice) |
| 14:36.01 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 14:36.19 | cytrak | brettnem: I don't see IRQ 4 or 13 been used though |
| 14:36.23 | brettnem | cytrak: it's not always a problem, but it'd be best if you can eliminate that sharing |
| 14:36.32 | tangel | brettnem, how's that? i don't know how to forward udp over ssh |
| 14:36.38 | [TK]D-Fender | tangel : Not really... SIP & IAX2 are UDP, SSH = TCP. |
| 14:36.58 | tangel | i see some versions of openssh have a -w option to setup a tun device |
| 14:37.01 | cytrak | does voip-info has any docs on how to do so ? |
| 14:37.06 | tangel | ugh.. i wish there was a tcp version of iax |
| 14:37.17 | [TK]D-Fender | tangel : No you don't :) |
| 14:37.24 | robl^ | you can do ppp over ssh.. but be prepare for a headache ;-) |
| 14:37.42 | tangel | i suspect you're right |
| 14:37.44 | tangel | damnit. |
| 14:38.01 | tanvalmg | Hi there, I have a question on asterisk set-up...is there a way to access the voicemail greetings of each phone in the network and make it as sort of directory (as part of auto-attendant function)? |
| 14:38.09 | brettnem | tangel: if you don't know how to forward stuff over ssh, you are way in over your head.. honestly, the forwarding is the easy part. then you have to get the SDP right |
| 14:38.16 | robl^ | tangel if you want simple.. look into hamachi :-) |
| 14:38.57 | *** join/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 14:39.06 | brettnem | hamachi does look cool |
| 14:39.21 | *** part/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 14:39.56 | dyn | is there any reason why all example uses ignorepat for the number '9'? can I use '0' as the dial-out prefix? |
| 14:40.02 | robl^ | hamachi is fairly neat.. but I can't use it at work. they firewall everything.. only port 80 and 443 go to the outside world. |
| 14:40.12 | stoffell | anyone knows why i see AUTH traffic from the * server to my polycom phones? (tcp port 113) |
| 14:40.26 | robl^ | I can't even ssh from work. they are evil. |
| 14:40.33 | Shark_y | can anybody help me about mISDN? |
| 14:40.58 | tangel | i've managed to tunnel my way past the firewall here (requires 3 hops) |
| 14:41.03 | tangel | i'll look at hamachi.. sounds interesting |
| 14:41.49 | brettnem | robl^: set up a proxy on an outside server on port 80. ;) |
| 14:42.10 | robl^ | brettnem: I did. ;-) |
| 14:43.09 | robl^ | I am VNCing my Mac via an ssh tunnel via an ssl proxy I set up on port 443 ;-) |
| 14:43.13 | tangel | that only works well if your corporate proxy allows CONNECT |
| 14:43.43 | ghenry | yeah! Connection via 2 * boxes over IAX2 working fine! Nice. |
| 14:43.55 | tangel | hehe.. i'm glad i'm not the only one to go to such lengths at work |
| 14:44.01 | robl^ | as far as the firewall here at work is concerned.. I just spend a lot of time on an a secure website ;-) |
| 14:44.03 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org/ -=- Happy Birthday oej!!! |
| 14:44.24 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.216) |
| 14:44.34 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:46.17 | dyn | is there any reason why all example uses ignorepat for the number '9'? can I use '0' as the dial-out prefix? |
| 14:46.58 | [TK]D-Fender | dyn : Sure you can use 0, 9 is just traditional. And ignorepat is for analog zap channels so that you continue to get dial-tone after dialing the leading 9. |
| 14:47.08 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:47.20 | [TK]D-Fender | dyn : You don't even NEED a prfix. its just a common mechanism for other matters of convenience. |
| 14:47.34 | dyn | [TK]D-Fender: ok i get what's ignorepat about, I just find more convenient to prefix outgoing calls by 0 (working the same as most analogue lines) |
| 14:47.51 | dyn | yeah I see I could bind specific extensions to outgoing numbers directly |
| 14:48.38 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:50.17 | [TK]D-Fender | dyn : the advantage for a dial-out prefix is so that lets say you have a 4 digit extension system in the 5000-5999 range, you can have it so the moment the 4th digit arrives it will accept the # as complete and proccess it instantly. The same fo other features like * code functionality. |
| 14:50.45 | dyn | okay, that's reasonable |
| 14:51.19 | dyn | and what if I don't know exactly how many digits the outgoing call's number will be? |
| 14:51.45 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 14:51.53 | dyn | eg we can have 6 or 7 digits for in-city calls or 12 digits for in-country calls, etc? |
| 14:51.57 | [TK]D-Fender | dyn : frankly you can just use a standand NXXXXXX type patter and forget the convenience factor and make it all just feel natural. |
| 14:52.06 | salviadud | ${EXTEN} |
| 14:52.17 | [TK]D-Fender | dyn : it'd just add a small delay before arbitrarily accepting that you are finished your input. |
| 14:52.18 | dyn | [TK]D-Fender: yeah but i need to specify the exact amont of digits in the pattern matching, right? |
| 14:52.26 | [TK]D-Fender | dyn : exactly |
| 14:52.52 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.172) |
| 14:52.53 | dyn | err |
| 14:52.55 | dyn | getting lost |
| 14:53.00 | [TK]D-Fender | dyn : You can tell * exactly what you want to accept like lets say 6 or MORE digits = _XXXXXX. |
| 14:53.00 | dyn | exactly what? ${EXTEN}? :) |
| 14:53.10 | dyn | ahh |
| 14:53.10 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 14:53.17 | dyn | must have been missed that part in the docs |
| 14:53.21 | [TK]D-Fender | dyn : You need to do some "primary" reading on how * works. Go DL THEBOOK now. |
| 14:53.24 | [TK]D-Fender | ~thebook |
| 14:53.26 | jbot | it has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 14:53.30 | dyn | i though XXXXXX would mean exactly 6 digits |
| 14:53.36 | dyn | yeah i'm on the book already :) |
| 14:53.43 | salviadud | X. means 6 digits too |
| 14:53.50 | [TK]D-Fender | dyn : The period at the end is intentional, not jsut part of my sentence :) |
| 14:53.51 | *** join/#asterisk dcornewell (n=dcornewe@proxy5.med-web.com) |
| 14:53.52 | salviadud | or 200 |
| 14:53.55 | dyn | lol |
| 14:54.00 | dyn | okie, thank you :) |
| 14:55.13 | [TK]D-Fender | dyn : so for example : exten => _XXXXXX.,1,Dial(Zap/g1/${EXTEN}) would take anything 6 digits or longer and dial it our the first Zap channel available in Group 1. |
| 14:55.24 | dyn | excellent |
| 14:55.34 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 14:55.36 | dyn | and.. I use Zap for the ISDN line, right? |
| 14:55.55 | [TK]D-Fender | dyn : Not sure about ISDN. Very rare here and never touched it personally. |
| 14:56.04 | dyn | hm strange |
| 14:56.14 | dyn | i thought most of you guys are using isdn for the outgoing calls |
| 14:56.33 | [TK]D-Fender | dyn : ISDN is only a reality in parts of Europe... |
| 14:56.46 | dyn | okay |
| 14:57.01 | [TK]D-Fender | dyn : The majority of the world is on POTS, T1/E1, and VoIP. |
| 14:57.06 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 14:57.22 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:57.23 | mut | http://mdn.mainichi-msn.co.jp/national/news/20060513p2a00m0na027000c.html |
| 14:58.24 | pythos | mornin' gang |
| 14:59.00 | pythos | I have an old DTA310 from when I had Packet8, long ago.. Can I use it for my IP phone under asterisk? |
| 15:01.09 | tzafrir | stoffell, here? |
| 15:01.12 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 15:01.38 | stoffell | tzafrir; ack |
| 15:01.42 | stack_ | I switched echo cancellers for my Zaptel cards... do I need to reload the kernel modules or will ztcfg handle it? |
| 15:02.34 | [TK]D-Fender | mut : Peri-air! |
| 15:03.29 | cytrak | brad_mssw: lspci -vb show a lot more sharing than /proc/interrupts |
| 15:03.56 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:03.56 | stoffell | stack_, i always do a complete ztcfg -s and unload of modules, so maybe that's best yeah. |
| 15:05.03 | dcornewell | Has anyone had the call parking feature lock up on them? transferring someone to park never announces park number and "show channels" shows a parked call that doesn't exist. I have to "shutdown now" and restart to get it back. We've been having this since 1.2.4. |
| 15:06.15 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 15:06.59 | stack_ | stoffell: ok, thanks |
| 15:10.57 | *** part/#asterisk robby (n=robby@host23-229.pool8252.interbusiness.it) |
| 15:11.50 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 15:12.14 | *** join/#asterisk www2 (n=www2@cd4400448.cable.wanadoo.nl) |
| 15:12.24 | cytrak | how do I know if my IDe drive is ATA33 or ATA44 ? |
| 15:13.16 | cytrak | my hdparm -i gives me this |
| 15:13.19 | cytrak | Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2: |
| 15:14.25 | docelmo | And your asking this of us why? |
| 15:14.30 | docelmo | were not #harddrive |
| 15:14.53 | *** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:14.54 | demian_ | there is no ATA44 AFAIK |
| 15:15.08 | cytrak | I'm trying to figure out if that causes any problem with my IRQ issues |
| 15:15.27 | cytrak | according to voip-info.org there is |
| 15:15.37 | Ahrimanes | hm running production on ide drives? |
| 15:15.38 | *** join/#asterisk steve12345 (n=steve@store-fw.porchlight.ca) |
| 15:15.45 | docelmo | I can tell ya HD's are IRQ 14/15 exclusively |
| 15:15.57 | docelmo | I use sata.. |
| 15:15.59 | cytrak | SATA didn';t work |
| 15:16.06 | docelmo | cheaper and better bang for yer buck |
| 15:16.10 | Ahrimanes | sata's good.. ata not good |
| 15:16.16 | *** join/#asterisk lorinc (n=ang@caracas-3821.adsl.interware.hu) |
| 15:16.19 | *** join/#asterisk Celery_General (n=chiv@enforcer.solutionip.com) |
| 15:16.24 | Ahrimanes | well, assuming you buy actual sata drives, not ata with sata interface |
| 15:16.33 | robl^ | I am running production on... compact flash ;-) |
| 15:16.43 | docelmo | Nope actual SATA's |
| 15:16.45 | Ahrimanes | better than ide ;) |
| 15:16.49 | Ahrimanes | well.. ata |
| 15:17.12 | dcornewell | compact flash! watch out. we had some routers on that. it tends fry itself on constant power unless you spend big bucks on it. |
| 15:17.40 | Ahrimanes | hm i run lots of embedded firewalls on cf, havent seen that |
| 15:17.47 | Ahrimanes | long as you dont write to them all the time |
| 15:17.56 | Celery_General | Rules anywhere i should read? or can i just fire off a question to you knowledged ones? |
| 15:18.05 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 15:18.26 | docelmo | Celery_General, 2 rules.. dont flood and only ask questions about asterisk |
| 15:18.36 | robl^ | dcornewell: never seen that. and I keep a backup image of the CF so I can bring it up with a new card in 5 minutes. it's a home / small office system.. not enterprise. 10 phones |
| 15:18.38 | dcornewell | we had like 10 mini atx boards booting from CF. after a year or so the CF was just not readable. |
| 15:18.39 | docelmo | :) |
| 15:19.44 | Ahrimanes | dcornewell: hm did you have /tmp and /var on the cf's ? hehe |
| 15:19.57 | [TK]D-Fender | Celery_General : ... |
| 15:20.00 | [TK]D-Fender | ~suggestions |
| 15:20.02 | jbot | extra, extra, read all about it, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very ... |
| 15:20.22 | dcornewell | ours were in medical offices. driving 10 minutes + 5 minute load is too long ;] we did buy cheap CF. whole install was on it. 512MB CF prolly did have some writes. i think logging was at a minimum for space. |
| 15:20.26 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:21.17 | Ahrimanes | dcornewell: ok, all my cf's are mounted read-only unless at shutdown time.. that helps a lot |
| 15:21.43 | tanvalmg | Is there a eay to access and use voicemail names (greetings) for auto-attendant? |
| 15:21.45 | dcornewell | makes sense. |
| 15:21.46 | Makenshi | I think it would have been nice if they'd made minidisc usable as a data medium |
| 15:22.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:22.28 | Ahrimanes | Makenshi: hm yeah if they'd done it when they launched minidisc.. now it's a bit too low capacity i think.. |
| 15:24.33 | Celery_General | Alright tks... Given having only 1 (ONE) FXO module on a digium card is it possible to transfer a call off to an external phone number from an internal extension?? Normally, outside of asterisk, one could: Flash, Dial, Flash, Disconnect to transfer...... am I explaining myself alright? |
| 15:25.04 | [TK]D-Fender | tanvalmg : Yes, the recordings are easily usable. |
| 15:25.33 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 15:25.35 | [TK]D-Fender | tanvalmg : like "Playback(/var/spool/asterisk/voicemail/default/0001/busy) |
| 15:25.42 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 15:25.43 | [TK]D-Fender | Celery_General : no. |
| 15:26.21 | [TK]D-Fender | Celery_General : forgetabout Flash related functionality on Zap lines. Tha means no call-waiting / transfer or anything else on the, |
| 15:26.58 | froguz | i have configued an E1 PRI as span =>1,1,0,css,hdb3 but when i run zttool it says that the clock is internal |
| 15:27.38 | froguz | i'm also been flooded with HDLC warnings and yellow alarms |
| 15:28.04 | robl^ | ahhh.. mine has minimal writing.. /tmp and /var is a RAM disk. config files don't change often. astdb and cdr only gets written to the CF at shutdown. Even if I have to replace the CF card eery 9 months, its worth it to not have to worry about having a full size PC |
| 15:28.35 | Ahrimanes | yep |
| 15:28.43 | Ahrimanes | same sort of setup on my firewalls :D |
| 15:28.54 | froguz | any clue? |
| 15:29.36 | robl^ | when my asterisk server is 10 ft from my bed, I don't want to listen to the roar of 20 cooling fans at 3am ;-) |
| 15:30.07 | Ahrimanes | robl^: haha |
| 15:30.34 | [TK]D-Fender | froguz : All related I'm sure. Your card on its own IRQ? |
| 15:31.52 | froguz | [TK]D-Fender, the card is sharing the IRQ with some others devices |
| 15:32.11 | froguz | i can't get it taking is own IRQ |
| 15:32.22 | [TK]D-Fender | froguz : Thats a serious problem. |
| 15:32.42 | [TK]D-Fender | froguz : and I'm betting its killing your ability to synch with the telco clock and causing everything else. |
| 15:33.22 | De_Mon | I want to setup an extension to call multiple devices, and then a macro to call those same resources and leave voicemail if no-one answers. |
| 15:33.22 | Sonderblade | is there a way in asterisk to setup so that if a connection attempt takes more than X seconds, asterisk hangs up on you? |
| 15:33.26 | froguz | mmmm... should i replace the motherboard? |
| 15:34.36 | pythos | heh, think I've found a firmware to make my old packet8 310 into something usefull under asterisk!!! |
| 15:34.53 | pythos | flashing firmware now |
| 15:34.57 | [TK]D-Fender | froguz : Play areound with it a bit. change slost, look in the BIOS, pray to whatever God you believe in.... etc.... |
| 15:35.13 | froguz | hahaha |
| 15:35.25 | tzanger | http://www.eff.org/legal/cases/att/att_splash.png |
| 15:35.46 | [TK]D-Fender | tzanger : Death Star is a nice touch ;) |
| 15:35.51 | tzanger | yeah |
| 15:35.55 | dcornewell | So, we've been having this problem. Every so often call parking just quits on us. Transferring someone to 700 never announces park number and "show channels" shows a parked call that doesn't exist. I have to "shutdown now" and restart to get it working again. We've been having this since 1.2.4. And after I "show parkedcalls" the console (asterisk -r) stops responding to commands. every command after that just brings back the prompt. Anyone |
| 15:36.04 | pythos | ahh, now I have a LeadTek BVA8051 !!! |
| 15:36.10 | tzanger | pythos: what's that |
| 15:36.12 | froguz | maybe the card is just broken |
| 15:36.28 | robl^ | [TK]D-Fender: that is God(s)/Goddess[es]/other spiritual super natural entities. (be politically correct). ;-) |
| 15:36.40 | Celery_General | [TK]D-Fender: I don't necessairly need to use flash from the internal extension.... but asterisk can't 'flash' the external line connected to the single FXO (on which there is a call in progress) and then dial an external line to transfer to... ? Or does 'No' mean No? ;) |
| 15:36.58 | [TK]D-Fender | Celery_General : NO <- |
| 15:37.29 | Celery_General | alright thanks. |
| 15:37.43 | froguz | [TK]D-Fender, is it mandatory that the card has is own unshared IRQ? i can't get it to work not even for a second |
| 15:38.12 | pythos | it used to be a packet8 DTA310, useless unless using packet8 as VIOP provider, but now its an adapter I can use that goes between a POTS handset, and an asterisk server! |
| 15:38.29 | robl^ | shared IRQs are bad. VERY VERY bad. |
| 15:39.46 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 15:40.03 | *** join/#asterisk keyhack (n=keyhack@68.236.93.234) |
| 15:40.05 | Dr-Linux | how cna i convert Unix time to simple time? |
| 15:40.10 | Dr-Linux | 1147463372|1147463337.13935|mcp-support|SIP/serverA-89c6|COMPLETECALLER|5|16 |
| 15:40.15 | Nugget | unix time is simple! :) |
| 15:40.21 | Dr-Linux | what does this mean |
| 15:40.36 | Dr-Linux | okey tell me what is this time? >>> 147463372|1147463337.13935 |
| 15:41.01 | Nugget | google for "unix epoch time" |
| 15:41.10 | brettnem | perl localtime. :) |
| 15:41.56 | [TK]D-Fender | froguz : Seems pretty madatory to me.... you know everyone tells you these things all the time.. you should start believing them.... |
| 15:43.50 | froguz | well, i will have to play a lot with it them |
| 15:43.51 | froguz | thanks |
| 15:44.03 | [TK]D-Fender | froguz : What card are you using? |
| 15:45.26 | robl^ | also, PCI 2.2 is a requirement |
| 15:45.51 | dyn | is someone on ubuntu here? |
| 15:46.14 | Celery_General | I am.... |
| 15:46.44 | dyn | Celery_General: if I load hisax + isdn modules with the correct parameters matching my hardware, the /dev/ttyI0 and /dev/isdn/* devices should get created, no? |
| 15:46.57 | dyn | Celery_General: or how do I get those devices created correctly? |
| 15:47.34 | dyn | i believe my gentoo box created the devices as I loaded the kernel module, but I had to move my asterisk experiments to an ubuntu server box |
| 15:47.51 | Celery_General | Sry :)... any takers?? (Don't use asterisk + ubuntu) |
| 15:49.20 | Nivex | A friend of mine runs Asterisk on Ubuntu and doesn't have any problems that I know of. |
| 15:49.42 | dyn | Nivex: i guess he's not on irc, is he :) |
| 15:50.46 | *** join/#asterisk zuobf (n=colinzuo@61.149.156.194) |
| 15:52.58 | redondos | My server seems to be having problems playing audio files. The timer in my soft phone keeps going, but I cannot listen to any audio. Here's the log for one attempt: http://pastebin.com/718917 |
| 15:53.45 | redondos | [TK]D-Fender: Yesterday you helped me fix music on hold. It worked fine. Today I wake up and it's not working anymore. |
| 15:54.12 | [TK]D-Fender | redondos : What kind of file is that? |
| 15:55.39 | redondos | [TK]D-Fender: gsm |
| 15:55.59 | [TK]D-Fender | redondos : Is there any more you can paste about that call? |
| 15:56.04 | redondos | [TK]D-Fender: But the music on hold also doesn't work (with an MP3) |
| 15:56.15 | [TK]D-Fender | redondos : Hmmm |
| 15:57.37 | mercestes | Could it be one way Audio? Ie: it's working, but you just can't hear it? (my random guess) |
| 15:57.51 | redondos | [TK]D-Fender: Here's a bigger log: http://pastebin.com/718929 |
| 15:58.09 | redondos | mercestes: Why would it? It hasn't ever bothered me. |
| 16:01.07 | [TK]D-Fender | redondos : I still doub the quality of your MOH files... |
| 16:02.06 | redondos | [TK]D-Fender: Could that affect the playback of my GSM files? |
| 16:03.03 | file | rumor has it that if you spin around three times, your hair catches on fire |
| 16:03.23 | *** join/#asterisk wm4k (n=wm4k@dsl213-218-233-192.as15444.net) |
| 16:03.34 | tzanger | hmm |
| 16:03.37 | redondos | file: is that a suggestion? :) |
| 16:03.47 | tzanger | iax2 register a peer with a mailbox |
| 16:03.50 | tzanger | I see the MWI there |
| 16:03.53 | file | perhaps. |
| 16:03.58 | tzanger | now if I have a zap channel I want to stutter-tone |
| 16:04.07 | tzanger | I wonder |
| 16:04.33 | *** join/#asterisk heka (n=heka@82.114.68.123) |
| 16:05.40 | file | eep |
| 16:05.49 | trelane_ | how's it going? |
| 16:05.59 | file | meh |
| 16:06.03 | file | slightly hyper |
| 16:06.56 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 16:07.08 | file | and they slowly appear... |
| 16:08.08 | mog_work | oucha |
| 16:10.42 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 16:15.18 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:16.05 | froguz | if i'm connecting an asterisk with a siemens generated E1 line, should i set pri_net or pri_cpe? |
| 16:16.39 | *** join/#asterisk theNOTO (n=biggs@cpe-65-24-200-245.insight.res.rr.com) |
| 16:16.48 | mog_work | most likely cpe |
| 16:16.54 | mog_work | but its real easy to test |
| 16:18.24 | tzanger | wow every time I see that name I think of a nasty joke |
| 16:18.43 | froguz | mog_work, how can i test it? |
| 16:18.50 | mog_work | run it in that mode |
| 16:18.52 | mog_work | if its wrong |
| 16:18.59 | mog_work | asterisk will print out tons of messages saying |
| 16:19.05 | mog_work | hey we are the cpe and so are they |
| 16:21.25 | froguz | thanks |
| 16:21.34 | mog_work | also d chan wont come up |
| 16:21.41 | mog_work | if it is right it will most likely come up |
| 16:23.16 | websae | PRI = fun |
| 16:23.50 | pythos | hi websae |
| 16:24.18 | websae | pythos: hey :) |
| 16:25.35 | pythos | Im still mucking about not knowing enough to even know what I want to do.. BUT, I found an old packet8 DTE310 that I flashed to make into a Leadtek BVA8051 to try it out connecting to asterisk. IM having a bit of a time of it tho. |
| 16:25.51 | *** join/#asterisk ToTo (n=ToTo@host46-89.pool8256.interbusiness.it) |
| 16:26.00 | pythos | BTW, I got dialtone on my two FXO lines :-) |
| 16:26.07 | pythos | no, FXS lines |
| 16:26.15 | websae | well that's good then :) |
| 16:26.20 | pythos | of course there is dialtone ont he POTS lines.. derp |
| 16:26.32 | websae | dial tone is always a good sign |
| 16:26.39 | pythos | indeed |
| 16:26.45 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:27.07 | pythos | I had some static at first, but after I pulled th emodules off and re-inserted them, it cleared up. |
| 16:29.08 | pythos | Now Im trying to config for sending dialtone to that IP handset |
| 16:30.29 | *** join/#asterisk Xen^ (n=linux@203.101.164.31) |
| 16:31.20 | *** part/#asterisk Xen^ (n=linux@203.101.164.31) |
| 16:32.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.76.181) |
| 16:33.54 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 16:34.17 | asteriskmonkey | [TK]D-Fender: I have version 2.x of firmware now |
| 16:35.16 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 16:35.21 | *** join/#asterisk ronn (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 16:35.31 | carlos-the-man | Happy Birthday oej |
| 16:35.43 | *** part/#asterisk dcornewell (n=dcornewe@proxy5.med-web.com) |
| 16:36.15 | carlos-the-man | thanks to computer we are nicer guys because we are remembering everybodys birthdays :D |
| 16:36.33 | salviadud | carlos your english is funny |
| 16:36.55 | salviadud | where you from vato? |
| 16:37.45 | salviadud | ah, jalisco |
| 16:37.48 | carlos-the-man | yes |
| 16:37.58 | salviadud | inche loco |
| 16:38.01 | [TK]D-Fender | asterboy : 2.x of what? Beta PC? :) |
| 16:38.02 | carlos-the-man | lived in usa for a couple years :) |
| 16:38.25 | asteriskmonkey | polycom firmware |
| 16:38.42 | [TK]D-Fender | asteriskmonkey : Beta or official? |
| 16:38.51 | asteriskmonkey | Beta |
| 16:39.12 | [TK]D-Fender | asteriskmonkey : ETA for public consumption? |
| 16:39.14 | *** join/#asterisk Lino` (n=Lino@i577BCC52.versanet.de) |
| 16:39.34 | [TK]D-Fender | asteriskmonkey : Got a release notes file you can forward me on it? |
| 16:39.58 | asteriskmonkey | gah ill put it on that link when i have some time |
| 16:41.05 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 16:42.09 | [TK]D-Fender | asteriskmonkey : Cool... let me know. |
| 16:42.30 | asteriskmonkey | k |
| 16:42.45 | De_Mon | I can change global varialbes from the dialplan, can't I? |
| 16:43.58 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 16:44.25 | *** join/#asterisk viperdude (n=jon@84.45.193.6) |
| 16:44.42 | salviadud | in extensions.conf |
| 16:44.46 | salviadud | you can set globals |
| 16:44.54 | *** join/#asterisk watchy (n=watchy@70.238.57.237) |
| 16:45.05 | watchy | anyone tried the new polycom sip 1.6.6? |
| 16:45.08 | viperdude | hi anyone familiar with forwarding a NAT'd client from ser to asterisk ? |
| 16:45.14 | De_Mon | I want to 'set'/change them in the dialplan |
| 16:45.20 | *** part/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 16:46.17 | De_Mon | there it is under cmd Set. the wiki is so ancient |
| 16:47.09 | [TK]D-Fender | watchy : I have |
| 16:47.41 | De_Mon | someone really needs to break out the 'using variables in asterisk dialplans' by branch (1.1, 1.2 etc) |
| 16:47.56 | watchy | does buddy watch work with more then like 7? |
| 16:48.36 | [TK]D-Fender | watchy : 2 modules + fully loaded and glorious :) |
| 16:48.57 | watchy | you mind if i msg you for a sec? |
| 16:49.02 | [TK]D-Fender | watchy : and with the subscription flush on reload fixed life is good |
| 16:49.09 | [TK]D-Fender | watchy : sure |
| 16:49.38 | watchy | i;m so tired of being sick. i cant get better i think i;m dying |
| 16:49.58 | mercestes | OMG.....*goes googling for Sip 1.6.6 |
| 16:51.13 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 16:51.39 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:51.46 | tzanger | 1.6.6? as in new polycom firmware? |
| 16:51.54 | watchy | yea |
| 16:52.00 | CunningPike | Good morning |
| 16:52.10 | tzanger | watchy: got a changelog? |
| 16:52.23 | *** join/#asterisk dasenjo (i=www-data@208.195.214.137) |
| 16:52.46 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 16:54.10 | *** part/#asterisk heka (n=heka@82.114.68.123) |
| 16:54.32 | dasenjo | Hi, I have a problem setting the calling party number on a pri, I use the SetCIDNum witjout errors but does not work .. *.1.0.10 .. |
| 16:54.35 | dasenjo | can you help me? |
| 16:54.39 | [TK]D-Fender | tzanger : yup |
| 16:55.18 | Damin | dasenjo: Does the Telco that you are sending the PRI to allow you to sed the CallerID information? If not, call them and request it. |
| 16:55.35 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 16:55.44 | tzanger | wow I didn't know sed was a PRI option! |
| 16:57.01 | dasenjo | Damin, I have a pri divided in PBXs .. there are tariffication problems .. and they said me that the need the calling party number info, so I supposed they are expecting it .. |
| 17:01.42 | Hmmhesays | heh |
| 17:02.08 | CunningPike | dasenjo: What switchtype are you using? |
| 17:02.46 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:03.15 | tzanger | [TK]D-Fender: got a link to the changelog? |
| 17:03.33 | [TK]D-Fender | tzanger : Not a link, just the file itself |
| 17:03.47 | tzanger | [TK]D-Fender: can you throw it up somewhere? |
| 17:03.55 | dasenjo | CunningPike: switchtype=euroisdn, pridialplan=national |
| 17:04.02 | [TK]D-Fender | tzanger : Yes, I can vomit on cue... |
| 17:04.06 | blitzrage | [TK]D-Fender: ! ! ! |
| 17:04.13 | tzanger | [TK]D-Fender: :-D |
| 17:04.30 | *** join/#asterisk _Paulo_ (n=Paulo@c906d4b0.virtua.com.br) |
| 17:04.55 | mercestes | Dude! I so want that! |
| 17:04.57 | _Paulo_ | hi people! |
| 17:05.11 | mercestes | That would be awesome....spray stomach acid in someone's eyes when necessary. |
| 17:05.22 | [TK]D-Fender | blitzrage : I don't want to know your name! |
| 17:07.05 | CunningPike | dasenjo: I wonder does EuroISDN transmit CID properly. Can you try other switchtypes with your telco? |
| 17:07.18 | *** part/#asterisk Vhata (i=vhata@shell.rucus.ru.ac.za) |
| 17:07.32 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 17:07.36 | salviadud | -thebook |
| 17:07.40 | salviadud | pfffff |
| 17:07.42 | salviadud | -book |
| 17:07.47 | *** join/#asterisk XARiUS (n=bdarcy@adsl-71-154-210-28.dsl.sndg02.sbcglobal.net) |
| 17:07.50 | salviadud | dammit! |
| 17:07.56 | [TK]D-Fender | ~thebook |
| 17:07.57 | jbot | somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 17:08.13 | salviadud | heeeeeeeey, how come only you can do that? |
| 17:08.26 | [TK]D-Fender | salviadud : Devil's in the details ;) |
| 17:08.31 | blitzrage | you need the tilde -- not hyphen |
| 17:08.40 | dasenjo | CunningPike .. I dont think so .. Im in Colombia .. I did not have use another working switchtypes .. |
| 17:09.07 | salviadud | oh, i was thinking Fender was a BOFH... but now I know he's a good gu |
| 17:09.09 | salviadud | guy |
| 17:09.14 | XARiUS | hey guys, quick q, hopefully someone's seen this before.. production * system, just this morning started throwing back circuit busy on sip devices when called from a QUEUE.. direct sip --> * --> sip device works just fine however? |
| 17:11.23 | CunningPike | dasenjo: Not saying that's your problem, but a quick google search threw up a few posts from people having CID problems with euroisdn. No answers, I'm afraid |
| 17:11.57 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 17:12.02 | salviadud | has somebody succesfully created a .call file with FWD? |
| 17:12.05 | *** join/#asterisk hgaillac (n=Harry@224.15.119-80.rev.gaoland.net) |
| 17:12.15 | hgaillac | Hello, |
| 17:12.55 | salviadud | and why would * not recognize dtmf signals from a party called on voipbuste? |
| 17:13.00 | salviadud | voipbuster |
| 17:13.02 | CunningPike | XARiUS: Are the SIP devices listed as queue members? |
| 17:13.25 | hgaillac | <PROTECTED> |
| 17:13.34 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 17:13.37 | dasenjo | CunningPike: thank you |
| 17:13.49 | CunningPike | dasenjo: np |
| 17:13.53 | salviadud | isn't H323 old as marbles? |
| 17:14.21 | salviadud | i've never configured that audio chan... |
| 17:15.06 | CunningPike | hgaillac: Nope - sorry - not using h323 here |
| 17:15.46 | *** join/#asterisk Tili (i=Tili@202.133.67.64) |
| 17:16.02 | hgaillac | thank CunninPike may somebody here use H323 |
| 17:18.19 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 17:20.48 | hgaillac | anybody is able to configure netmeeting or gnomeeting ? |
| 17:24.40 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:24.45 | algorithmn | * blf w/grandstream problems.. any1 get it working? show hints shows hints.. sip show subscriptions shows the BLF enabled phones... |
| 17:25.49 | algorithmn | but still the lights do nothing |
| 17:26.18 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 17:26.37 | algorithmn | i have no 1000,n(Dial),Dial... set... have seen 0 docs on it |
| 17:27.32 | shanky | hi, just a simple question, I have an asterisk which I can't see any message in the cli |
| 17:27.38 | shanky | any suggestion? |
| 17:27.47 | hgaillac | TCHAO!! |
| 17:28.09 | algorithmn | are you local or remote on the * box? |
| 17:28.46 | shanky | algorithmn: connected usin ssh |
| 17:29.04 | algorithmn | check your /var/log/asterisk/messages |
| 17:29.13 | algorithmn | c if anything relevent is shown in there |
| 17:29.51 | algorithmn | also make sure you connect with /usr/sbin/asterisk -vvvr |
| 17:29.57 | Zodiacal | skype is free? http://www.digg.com/technology/SkypeOut_now_FREE_for_US_Canada_ |
| 17:30.03 | Zodiacal | think that will work with asterisk some how? |
| 17:30.09 | Zodiacal | i have never used skype b4 |
| 17:31.19 | shanky | algorithmn: I can't see anything related to logging/verbose/debugging, and I use asterisk -vvvvr |
| 17:31.28 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 17:31.31 | algorithmn | interesting enough |
| 17:31.39 | Assid | u kidding me? |
| 17:31.44 | Assid | skype free? |
| 17:31.49 | Zodiacal | thats what it says |
| 17:31.49 | zoa | yes |
| 17:31.51 | zoa | skype free |
| 17:31.59 | zoa | us and canada |
| 17:32.01 | zoa | 100% free |
| 17:32.03 | Zodiacal | for usa and canada |
| 17:32.19 | Zodiacal | so can it work with voip phones? |
| 17:32.23 | Zodiacal | & *? |
| 17:32.29 | Assid | u sure of that? |
| 17:32.36 | zoa | yes |
| 17:32.44 | Assid | zoa: i dont see anything like that on their site |
| 17:32.48 | zoa | i did |
| 17:32.52 | zoa | look at digg.com |
| 17:32.59 | Zodiacal | till end of 2006 |
| 17:32.59 | zoa | http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html |
| 17:33.11 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 17:33.11 | websae | til the end of the year |
| 17:33.13 | websae | lol |
| 17:34.05 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 17:34.06 | Shark_y | Is there a place where I can find good dialplan-templates and macros? THX |
| 17:34.16 | Assid | "Both the caller and the number called must be in either the US or Canada" ? |
| 17:34.21 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:34.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.78.18) |
| 17:34.31 | schirpich | Is there a way to manually (through the asterisk console) reset a voicemailbox? |
| 17:34.58 | syle | hmmm |
| 17:35.09 | syle | can we tie into the skype network with asterisk at all |
| 17:35.36 | algorithmn | if a skype usb phone has an rj-11 type connection.. then sure |
| 17:35.51 | schirpich | i have two cisco 7940's that show they have voicemail, but when you log into the vm box to check it, there is actually nothing there |
| 17:36.20 | *** join/#asterisk RoyK (n=roy@212.17.141.54) |
| 17:36.34 | syle | algorithmn : do elaborate |
| 17:36.55 | algorithmn | syle, on blf? |
| 17:36.57 | [TK]D-Fender | schirpich : Thats Cisco "not getting the picture", not *'s fault |
| 17:37.12 | algorithmn | syle, on u mean skype |
| 17:37.18 | syle | yes |
| 17:37.20 | algorithmn | k |
| 17:37.33 | schirpich | [TK]D-Fender, i dont doubt it :P but how do i fix it ! |
| 17:37.47 | blitzrage | OT: http://video.google.com/videoplay?docid=-869183917758574879 |
| 17:37.48 | algorithmn | if there is a skype usb phone w/an rj-11 type connection.. that supports connecting a standard analouge phone... |
| 17:38.00 | algorithmn | feed it into a fxo port |
| 17:38.15 | Assid | that sucks.. i gotta be in us to get it |
| 17:38.18 | CunningPike | shanky: Do you have a pending 'stop when convenient'? |
| 17:38.20 | *** join/#asterisk saftsack (n=saftsack@p54A7CCE4.dip.t-dialin.net) |
| 17:38.28 | algorithmn | then bam.. set up your extensions for outbound then start using skype w/* |
| 17:38.42 | algorithmn | im unsure of skype hardware though |
| 17:38.59 | syle | thats one way to do it for sure |
| 17:39.02 | algorithmn | that is the dependant variable... or you can write a chan_skype... |
| 17:39.20 | algorithmn | i know im not going to write chan_skype... |
| 17:39.22 | algorithmn | lol |
| 17:39.23 | syle | i beleive their closed source, i think that would be difficult |
| 17:39.42 | algorithmn | im for sure mr spenser didn't say that about gaim |
| 17:39.45 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 17:40.14 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 17:40.24 | syle | there were already modules for connecting to MSN AOL etc on CPAN by then, i don;t think he had much problems |
| 17:40.34 | algorithmn | i didn't know that |
| 17:40.35 | saftsack | hi |
| 17:40.47 | algorithmn | hey |
| 17:40.48 | saftsack | are there some news related to the b410p card? |
| 17:40.52 | algorithmn | ? |
| 17:41.03 | saftsack | it is a isdn card from digium |
| 17:41.03 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 17:41.06 | algorithmn | syle, pm me if you find a skype phone w/those requirements |
| 17:41.25 | Assid | bah.. thats ridiculous.. why limit the free calls from the US only |
| 17:42.16 | zoa | because they also dont need to pay for them in the us |
| 17:42.21 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 17:42.27 | Zodiacal | ahh |
| 17:42.32 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 17:42.58 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 17:44.40 | tzanger | heh |
| 17:44.49 | tzanger | ther'es something funny about a guy named "winkie" with a username "urmom" |
| 17:46.11 | docelmo | ??? |
| 17:46.12 | docelmo | huh? |
| 17:46.19 | docelmo | winkie urmom |
| 17:47.11 | tzanger | 13:43 -!- Winkie [n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com] has joined #asterisk |
| 17:47.48 | *** join/#asterisk ckwall (n=ckwall@65.218.229.224) |
| 17:47.52 | syle | kinda interesting |
| 17:48.10 | syle | now watch every voip provider use a skype phone for some of their calls outgoing lol |
| 17:48.14 | ckwall | has anyone out there successfully set up Windows ISA firewall to allow SIP traffic into the asterisk server? |
| 17:48.35 | ckwall | I hear that this is not possible... is that true? |
| 17:49.03 | ckwall | I have opened ports 5060 and 10000-20000 and am still not getting through |
| 17:49.13 | ckwall | It is dropping the packets at 5060 |
| 17:49.53 | docelmo | eww winDOZ |
| 17:49.59 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:50.02 | brettnem | ISA? gag |
| 17:50.05 | ckwall | I agree |
| 17:50.15 | docelmo | hell get a PIX its not much better.. |
| 17:50.20 | docelmo | Im bout ready to get rid of mine |
| 17:50.24 | CunningPike | ckwall: http://www.voip-info.org/wiki-Asterisk+firewall+rules |
| 17:50.36 | CunningPike | ckwall: There's a section in there about ISA |
| 17:50.41 | CunningPike | ~google |
| 17:50.47 | jbot | somebody said google was a search engine found at http://www.google.com/ |
| 17:50.51 | CunningPike | ;) |
| 17:51.30 | ckwall | i am running asterisk on a linux box... isa server is infront of it |
| 17:52.03 | ckwall | been googling for 2 days... nothing says it is possible... |
| 17:52.11 | syle | algorithmn: http://share.skype.com/directory/ws:3200_voip_usb_to_rj11_connector/view/ |
| 17:52.24 | ckwall | but I dont understand how that can be, isnt tcp, udp all standard? |
| 17:52.44 | CunningPike | ckwall: It is. However, MS breaks those standards all the time |
| 17:53.07 | schirpich | how can you find out more information about the 911 services that a asterisk box is connected to |
| 17:53.34 | ckwall | so am i misunderstanding the isa portion of the wiki-asterisk? but that seems to be how to set up asterisk on the ISA server. |
| 17:54.09 | CunningPike | ckwall: You're right - my bad |
| 17:54.33 | MikeJ[Laptop] | ckwall, thats not likely.. |
| 17:54.54 | MikeJ[Laptop] | did somone really say microsoft breaks the standards for udp? |
| 17:54.55 | brettnem | schirpich: ask your provider |
| 17:55.20 | syle | why not they broke them for icmp |
| 17:55.28 | *** join/#asterisk keyhack (n=keyhack@68.236.93.234) |
| 17:55.36 | CunningPike | syle: Exactly |
| 17:55.38 | ckwall | so has anyone done this successfully? |
| 17:55.46 | schirpich | i basicly just want to test it and make sure that it works. but how do you call 911 and say, "this is just a test?" |
| 17:55.48 | websae | schirpich: who is your provider? |
| 17:55.48 | keyhack | Is there a way in Asterisk to limit by MAC address? |
| 17:55.52 | brettnem | ckwall: it sounds like an overwhelmingly bad idea |
| 17:55.56 | schirpich | i believe its XO Communications |
| 17:56.01 | CunningPike | schirpich: What we do it exactly that |
| 17:56.08 | CunningPike | s/it/is/ |
| 17:56.12 | brettnem | schirpich: call your provider and ask them.. it's different for everyone |
| 17:56.45 | brettnem | you shouldn't just call 911 and tell them it's a test.. 911 calls should be scheduled with the appropriate 911 agency administration |
| 17:56.56 | schirpich | yea, i dont want any police or fire dept showin up, i think i'll call XO |
| 17:57.09 | websae | don't call 911 to "test" |
| 17:57.21 | websae | check with your provider |
| 17:57.46 | brettnem | some 911 agencys have rules on when you can test.. Houston for example doesn't allow tests on monday or friday or on weekends and all tests have to be done before 2pm.. something like that.. |
| 17:58.09 | MikeJ[Laptop] | I've done 911 tests before.. it's fin |
| 17:58.11 | MikeJ[Laptop] | fun |
| 17:58.17 | file | or... phun? |
| 17:58.21 | MikeJ[Laptop] | no |
| 17:58.25 | file | :( |
| 17:58.27 | brettnem | they get something like 5,000 calls a day.. everything from heart attacks to asking for directions. |
| 17:58.48 | schirpich | how do you find out what the rules for testing are,,, the provider i suppose? |
| 17:58.49 | brettnem | I've done several hundred 911 tests.. |
| 17:59.01 | brettnem | the provider should give you some guidance.. |
| 17:59.14 | brettnem | I've worked directly with the agencies, but you should work with your provider |
| 17:59.15 | websae | if you can't call 911 --- then sue |
| 17:59.21 | websae | and use youre cell phone |
| 18:00.02 | lunk | 911 works against natural selection, so i can't support it. |
| 18:00.23 | brettnem | haha |
| 18:00.36 | brettnem | I dont' know if it actually does.. |
| 18:00.43 | sphing | whats a good voip provider that allows asterisk connections? |
| 18:00.47 | lunk | bike helmets also work against natural selection ;) |
| 18:00.52 | CunningPike | Oh, it does, believe me |
| 18:01.04 | brettnem | what about soap and toothpaste? |
| 18:01.08 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 18:01.21 | CunningPike | brettnem: Microsoft broke the SOAP standard too |
| 18:01.23 | MikeJ[Laptop] | shido6, there are lists on the wiki |
| 18:01.24 | lunk | brettnem: bad breath only kills others :D |
| 18:01.37 | MikeJ[Laptop] | err.. sphing, that was for you.. |
| 18:01.39 | MikeJ[Laptop] | shido6, hi. |
| 18:01.46 | CunningPike | OK - this is getting silly - back to work, everyone ;) |
| 18:01.53 | algorithmn | syle, that looks like it will do it simply |
| 18:02.05 | sphing | MikeJ[Laptop]: asterisk wiki or voip-info wiki? |
| 18:02.49 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 18:03.13 | file | MikeJ[Laptop]: fyi you are hereby required to run around in circlez |
| 18:05.42 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 18:06.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:08.59 | keyhack | Is there a way in Asterisk to limit access by MAC address? |
| 18:09.21 | sphing | limit access? |
| 18:09.28 | sphing | as in stop access |
| 18:09.38 | CunningPike | Ouch |
| 18:12.37 | ManxPower | keyhack, NO! Since the MAC address does not get sent across routers |
| 18:13.22 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 18:13.25 | keyhack | so how do I prevent someone from figuring out the user/pass to register to the extension and receiving phone calls? |
| 18:13.55 | ghenry | damn my aatra 9133i phones arrived today, with no sys-admin install guide, only a user guide. what gives! |
| 18:14.17 | ManxPower | keyhack, Um, how do you prevent people from figuring out ANY username/password? |
| 18:14.55 | sevard | hhaha |
| 18:15.26 | ghenry | oh, it is 165 pages acordin got their website, so that's why ;-) |
| 18:15.26 | keyhack | ManxPower: You add additional security measures |
| 18:15.29 | ManxPower | What we do is set the username/password of each SIP device to be the MAC address. Since MAC addresses are not sent across routers, the only issue is if someone has access to our local network, and if that's the case we have much larger problems that someone knowing the auth information of the IP phones. |
| 18:15.36 | ManxPower | keyhack, such as? |
| 18:16.01 | keyhack | ManxPower: I don't know what applies to SIP devices and Asterisk |
| 18:16.47 | sevard | I think he's asking if there is any sort of SIP encryption so the passwords aren't sent plain text |
| 18:16.48 | ManxPower | SIP is MORE secure in some ways because there is no actual reason for the user to know what the auth information is. |
| 18:16.58 | ManxPower | have the auth information preconfigured and don't let the users change it. |
| 18:17.13 | cytrak | hm I'm checking a file called codecs.conf on asterisk which I haven't yet messed with and I see something about codec complexity . |
| 18:17.14 | ManxPower | sevard, you mean like the stuff in sip.conf.sample? |
| 18:17.15 | websae | if anyone is looking for good deal on SIP phones or ATAs ....should talk to me hehehe, have lots :) |
| 18:17.22 | cytrak | mine is set to 2 |
| 18:17.23 | keyhack | ManxPower: And what if I was to run a PBX out of a colo and allow a few small businesses rent the PBX from me rather than investing in one themselves? |
| 18:17.26 | ManxPower | I recall MD5, but chan_sip might support RSA auth now. |
| 18:17.36 | cytrak | is that ok ? |
| 18:17.37 | sevard | ManxPower: I don't know what you're referencing, i'm just trying to decypher his question |
| 18:18.07 | ManxPower | keyhack, Then I guess it's up to those businesses to keep their sip auth information secure. |
| 18:18.30 | keyhack | ManxPower: It's being sent in the clear, how could they do that? |
| 18:18.45 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:18.46 | ManxPower | keyhack, it does not have to be sent in the clear. |
| 18:18.50 | sevard | SIP uses tolkins like http, correct? |
| 18:19.10 | ManxPower | keyhack, please read /path/to/src/asterisk/configs/sip.conf.sample Then come back |
| 18:19.23 | keyhack | ManxPower: Typical. |
| 18:19.48 | ManxPower | keyhack, better make sure their SMTP, IMAP, POP3, and HTTP passwords are not sent cleartext as well |
| 18:19.58 | sevard | keyhack: I'm sorry guy, but if RTFM is all you get for an answer you're asking the wrong questions |
| 18:20.32 | keyhack | yeah, because asking about sharing the PBX with multiple companies and trying to make it secure as possible is the wrong question to ask |
| 18:20.42 | keyhack | who cares? let it go plaintext |
| 18:20.59 | keyhack | its not like you care if some kid can register and start receiving your businesses phone calls |
| 18:21.03 | sevard | I find it a little alarming that RTP streams aren't encrpyted yet. THat's my only beef |
| 18:21.12 | ManxPower | keyhack, if you want to. Personally I try not to send plain text passwords out across the internet. |
| 18:21.16 | sevard | keyhack: send all your information over a VPN |
| 18:21.31 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 18:21.57 | ManxPower | You can use MD5 if you don't want plaintext (as anyone would know that looked at sip.conf.sample) and if you want REAL security then, of course, you would want some kind of VPN. |
| 18:22.08 | Zodiacal | do you guys think skype can be used by those wifi phones? |
| 18:22.15 | keyhack | ManxPower: Wouldn't that require the device supporting MD5 then? |
| 18:22.28 | keyhack | ManxPower: And whats stopping someone from seeing the MD5, and just using that? |
| 18:22.36 | ManxPower | keyhack, most devices these days seem to support it. |
| 18:22.38 | ManxPower | You can use MD5 if you don't want plaintext (as anyone would know that looked at sip.conf.sample) and if you want REAL security then, of course, you would want some kind of VPN. |
| 18:23.16 | ManxPower | keyhack, no matter what you do, if you want to have secure passwords then you need both sides to support it. |
| 18:23.49 | ManxPower | permit/deny will let you limit access by IP address and/or network block, MD5 passwords, VPN, etc. |
| 18:23.51 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-58.apple.com) |
| 18:24.08 | keyhack | ManxPower: Of course, but having MD5 and just merely submitting a MD5 hash is no more secure than submitting a plaintext password if its just a flat out MD5 |
| 18:24.16 | ManxPower | it's not rocket science, and it's not a new topic. It's talked about every few months on the mailing lists. |
| 18:24.44 | sevard | keyhack: the most secure thing you could do would be to put up a VPN |
| 18:24.51 | ManxPower | keyhack, You have valid points. What do you expect people to do about it? |
| 18:24.59 | keyhack | sevard: ManxPower: I'm just curious to how people like BroadVoice and Vonage do it |
| 18:25.02 | sevard | i read an article that said VPN encapsulation actually inscreases the quality of the call for some reason |
| 18:25.02 | keyhack | or maybe they don't |
| 18:25.20 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-58.apple.com) |
| 18:25.22 | keyhack | maybe the registration does occur in plaintext |
| 18:25.25 | ManxPower | keyhack, do a packet dump. |
| 18:25.35 | keyhack | ManxPower: Thanks, didn't think of that one. |
| 18:26.09 | ManxPower | keyhack, any time you say "I wonder how" when refering to something using IP, the answer is "do a packet dump" |
| 18:26.26 | sevard | keyhack is super smart. maybe his genious will think of a way to develop to make SIP more secure out of the box |
| 18:26.27 | tamp4x | any t38 gurus here |
| 18:26.35 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:27.29 | _Paulo_ | * has t.38 support now? |
| 18:27.41 | ManxPower | _Paulo_, no |
| 18:28.10 | *** join/#asterisk jonson_ (i=ident@81.12.217.11) |
| 18:28.20 | jonson_ | hello ppl |
| 18:28.29 | _Paulo_ | I mess with fax a lot. |
| 18:28.46 | ManxPower | I do too, but I almost never have problems with faxes. |
| 18:29.09 | jonson_ | someone can explain me how to set my asterisk to accept push to talk clients from nokia phones? |
| 18:29.11 | sevard | ManxPower: I get about a 60%-75% goodness on fax |
| 18:29.20 | sevard | ManxPower: have you ever tried a credit card machine? |
| 18:29.35 | ManxPower | sevard, Those work fine too. What to know why? |
| 18:29.47 | *** join/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 18:29.49 | _Paulo_ | nor do I, I let hylafax do the fax routing, |
| 18:29.51 | sevard | ManxPower: Yeah |
| 18:29.54 | ManxPower | I send those calls out local analog lines. |
| 18:30.03 | sevard | hahaha |
| 18:30.04 | ManxPower | I NEVER send data over voice over ip |
| 18:30.31 | sevard | How about SIP -> * -> analog |
| 18:30.38 | ManxPower | I always have 1 - 2 local analog lines at client locations for fax, modem, CC, and 911 |
| 18:31.05 | ManxPower | those analog lines are also never tied into Asterisk |
| 18:31.29 | jonson_ | someone show my question?:) |
| 18:31.33 | robl^ | who uses FAX? even in a lawfirm, we use scanners -> PDF to email |
| 18:32.03 | sevard | robl^: a _lot_ of people still use fax machines even though they cost a million times more than a damn email |
| 18:32.05 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 18:32.22 | VoicePulse | A notice to any VoicePulse Connect for Asterisk users still using the gw5.voicepulse.com server scheduled to be decommissioned today -- in a few minutes, outgoing calls will stop working through gw5. Please login to the Account Center for instructions on migrating to a newer server if you have not done so already. |
| 18:32.39 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 18:32.46 | tamp4x | i use ser for fax t38 routing |
| 18:32.48 | sevard | VoicePlus has a bot in #asterisk? |
| 18:32.51 | sevard | that's freaking awesome |
| 18:32.52 | _Paulo_ | people here in the office uses the fax as a scanner... :-) |
| 18:33.07 | VoicePulse | I wouldn't say I'm a bot, though I often talk like one :b |
| 18:33.08 | ManxPower | most of our users are too stupid to use a scanner |
| 18:33.16 | sevard | Hahahaha |
| 18:33.27 | b00mer_ | anyone know how to upgrade to SIP on a really old version of non-sip software on a cisco 7960? |
| 18:34.00 | b00mer_ | usually... it checks the OS79XX.TXT and does it... but this one is very very old |
| 18:34.00 | ManxPower | sevard, And I quote: "I don't want to be notified of voicemail with a text message. It's too complicated to read text messages on my cell phone." |
| 18:34.18 | sevard | oih |
| 18:34.19 | sevard | my |
| 18:34.20 | robl^ | hehehe. we have several HUGE Xerox Document Centers (like a large industrial photocopier) that will scan documents to PDF and email it to the user's Inbox.. then it can be forwarded anywhere as needed. |
| 18:34.20 | sevard | gawd. |
| 18:34.27 | sevard | shoot your users ManxPower |
| 18:34.40 | ManxPower | sevard, don't tempt me. |
| 18:34.49 | Assid | ManxPower: how do you sent voicemail as text message? |
| 18:35.20 | jonson_ | someone can explain me how to set my asterisk to accept push to talk clients from nokia phones? |
| 18:35.21 | dpryo | ManxPower: The solution is ofcourse, have festival call the person and text-to-speech the text message about his voicemails :D |
| 18:35.26 | ManxPower | Assid, But we don't send voicemail via a text message, we (would) send NOTIFITCATION via a text message |
| 18:35.44 | Assid | so the ligth just blnks and it shows as 1 message? |
| 18:35.44 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 18:35.58 | ManxPower | dpryo, the system calls their cell phone and TELLS them they have voicemail and let them get it |
| 18:36.05 | Assid | okay i wanna be able to send text messges through the phone |
| 18:36.31 | robl^ | I send voicemails as soundfile attatchemnts to my cell phone's email. saves on my cell minutes |
| 18:36.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:36.59 | sevard | only if your client has a good data plan |
| 18:37.09 | mercestes | ......WHy don't you ppl just forward your phones??? I have my voicemails forwarded to my work email as an attachment and I get to them when I can check my email. |
| 18:37.16 | ManxPower | for the most part the magic words to get users off my back is "I'll get you a quote for a custom programming project to implement your request." |
| 18:37.32 | robl^ | I have unlimitted data. ;-) |
| 18:37.42 | ManxPower | mercestes, many of my users don't have computers |
| 18:37.53 | Assid | hrmm.. im trying this 123 example on : http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText , but it doesnt seem to work for me |
| 18:38.00 | Assid | i dont see the hello world |
| 18:38.02 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:38.04 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:38.06 | Assid | but asterisk says it sent it |
| 18:38.08 | Hmmhesays | i'll unlimit your data |
| 18:38.14 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:38.15 | ManxPower | mercestes, the RECEPTIONIST at 2 of the offices can't forward the front desk phone correctly more than %80 of the time. |
| 18:38.24 | ManxPower | usually she forgets the leading "9" and never tests ot |
| 18:38.47 | *** part/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:38.50 | ManxPower | Assid, I do not know of any SIP phones that support send text |
| 18:38.53 | sevard | you need a receptionist with an iq higher than a brick's |
| 18:39.00 | ManxPower | well, I'm off to the bank |
| 18:39.01 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:39.17 | Assid | ManxPower: dont polycomes support text? |
| 18:39.23 | robl^ | ManxPower: bring me back some money |
| 18:39.26 | [TK]D-Fender | ManxPower : You should abolish leading digits... just dial everything naturally. |
| 18:39.39 | [TK]D-Fender | Assid : Polycom yes, * no. |
| 18:39.50 | ManxPower | assaf, Yes, via MSN, SIP, or AIM, not SendText |
| 18:39.57 | robl^ | Asterisk can SEND text.. just not receive it |
| 18:40.15 | ManxPower | SendText is an Asterisk only protocol. |
| 18:40.20 | *** join/#asterisk dwclarkNU (i=dwclarkN@155.33.196.139) |
| 18:40.23 | robl^ | Snom's support SendText |
| 18:40.39 | sevard | say ManxPower before you leave |
| 18:40.43 | Assid | [TK]D-Fender: but if asterisk is saying sending text.. shouldnt my phone get it ? |
| 18:40.50 | keyhack | [TK]D-Fender: I have having to dial 9 first, annoying |
| 18:40.54 | sevard | would you happen to have any cheap used Sipura ATAs for sale? |
| 18:41.07 | keyhack | [TK]D-Fender: And cell phones destroyed the habit of dialing 1 first as well |
| 18:41.27 | [TK]D-Fender | Assid : No Idea... never tested it... |
| 18:41.29 | sevard | keyhack: then match and prepend |
| 18:41.37 | [TK]D-Fender | keyhack : why is that? |
| 18:41.38 | keyhack | sevard: I know, if I ruled the world, but I don't |
| 18:41.46 | keyhack | [TK]D-Fender: Why is what? |
| 18:41.53 | Assid | [TK]D-Fender: so theres no way to flash some text on the phones' screen? |
| 18:41.57 | sevard | keyhack: or you could pipe the line into your * and start ruleing that |
| 18:42.16 | [TK]D-Fender | key2 : why do you have to dial 9 first? |
| 18:42.37 | [TK]D-Fender | Assid : Yeah you can force a way, but you can't use the phone built-in IM function to SEND it. |
| 18:42.54 | keyhack | [TK]D-Fender: Because someone persistent jerk keeps insisting we have it that way because "I've never worked at a company where you didn't have to dial 9 first to get an outside line" |
| 18:43.02 | justinu | lol |
| 18:43.07 | robl^ | Assid: you can flash text messages on Snom phones from asterisk |
| 18:43.13 | *** join/#asterisk Uchiha_Itachi (i=jjjjj@93-239-222-201.adsl.terra.cl) |
| 18:43.14 | justinu | what an excellent reason |
| 18:43.30 | Assid | hrmm |
| 18:43.34 | keyhack | justinu: Yes, I love it |
| 18:43.35 | JonR800 | yes.. because if everyone else jumped off a bridge... |
| 18:43.55 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 18:43.56 | dwclarkNU | keyhack: sounds like a real jerk |
| 18:44.03 | keyhack | dwclarkNU: Yeah, he does ;-) |
| 18:44.23 | [TK]D-Fender | keyhack : Dumb reason. Go change that behind his back! |
| 18:44.30 | JonR800 | tell him it's a measure to reduce RSI... |
| 18:44.32 | sevard | keyhack: if it's an asterisk box create an extension that dials 9 out and create matching and prepending exensions for yourself |
| 18:44.36 | keyhack | dwclarkNU: Funny how he also wanted 4 digit extensions for a 2 person company! |
| 18:45.09 | techman97_andy | hey all - I have one perplexing situation. I have a workstation that is using xLite, it's working fine. Just shipped out a GrandStream GXP-2000 that I setup at our office with the SIP account settings, etc - and it successfully logged in and made / received calls. Now at this guy's house, he plugs the phone in and I get a "wrong password" message in the * CLI when he powers the phone up. Any ideas? |
| 18:45.11 | dwclarkNU | keyhack: probably for expansion later, so the extneions would have to be reconfigured when the 1000th person joined |
| 18:45.12 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 18:45.28 | techman97_andy | I've reset the password in sip.conf and reloaded the configuration...and changed it on the phone to no avail. |
| 18:45.39 | techman97_andy | the xLite phone can be turned back on and logs in just fine |
| 18:45.40 | techman97_andy | wtf? |
| 18:45.48 | keyhack | dwclarkNU: Possibly, but then again, the chances of having over 100 employees within the next ten or so years is slim to none, and the comapny would reorganize and shift around so much before then it wouldn't make sense anyways |
| 18:45.56 | Hmmhesays | its the damn gremlins |
| 18:46.07 | sevard | sup bitch |
| 18:46.35 | Hmmhesays | why does it take the cable people a fucking week to get here |
| 18:46.47 | justinu | cable? ouch |
| 18:46.52 | Hmmhesays | hey, its free |
| 18:46.57 | Hmmhesays | 5meg down |
| 18:46.59 | MikeJ[Laptop] | Hmmhesays, becuase you live in the middle of nowhere...it's a long drive? |
| 18:46.59 | justinu | cable companies aren't used to doing things fast, or right... because it was just TV |
| 18:47.06 | sevard | I live in a tiny town and it still takes the cable guys a week to get to your house |
| 18:47.07 | gmfm | better than the phone company... they would take a month |
| 18:47.19 | justinu | telco here is a lot more serious about things |
| 18:48.16 | Hmmhesays | yeah wtf town is that |
| 18:48.26 | *** join/#asterisk RF_MIA (n=unknown@adsl-070-147-214-250.sip.mia.bellsouth.net) |
| 18:48.45 | techman97_andy | any ideas on my issue - I'm pulling my hair out on this thing |
| 18:48.48 | techman97_andy | ? |
| 18:49.02 | sevard | Hmmhesays: you still haven't figured that shit out? |
| 18:49.09 | Hmmhesays | its not foremost on my mind either |
| 18:49.19 | sevard | it is if you want a night of free beer |
| 18:49.27 | sevard | as in four |
| 18:49.30 | sevard | my budget is limited |
| 18:49.31 | sevard | ;) |
| 18:50.00 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 18:50.42 | CunningPike | techman97_andy: What does a 'sip debug' say? |
| 18:50.48 | techman97_andy | running that right now |
| 18:50.55 | CunningPike | ;) |
| 18:50.59 | techman97_andy | =) |
| 18:51.13 | sevard | CunningPike Hmmhesays [TK]D-Fender do any of you have a cheap used ATA you could pawn off on me |
| 18:51.37 | CunningPike | sevard: Not really - we're using all the ones we have - sorry |
| 18:51.43 | sevard | bah |
| 18:51.46 | Hmmhesays | i've got an h.323 one |
| 18:51.49 | CunningPike | sevard: eBay? |
| 18:51.58 | sevard | I can get a SIP 1000 for 37.00 on ebay |
| 18:52.06 | sevard | I was just looking for a better price on maybe a used one |
| 18:52.09 | [TK]D-Fender | sevard : the shipping would kill the value :) |
| 18:52.14 | Hmmhesays | haha not from here |
| 18:52.16 | sevard | [TK]D-Fender: where you at |
| 18:52.20 | Hmmhesays | since i'm like an hour away |
| 18:52.25 | [TK]D-Fender | sevard : Montreal, QC :) |
| 18:52.44 | sevard | yeah you're about an hour away |
| 18:52.46 | Hmmhesays | i have a bunch of h.323 only ata's |
| 18:52.47 | techman97_andy | oh wtf....no password and it works. Setting a password now. wtf. |
| 18:52.49 | robl^ | mmmmmmmmmm....BEER! |
| 18:52.50 | sevard | I' |
| 18:53.05 | sevard | I've never used h.323 and didn't know they made only h.323 atas |
| 18:53.07 | [TK]D-Fender | robl^ : Beer GOOOOOODDDD! |
| 18:53.22 | sevard | [TK]D-Fender: Montreal isn't _that_ far |
| 18:53.23 | Hmmhesays | before sip rose to power h.323 was king |
| 18:53.24 | [TK]D-Fender | robl^ : Go check out the Camp Chaos Metallica SWF's :) |
| 18:53.31 | robl^ | esp when hacking on a dialplan ;-) |
| 18:53.33 | [TK]D-Fender | sevard : You're in CA, no? |
| 18:53.39 | sevard | [TK]D-Fender: close enough |
| 18:53.48 | sevard | Hmmhesays: what about a firmware upgrade? :\ |
| 18:53.51 | Hmmhesays | [TK]D-Fender he's in the same frozen wasteland i'm in |
| 18:54.09 | sevard | Hmmhesays: have you been getting the month long rainy day too? |
| 18:54.20 | sevard | It stopped raining long enough for me to cut my grass yesterday |
| 18:54.25 | sevard | I love the rain but this is getting retarded |
| 18:54.37 | [TK]D-Fender | Hmmhesays : Yeah you're remote enough to qualify for that, otherwise I'd say Montreal has it worse.... |
| 18:55.26 | sevard | I hate to say but Montreal is kind of a shit-hole |
| 18:55.43 | Hmmhesays | sevard yeah |
| 18:56.00 | sevard | i'd rather go to north dakota ! |
| 18:56.01 | Hmmhesays | you can't even get a good romp in on the golf course right now its been rainging so much |
| 18:56.02 | sevard | :| |
| 18:56.12 | Hmmhesays | she just whines because the grass is cold |
| 18:56.14 | sevard | Hmmhesays: you're a golfer? uhh, fun. |
| 18:56.36 | CunningPike | techman97_andy: Pastebin your sip.conf and the output from 'sip debug' |
| 18:56.41 | sevard | call me when you pick up soccer :) |
| 18:56.45 | Hmmhesays | no I don't golf |
| 18:57.01 | sevard | uh, what |
| 18:57.07 | Hmmhesays | doesn't mean you can't sneak a quicky in on the green |
| 18:57.12 | sevard | hahaha |
| 18:57.19 | CunningPike | How many Canucks are here anyway? |
| 18:57.21 | sevard | i gotcha |
| 18:57.21 | sevard | hahaha |
| 18:57.42 | sevard | i remember those days :) |
| 18:57.46 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:57.53 | sevard | sneaking in the dark in highschool |
| 18:58.04 | Hmmhesays | call me when you're not gay |
| 18:58.15 | sevard | "shh!" "did you hear a dog?" "no fuck you" |
| 18:58.26 | sevard | call me when i can take a shit in your mouth |
| 18:58.28 | Hmmhesays | clerks II is coming out next onth |
| 18:58.36 | Flauto | Junk at the beginning 49443303 |
| 18:58.36 | Flauto | Warning, flexibel rate not heavily tested! |
| 18:58.45 | Flauto | what is this? |
| 18:58.59 | sevard | Hmmhesays: i loved the first but the second is goin to suck |
| 18:59.06 | *** part/#asterisk ckwall (n=ckwall@65.218.229.224) |
| 18:59.13 | Hmmhesays | no it'll be watcheable |
| 18:59.21 | sevard | 'watchable' is suck |
| 18:59.28 | mercestes | OMG..my virgin ears..:( |
| 18:59.48 | sevard | those wern't virgin! |
| 18:59.50 | lunk | one of the only cool things to happen in this college town was Kevin Smith coming to speak |
| 19:00.02 | lunk | talked with the crowd for like 4+ hours |
| 19:00.07 | sevard | lunk: sweet. |
| 19:00.09 | mercestes | >.> |
| 19:00.17 | mercestes | uh.....yea they were.... |
| 19:00.32 | sevard | i didn't get any resistance |
| 19:00.46 | lunk | i recently watched Clerks with a first timer, i don't think she kne what she was getting into |
| 19:00.50 | mercestes | There is a reason for that... |
| 19:00.58 | lunk | and she hasn't called since, haha |
| 19:00.59 | mercestes | you have to reach a certain depth to hit resistance and...well, you just feel short. |
| 19:01.08 | mercestes | >.> |
| 19:01.09 | sevard | mercestes: lulz |
| 19:01.10 | C4T3l | ?? |
| 19:01.13 | sevard | hahahahahah |
| 19:01.16 | sevard | ??? |
| 19:01.16 | sevard | hahahaha |
| 19:01.37 | mercestes | Umm.>.> yea, so my * box has choppy calls..>.> *gets back on topic* |
| 19:01.52 | techman97_andy | rofl |
| 19:02.08 | techman97_andy | btw - my SIP authentication issue...all of the sudden fixed itself |
| 19:02.09 | techman97_andy | wtf |
| 19:02.10 | techman97_andy | oh well |
| 19:02.14 | *** join/#asterisk backblue (n=moo@87-196-7-245.net.novis.pt) |
| 19:02.14 | CunningPike | Flauto: Sounds like you're using variable-rate encoding for your files |
| 19:02.43 | CunningPike | Flauto: See http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf - about a third of the way down |
| 19:03.03 | Flauto | thanks |
| 19:04.30 | Flauto | random=>yes? |
| 19:04.56 | CunningPike | Flauto: ? |
| 19:05.05 | [TK]D-Fender | sevard : Shit-hole? Far from! Try New England. The instant cure for insomnia! |
| 19:05.14 | sevard | hahaha |
| 19:05.56 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 19:06.16 | Flauto | cumingpike, that is the address you gave me |
| 19:06.21 | Flauto | itis about music on hold |
| 19:06.31 | Flauto | okay |
| 19:06.34 | Flauto | give me one sec |
| 19:06.38 | C4T3l | cumingpike? |
| 19:07.39 | *** join/#asterisk jvigil (n=root@ip21.farheap.net) |
| 19:07.57 | mercestes | [global] config => +nocomment |
| 19:08.08 | CunningPike | C4T3l: Speak, my child |
| 19:08.52 | CunningPike | Flauto: Sorry - there's a paragraph about flexibel rate about 1/3 of the way down |
| 19:09.18 | CunningPike | Flauto: Basically, you need to use a fixed rate for encoding your MOH files |
| 19:09.19 | Flauto | thanks cunningpike |
| 19:09.32 | Flauto | like 128 |
| 19:09.35 | Flauto | or160 |
| 19:09.41 | Flauto | stuff like that? |
| 19:09.42 | CunningPike | Flauto: no problem - are you using native MOH? |
| 19:10.26 | CunningPike | Flauto: Correct, although they will be way to high and be wasted on a phone - are you using sox? |
| 19:11.00 | Flauto | i am not using native moh |
| 19:11.07 | CunningPike | s/to high/too high quality/ |
| 19:11.08 | Flauto | just mohmp3 |
| 19:11.26 | Flauto | my mp3 files are all higher bitrate |
| 19:11.47 | Flauto | it plays okay though just i got this message once in a while |
| 19:11.48 | CunningPike | Flauto: I highly recommend native MOH - this has allowed us to use ulaw files for our MOH and avoid transcoding |
| 19:12.09 | CunningPike | Flauto: Yup - it'll play the way it is - but native MOH is the way to go |
| 19:12.50 | Flauto | that means i need to use ulaw format? |
| 19:13.18 | CunningPike | Flauto: No - you basically have the same choices as you would for Playback and BackGround |
| 19:13.33 | Flauto | oh |
| 19:13.49 | CunningPike | Flauto: We chose ulaw because most of our calls are |
| 19:13.55 | Flauto | yes |
| 19:14.00 | *** join/#asterisk ikey (i=ikey@220.226.22.14) |
| 19:14.41 | CunningPike | Flauto: I may be wrong, but I can't think of a reason not to use native MOH over mpg123 |
| 19:15.05 | [TK]D-Fender | CunningPike : You know what I find ironic? * Comes with sample MP3's for MoH, yet the DEFAULT is to still use MPG123, and even if you were to switch to Native, you'd then have to download the Add-ons package to add support for them! |
| 19:15.26 | CunningPike | [TK]D-Fender: I know - I'm hoping they will change that for 1.4 |
| 19:15.41 | Flauto | cunningpike, i have asterisk-addons installed |
| 19:15.56 | CunningPike | OK good - then you have all you need |
| 19:15.59 | [TK]D-Fender | CunningPike : At least provide them in uLAW :) I understand the lack of MP3 support however... just that it won't work by DEFAULT. |
| 19:16.01 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 19:16.21 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 19:16.22 | Flauto | the only thing is to create the folder |
| 19:16.50 | dlynes_office | I'm just curious if anyone would happen to know what would be causing echo on a voip line, but not on an analog line |
| 19:17.13 | dlynes_office | The particular issue is the person on the voip phone can hear themselves echo back...no other echo, besides that |
| 19:17.25 | dlynes_office | and again, it doesn't happen when they're on an analog line |
| 19:17.39 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:17.45 | CunningPike | dlynes_office: In my experience, direct SIP-to-SIP echo is nearly always acoustic coupling in the phone |
| 19:18.20 | sevard | Where could one get a list of the Cause Codes for IAX? I've searched the wiki |
| 19:18.23 | CunningPike | C4T3l: Did you have a question for me? |
| 19:18.40 | justinu | CunningPike: it has to be, since there's no hybrid... no? |
| 19:19.24 | [TK]D-Fender | dlynes_office : What phone? |
| 19:19.26 | CunningPike | justinu: That's pretty much my understanding - I've heard vague references to poor network QoS being an issue, but never experienced it |
| 19:19.46 | [TK]D-Fender | dlynes_home : GS has had some SUPER shitty firmware problems that caused nasty echo. |
| 19:20.13 | sevard | or... is this stuff not documented? |
| 19:20.57 | justinu | CunningPike: well, network qos (latency) will exacerbate the acoustic coupling problems |
| 19:21.07 | C4T3l | CunningPike: what is the meaning of life? |
| 19:21.16 | CunningPike | C4T3l: 42, of course |
| 19:21.17 | [TK]D-Fender | C4T3l : 42 |
| 19:21.22 | justinu | if latency is super low, the echo might just seem like normal sidetone |
| 19:21.22 | [TK]D-Fender | :D |
| 19:21.41 | CunningPike | C4T3l: 42 the Meaning of Life is |
| 19:21.41 | sevard | C4T3l: what is six by nine? |
| 19:21.42 | Flauto | cunningpike, i installed addons/mp3_files |
| 19:21.47 | Flauto | i made install there |
| 19:21.51 | sangee | how do i setup digium card? 4T1 |
| 19:21.57 | [TK]D-Fender | sevard : 54? |
| 19:22.05 | sevard | wrong, 42. |
| 19:22.10 | C4T3l | haha |
| 19:22.12 | jonson_ | someone can explain me how to set my asterisk to accept push to talk clients from nokia phones? |
| 19:22.21 | [TK]D-Fender | sangee : Try reading the WIKI and THEBOOK first before asking a generic question like that... |
| 19:22.23 | [TK]D-Fender | ~docs |
| 19:22.26 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 19:22.28 | [TK]D-Fender | ~thebook |
| 19:22.29 | jbot | somebody said thebook was somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 19:22.44 | sevard | [TK]D-Fender: perhaps you can tell me what Cause Code 29 is in IAX2 |
| 19:22.58 | CunningPike | Flauto: We installed format_mp3.so - that's all the add-ons we needed |
| 19:23.38 | [TK]D-Fender | sevard : I could, but then it'd be a completely blind guess :) |
| 19:23.42 | sangee | ok,can u give the link? |
| 19:23.59 | [TK]D-Fender | sangee : JBOT just pasted them all for you. |
| 19:24.05 | sangee | thx |
| 19:24.16 | Flauto | i will try to create the folder now. |
| 19:24.21 | sangee | ok |
| 19:24.23 | CunningPike | Ok - lunchtime - bbl |
| 19:24.49 | justinu | anyone familiar with OriginateAction in AMI? |
| 19:25.03 | [TK]D-Fender | sevard : six by nine IS 54. You're thinking of 6 by SEVEN. |
| 19:25.33 | [TK]D-Fender | sevard < math |
| 19:25.46 | justinu | i'm having an issue with the CDR that gets output when I make a call with OriginateAction |
| 19:26.22 | SpaceBass | no one has dugg my story on digg.com :( |
| 19:26.25 | justinu | src, dst aren't filled in |
| 19:26.46 | sevard | [TK]D-Fender: 6 by 9 is 42 |
| 19:28.03 | *** join/#asterisk riogeed (n=riogeed@66-178-161-84.generic.rio.com) |
| 19:28.08 | riogeed | Hello everyone |
| 19:28.16 | techman97_andy | hello |
| 19:28.20 | [TK]D-Fender | sevard : Pull out a calculator, you've clearly run out of fingers and toes... |
| 19:28.25 | *** join/#asterisk doolph (n=doolph@cpe-00c0a8f7b32d.cpe.cableonda.net) |
| 19:28.29 | doolph | hi |
| 19:28.32 | sevard | [TK]D-Fender: I guess you don't appreciate jokes in base 13 |
| 19:28.33 | riogeed | I had a question about asterisk CDR. I notice that the csv by default has a start, answer, end timestamp |
| 19:28.33 | techman97_andy | hello |
| 19:28.34 | doolph | anyone know what is TDM2400P¿ |
| 19:28.55 | riogeed | but it seems the postgresql schema does not include fields for these. Is there any reason why ? |
| 19:29.11 | [TK]D-Fender | sevard : Only 10 kinds of people in the world.. those that understand binary and those that don't.... |
| 19:29.24 | sevard | :) |
| 19:29.42 | SpaceBass | i've seen that on a tee shirt |
| 19:29.43 | mercestes | LOL @ D-Fender |
| 19:29.45 | [TK]D-Fender | sevard : Is that base-13 reference actually from the books? |
| 19:29.45 | SpaceBass | i love that quite |
| 19:29.47 | SpaceBass | quote |
| 19:29.54 | riogeed | SpaceBass: thinkgeek has got it. |
| 19:30.00 | SpaceBass | yeah |
| 19:30.24 | sevard | [TK]D-Fender: I may be a pretty sad person but at least I don't make jokes in base 13. |
| 19:30.43 | riogeed | anyway I am pretty sure I need the start, answer and end timestamps but would like to use pgsql instead of csv. Is there a way to achieve this as it seems they are only logged when using csv ? |
| 19:33.51 | riogeed | not many postgres users I take it :) |
| 19:33.59 | websae | mssql here :) |
| 19:35.43 | cytrak | is it possible to call our own number from an outside and say dial "*" when we reach the voicemail in order to get in to the adminitrative section ? |
| 19:36.12 | cytrak | like I can with vonage or my cell |
| 19:36.53 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 19:37.34 | [TK]D-Fender | cytrak : sure |
| 19:37.59 | [TK]D-Fender | cytrak : Go read up on the "a" standard extension. |
| 19:38.09 | cytrak | ok |
| 19:38.53 | riogeed | websae: have you got fields for start, answer and end times ? |
| 19:39.18 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 19:39.25 | *** join/#asterisk ikey1 (i=ikey@220.226.22.246) |
| 19:39.26 | riogeed | websae: from the looking I did it seemed all non csv storage had those missing. |
| 19:40.03 | exonic | Is anyone aware of asterisk@home containing any major security flaws? One in which would allow hosting of those pesky paypal fraud sites? I have a customer who recently was just hacked. |
| 19:40.05 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 19:40.17 | Flauto | cunningpike, what do i do with the old mohmp3 folder than, i can delete it and unistall mpg123? |
| 19:40.20 | dlynes_office | [TK]D-Fender: It's on an Aastra 9133i |
| 19:40.38 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:41.04 | [TK]D-Fender | dlynes_home : Oh, guess you should have gone Polycom then ;) |
| 19:41.14 | dlynes_office | [TK]D-Fender: whatever :) |
| 19:41.17 | [TK]D-Fender | :D |
| 19:41.30 | exonic | Any asterisk at home folks in here? |
| 19:41.35 | [TK]D-Fender | I know.. I should be paid for the amount I prop them for.... |
| 19:41.41 | Qwell[] | exonic: #asteriskathome |
| 19:41.43 | syle | this is not #asteriskathome |
| 19:41.46 | exonic | Qwell, aye, thx |
| 19:41.52 | [TK]D-Fender | exonic : No, we just finished burying the body of the last guy! |
| 19:41.57 | exonic | good |
| 19:41.59 | exonic | I hate it too |
| 19:42.06 | exonic | but my moron customer can't setup a real asterisk box |
| 19:42.24 | [TK]D-Fender | exonic : If he's your customer then its YOUR job! |
| 19:42.27 | Nivex | exonic: that's why they should pay you to do it |
| 19:42.36 | exonic | he doesnt want to pay me/us |
| 19:42.48 | Qwell[] | Then you can pay me/us |
| 19:42.50 | Nivex | no $$$, no support |
| 19:43.11 | *** part/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:43.12 | exonic | exactly, now I am sitting here with a hacked asterisk@home in my network :) |
| 19:43.30 | mercestes | and in the wrong IRC channel complaining about it. |
| 19:43.47 | exonic | nobody is in #asteriskathome |
| 19:43.54 | mercestes | there is a good reason for that. |
| 19:43.56 | Nivex | This should tell you something. |
| 19:43.57 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:43.58 | Qwell[] | exonic: that really says something |
| 19:43.59 | riogeed | websae: you still here ? |
| 19:44.01 | exonic | ;) |
| 19:44.04 | exonic | i'll shutup now |
| 19:44.04 | file | TROLLS |
| 19:44.06 | mercestes | congratulations..your now the moderator for *@home. |
| 19:44.23 | Qwell[] | file: nub |
| 19:44.28 | file | :( |
| 19:44.32 | Qwell[] | :D |
| 19:44.35 | exonic | i'm not trolling, I personally hate *@home |
| 19:44.42 | SpaceBass | I personally love it |
| 19:44.42 | Qwell[] | exonic: no, we're trolling |
| 19:44.49 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 19:44.51 | SpaceBass | and support can be found at #freepbx |
| 19:45.01 | exonic | heh |
| 19:45.14 | Assid | err.. how do you support the im/text messaging of a phone? |
| 19:45.27 | file | in Asterisk? you don't. |
| 19:45.32 | [TK]D-Fender | exonic : got to the Voxilla forums. Their * msg board may as well be an A@H board.... |
| 19:45.33 | *** join/#asterisk ToTo (n=ToTo@host46-89.pool8256.interbusiness.it) |
| 19:45.39 | SpaceBass | Asterisk can do SMS |
| 19:45.42 | dlynes_office | [TK]D-Fender: btw...just got those sangoma cards today...they look pretty snazzy |
| 19:46.00 | Assid | err.. what about the clients? |
| 19:46.08 | [TK]D-Fender | dlynes_office : Cool, which models? |
| 19:46.18 | SpaceBass | Some phones can do it...like the Hitachi IP5000 |
| 19:46.20 | Assid | like xten / polycoms.. cant they send/receive |
| 19:46.21 | syle | who cares what they look like lol, how do they work? |
| 19:46.30 | CrummyGummy | SpaceBass, Can Asterisk do MMS? |
| 19:47.24 | riogeed | is anyone here using asterisk with cdr logging to something other than CSV ? |
| 19:47.38 | Assid | i get like .. ";tag=c205251b, dropped it..." |
| 19:47.47 | file | Asterisk does not support it yet. |
| 19:47.51 | [TK]D-Fender | riogeed : many use MySQL/PGSQL, etc |
| 19:47.52 | file | (messaging) |
| 19:47.55 | Qwell[] | keyword; yet |
| 19:47.55 | dlynes_office | [TK]D-Fender: both a200d's...one with four fxo, one with 2 fxs/2 fxo |
| 19:47.58 | Qwell[] | patches accepted |
| 19:48.01 | CrummyGummy | Pity, it'd be useful |
| 19:48.15 | [TK]D-Fender | dlynes_home : You mean 1 card with remora's to expant it, right? |
| 19:48.24 | [TK]D-Fender | dlynes_home : So it IS effectively 1 card. |
| 19:48.26 | dlynes_office | [TK]D-Fender: going to try faxing on the fxo/fxs one, and going to replace our nortel keysystem with the four fxo one |
| 19:48.30 | SpaceBass | CrummyGummy, no idea, but i'd suspect not |
| 19:48.33 | riogeed | [TK]D-Fender: from what I can tell all methods other than csv are missing fields for start, answer and end timestamps. I am trying to confirm this |
| 19:48.42 | CrummyGummy | SpaceBass, Thanks |
| 19:48.45 | dlynes_office | [TK]D-Fender: two cards...one with 1 fxs module, 1 fxo module, and another one with 2 fxo modules |
| 19:49.17 | SplasPood | Can anyone suggest why my sip termination provider might be generating this asterisk console message when the number I'm dialing returns CONGESTION? May 15 15:48:19 WARNING[22412]: chan_sip.c:9559 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"James Brinkerhoff" <sip:2128124187@voice1.lga1.us.voxel.net>;tag=as289d0762' |
| 19:49.33 | [TK]D-Fender | dlynes_home : But if you ordered it right it should only use up 1 PCI interface, but 2 back-plane slots. |
| 19:49.37 | SplasPood | the name/number you see in the sip uri is my CID |
| 19:49.52 | dlynes_office | [TK]D-Fender: they're both going into different machines |
| 19:50.03 | [TK]D-Fender | dlynes_office : Oh, ok, nvm :) |
| 19:50.10 | dlynes_office | [TK]D-Fender: not using hte backplane on either one...one will replace our keysystem, the other will be for demo purposes |
| 19:50.15 | [TK]D-Fender | dlynes_home : have fun.. great cards, very funky. |
| 19:50.35 | dlynes_office | yeah...the three swooshes on the card make it look cool too =) |
| 19:50.36 | Assid | how much do sms gateways charge anwayys? |
| 19:51.16 | [TK]D-Fender | dlynes_office : Oh they have the CardShark "gills" on it too? That A104d had them.... very nice fit&finish. |
| 19:51.31 | dlynes_office | yeah..not sure what the purpose of them is though |
| 19:52.23 | dlynes_office | man those bastards on ebay are cheap |
| 19:52.35 | dlynes_office | only bidding $5 on my damned phone :(( |
| 19:52.47 | file | people. are. cheap. |
| 19:52.51 | SpaceBass | lol |
| 19:52.57 | file | what, it's true |
| 19:53.01 | sevard | i'll give you 10 bucks |
| 19:53.06 | Assid | $5 for a phone? which one? |
| 19:54.00 | SpaceBass | i've bought a lot of phones off ebay |
| 19:54.06 | dlynes_office | This one: http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&item=9724568965&rd=1&sspagename=STRK%3AMEWA%3AIT&rd=1 |
| 19:54.13 | SpaceBass | including 2 of these: http://www.cashintelecom.ie/ipicasso.htm which turned out to not have firmware |
| 19:54.20 | Assid | so what exactly is the use fo the SendText cmd if it doesnt send text? |
| 19:55.44 | dlynes_office | SpaceBass: and you couldn't get firmware developer's kits from the manufacturer, either? |
| 19:56.03 | sevard | dlynes_office: my roomate is looking for a cheap phone / ata for his gf |
| 19:56.11 | sevard | i'll give you 10 bucks |
| 19:56.21 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 19:56.21 | dlynes_office | Well, make an offer then :) |
| 19:56.26 | dlynes_office | put it on ebay :) |
| 19:56.27 | sevard | i just did |
| 19:56.37 | Assid | not bad |
| 19:56.44 | paolob | Hi guys! any idea about where finding config help for the zoom 5801? thank you! |
| 19:56.49 | SpaceBass | dlynes_office, out of business apparently |
| 19:56.55 | sevard | dlynes_office: want to cancel the auction? |
| 19:56.58 | dlynes_office | they're not great phones by any stretch of the imagination |
| 19:57.09 | dlynes_office | sevard: nah...cause i've about another 80 phones to go |
| 19:57.13 | sevard | I know, we're just looking for cheap |
| 19:57.25 | sevard | you're selling 80 phones? |
| 19:57.34 | dlynes_office | sevard: at $10, i'm still losing $90 a phone |
| 19:57.49 | sevard | ouch, they're $100 phones? |
| 19:57.54 | sevard | i wouldn't want to put you at a loss |
| 19:57.57 | dlynes_office | Were |
| 19:58.11 | sevard | are there yours personally or a company you're working for |
| 19:58.12 | dlynes_office | about 1-1/2 years ago |
| 19:58.17 | dlynes_office | Company's |
| 19:58.25 | sevard | I'll give you 10 bucks for 5 of them |
| 19:58.26 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 19:58.31 | dlynes_office | lol |
| 19:58.35 | dlynes_office | He'd never go for it |
| 19:58.37 | sevard | come on mate they're not yours |
| 19:58.37 | a1fa | how much memory does * use per sip peer? |
| 19:58.40 | dlynes_office | He'd rather put them in the land fill :) |
| 19:58.43 | sevard | a1fa: like a gig |
| 19:58.47 | Assid | haha |
| 19:58.54 | a1fa | sevard : yeah ;) ***** u |
| 19:58.54 | Assid | a1fa: memory is never the problem |
| 19:58.58 | a1fa | cpu time |
| 19:59.02 | Assid | nothing |
| 19:59.07 | a1fa | what is the problem? |
| 19:59.08 | Assid | its only when a call is in process |
| 19:59.13 | a1fa | thats what i ment |
| 19:59.20 | a1fa | on G711? |
| 19:59.34 | sevard | bandwidth is the problem |
| 19:59.35 | Assid | doesnt matter.. is there any transcoding to/from 711 ? |
| 19:59.42 | a1fa | cpu time & memory per ulaw sip peer? |
| 19:59.43 | Assid | hrmm |
| 19:59.49 | Assid | im gonna go play with openser |
| 19:59.49 | a1fa | no |
| 19:59.53 | a1fa | its between ulaw and ulaw |
| 19:59.58 | Assid | find a way to offload sip calls onto openser |
| 20:00.13 | a1fa | sevard : i got nuff bandwidth |
| 20:00.17 | *** join/#asterisk stealthmethod (n=unknown@adsl-068-153-171-051.sip.mia.bellsouth.net) |
| 20:00.20 | sevard | dlynes_office: well i'm poor as fuck, if you can throw me a reasonable price i'll pick a few up |
| 20:00.26 | a1fa | sevard : about 100mbit |
| 20:00.46 | a1fa | 100 mibt will handle at least 100 stim. calls |
| 20:00.47 | sevard | a1fa: give me a shell |
| 20:00.57 | a1fa | sevard : i will give you my.. |
| 20:01.08 | a1fa | so you can x) on it |
| 20:01.10 | Assid | what is the difference in cpu usage with ser/* handling calls? |
| 20:01.12 | Assid | anyone know? |
| 20:01.24 | sevard | a1fa: say what |
| 20:01.26 | a1fa | Assid : so ulaw to ulaw? |
| 20:01.33 | Assid | a1fa ? |
| 20:01.36 | a1fa | how much cpu time, memory am I looking for? |
| 20:01.43 | Assid | nothing much |
| 20:01.56 | sevard | like dual 4ghz processors with 30gb of ram for about 10 calls |
| 20:02.00 | dlynes_office | sevard: well, i've got a bunch of grandstream budgetones i can offload for about $50 Cdn each |
| 20:02.05 | sevard | i'll manage your server for free |
| 20:02.05 | tzanger | sevard: only if you're poor |
| 20:02.11 | tzanger | for 10 calls you want at least a quad processor board |
| 20:02.20 | Assid | you should be able to manage tons of calls with no transcoding |
| 20:02.23 | sevard | dlynes_office: you DO know the exchange rate is absolute ass right now |
| 20:02.28 | cytrak | [TK]D-Fender: is that "a" extension supposed to also work within a macro ? |
| 20:02.28 | a1fa | tzanger : no fucking way? |
| 20:02.38 | Assid | just trying to see if i can get any difference with openser |
| 20:02.38 | tzanger | a1fa: do you think that call processing is easy on the CPU? |
| 20:02.44 | justinu | heh |
| 20:02.45 | a1fa | i thinks it is about a 100 stim calls on 1ghz, 512mb ram |
| 20:02.45 | Assid | i know you can scale more calls with that |
| 20:02.52 | dlynes_office | sevard: so assassinate your crappy president then :) |
| 20:02.58 | a1fa | tzanger : stfu, i aint doin any encoding |
| 20:03.04 | sevard | dlynes_office: send a gun instead of a phone then |
| 20:03.07 | tzanger | a1fa: I said voice processing, not transcoding |
| 20:03.08 | a1fa | its all ulaw |
| 20:03.21 | tzanger | echo cancellation, tone detection, tone generation |
| 20:03.26 | tzanger | there is a reason DSPs were designed |
| 20:03.29 | a1fa | so quad cpu |
| 20:03.40 | a1fa | lol |
| 20:03.42 | a1fa | dude |
| 20:03.55 | a1fa | with quadcpu i can mp4 encode a dvd in 1 minute |
| 20:04.17 | tzanger | yes but how many DVDs can you mp4 encode at once? |
| 20:04.21 | tzanger | you're asking for parallel processing here |
| 20:04.21 | Assid | okay anyone here got ser running with * ? |
| 20:04.30 | tzanger | not single-stream processing like DVD re-encode |
| 20:04.32 | sevard | dlynes_office: I was more looking for a used Sipura, I found a new sip-1000 on ebay for 37.00, i've heard bad things about the grandstream's quality |
| 20:04.43 | a1fa | tzanger : about 20 in 5 minutes |
| 20:04.45 | [TK]D-Fender | cytrak : Yup, thats the best way to run your phone extensions... |
| 20:05.02 | tzanger | you've encoded 20 DVDs in 5 minutes with a quad-processor box? |
| 20:05.09 | tzanger | I'd like ot see your test report |
| 20:05.12 | Flauto | May 15 15:04:25 WARNING[503]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 20:05.26 | a1fa | tzanger : not really, but I am saying, it would be possible |
| 20:05.35 | tzanger | until you benchmark it, it's all hand-waving |
| 20:05.40 | sevard | i agree |
| 20:05.44 | sevard | give me root on your box |
| 20:05.44 | dlynes_office | sevard: we've got some sipura 2000's, too |
| 20:05.47 | sevard | i'll benchmark it for you |
| 20:05.58 | Assid | hey tkd, you tried running ser with * ? |
| 20:06.02 | a1fa | sevard : why dont you give me your sisters door number, i will benchmark her |
| 20:06.06 | stoffell | hand-waving, is that the correct term for "just taking a guess" ? (just curious here) |
| 20:06.09 | dlynes_office | a1fa: lol |
| 20:06.10 | sevard | dlynes_office: what's the going price on that? i know the 2000 series is a bit more pricey |
| 20:06.21 | sevard | a1fa: let my first benchmark your face |
| 20:06.28 | dlynes_office | sevard: they're currently going at about $90-110USD |
| 20:06.28 | a1fa | first? |
| 20:06.28 | tzanger | stoffell: yes. watch a pair of theoretical physics professors have a conversation sometime |
| 20:06.32 | sevard | FIST |
| 20:06.40 | ikey1 | hi |
| 20:06.44 | a1fa | sevard : i will leave fisting up for your sister |
| 20:06.45 | sevard | dlynes_office: i wish i had that kind of money to spend |
| 20:06.51 | stoffell | tzanger, lol, okay, thanks for the tip.. |
| 20:07.06 | sevard | a1fa: as long as i can have your mom, with you tied to the bed forced to watch |
| 20:07.14 | dlynes_office | sevard: we're just getting rid of excess inventory...we have too much money tied up in inventory; we're wanting to free some up so we've got more working capital |
| 20:07.21 | a1fa | sevard : what r u, 12 |
| 20:07.25 | syle | tzanger no different than 2 stoned programmers talking |
| 20:07.27 | a1fa | i gotta go |
| 20:07.30 | tzanger | syle: hahaha |
| 20:07.31 | sevard | a1fa: just upstairs |
| 20:07.34 | a1fa | dont have time for this punk ass bitch |
| 20:07.55 | sevard | dlynes_office: So you're looking to make some actual money not free up space |
| 20:08.08 | Flauto | May 15 15:04:25 WARNING[503]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 20:08.14 | dlynes_office | sevard: obviously we don't want to lose a crapload of money |
| 20:08.22 | Flauto | is it because of the mp3 files bitrate? |
| 20:08.26 | dlynes_office | sevard: just want to free up some money, not space |
| 20:08.32 | sevard | dlynes_office: I just figured you were working for a big corporation and could sluff a few off :) |
| 20:08.36 | dlynes_office | sevard: space would be nice, too |
| 20:08.43 | SpaceBass | i only have one dig :( http://digg.com/search?search=itunes+asterisk&submit=Submit |
| 20:08.44 | dlynes_office | sevard: then our rent wouldn't be so high :) |
| 20:09.14 | sevard | dlynes_office: but if you're in a position in the company to worry about money I'm in no position to get things dirt cheap from you |
| 20:09.47 | syle | laughing at the post you guys pasted here earlier |
| 20:09.50 | syle | http://www.digg.com/technology/SkypeOut_now_FREE_for_US_Canada_ |
| 20:09.57 | tzanger | sevard: that sounds very close to "a lack of planning on your part does not constitute and emergency on my part" :-) |
| 20:10.05 | syle | that guy is getting taken up on his offer to give free blowjobs lol |
| 20:10.15 | sevard | tzanger: heh |
| 20:10.59 | dlynes_office | sevard: heh...we're only opening the stuff to auction so that we can get the ebay points we need to use buy it now pricing |
| 20:11.24 | *** part/#asterisk ikey1 (i=ikey@220.226.22.246) |
| 20:11.24 | sevard | dlynes_office: so... if one happens to get a phone for 20 bucks... you'll send it? |
| 20:11.38 | dlynes_office | sevard: yeah...don't want to sell it that cheap |
| 20:11.46 | dlynes_office | sevard: but don't want a bad ebay rating, either :) |
| 20:11.54 | sevard | ;-) |
| 20:12.03 | [hC] | so if im not mistaken, on the -dev list it looks like code has been introduce that will allow native transfers with IAX2 now? :) |
| 20:12.39 | tzanger | [hC]: media-only transfers, yes |
| 20:13.29 | [hC] | tzanger: awesome. Id been wanting that for a long time. Eliminating unnecessary bandwidth certainly helps, but the CDR issue just made it impossible. |
| 20:13.39 | tzanger | yep |
| 20:13.49 | [hC] | tzanger: Id been debating moving to SIP entirely for a ling time, to get around some of IAX's p[roblems |
| 20:13.53 | [hC] | problems* |
| 20:14.01 | [hC] | but, this is one more step in the right direction! |
| 20:14.44 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 20:15.29 | carlos-the-man | zzzzzzzzzzzzzzzzzzzzzzzzzzz |
| 20:17.07 | tzanger | [hC]: well there are still troubles with DTMF on IAX which is relatively new |
| 20:17.21 | Assid | iax has dtmf problems? |
| 20:17.37 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 20:17.42 | tzanger | with the jitter buffer, yes |
| 20:18.10 | Assid | oh |
| 20:18.19 | Assid | hrmm.. that would explain a few things |
| 20:18.37 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 20:19.15 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:19.28 | lullabud | i have a question concerning centrally managed deployment of voip phones... |
| 20:19.47 | dlynes_office | CunningPike: so that problem would probably just be the handset, even though the problem doesn't manifest itself on analog lines? |
| 20:20.10 | lullabud | right now i'm running asterisk and we've been doing per-phone configurations on the phones themselves. i want to move to a centrally managed, all-in-one type solution. |
| 20:20.21 | lullabud | something that integrates the asterisk side with the phone configuration side. |
| 20:20.21 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:20.27 | lullabud | does anybody know anything like that? |
| 20:20.40 | dlynes_office | lullabud: you want to get something that does autoprovisioning through ftp, tftp, http, or https, depending on your needs |
| 20:20.49 | lullabud | free or commercial, even. |
| 20:20.56 | CunningPike | dlynes_office: That would be my guess...... |
| 20:21.15 | dlynes_office | CunningPike: so any idea why the problem wouldn't manifest itself on the analog line, but it does on the voip line? |
| 20:21.51 | lullabud | dlynes_office: yeah, that's pretty much it. what i'd really like is the ability to tie mac addresses into asterisk so that phones auto-configure with the extension settings i've set up in there. |
| 20:21.51 | CunningPike | dlynes_office: Don't forget echo is a far end problem - there is EC in your PSTN path |
| 20:21.56 | dlynes_office | lullabud: most decent phones/atas have autoprovisioning tools...some of them only have the tools if you buy the devices in bulk |
| 20:22.06 | CunningPike | Gotta run - biab |
| 20:22.12 | *** join/#asterisk Smi|k (n=smilk@netblock-72-25-103-165.dslextreme.com) |
| 20:22.16 | lullabud | we're buying polycom 501's and 301's in bulk and they're not coming with any tools... |
| 20:23.01 | dlynes_office | Anyone know if Polycom has provisioning tools? And what's the minimum order you need to do to get them? |
| 20:23.01 | lullabud | our asterisk solution is provided by switchvox, but they don't provide any auto-provisioning of the phones. |
| 20:23.21 | lullabud | so, basically i'm looking for other options. |
| 20:23.30 | dlynes_office | lullabud: the autoprovisioning isn't done by asterisk; it's done by a tftp/http/https/ftp server |
| 20:23.41 | dlynes_office | lullabud: and yes, it's based on mac addresses |
| 20:23.44 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 20:23.45 | lullabud | dlynes_office: right, but i want a solution that ties those two together. |
| 20:24.13 | lullabud | dlynes_office: i'm doing some of our phones off ftp right now, but it's a nightmare to configure. |
| 20:24.14 | *** join/#asterisk dasenjo (i=www-data@208.195.214.137) |
| 20:24.30 | [TK]D-Fender | lullabud : There are apps out there to generate configs for them. ScopServ (an * GUI) has a module for that. |
| 20:24.37 | lullabud | dlynes_office: it seems ridiculous that you'd have to enter that information in two places. there's got to be a solution that ties the asterisk config into the phone provisioning. |
| 20:24.42 | dlynes_office | lullabud: you mean you want one where you can just drop in a new mac address, and it'll autoprovision it as soon as it asks for a dhcp address, and then also autoconfigure asterisk? |
| 20:24.52 | jsaunders | I noticed a Shortel phone connected to a Shoretel Shoregear system obtained it's configuration via ftp when booting up, never looked into it any further. With some cleverness you think I could rig it to work w/ *? :D |
| 20:25.43 | lullabud | dlynes_office: basically i want to be able to use asterisk just like i am now, then have something that serves up auto-provisioning configs for my polycom phones using the asterisk config as its reference. |
| 20:26.24 | lullabud | dlynes_office: so, when i say ext156 is john doe in asterisk, i want to be able to enter a mac address and have that phone auto-config with the info from within asterisk. |
| 20:26.43 | dlynes_office | lullabud: where are you entering the mac address? |
| 20:26.45 | brettnem | hey, anyone in here using native MOH? |
| 20:26.54 | lullabud | dlynes_office: right now, nowhere. ;-) that's the problem |
| 20:27.00 | dlynes_office | brettnem: yep...lotsa people |
| 20:27.19 | dlynes_office | lullabud: let's rephrase that...where is it that you want to enter it? |
| 20:27.23 | brettnem | dlynes_home: I'm having trouble with the MOH volume.. tried sox -v .5 but that didn't actually do anything.. any ideas? |
| 20:27.28 | stoffell | anyone knows why the auth port (tcp 113) is being used between * server and polycom phones? |
| 20:27.40 | lullabud | dlynes_office: let's step back and look at the broad picture.... |
| 20:27.42 | dlynes_office | brettnem: afaik, there's isn't a volume control |
| 20:28.03 | lullabud | dlynes_office: right now i have to generate config files for phones. i also have to configure asterisk to have the proper extensions set up for each employee... |
| 20:28.06 | dlynes_office | lullabud: btw...what you're asking is also technically impossible without a shitload of work, if it's going to be generic |
| 20:28.11 | brettnem | dlynes_home: right, I'm just trying to reduce the volume.. I tried editing the file with sox.. it's way too loud |
| 20:28.13 | lullabud | dlynes_office: i want to unify those two configurations, however it is possible. |
| 20:28.51 | brettnem | lullabud: just write a perl script to do it.. shouldn't be too hard. |
| 20:29.07 | dlynes_office | lullabud: mainly because every phone and/or ata has its own binary profile configuration files that you have to content with |
| 20:29.15 | lullabud | brettnem: it wouldn't be if i weren't so rusty on perl, and wasn't paying for a supported version of asterisk which doesn't give me ssh access. |
| 20:29.17 | dlynes_office | s/content/contend/ |
| 20:29.46 | lullabud | dlynes_office: yeah, i know what you mean. i was hoping there would be something specifically for polycom. |
| 20:30.02 | brettnem | lullabud: what does that mean? you have asterisk without ssh access?! |
| 20:30.04 | syle | brettmen : i suggest you use madplay, you can control volume through switches that way |
| 20:30.13 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 20:30.15 | dlynes_office | lullabud: well, then you run into licensing issues on the provisioning tools I would imagine, too |
| 20:30.28 | brettnem | syle: I saw some stuff about madplay on the wiki.. I will check it out.. |
| 20:30.44 | lullabud | brettnem: we're paying for a supported packaged solution of asterisk through www.switchvox.com |
| 20:30.50 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:30.54 | dlynes_office | brettnem: yeah...madplay is 100x more stable than mpg123 |
| 20:31.03 | brettnem | lullabud: sounds like an issue for your service provider!! |
| 20:31.05 | lullabud | brettnem: they provide hardware replacements and software support and a web-based gui. |
| 20:31.07 | dlynes_office | brettnem: mpg123 is the most unstable piece of crap |
| 20:31.22 | brettnem | dlynes_office: yes I know.. which is why I'm doing native. which I'm not real happy with |
| 20:31.31 | lullabud | brettnem: yeah... i think you might be right. this is totally something they should do. maybe i'll run it by them... |
| 20:31.40 | dlynes_office | brettnem: yeah...i'm doing native...never had a problem with the volume |
| 20:31.41 | lullabud | brettnem: seems like it'd be a great service to provide too. |
| 20:31.49 | brettnem | lullabud: I'm familiar with them.. I wouldn't expect them to create an auto provisioning gui.. |
| 20:31.51 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:31.53 | dlynes_office | brettnem: have you tried reducing the volume using audacity? |
| 20:32.09 | schirpich | how can i force 911 to dial externally? |
| 20:32.25 | dlynes_office | schirpich: yes....read up on the wiki for dial plans |
| 20:32.25 | lullabud | brettnem: well, they only sell polycom phones and sipura adapters. i'd think they could come up with something to support those two devices, or even just the polycom. |
| 20:32.36 | brettnem | dlynes_office: I tried it with sox.. which really ought to do it |
| 20:32.38 | lullabud | brettnem: i'd think it'd be a new revenue channel for them. |
| 20:32.43 | dlynes_office | schirpich: there's plenty of 911 dial plan examples |
| 20:32.56 | brettnem | lullabud: nah.. why bother with that |
| 20:33.00 | dlynes_office | brettnem: yeah...don't know...i know audacity will do it...that's what I always use |
| 20:33.07 | lullabud | brettnem: well, if it makes my job easier..... |
| 20:33.18 | syle | you don't use one program to fix another, just use madplay |
| 20:33.25 | dlynes_office | brettnem: you just tell audacity to lower the amplitude across the entire wav file |
| 20:33.26 | brettnem | lullabud: keep an eye on the phonecall project.. it'll eventually have what you describe in it. |
| 20:33.38 | brettnem | dlynes_office: sox -v does the same thing |
| 20:33.45 | lullabud | brettnem: oh? got a url? |
| 20:33.57 | brettnem | lullabud: http://www.vecsector.com/phonecall |
| 20:34.01 | brettnem | doesn't have it yet, but it will |
| 20:34.02 | lullabud | gracias |
| 20:34.15 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:34.24 | schirpich | btw, for those who were here earlier when i was asking about the 911 stuff, i called it (deemed ok to test by XO) the lady nearly had a heart attack that i would call just to test |
| 20:34.25 | brettnem | syle: that isn't using one program to fix another.. I'm not using mpg123 |
| 20:34.31 | hensema | I'm trying to setup voicemail, but voicemail messages just contain some clicking noises; the recorded prompts are fine |
| 20:34.35 | hensema | what could be wrong? |
| 20:34.48 | dlynes_office | cockroaches have infested your voicemail! |
| 20:34.55 | hensema | lol |
| 20:34.58 | lullabud | schirpich: that seems ridiculous.... i'd assume everybody should test it. wtf if it doesn't work? |
| 20:35.11 | docelmo | there's your bugs dlynes_office |
| 20:35.29 | lullabud | ok, i have another question concerning providers... |
| 20:35.36 | docelmo | Im a provider.. :) |
| 20:35.39 | dlynes_office | i didn't know i had any bugs ;) |
| 20:35.41 | lullabud | right now we've got a dedicated t1 for our voip stuff... |
| 20:35.52 | docelmo | data/voice? |
| 20:36.09 | brettnem | docelmo: where are you located? |
| 20:36.16 | docelmo | physically? |
| 20:36.19 | lullabud | docelmo: just for voice right now. |
| 20:36.19 | nextime | in your opinion, can a slower 233 Mhz arm processor with 32 megs of ram with uclinux run a stripped down version of * that must manage only one channel a time without transcoding? |
| 20:36.19 | brettnem | no, mentally |
| 20:36.26 | lullabud | docelmo: cupertino, ca |
| 20:36.43 | docelmo | Physically Tampa, FL |
| 20:36.44 | schirpich | i know, I explained to her i was setting up a phone system for a corporation and that 911 services had to be tested. she grumbled a bit and got off the phone, about a 1/2 hour later HR director comes in, "did you call 911?" aparently 911 called back to check. even though i told the lady, " i am testing to see if 911 works" |
| 20:37.11 | schirpich | which is good i guess |
| 20:37.16 | lullabud | so... i'm not sure what the terms are to say this, i'm new to all of this... |
| 20:37.30 | dlynes_office | schirpich: well, that confirmed your suspicions that it really was 911, right? |
| 20:37.32 | lullabud | basically, we have a t1 that comes in and provides us 24 channels of voice. |
| 20:37.40 | schirpich | haha, yes |
| 20:37.41 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 20:37.43 | brettnem | "I would like to make a 911 test call to ensure my 911 routing is correct" |
| 20:37.54 | lullabud | what i'd like is to do sip peering to our provider. |
| 20:38.28 | docelmo | what's your internet connection like? |
| 20:38.36 | lullabud | multiple dedicated t1's. |
| 20:38.48 | lullabud | thinking of going to t3 or ds3 |
| 20:38.55 | docelmo | QoS setup? |
| 20:39.11 | lullabud | IP QoS? |
| 20:39.20 | docelmo | yes |
| 20:39.22 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 20:39.33 | lullabud | just rudimentary right now, but we have it. |
| 20:39.46 | lullabud | we're a growing company so stuff changes pretty rapidly. |
| 20:39.59 | docelmo | Call yer provider up and tell em you would like to know if they do SIP and would like to failover and interconnect with them |
| 20:41.43 | lullabud | alright |
| 20:42.05 | lullabud | docelmo: is there a link that you know of where i can get some theory on these setups? |
| 20:42.05 | docelmo | if not I know a couple good providers you can fail over to |
| 20:42.21 | docelmo | www.voip-info.org |
| 20:42.33 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:42.45 | lullabud | ok, cool. thanks. |
| 20:42.52 | docelmo | ~wiki |
| 20:42.58 | lullabud | yeah, i'm a member. ;-) |
| 20:43.06 | lullabud | been doing grammatical corrections... :P |
| 20:43.21 | docelmo | ~wiki irc |
| 20:43.30 | brettnem | ~seen ctooley |
| 20:43.44 | jbot | ctooley <n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com> was last seen on IRC in channel #asterisk, 5d 3h 52m 56s ago, saying: 'What's the extension that gets jumped to after a call is hung up? it's like context,h,1 right?'. |
| 20:44.20 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:44.40 | docelmo | ~seen docelmo |
| 20:44.52 | jbot | docelmo is currently on #asterisk (4d 1h 26m 18s). Has said a total of 49 messages. Is idling for 12s, last said: '~seen docelmo'. |
| 20:44.53 | docelmo | ~seen docelm0 |
| 20:44.59 | jbot | docelm0 is currently on #asterisk (1d 13h 12m 36s). Has said a total of 178 messages. Is idling for 23h 51m 37s, last said: 'FXO == CO FXS == phone'. |
| 20:45.11 | brettnem | ok, polycom phones are a major pain in the ass |
| 20:45.13 | lullabud | that's neat |
| 20:45.36 | tzanger | well that's not quite right |
| 20:45.36 | lullabud | does it monitor all chat rooms? |
| 20:45.36 | brettnem | haha |
| 20:45.36 | tzanger | FXO connects to a CO |
| 20:45.36 | x86 | ~seen my_feet |
| 20:45.41 | jbot | i haven't seen 'my_feet', x86 |
| 20:45.41 | tzanger | FXS emulates a CO |
| 20:45.41 | tzanger | x86 don't be starting that shit again :-) |
| 20:45.45 | x86 | ;) |
| 20:45.47 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 20:45.50 | brettnem | ~seen bigfoot |
| 20:45.51 | jbot | bigfoot <~simon@80.88.192.113> was last seen on IRC in channel #asterisk, 588d 11h 28m 44s ago, saying: 'Anyone know if there is a way to reset the GS BT101 other than the menu ? i have 2 phones that will not boot'. |
| 20:45.52 | docelmo | ~seen my_arse |
| 20:45.53 | jbot | i haven't seen 'my_arse', docelmo |
| 20:45.53 | x86 | jbot: lose some weight then you cow |
| 20:45.55 | jbot | some weight then you cow: You Lose. |
| 20:45.55 | tzanger | ~seen my dick in three years, and god is it depressing |
| 20:45.58 | jbot | tzanger: i haven't seen 'my dick in three years, and god is it depressing' |
| 20:45.58 | lullabud | hahaha |
| 20:46.06 | x86 | tzanger: rofl |
| 20:46.10 | docelmo | HAHAHA! |
| 20:46.20 | syle | ~seen my_balls_since_i_was_casterated |
| 20:46.21 | jbot | i haven't seen 'my_balls_since_i_was_casterated', syle |
| 20:46.40 | ManxPower | There's nothing wrong with playing with the bot, as long as you do it in private and wash your hands after |
| 20:47.14 | ManxPower | ~fxsfxo |
| 20:47.16 | jbot | fxsfxo is, like, an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 20:48.25 | *** join/#asterisk tuxd00d (n=tuxinato@69-169-11-49.lmdaca.adelphia.net) |
| 20:48.52 | *** join/#asterisk wilsto (n=icechat5@85.230.55.66) |
| 20:49.52 | brettnem | jbot: lose survey says |
| 20:49.53 | jbot | survey says: You Lose. |
| 20:50.30 | robl^ | ~robl^ |
| 20:50.32 | jbot | methinks robl^ is the enemy |
| 20:50.52 | fourcheeze | when a call on a Local channel is made should all the existing variables be passed on to it? |
| 20:51.07 | ManxPower | fourcheeze, yes. no. |
| 20:51.15 | fourcheeze | ManxPower :-) |
| 20:51.20 | ManxPower | see chanlocal.txt in the asterisk dource docs dir |
| 20:51.22 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 20:51.28 | ManxPower | also see README.variables |
| 20:51.35 | fourcheeze | ManxPower ok |
| 20:53.24 | fourcheeze | ManxPower I don't see anything about it in README.variables |
| 20:53.35 | fourcheeze | don't seem to have chanlocal.txt here :/ |
| 20:54.08 | ManxPower | /home/software/asterisk/asterisk-1.2/doc/localchannel.txt |
| 20:54.13 | fourcheeze | ahh |
| 20:54.45 | *** join/#asterisk SrFr (n=jeroenz@83.119.27.39) |
| 20:55.59 | fourcheeze | ManxPower so do I want /n ? |
| 20:56.33 | ManxPower | fourcheeze, or use the _ and __ features in 1.2.x to make the variables be inherited. |
| 20:58.37 | a1fa | yo yo yo |
| 20:58.44 | a1fa | back to my question |
| 20:59.05 | a1fa | whats the cpu time, memory vs. ulaw sip call? |
| 20:59.14 | a1fa | i cant find anything on google |
| 20:59.24 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 21:01.54 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:02.00 | a1fa | [TK]D-Fender : sup bud? |
| 21:02.03 | [TK]D-Fender | ntm |
| 21:02.18 | [TK]D-Fender | just relaxing this week. recovering from lasik on friday |
| 21:02.46 | syle | i had that to |
| 21:02.51 | syle | back at work in 2 days |
| 21:03.18 | schirpich | do they have full ten digit numbers that route to emergency 911? |
| 21:03.20 | syle | first 2 days i spent trying to focus on my screens lol |
| 21:03.32 | schirpich | just asking if there is such a thing |
| 21:03.40 | [TK]D-Fender | syle : same here, just have foxus issues and eys dry out fast |
| 21:03.43 | file | [TK]D-Fender: how did that go? |
| 21:03.47 | a1fa | [TK]D-Fender : quick question.. how much memory + cputime you need per sip peer? |
| 21:04.02 | [TK]D-Fender | a1fa : Um, some, and more? :) |
| 21:04.14 | syle | yeah man keep putting those eye drops in everytime your eyes feel dry, don;t screw them up :) |
| 21:04.19 | a1fa | lets say, with 96mb, how many active calls can i handle? |
| 21:04.28 | [TK]D-Fender | a1fa : not a clue |
| 21:04.32 | Qwell[] | several. test it |
| 21:04.34 | a1fa | anybody in here? |
| 21:04.41 | a1fa | Qwell: i tested with 4 calls |
| 21:04.46 | a1fa | i cant test with more |
| 21:04.46 | Qwell[] | so test with 8 |
| 21:04.52 | Qwell[] | then 4 is the ilmit |
| 21:04.52 | a1fa | how? |
| 21:04.54 | Qwell[] | limit too |
| 21:04.58 | [TK]D-Fender | file : went well I guess, takes some time to stabilize. I'm due in for a checkup on thrursday and again a few times through the year |
| 21:05.05 | obiwanmikenolte | Is there any way to simulate call volume? |
| 21:05.14 | a1fa | obiwanmikenolte : that would be good |
| 21:05.20 | a1fa | i tested with 4 users |
| 21:05.27 | a1fa | i dont have any more users to test with |
| 21:05.28 | syle | i retrained my eyes in close range starting at computer screen, then at far distances again staring outside on car drives :) |
| 21:05.36 | a1fa | it should handle at least 20-30 |
| 21:06.08 | a1fa | there should be a formula somewhere |
| 21:06.26 | a1fa | x=cpu, y=memory z=number of stim sip calls |
| 21:06.29 | syle | one thing is true, its exactly like wearing contacts full time hehe |
| 21:06.35 | obiwanmikenolte | There are too many variables in the dialplan for a formula, really. It all depends on what your calls are doing |
| 21:06.35 | [TK]D-Fender | a1fa : with or without memory leaks? ;) |
| 21:06.53 | Assid | err.. how do you set a limit on channels per sip peer? |
| 21:07.38 | a1fa | Gumstix-based SIP-to-IAX proxy server can route about 40 calls concurrently (no transcoding, Gumstix 400XM, voicemail on CF card) |
| 21:07.44 | a1fa | ok |
| 21:07.45 | syle | Fender: i always tell people the best investments are something you will use everyday, and that had to be my best one :) |
| 21:07.50 | a1fa | i found a document on the voip-info |
| 21:07.58 | a1fa | if gumstix can handle 40 concurent calls |
| 21:08.04 | a1fa | i am sure I can handle 100 concurent calls |
| 21:08.07 | Qwell[] | a1fa: comparisons like that are impossible |
| 21:08.07 | a1fa | on 96mb of ram |
| 21:08.12 | a1fa | :) |
| 21:08.12 | justinu | haha |
| 21:08.17 | a1fa | Qwell : damn you |
| 21:08.36 | [TK]D-Fender | syle : nice approach. I took mine as tired of glasses, I just broke them and I don't pay to PATCH problems, I pay to FIX them. |
| 21:09.48 | a1fa | Qwell: ok, i am thinking at least 30+ |
| 21:09.54 | syle | i think by time we all develop cataracts they will have artificial eyes, some clinics exist already for it in the US, perfect sight even in old age |
| 21:09.56 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
| 21:10.02 | Qwell[] | a1fa: only way to know, is to test |
| 21:10.23 | a1fa | Qwell[] : yeah, i will test |
| 21:10.35 | Qwell[] | get something like sipp to generate some calls |
| 21:10.43 | a1fa | :P |
| 21:10.54 | Assid | i want a free did :P |
| 21:11.09 | *** join/#asterisk ad3 (n=ad3@walden.acns.colostate.edu) |
| 21:11.14 | a1fa | http://www.astertest.com/ |
| 21:11.17 | a1fa | oh wow |
| 21:11.26 | Nivex | Assid: http://www.ipkall.com/ |
| 21:12.16 | syle | interesting statement Fender, although you can;t pay to fix something when a technology don;t exist, like "fix my cancer", might want to revise that :) |
| 21:12.30 | a1fa | dang dude |
| 21:12.39 | a1fa | i have 3ghz available and only 96mb :) |
| 21:12.47 | a1fa | i need to update ram, asap |
| 21:13.07 | [TK]D-Fender | syle : I never said I am able to APPLY it to all things did I? :) |
| 21:13.11 | a1fa | i will just add another 512 mb to my server |
| 21:13.28 | Qwell[] | You have a 3ghz machine with 96mb ram? |
| 21:13.38 | syle | who said the best things in life are free lol/ i give it a 50/50 |
| 21:13.54 | achandra | hello..any good sites on how to create various pcap files? other than sipp site...its kinda light in there..more of a curiosity question.. |
| 21:14.06 | Qwell[] | achandra: yeah, the tcpdump site, I think |
| 21:14.14 | syle | ok 80/20, having kids gives almost the full 80 i think |
| 21:15.16 | a1fa | Qwell[] : shared memory :) it has about 4gb |
| 21:15.35 | a1fa | shared cpu cycles, too.. |
| 21:15.51 | a1fa | <PROTECTED> |
| 21:15.56 | a1fa | but, i cant even feel it |
| 21:16.01 | a1fa | memory is the only problem |
| 21:16.12 | a1fa | it doesnt help that |
| 21:16.24 | a1fa | it runs 1000 processes |
| 21:16.25 | a1fa | :P |
| 21:16.40 | Qwell[] | and you plan on running a realtime pbx on that box? |
| 21:16.42 | Qwell[] | good luck |
| 21:16.48 | a1fa | it works great |
| 21:17.12 | a1fa | i never experienced any issue with it |
| 21:17.19 | a1fa | 4 users |
| 21:17.29 | websae | as long as you have no transcoding you'd be fine |
| 21:17.36 | a1fa | yup. its all sip |
| 21:17.44 | a1fa | err.. ulaw sip |
| 21:17.47 | websae | pasthru should be a breeze then |
| 21:18.51 | *** join/#asterisk tamp4x (n=Lab@64.201.13.172) |
| 21:18.59 | *** join/#asterisk RF_MIA (n=unknown@adsl-070-147-214-250.sip.mia.bellsouth.net) |
| 21:19.19 | a1fa | when i had 4 stim phone calls |
| 21:19.27 | a1fa | my idle was at 99.9% |
| 21:19.34 | a1fa | weerd, aint it |
| 21:21.45 | *** part/#asterisk ad3 (n=ad3@walden.acns.colostate.edu) |
| 21:22.43 | *** part/#asterisk RF_MIA (n=unknown@adsl-070-147-214-250.sip.mia.bellsouth.net) |
| 21:24.41 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:25.09 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 21:26.35 | *** part/#asterisk SrFr (n=jeroenz@83.119.27.39) |
| 21:28.10 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 21:31.30 | *** join/#asterisk Skarmeth (n=Skarmeth@200164212156.user.veloxzone.com.br) |
| 21:31.42 | Skarmeth | hi all |
| 21:32.48 | Skarmeth | I'm trying to set up CDR to a PostgreSQL database and I just get a failed ident error. Looking at postgresql logs, I can see that the user it's trying to use ident... how can I force it to use user and password? |
| 21:33.08 | Skarmeth | what king of auth mech the cdr_pgsql.so module supports? |
| 21:34.06 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:34.51 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 21:34.58 | diclophis | y0 |
| 21:35.05 | diclophis | so... what is the latest with asterisk + faxing |
| 21:35.25 | diclophis | mercestes... any news or further insight? |
| 21:38.15 | Skarmeth | May 15 18:39:46 ERROR[27602]: cdr_pgsql.c:299 process_my_load_module: cdr_pgsql: Reason: FATAL: Ident authentication failed for user "asterisk" |
| 21:38.34 | kay2 | could someone tell me when I register with asterisk on an other asterisk using IAX2, and there is an incomming call from the other asterisk, in what context does it come ? |
| 21:38.47 | Skarmeth | any way to put cdr_pgsql.so auth using md5 or so one? |
| 21:38.57 | *** join/#asterisk implicit (n=implicit@ip72-197-16-68.sd.sd.cox.net) |
| 21:38.59 | implicit | yone in? |
| 21:38.59 | obiwanmikenolte | kay2: you define context in iax.conf |
| 21:39.01 | implicit | anyone in? |
| 21:39.04 | implicit | :) |
| 21:39.11 | kay2 | obiwanmikenolte: I get that: NOTICE[3438]: chan_iax2.c:6796 socket_read: Rejected connect attempt from 213.161.201.200, who was trying to reach '0176606571@' |
| 21:39.14 | justinu | dave's not here man |
| 21:39.16 | ManxPower | kay2, The ONLY thing registration does is inform the other side the IP address associated with the username/secret pair. |
| 21:40.15 | kay2 | and when the other side says that there is a call comming |
| 21:40.45 | kay2 | ManxPower: I'm supposed to make an account for the otherside in iax.conf with incomming=something |
| 21:40.50 | ManxPower | kay2, looks like an auth poblem. |
| 21:41.16 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:41.23 | ManxPower | what does the Dial look like? |
| 21:42.02 | kay2 | ManxPower: I don't have it on the other side |
| 21:42.09 | kay2 | it's my IAX provider |
| 21:42.24 | kay2 | I register with my asterisk to my iax provider |
| 21:42.31 | kay2 | and when I call the number, it sends that to my asterisk |
| 21:42.32 | ManxPower | Ah. Your iax.conf is screwed up. Read the provider instructions again |
| 21:43.03 | ManxPower | For example, NuFone sends all calls to the NuFone user on the customer's server. |
| 21:43.03 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:43.12 | ManxPower | other providers do similar things. |
| 21:44.58 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 21:45.41 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 21:47.21 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 21:48.22 | kay2 | ManxPower: could you give me an example iax.conf with nuphone ? |
| 21:48.28 | kay2 | ManxPower: or a link ? |
| 21:48.46 | ManxPower | kay2, the info is in the e-mail they sent me when I signed up |
| 21:48.52 | ManxPower | ARE you using NuFone? |
| 21:50.54 | robl^ | NuFone has examples when you login to the customer control panel |
| 21:51.06 | *** join/#asterisk mgob (n=goldenol@65.171.196.23) |
| 21:51.07 | mgob | hi |
| 21:51.40 | mgob | is there a significant difference between the TDM2403 B/E cards .... trying to decide if I want to kickup the extra money for the echo cancellation when echo cancellation is built into Asterisk |
| 21:53.22 | robl^ | going home.. back in about 60 mins |
| 21:54.14 | diclophis | how do i get the return value of a dialplan app |
| 21:54.14 | diclophis | ? |
| 21:54.15 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 21:54.22 | Qwell[] | diclophis: What app? |
| 21:54.31 | Qwell[] | Dial() has ${DIALSTATUS} |
| 21:54.45 | Qwell[] | show application someApp, will show you what variable(s) it sets |
| 21:55.00 | diclophis | mmm well its the app_txfax one |
| 21:55.07 | kay2 | ManxPower: so you think it's an auth pb ? |
| 21:55.07 | Qwell[] | probably doesn't |
| 21:55.23 | diclophis | yea... it says it doesnt |
| 21:55.25 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 21:55.30 | ManxPower | kay2, It os. |
| 21:55.42 | ghenry | whats a busy lamp field for? |
| 21:55.46 | kay2 | ManxPower: and to what parameter is is related ? |
| 21:55.50 | kay2 | the permit is ok |
| 21:55.58 | Qwell[] | ~blf |
| 21:56.01 | jbot | from memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 21:56.05 | ManxPower | kay2, the [whatevertheprovidertellsyoutoputhere] section of iax.conf |
| 21:56.14 | kay2 | ohh |
| 21:56.20 | Qwell[] | ghenry: that help? |
| 21:56.22 | riogeed | for anyone here using a cdr logging method other than csv, how are you finding the start, answer and end times ? I can't find them in the pgsql structure or the other db's for that matter. |
| 21:56.23 | ManxPower | Now WHAT provider are you using? |
| 21:56.29 | kay2 | ManxPower: and If i do a iax2 debut, I can see it ? |
| 21:56.44 | Qwell[] | iax2 made it's debut years ago...you're too late |
| 21:56.53 | ManxPower | kay2, you would see the iax2 username, but nothing else that's required. |
| 21:56.55 | jsaunders | heheh |
| 21:57.46 | ghenry | Qwell: thanks. |
| 21:57.49 | ManxPower | Is your provider that much of a secret? |
| 21:59.27 | [Airwolf] | Does anyone has some experience with a Panasonic kx-tda100 and connecting a Asterisk box to it, with h323 ? |
| 21:59.43 | riogeed | time to run |
| 21:59.47 | *** part/#asterisk riogeed (n=riogeed@66-178-161-84.generic.rio.com) |
| 22:01.09 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:01.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:02.31 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:02.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:02.51 | Skarmeth | that cdr_pgsql.so module are crazy, I can connect to a postgresql database using ident auth with asterisk user, I can do select's on database tables, etc... but the module can't connect... and say it cannot connect because it failed to auth using ident... |
| 22:03.13 | dlynes_office | Skarmeth: use cdr_odbc instead...cdr_pgsql's not in active development |
| 22:03.35 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:03.42 | justinu | cdr_pgsql works fine for me |
| 22:04.03 | dlynes_office | yeah, but it's still not in active development |
| 22:04.04 | Skarmeth | juanjoc, using native access or using odbc? |
| 22:04.23 | Skarmeth | justinu, using native access or using odbc? |
| 22:04.51 | justinu | native |
| 22:05.04 | Skarmeth | can you share you configs with me? |
| 22:05.08 | Skarmeth | PVT... |
| 22:05.50 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:06.23 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 22:07.36 | ghenry | what DTMF Method should I use? Setting up a Aastra 9133i. * doesn't hear anything when I enter voicemail password etc. |
| 22:07.58 | ghenry | I have a choice of RTP, SIP INFO or BOTH |
| 22:08.00 | justinu | rfc2833 |
| 22:08.04 | justinu | RTP, in that case |
| 22:08.14 | Qwell[] | rtp...would that be inband? |
| 22:08.24 | fourcheeze | can't go wrong with both |
| 22:08.30 | Qwell[] | fourcheeze: yes you can |
| 22:08.31 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:08.36 | fourcheeze | ok |
| 22:08.40 | fourcheeze | you can go wrong with both |
| 22:08.41 | ghenry | This is ticked already "Force RFC2833 Out-of-Band DTMF" |
| 22:08.41 | Qwell[] | fourcheeze: You'll get double dtmf |
| 22:08.52 | fourcheeze | ghenry that's the one you want |
| 22:08.56 | Qwell[] | ghenry: That's the one you want. Just set it to rfc2833 in * |
| 22:09.07 | Qwell[] | sip.conf, dtmfmode |
| 22:09.18 | fourcheeze | or to be fair INFO isn't bad for some dodgy clients |
| 22:09.36 | ghenry | thanks, checking what's there |
| 22:10.12 | *** join/#asterisk jeffgus (n=jeffgus@alpha.zimage.com) |
| 22:10.28 | ghenry | I already set that, cool. |
| 22:13.00 | *** part/#asterisk stealthmethod (n=unknown@adsl-068-153-171-051.sip.mia.bellsouth.net) |
| 22:14.26 | *** join/#asterisk djacob-wcgs (i=djacob-w@c-68-32-12-249.hsd1.pa.comcast.net) |
| 22:16.03 | ghenry | I can't get * to recognise any DTMF for passwords etc. hmmmm |
| 22:16.32 | *** join/#asterisk ozverenm (n=ozverenm@204.27.103-84.rev.gaoland.net) |
| 22:18.53 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:24.38 | kay2 | is this correct: exten _01XXXXXXXX,1,Dial(IAX2/5947572280/${EXTEN}) |
| 22:25.04 | kay2 | If I want all the number that starts with 01 and are 10 digit long to be dialed from this IAX2 interface |
| 22:25.11 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 22:26.46 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 22:27.12 | obiwanmikenolte | That looks right. Where are you that your numbers start with 01? |
| 22:28.57 | *** part/#asterisk tessier (n=treed@66-162-45-90.gen.twtelecom.net) |
| 22:29.08 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 22:31.59 | ManxPower | kay2, only if you have a [5947572280] section in sip.conf |
| 22:32.14 | ManxPower | Sorry, you won't tell us your provider, so I cannot help your curther. |
| 22:32.16 | ManxPower | or further |
| 22:32.27 | hads|home | Does anyone know what driver the Digium B410P will be using? |
| 22:32.59 | *** join/#asterisk terrapen (n=cjs@166.70.183.109) |
| 22:33.32 | Qwell[] | B410P? |
| 22:33.44 | Qwell[] | Was there a product announcement that I missed? |
| 22:34.00 | Qwell[] | quad BRI interface? |
| 22:34.14 | hads|home | http://www.google.co.nz/search?q=B410P&ie=UTF-8&oe=UTF-8 |
| 22:34.15 | *** join/#asterisk ozverenm01 (n=ozverenm@204.27.103-84.rev.gaoland.net) |
| 22:34.16 | ManxPower | It will be whatever driver is listed in the zaptel README |
| 22:34.20 | Flauto | i was able to call toll free numbers through enumlookup, it is not working now, any idea? |
| 22:34.22 | ozverenm01 | hello all |
| 22:34.28 | Qwell[] | chances are, the driver will be named b4xxp |
| 22:34.34 | Qwell[] | if it's real |
| 22:34.36 | ozverenm01 | i have big problem with asterisk and PRI |
| 22:34.54 | hads|home | I just grepped through zaptel svn and didn't see anything. |
| 22:34.56 | ozverenm01 | my E1 link go down and up ... |
| 22:35.06 | ozverenm01 | I dont know how |
| 22:35.40 | ManxPower | hads|home, then the driver is not in the source yet |
| 22:35.53 | ozverenm01 | When I active debug I show SABME received .. |
| 22:36.06 | ozverenm01 | but no SETUP pass |
| 22:36.08 | macTijn | Qwell[]: it's real, it's on digium.com |
| 22:36.15 | cytrak | does * need the sound card on ? |
| 22:36.15 | macTijn | http://www.digium.com/en/mediacenter/news/viewpress.php?id=BRICard |
| 22:36.19 | Qwell[] | hmm |
| 22:36.23 | hads|home | Fair enough :) I was just wondering (hoping) that it was going to be zaptel. |
| 22:36.28 | Qwell[] | I completely missed that announcement...and I was there |
| 22:36.35 | Qwell[] | cytrak: no |
| 22:36.40 | macTijn | Qwell[]: I wonder if it's a junganns.net card |
| 22:36.41 | cytrak | I'm trying to disable most the stuff that is taking up my IRQS |
| 22:36.57 | Qwell[] | cytrak: usb, serial |
| 22:37.44 | cytrak | One of my digium's card is using irq 11 which according to lspci -vb my nvidia and network controller also use |
| 22:37.49 | file | the BRI card is not available yet |
| 22:37.53 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 22:38.10 | macTijn | file: so I see, but what kind of card is it? what chip? |
| 22:38.16 | CunningPike | cytrak: Ugh |
| 22:38.17 | cytrak | proc/interrupts only showed usb controllers though |
| 22:38.27 | ozverenm01 | anyone can help me ? |
| 22:38.46 | file | if I was in the hardware department... stuff would catch on fire more often |
| 22:38.54 | hads|home | lol |
| 22:38.55 | macTijn | file: haha :) |
| 22:38.59 | Qwell[] | file: It says it's available on the press release :p |
| 22:39.04 | CunningPike | ozverenm01: Not really - we'd need more details - maybe a pri debug span or something |
| 22:39.06 | cytrak | if anyone know how I can setup the irqs on a digium card please let me know |
| 22:39.10 | cytrak | jumpers ? |
| 22:39.11 | file | don't remind me |
| 22:39.22 | *** part/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
| 22:39.22 | Qwell[] | The Digium B410P is available now through Digium.s worldwide distributors and through the Digium website. |
| 22:39.28 | diclophis | anyone know of a miliwat test number? |
| 22:39.35 | file | Qwell[]: don't remind me |
| 22:39.38 | Qwell[] | diclophis: use app_milliwatt |
| 22:39.39 | CunningPike | cytrak: In the PC BIOS usually - you need to try and find an IRQ that the card can occupy by itself |
| 22:39.41 | Qwell[] | file: bad juju? |
| 22:39.50 | ozverenm01 | i have big problem with asterisk and PRI |
| 22:39.52 | file | silly marketing |
| 22:39.56 | ozverenm01 | my E1 link go down and up ... |
| 22:39.57 | Qwell[] | file: ahh |
| 22:40.03 | ozverenm01 | I cannot make calls |
| 22:40.21 | Qwell[] | file: They need people to review their stuff :p |
| 22:40.28 | macTijn | anyway |
| 22:40.36 | cytrak | me too .. my calls are now been dropped maybe every 45 minutes |
| 22:40.47 | cytrak | better than 3 minutes like it was before |
| 22:40.51 | CunningPike | cytrak: The trick is that many PCs share the PCI slot with other stuff on the same IRQ. We have to disable USB on our server, for example |
| 22:40.53 | macTijn | nn all |
| 22:41.14 | cytrak | CunningPike: yeah I'm doing so now |
| 22:41.35 | cytrak | CunningPike: but i think my main problem may be the nic and video |
| 22:41.43 | cytrak | I'll see what happens |
| 22:41.54 | CunningPike | cytrak: We did as Qwell said - USB and serial disabled. How many PCI slots do you have? |
| 22:42.14 | cytrak | 2 |
| 22:42.27 | CunningPike | cytrak: Try the other one |
| 22:42.42 | cytrak | ok i'll do that now ...brb |
| 22:44.43 | ozverenm01 | any help ? |
| 22:47.33 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 22:49.06 | [hC] | Hmm. I wonder what i should charge a client who's asking for hosted meetme.... :P |
| 22:49.23 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 22:49.24 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:49.54 | Qwell[] | [hC]: god...a lot |
| 22:50.01 | Qwell[] | [hC]: ever priced global crossing? |
| 22:50.13 | Qwell[] | we (you know where I work) pay $.10/min, PER connection |
| 22:50.20 | kay2 | how can 3 people conference at the same time |
| 22:50.24 | kay2 | is there an app for that ? |
| 22:50.37 | justinu | most phones can do 3 way calling internally |
| 22:50.40 | justinu | for more, app_meetme |
| 22:50.52 | Qwell[] | We have massive amounts of conf calls at a time...hundreds of thousands of minutes/month, no doubt, and we still only get $0.10/min |
| 22:51.15 | brif8 | hi all. If the network has spikes of non VoIP traffic eg FTP of 182 GB at < 840 kbps (1) will this effect quality (2) how can this be worked around so as not to effect quality ? |
| 22:51.28 | Qwell[] | brif8: QoS |
| 22:51.45 | Qwell[] | and yes, it'll kill voip without qos |
| 22:51.47 | [hC] | Qwell: no, but im looking at some canadian based conferencing systems who charge anywhere from 15 to 22c/min... no setup fees, but i guess they make that up per minute. |
| 22:51.55 | Qwell[] | [hC]: yeah |
| 22:51.56 | brif8 | Qwell[]: using a kentrox Q1300 Diffserv appliance for QoS, what else ? |
| 22:52.06 | Qwell[] | [hC]: It's an ungodly amount you can charge |
| 22:52.22 | Qwell[] | [hC]: YOu'd be competitive at $0.10/min, no setup |
| 22:52.43 | Qwell[] | but..dunno |
| 22:52.51 | Qwell[] | if it's high volume...do it, heh |
| 22:53.04 | CrashHD | .10 per user connected? |
| 22:53.11 | CrashHD | or .10 per minute per conference |
| 22:53.12 | CrashHD | ? |
| 22:53.31 | diclophis | is it possible to change the txgain in realtime? |
| 22:53.38 | Qwell[] | CrashHD: per connected user |
| 22:53.42 | Qwell[] | insane |
| 22:53.50 | CrashHD | $$ |
| 22:53.51 | CrashHD | :) |
| 22:53.56 | CrashHD | fun |
| 22:54.07 | Qwell[] | plus, it wouldn't surprise me if there is a monthly charge |
| 22:54.19 | CrashHD | bend over please |
| 22:54.26 | CrashHD | :) |
| 22:55.43 | CrashHD | who would have thought |
| 22:55.48 | CrashHD | communication would cost so much |
| 22:55.58 | CrashHD | to keep on the same page with people |
| 22:57.02 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 22:57.34 | supjigatr | Hello |
| 22:57.45 | *** join/#asterisk tuxd00d (n=tuxinato@69-169-11-49.lmdaca.adelphia.net) |
| 22:57.55 | brif8 | Qwell[]: I'm using a kentrox Q1300 a Diffserv type of appliance for QoS, what else could I do? |
| 23:00.06 | diclophis | hey.. so what are your alls rx and txgain settings? |
| 23:00.26 | dlynes_office | diclophis: 42 |
| 23:00.41 | diclophis | nice |
| 23:00.43 | diclophis | 42 |
| 23:00.48 | diclophis | the answer to everything |
| 23:00.55 | Qwell[] | no, not everything |
| 23:01.02 | Qwell[] | just rx and tx gains |
| 23:01.06 | dlynes_office | it's the answer to the universe, not everything :) |
| 23:01.20 | Qwell[] | dlynes_office: life, the universe, AND everything |
| 23:01.22 | diclophis | mm i thought it was the answere to to life universe and everything |
| 23:01.37 | CrashHD | ya just rx and tx, the answer to everything else is 42 + 1 |
| 23:01.43 | diclophis | i think it would be criminal to run at 42 txgain |
| 23:01.44 | dlynes_office | so as you can see, it's not just everything |
| 23:01.54 | diclophis | my line sounds like scratch at 10 txgain |
| 23:02.09 | dlynes_office | well, you asked for everyone's gains |
| 23:02.17 | dlynes_office | not just one person's :) |
| 23:02.47 | CrashHD | isn't scratch a very? |
| 23:02.49 | CrashHD | verb? |
| 23:02.54 | dlynes_office | anyways...it all depends on your line conditions what gain you will need to use |
| 23:02.58 | diclophis | you can hear a scratch? |
| 23:03.03 | dlynes_office | well, that and what kind of fxo/fxs adapter you're using |
| 23:03.09 | diclophis | te411p |
| 23:03.26 | dlynes_office | or t1 adapter, for that matter |
| 23:03.26 | [hC] | in a couple installs, ive used settings like 4/4 -5/10, 14/16 |
| 23:03.26 | diclophis | and it is related to a fax problem |
| 23:03.26 | diclophis | i cant get it to jive |
| 23:03.26 | [hC] | it really ranges. |
| 23:03.31 | supjigatr | diclophis? SPandsp? |
| 23:03.34 | diclophis | yea |
| 23:03.39 | diclophis | like rx works perfectly |
| 23:03.47 | diclophis | tx craps out after a page, if i am lucky |
| 23:03.59 | implicit | do u know what could cause audio prompts being played over a SIP channel to go super fast? |
| 23:04.01 | dlynes_office | diclophis: how are your interrupts? are they shared? |
| 23:04.06 | implicit | having that problem with voicemail |
| 23:04.18 | diclophis | mmm, technically speaking my interups for the digium card are shared with the raid controller |
| 23:04.20 | justinu | implicit: wrong RTP packetization value? |
| 23:04.29 | diclophis | /... is that going to be a _major_ problem? |
| 23:04.32 | diclophis | rx works fine |
| 23:04.33 | diclophis | ... |
| 23:04.33 | dlynes_office | diclophis: make it so it's not shared |
| 23:04.45 | dlynes_office | diclophis: faxing is very picky even when you don't have shared interrupts |
| 23:04.46 | diclophis | how can rx work fine and tx not though? |
| 23:04.56 | diclophis | can i manually set the irqs of devices |
| 23:05.02 | diclophis | i scoped the bios and i ran out of options |
| 23:05.17 | dlynes_office | diclophis: move the card to a different pci slot, then |
| 23:05.27 | diclophis | mmm its a 1u |
| 23:05.48 | diclophis | http://pastebin.com/719710 is my output from cat /proc/interupts |
| 23:06.11 | diclophis | is APIC important? |
| 23:06.14 | diclophis | i turned that off |
| 23:06.18 | justinu | APIC should help you |
| 23:06.25 | dlynes_office | It'll give you more interrupts |
| 23:06.26 | diclophis | i also turned off ACPI |
| 23:06.30 | justinu | ACPI isn't related |
| 23:06.30 | dlynes_office | acpi is bad bad bad |
| 23:06.32 | diclophis | mmm but they are virtual |
| 23:06.40 | diclophis | ok, i have acpi off |
| 23:06.48 | dlynes_office | virtual? you're on a virtual server? |
| 23:06.53 | diclophis | but are not APIC interupts 'virtual' |
| 23:06.58 | diclophis | no. dedicated hardware |
| 23:07.15 | diclophis | at least.. its presented to me as dedicated hardware |
| 23:07.17 | justinu | what's iop0? |
| 23:07.22 | diclophis | raid controller |
| 23:07.24 | justinu | ah |
| 23:07.32 | justinu | tried moving your card to a different slot? |
| 23:07.33 | dlynes_office | diclophis: have you run zttest? |
| 23:07.39 | dlynes_office | justinu: it's a 1U |
| 23:07.45 | justinu | so one slot? |
| 23:07.45 | diclophis | yea, 99.987 |
| 23:07.55 | dlynes_office | diclophis: how long did you let it run for? |
| 23:07.58 | supjigatr | I have same problems on 2U MPC. |
| 23:08.04 | diclophis | mmm 5 minutesd or so |
| 23:08.08 | justinu | APIC enabled, and SMP kernel might help you |
| 23:08.12 | diclophis | i will start it now while i investigate some other things |
| 23:08.13 | justinu | (even on a non SMP machine) |
| 23:08.14 | dlynes_office | diclophis: did you run patlooptest? |
| 23:08.28 | diclophis | i have a dual chip box with an smp kernil |
| 23:08.31 | diclophis | er kernel |
| 23:08.36 | robl^ | if you get unsolicited email from a member of congress, can ya sue for spamming? |
| 23:08.41 | diclophis | patlooptest ? |
| 23:08.54 | dlynes_office | diclophis: it's in your zaptel source directory...it needs to be compiled |
| 23:09.00 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 23:09.24 | diclophis | why the hell dont they include that in the build? |
| 23:09.25 | diclophis | !!! |
| 23:09.29 | dlynes_office | diclophis: you also need to use the clear= statement in your zaptel.conf for that span, and have a loop connector inserted into the jack |
| 23:09.44 | dlynes_office | diclophis: they don't include zttest in the build, either, but yet you managed to find that |
| 23:09.52 | diclophis | whoa whoa.. loop conector.. clear setting |
| 23:10.04 | diclophis | i dont have access to this machine |
| 23:10.14 | dlynes_office | ah...then patlooptest is out of the question |
| 23:10.20 | diclophis | what does it do though? |
| 23:10.44 | dlynes_office | It does a looped pattern test on the t1 circuit for that span |
| 23:11.03 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 23:11.17 | dlynes_office | it'll let you know if you have a bad port, or not |
| 23:11.57 | diclophis | mmm, what about zttool |
| 23:11.59 | diclophis | or ztdiag? |
| 23:12.08 | diclophis | i notice on zttool there is no 'output' |
| 23:12.18 | diclophis | it says the alarms are OK.. but the numbers never change |
| 23:12.23 | dlynes_office | diclophis: one of those requires some third party software |
| 23:12.30 | dlynes_office | diclophis: read the README file |
| 23:12.34 | diclophis | no, like i see the app |
| 23:12.45 | dlynes_office | it mentions some stuff in there about the ztdiag and zttool programs |
| 23:13.03 | diclophis | yea i see the interface and stuff |
| 23:13.09 | diclophis | but there are no numbers or anything |
| 23:13.16 | dlynes_office | 42 |
| 23:13.57 | diclophis | arrrg |
| 23:14.35 | dlynes_office | diclophis: have you contacted digium support about your problem? |
| 23:14.43 | diclophis | can i set the irqs in software? |
| 23:14.54 | dlynes_office | diclophis: you get 2 years of support for it |
| 23:14.54 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 23:15.05 | dlynes_office | diclophis: recompile your kernel, adding acpi support |
| 23:15.13 | dlynes_office | diclophis: you might get lucky |
| 23:15.16 | diclophis | mmm, i thought the official word from digium was faxing doesnt work |
| 23:15.21 | diclophis | you mean APIC ? |
| 23:15.26 | dlynes_office | erm yeah |
| 23:15.31 | dlynes_office | dyslexia's kicking in |
| 23:16.01 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 23:16.05 | diclophis | what is irqbalancer? |
| 23:16.07 | dlynes_office | the server's not even in the same city as you? |
| 23:16.48 | diclophis | nope, it is 2 hours away,. and i dont have a car |
| 23:16.54 | diclophis | i dont even think i have a loop tester |
| 23:16.58 | diclophis | nor does my it dept |
| 23:17.07 | justinu | those you can make |
| 23:17.19 | dlynes_office | steal one from the demarc closet :) |
| 23:17.51 | CrashHD | probably doesn't have a demarc closet either |
| 23:17.53 | CrashHD | :) |
| 23:18.08 | dlynes_office | every building's got a demarc closet |
| 23:18.12 | diclophis | i dont even know what a demarc closet is |
| 23:18.17 | CrashHD | heh see |
| 23:18.17 | dlynes_office | lol |
| 23:18.33 | *** part/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 23:18.34 | dlynes_office | demarcation closet...where your phone company terminates the phone lines for the building |
| 23:19.14 | dlynes_office | each floor usually has a riser closet, too, which is where the lines are terminated on each floor, and then jumpered to each office on the floor |
| 23:19.16 | diclophis | what about jitterbuffers? |
| 23:19.31 | dlynes_office | diclophis: jitterbuffers on a t1? |
| 23:19.35 | diclophis | yea |
| 23:19.40 | diclophis | enable or what? |
| 23:19.50 | dlynes_office | never even heard of them |
| 23:20.08 | dlynes_office | but yeah...if zapata.conf allows you to specify them, it might help |
| 23:20.19 | dlynes_office | it'll reorder the packets in the proper orders |
| 23:20.20 | diclophis | should i have alot or little of thems |
| 23:20.55 | CrashHD | medium:) |
| 23:20.56 | CrashHD | hah |
| 23:21.04 | CrashHD | in between |
| 23:22.18 | dlynes_office | Well, voip-info.org doesn't even mention the jitterbuffer on zaptel channels :) |
| 23:22.33 | dlynes_office | that sounds promising :) |
| 23:23.04 | dlynes_office | try jitterbuffers=42 |
| 23:23.14 | CrashHD | hah |
| 23:23.24 | CrashHD | it's so much fun fekking with people in here sometimes |
| 23:23.39 | CrashHD | I <i>almost</i> feel bad sometimes |
| 23:23.44 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:23.46 | dlynes_office | well, zapata.conf-dist seems completely useless for an example |
| 23:24.02 | diclophis | this is hopeless |
| 23:24.08 | jsaunders | jitter buffer on zapchan? is that an oxymoron? |
| 23:24.22 | dlynes_office | jsaunders: yeah...i'm not even sure why there would be one |
| 23:24.29 | dlynes_office | jsaunders: it's analog |
| 23:24.37 | dlynes_office | jsaunders: no packets on analog |
| 23:24.42 | CrashHD | pri/t1 is digital |
| 23:24.45 | CrashHD | although |
| 23:24.47 | CrashHD | using tdm |
| 23:24.56 | CrashHD | would make it useless |
| 23:24.57 | CrashHD | so I Dunno |
| 23:24.58 | CrashHD | weird |
| 23:25.23 | dlynes_office | Maybe someone felt like playing a joke on some poor hapless soul that would happen to encounter that when looking at the sample config |
| 23:25.30 | justinu | yeah, jitterbuffer is only important when going from VoIP to TDM/analog |
| 23:25.34 | dlynes_office | I didn't even know that option existed |
| 23:26.36 | diclophis | forget it theb |
| 23:26.40 | diclophis | do you guys have fax working? |
| 23:26.49 | dlynes_office | diclophis: You got it working more than I did |
| 23:27.06 | dlynes_office | diclophis: i didn't even get receive working very well |
| 23:27.19 | diclophis | mmm, i got rx fine i think |
| 23:27.24 | dlynes_office | diclophis: that's on a pri though |
| 23:27.24 | diclophis | upto 18 pages at a time tested |
| 23:27.36 | diclophis | yea, i tried on a sip connection |
| 23:27.39 | diclophis | didnt work for shit |
| 23:27.48 | CrashHD | no jitter on sip |
| 23:27.51 | CrashHD | sip = no fax support |
| 23:27.53 | dlynes_office | I just got a sangoma 2 fxs/2 fxo a200d today that i'm going to try testing faxing passthrough on |
| 23:27.53 | CrashHD | bad bad sip |
| 23:28.54 | dlynes_office | There's two main things I want to get working |
| 23:29.01 | dlynes_office | Fax passthrough on analog ports |
| 23:29.10 | dlynes_office | And fax transmission using call files on a pri |
| 23:29.10 | justinu | having trouble with that? or no time to test |
| 23:29.17 | dlynes_office | No time to test |
| 23:29.22 | justinu | you should be ok on both |
| 23:29.30 | dlynes_office | Almost finished my billing system now, though |
| 23:29.44 | dlynes_office | And starting testing on a new h323 provider shortly |
| 23:29.53 | justinu | eek |
| 23:29.57 | dlynes_office | Once I'm finished all that |
| 23:30.02 | dlynes_office | Then I can start testing faxing |
| 23:30.23 | dlynes_office | Well, I'm going to try testing it on my home machine first, not the production server |
| 23:30.27 | diclophis | how would i ramp up the disk activity on a raid mount? |
| 23:30.45 | dlynes_office | that way if it crashes, i don't lose a crapload of calls |
| 23:30.45 | justinu | there's some tools that will create disk activity |
| 23:31.00 | justinu | lemme take a look for the name |
| 23:31.14 | jsaunders | hdparm? |
| 23:31.31 | diclophis | thats for ata/ide |
| 23:31.32 | dlynes_office | i've heard of plenty of stability issues with h323 channel drivers, but nobody ever seems to mention which ones they're using |
| 23:31.34 | justinu | ah, i was using something called dbench |
| 23:31.46 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 23:31.56 | justinu | fortunately, i've avoided dealing with h323 so far |
| 23:32.36 | MRH2 | anyone here opened up a polycom phone ? |
| 23:32.45 | jsaunders | hdparm -T /dev/md0 is kinda fun. |
| 23:33.07 | dlynes_office | justinu: yeah, but this particular provider apparently has really good quality routes...that's why I'm going to try them out |
| 23:33.33 | dlynes_office | justinu: they do h323 for Canadian customers and SIP for their US customers |
| 23:33.41 | justinu | ah |
| 23:34.29 | jsaunders | /dev/md0: Timing cached reads: 3432 MB in 2.00 seconds = 1716.26 MB/sec |
| 23:34.29 | dlynes_office | h323, if it's stable is actually better in my mind |
| 23:34.29 | dlynes_office | It gives you much better error descriptors |
| 23:34.36 | Dr-Linux | h323 is getting down slowly slowly :P |
| 23:34.53 | dlynes_office | h323 is like pri on voip |
| 23:35.12 | Dr-Linux | :S |
| 23:35.26 | justinu | yeah - 323 came from the itu people, so makes sense |
| 23:35.46 | dlynes_office | all the signalling is similar to a pri, and it also uses q.831(?) |
| 23:37.25 | justinu | q.931? |
| 23:40.25 | dlynes_office | Yeah...that's what it was....that's why I put a (?) after it :) |
| 23:41.41 | justinu | yeah, i used to be intimately familiar with q931 |
| 23:42.25 | *** join/#asterisk delmar (n=delmar@203-114-178-231.static.inspire.net.nz) |
| 23:43.44 | dlynes_office | justinu: so do you happen to know if chan_h323.so that ships with asterisk requires openh323 or not? |
| 23:43.59 | MikeJ[Laptop] | dlynes_office, indeed |
| 23:44.13 | justinu | not sure |
| 23:44.13 | dlynes_office | ah...so chan_ooh323 is the only one that doesn't? |
| 23:44.41 | MikeJ[Laptop] | I beleive so |
| 23:44.48 | MikeJ[Laptop] | what's wrong with openh323? |
| 23:45.28 | dlynes_office | It depends on pwlib, which depends on openldap, sdl, and probalby some X11 libraries as well |
| 23:45.34 | *** join/#asterisk tuxd00d (n=tuxinato@69-169-11-49.lmdaca.adelphia.net) |
| 23:45.55 | dlynes_office | I'm just thinking i'm going to spend three or four days configuring everything just to get hello, world |
| 23:45.58 | justinu | wtf? x11? |
| 23:46.25 | dlynes_office | justinu: pwlib is a huge c++(?) framework library |
| 23:46.41 | dlynes_office | justinu: probably doesn't require all that crap to use it, but it does, to compile it |
| 23:47.49 | Flauto | i have an iax guest context and i am supposed to be able to get calls from that context, but it shows when a call is coming in and it hangs up right away. any idea on that problem? |
| 23:48.02 | justinu | that's annoting |
| 23:48.08 | justinu | s/annoting/annoying |
| 23:48.12 | dlynes_office | Flauto: it's probably going into a non-existent context |
| 23:48.18 | dlynes_office | Flauto: check your logs |
| 23:48.28 | Flauto | where to check the logs |
| 23:48.35 | dlynes_office | Flauto: default is /var/log/asterisk |
| 23:49.03 | dlynes_office | Flauto: i'm guessing you're using freepbx? |
| 23:49.47 | Flauto | i don't know what is freepbx |
| 23:49.53 | dlynes_office | amp? |
| 23:49.57 | Flauto | i am using mandriva and asterisk 1.2.7.1 |
| 23:50.16 | dlynes_office | ah...ok....i guess you must jsut be new to asterisk, then |
| 23:50.48 | Flauto | i see cdr-csv cdr-custom event_log messages queue_log |
| 23:51.12 | dlynes_office | Flauto: no full or debug? |
| 23:51.32 | Flauto | i am not very new but don't know much |
| 23:51.44 | Flauto | no |
| 23:51.46 | dlynes_office | Flauto: try the messages file then |
| 23:51.52 | Flauto | vi messages |
| 23:51.55 | dlynes_office | Flauto: if you can't find an error in there |
| 23:52.05 | justinu | dlynes_office: aren't you working pretty late? |
| 23:52.05 | tzafrir | view messages |
| 23:52.07 | dlynes_office | Flauto: then you'll need to edit your logger.conf file, and uncomment full |
| 23:52.08 | tzafrir | less messages |
| 23:52.21 | dlynes_office | justinu: It's only 4:50pm |
| 23:52.31 | justinu | oh, i thought you were in the UK |
| 23:52.40 | dlynes_office | justinu: no. Vancouver, Canada |
| 23:53.02 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:53.29 | dlynes_office | justinu: I live probably about 20 minutes away from CunningPike |
| 23:54.03 | Flauto | okay |
| 23:54.05 | justinu | ah |
| 23:54.05 | Flauto | let me try |
| 23:54.13 | justinu | i'm jealous |
| 23:54.18 | dlynes_office | justinu: how's that? |
| 23:54.20 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:54.32 | CunningPike | Living close to me :P |
| 23:54.42 | dlynes_office | CunningPike: look out...he's got the hots for you, man :) |
| 23:54.43 | justinu | living in a nice place like BC |
| 23:54.59 | dlynes_office | Yeah....BC kicks total ass |
| 23:55.11 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:55.18 | dlynes_office | I told my mother about five minutes after I moved here that I was never moving back to ontario :0 |
| 23:55.45 | dlynes_office | Now, she's decided she's moving out here this summer :) |
| 23:56.32 | Flauto | it seems that the messages file does not help much |
| 23:56.41 | justinu | i live in the shithole of southern california |
| 23:56.59 | dlynes_office | Flauto: if you look in /etc/asterisk/logger.conf, there'll be a line for full debugging that's commented out |
| 23:57.15 | dlynes_office | Flauto: uncomment it, do a reload, and then try your inbound call again |
| 23:57.16 | Flauto | i did change it just now |
| 23:57.31 | dlynes_office | Flauto: after that, you should have a file called 'full' in your /var/log/asterisk directory |
| 23:57.36 | dlynes_office | justinu: ah...San Diego? |
| 23:57.42 | justinu | burbs of LA |
| 23:57.54 | dlynes_office | Ah...everyone I know that's lived there said they loved it |
| 23:58.13 | justinu | i'm an unhappy native |
| 23:58.19 | dlynes_office | One of them moved back there again |
| 23:58.39 | dlynes_office | He lives somewhere near Beverley Hills |
| 23:58.46 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
| 23:59.12 | dlynes_office | But, after seeing the housing costs there, Vancouver's housing costs don't seem so bad |
| 23:59.18 | }btorch{ | why do I need to recompile asterisk when I recompile zaptel and libpri ? |
| 23:59.38 | dlynes_office | }btorch{: libpri depends on zaptel; asterisk depends on libpri |
| 23:59.53 | justinu | i think LA is a very stressful place to live... i'd like to kick back in Vancouver sometime |
| 23:59.58 | justinu | meet some more friendly people |
