| 00:01.08 | *** join/#asterisk crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net) |
| 00:02.25 | *** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com) |
| 00:07.04 | *** join/#asterisk forrestc (n=fwc@206.127.78.199) |
| 00:07.44 | forrestc | Has the name for cvs.digium.com changed? Or is DNS just broken right now....?? |
| 00:07.55 | Qwell[] | forrestc: it's dead Jim |
| 00:08.03 | Qwell[] | Don't you RTFML? |
| 00:08.13 | forrestc | Which manual? |
| 00:08.20 | Qwell[] | ML = mailing lists |
| 00:08.21 | pjchilds | ML... mailing list... |
| 00:08.22 | forrestc | The official digium one which was just updated? |
| 00:08.30 | forrestc | Ohhh Mailing list. |
| 00:08.42 | pjchilds | no more CVS... |
| 00:08.44 | Qwell[] | it was removed, in favor of svn |
| 00:08.49 | forrestc | great. |
| 00:08.55 | pjchilds | see www.asterisk.org/download or something for svn checkout instructions.... |
| 00:09.30 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 00:10.54 | forrestc | It would be nice if digium actually updated their guides... The Quick install guide (dated 3/13/06) still refers to CVS. |
| 00:10.59 | forrestc | Oh well... |
| 00:12.36 | pjchilds | the first casualty of software development companies is the documentation.... |
| 00:14.36 | file | no documentation foo you |
| 00:15.03 | Qwell[] | file: plenty of bar documentation though? |
| 00:15.32 | file | well, yes |
| 00:16.25 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:17.31 | Qwell[] | let's not even discuss the baz docs |
| 00:22.37 | *** join/#asterisk kardecallan (i=kardecal@201.14.101.23) |
| 00:29.20 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 00:29.43 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:32.27 | *** join/#asterisk wulfy814 (n=wulfy814@dsl093-061-214.pit1.dsl.speakeasy.net) |
| 00:32.47 | wulfy814 | evening folks, I'm having a strange issue |
| 00:33.10 | wulfy814 | Dell PE 430, CentOS, * 1.27, Sanomga Analog with HWEC |
| 00:33.11 | kernel20 | > |
| 00:33.23 | wulfy814 | when dialing LD sometimes it goes through sometimes it doesn't |
| 00:33.34 | wulfy814 | in the csv logs it always looks good |
| 00:33.47 | wulfy814 | although I often get "When dialing this number you must dial a 1" |
| 00:33.57 | Qwell[] | So dial a 1 |
| 00:33.59 | wulfy814 | even though the "1" was dialed and shows in the logs |
| 00:34.08 | Qwell[] | hmm |
| 00:34.17 | wulfy814 | sometimes it goes through fine |
| 00:34.24 | wulfy814 | seems completely random |
| 00:34.27 | pjchilds | perhaps you should pause before you dial... ie offhook, then pause, then dial... |
| 00:35.05 | wulfy814 | pjchilds: I'm just doing Dial(Zap/G1,{ARG1}) |
| 00:35.09 | wulfy814 | I think |
| 00:35.16 | wulfy814 | should I insert a pause |
| 00:35.28 | wulfy814 | maybe * is too fast for Analog Verizon line? |
| 00:36.27 | Qwell[] | wulfy814: make it w${ARG1} |
| 00:36.52 | Qwell[] | * is probably too fast for the sangoma. ;) |
| 00:38.50 | pjchilds | something about stones and glass houses... i can't remember... |
| 00:39.06 | pjchilds | heard a funny one yesterday... someone running a sip phone via citrix terminal services... |
| 00:39.13 | pjchilds | had some bad audio problems |
| 00:45.22 | Gabriel25 | can someone help me to install g729 codec in asterisk ? |
| 00:48.07 | *** join/#asterisk cyn0n (n=cyn0n@12-217-37-35.client.mchsi.com) |
| 00:48.23 | pjchilds | Gabriel25, have you purchased it from digium? |
| 00:51.49 | cyn0n | I tried following the instructions at http://astrecipies.net/index.php?q=AstRecipies/Installing%20the%20Linksys%20PAP2-NA%20port%20adapter with my unlocked pap2 and still don't see pap2 registering in the asterisk cli |
| 00:51.58 | cyn0n | anyone know where I can find some more info? |
| 00:53.23 | pjchilds | cyn0n, have you typed 'sip debug' in your asterisk console, then power cycled your PAP2 ? |
| 00:54.08 | *** join/#asterisk Splat (n=Splat@220-253-102-19.TAS.netspace.net.au) |
| 00:54.11 | cyn0n | I'll try that.. |
| 00:54.31 | docelm0 | someone tell me this is the right way to set callerID.. set(callerid(num)=NPANXXXXXX) right? |
| 00:55.39 | cyn0n | pjchilds: nothing showed up |
| 00:57.25 | Gabriel25 | pjchilds I want to try the dome version g729 |
| 00:57.44 | Gabriel25 | demo |
| 00:57.59 | docelm0 | anyone? |
| 00:58.22 | Bullseye_Network | docelm0: ${CALLERID(num)} |
| 00:58.40 | docelm0 | really when the hell did that change? |
| 00:58.53 | docelm0 | thats to get the number.. |
| 00:58.57 | docelm0 | I want to set it tho |
| 00:58.57 | *** join/#asterisk kardecallan (i=kardecal@201.14.101.23) |
| 00:59.06 | Bullseye_Network | or ,SetCIDnum(${CallerID}) |
| 00:59.21 | docelm0 | I thought that was depreciated.. but I will try it. |
| 00:59.26 | Bullseye_Network | it is |
| 01:00.18 | docelm0 | fukin amazing.. it still works |
| 01:01.05 | *** join/#asterisk gregbva (n=rufus@darkfibre.org.uk) |
| 01:01.41 | Bullseye_Network | depreciated for: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid |
| 01:02.08 | Bullseye_Network | so i guess: exten => s,1,Set(CALLERID(name)=Foo Fighters) |
| 01:02.15 | Bullseye_Network | you was right to begin with. |
| 01:02.58 | Bullseye_Network | well: set(callerid(numBER)= |
| 01:03.20 | Bullseye_Network | not just 'num' it shows 'number' in the page |
| 01:09.08 | docelm0 | really? |
| 01:09.12 | docelm0 | aint that some shit |
| 01:09.30 | docelm0 | ahh well |
| 01:10.01 | docelm0 | The reason I was asking is to keep that damn extension from showing up on my phone @ my home office |
| 01:10.14 | docelm0 | Now that plainvoip has phone numbers people can dial |
| 01:10.47 | dlynes_home | ~cvs |
| 01:10.48 | jbot | cvs stands for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS can be found at http://asterisk.espia-net.net/horde/chora/cvs.php. It is also Giving.....you.....the.......source......one..........bit.........at..............a.....................time. |
| 01:10.48 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-16-48.w86-207.abo.wanadoo.fr) |
| 01:11.08 | pjchilds | ~slap jbot |
| 01:11.09 | jbot | ACTION slaps jbot, keep your grubby fingers to yourself! |
| 01:11.18 | [hC] | dlynes_home: you were saying earlier about the a200 ec, that it doesnt appear that its something you can slap on after? |
| 01:11.28 | [hC] | dlynes_home: sorry i wasnt able to really read the details of your msg earlier |
| 01:11.43 | dlynes_home | jbot no cvs standard for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 01:11.44 | jbot | that's too long, dlynes_home |
| 01:11.57 | dlynes_home | jbot no, cvs standard for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 01:11.58 | jbot | that's too long, dlynes_home |
| 01:12.11 | dlynes_home | How do you update a jbot entry again? |
| 01:12.23 | dlynes_home | jbot no, cvs stands for concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 01:12.24 | jbot | dlynes_home: that's too long |
| 01:12.29 | docelm0 | kick em in the balls |
| 01:12.41 | [hC] | i think you need an 'is' instead of stands for |
| 01:12.43 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-211.cybersurf.com) |
| 01:13.19 | [hC] | jbot no, cvs is concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 01:13.20 | jbot | okay, [hC] |
| 01:13.24 | [hC] | like that. |
| 01:13.35 | *** join/#asterisk topping (n=topping@66.89.140.136.ptr.us.xo.net) |
| 01:13.43 | docelm0 | difference was? |
| 01:13.48 | docelm0 | ~cvs |
| 01:13.49 | jbot | cvs is probably concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 01:13.50 | [hC] | see my 'is' |
| 01:13.59 | docelm0 | ahh |
| 01:14.24 | dlynes_home | ah |
| 01:14.39 | [hC] | dlynes_home: what was the deal with the a200 echocan? |
| 01:14.46 | docelm0 | jbot no, mybutt is HUGE and stands for some funky stuff... |
| 01:14.48 | jbot | docelm0: okay |
| 01:14.48 | docelm0 | :) |
| 01:14.55 | docelm0 | ~mybutt |
| 01:14.57 | jbot | hmm... mybutt is HUGE and stands for some funky stuff... |
| 01:15.03 | docelm0 | haha |
| 01:15.23 | dlynes_home | [hC]: It seems to be a new bus soldered onto the main board, and then a new daughterboard plugged into that bus and fastened to the main board with a post |
| 01:15.59 | dlynes_home | [hC]: by bus i mean something like a 14 or 16 pin connection (7x2 or 8x2) |
| 01:16.11 | dlynes_home | [hC]: i'm not at the office, so I can't fully describe it to you |
| 01:16.33 | dlynes_home | [hC]: but, if you look at your existing main a200d board |
| 01:16.43 | dlynes_home | [hC]: you'll see the spot where it gets soldered onto |
| 01:17.30 | [hC] | dlynes_home: ahh yeah i think i know what you're referring to |
| 01:17.40 | [hC] | and it solders on there and the new daughterboard goes under the main pcb? |
| 01:17.50 | [hC] | as opposed to the standard remora thats bolted on top (With the two module slots) |
| 01:17.54 | dlynes_home | [hC]: it'll be under the bottom right-most corner of the module that's closest to the rj11 jacks |
| 01:18.11 | [hC] | ahh. |
| 01:18.39 | dlynes_home | [hC]: the ec daughterboard is sandwiched in between the main board and the remora daughterboard |
| 01:18.50 | [hC] | gotcha. |
| 01:18.58 | docelm0 | What are you guys working on? |
| 01:19.02 | [hC] | I thought the EC board was a module that went in one of the two FXO/FXS module ports |
| 01:19.20 | dlynes_home | [hC]: iow, you have the standard main board, the standard remora board, your standard modules, and an additional board that gets plugged into a new bus that's soldered onto the main board |
| 01:19.23 | [hC] | holy christ these messages are getting out of control |
| 01:19.23 | [hC] | <PROTECTED> |
| 01:19.33 | [hC] | from polycom 501's |
| 01:19.40 | dlynes_home | [hC]: nope |
| 01:19.40 | dlynes_home | [hC]: then that would cut you down to 22 ports, instead of 24 |
| 01:19.43 | [hC] | i presume its from the buddy watch stuff |
| 01:20.03 | [hC] | dlynes_home: 10-4.. makes sense. |
| 01:20.14 | dlynes_home | [hC]: i'm getting that crap from an Aastra 9133i, too |
| 01:21.09 | dlynes_home | [hC]: I sent a bug report into them today about it |
| 01:21.09 | dlynes_home | SIP error 500, and 498 |
| 01:21.16 | dlynes_home | Gave them a good three or four screen to thumb over |
| 01:21.23 | dlynes_home | s/screen/screens/ |
| 01:21.47 | [hC] | haha |
| 01:22.10 | [hC] | they seem to be harmless |
| 01:22.20 | [hC] | but i have 40 phones buddywatching, and they all spew it out |
| 01:22.24 | [hC] | seemingly |
| 01:22.30 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 01:22.53 | dlynes_home | [hC]: nah...the ones i've had spew it out spit up an Out of Service status message on the phone, and then go missing off the asterisk radar |
| 01:22.59 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:23.09 | dlynes_home | [hC]: and then every once in a while, they rear their ugly heads again |
| 01:23.39 | dlynes_home | [hC]: they all end up appearing as unreachable on asterisk |
| 01:24.20 | dlynes_home | [hC]: sangoma's been giving me pretty crappy service lately too, so i bitched and complained |
| 01:24.22 | [hC] | mine dont go unreachable. |
| 01:24.32 | dlynes_home | [hC]: they sent me a free sweatshirt with the last order :) |
| 01:24.32 | [hC] | sangoma's been great to me |
| 01:24.43 | [hC] | ooh maybe ill bitch too |
| 01:24.43 | [hC] | :) |
| 01:24.47 | dlynes_home | lol |
| 01:25.13 | dlynes_home | Yeah...all my shipments have gone out late |
| 01:25.19 | Qwell | I'm still waiting on my Sun shirt |
| 01:25.31 | *** join/#asterisk samer (n=pal@83.244.77.232) |
| 01:25.32 | dlynes_home | When they do ship something, the shipment gets here pretty fast though |
| 01:25.51 | dlynes_home | [hC]: it's roots brand, too :) |
| 01:25.57 | dlynes_home | [hC]: so it's pretty good quality |
| 01:26.27 | dlynes_home | wtf? |
| 01:26.38 | [hC] | nice :) I should try to get one out of them |
| 01:26.40 | wulfy814 | I'm looking at Master.csv (default) |
| 01:26.47 | wulfy814 | and duration doesn't seem right |
| 01:26.56 | [hC] | yeah these 500 errors have been happening for a long time with polycom's buddy watch system, and it doesnt seem to have any negative effect that i can see |
| 01:27.02 | [hC] | however i have read reports of it happening during call transfers |
| 01:27.03 | dlynes_home | [hC]: If your rep is doug vilim, you might have a hard time getting one :) |
| 01:27.06 | [hC] | with odd behaviour |
| 01:27.27 | dlynes_home | [hC]: well, maybe the aastra firmware is just buggy |
| 01:27.38 | dlynes_home | [hC]: and same for polycom |
| 01:27.50 | dlynes_home | [hC]: but polycom's fails predictably, and aastra's doesn't |
| 01:29.30 | [hC] | only people ive spoken with at sangoma are a kathryn something, Nenad corbic, and im not sure who else. |
| 01:29.44 | [hC] | what problems have you found with polycom's? |
| 01:30.06 | dlynes_home | [hC]: yeah i think i've spoken with kathryn too...she's in shipping or something, right? |
| 01:30.11 | [hC] | yeah. |
| 01:30.16 | dlynes_home | [hC]: none...i don't use polycom yet |
| 01:30.19 | [hC] | oh, apparently our rep is buddy cruz |
| 01:30.25 | dlynes_home | [hC]: we're probably going to use it on our next customer though |
| 01:30.36 | [hC] | I am leaning heavily towards polycom now |
| 01:30.41 | [hC] | i used to lean heavily on cisco |
| 01:30.44 | [hC] | i refuse to use aastra. |
| 01:30.55 | dlynes_home | cisco is severely overpriced |
| 01:30.59 | [hC] | indeed. |
| 01:31.03 | dlynes_home | plus you get dinged every time you want to upgrade |
| 01:31.04 | [hC] | underfeatured for the price |
| 01:31.06 | dlynes_home | it's ridiculous |
| 01:31.08 | [hC] | soiid, though. |
| 01:31.11 | [hC] | solid rather |
| 01:33.07 | pjchilds | anyone used the Linksys SPA-941/942... they 'looked' reasonable |
| 01:33.31 | pjchilds | or Snom 320 ? |
| 01:34.28 | [hC] | i use the 941/942 |
| 01:34.31 | dlynes_home | pjchilds: i've used the snom220, with the sidecars |
| 01:34.35 | [hC] | they're good for the price |
| 01:35.17 | dlynes_home | pjchilds: it's a good quality phone, but the handset is too light, and I swear they never actually tried using the user interface themselves |
| 01:36.47 | pjchilds | i've played with the snom 190.. and its nice.. but its just doesn't feel right to use (dialing, phonebook, etc...) |
| 01:37.12 | pjchilds | i did like doing SIP traces on the phone via the web browser .. that looked neat.. or updating the firmware straight from a web URL :) |
| 01:37.33 | dlynes_home | pjchilds: don't most sip phones support all that? |
| 01:38.51 | dlynes_home | [hC]: aren't the linksys phones roughly the same price as a polycom? |
| 01:39.24 | pjchilds | well the only other ones I've played with are a few atas, and the zytel wi-fi (older one)... |
| 01:39.30 | [hC] | I get the linksys phones for roughly 170 |
| 01:39.40 | [hC] | i forget what i get the polycom for |
| 01:39.57 | dlynes_home | [hC]: yeah...so like about a $10 or $20 price difference from the 501's |
| 01:40.10 | [hC] | id take a 501 over it for sure. |
| 01:40.14 | dlynes_home | I'd rather pay the extra 10 or 20 and get a polycom instead |
| 01:40.51 | dlynes_home | We were going with the aastras because we could get a 9133i for $135Cdn |
| 01:41.36 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 01:44.13 | dlynes_home | anyways...i'm out for a while |
| 01:45.51 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:50.43 | *** join/#asterisk oej (n=oej@65.246.174.67) |
| 01:51.45 | [hC] | Damn you polycom, why must you torment my console with your 500 internal server errors |
| 01:52.06 | bigmac4444 | give it whatfor |
| 01:52.53 | Bullseye_Network | anybody know what this is: !! Unknown IE 0 (cs6, Unknown Information Element) |
| 01:53.34 | bigmac4444 | isdn? |
| 01:54.15 | Bullseye_Network | The server has 2 405 cards... |
| 01:54.20 | Bullseye_Network | and we always get these |
| 01:54.50 | Bullseye_Network | since we upgraded to 1.2.7.1 we I also noticed we get: May 25 19:54:11 WARNING[18621]: cdr.c:550 ast_cdr_disposition: Cause not handled |
| 01:54.55 | Bullseye_Network | alot of these |
| 01:57.33 | [hC] | the apple.ca website has been screwed for days on the mac book pro order now button |
| 01:57.36 | [hC] | never takes me to the store |
| 01:58.32 | [hC] | interestingly enough i cant get to the canadian store at all! |
| 01:59.21 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 02:03.02 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 02:03.16 | Bullseye_Network | He all of a sudden im getting: pbx.c:5906 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 02:03.20 | Bullseye_Network | what is this? |
| 02:03.47 | Bullseye_Network | Never seen that one before |
| 02:04.13 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 02:04.35 | *** join/#asterisk RF_MIA (n=markmia@24-55-227-232.miamfl.adelphia.net) |
| 02:06.33 | *** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 02:06.50 | chino | what does "== Everyone is busy/congested at this time" |
| 02:06.57 | chino | mean ? |
| 02:07.18 | [TK]D-Fender | chino : Means nobody you dialed wants to hear about your call. |
| 02:07.36 | chino | what |
| 02:07.57 | chino | if i call my self it should loop right |
| 02:07.58 | [TK]D-Fender | chino It means exactly what it says.... |
| 02:08.16 | [TK]D-Fender | chino : what do you mean "loop"? |
| 02:08.22 | chino | hear my self |
| 02:08.28 | [TK]D-Fender | chino :No. |
| 02:09.16 | [TK]D-Fender | chino : If your phone supports more than one call it still won't KNOW you're calling yourself, jsut that its another incoming call. By ansewring the incoming line you are putting the other call on hold. |
| 02:09.43 | chino | what ports do i need open tcp/udp 5060 and tcp/udp 8000 ? |
| 02:10.22 | *** join/#asterisk vechers (n=svecher@64.61.117.138) |
| 02:10.43 | chino | for clients |
| 02:10.53 | *** part/#asterisk vechers (n=svecher@64.61.117.138) |
| 02:11.43 | pjchilds | udp 5060, plus all udp ports between rtpstart and rtpend in /etc/asterisk/rtp.conf and it _might_ work |
| 02:12.33 | chino | my soft phoen is not ringing |
| 02:13.17 | chino | why so many ports ? |
| 02:13.25 | chino | 10,000 ports! |
| 02:13.43 | chino | each port is an rtp session correct ? |
| 02:13.43 | [TK]D-Fender | chino : Try actually EXPLAINING your situation. Please provide actual details. You havn't told anything useful, like what you have tried. Who is trying to call you. From where. What is their setup like. Etc... |
| 02:14.02 | chino | just testing with gizmo |
| 02:14.06 | chino | im vnc'ed into a freinds pc |
| 02:14.20 | chino | im running an asterisk server |
| 02:15.01 | [TK]D-Fender | chino : Ok, you are behind NAT? |
| 02:15.03 | Qwell | [TK]D-Fender: YOU! |
| 02:15.11 | [TK]D-Fender | Qwell : ME! |
| 02:15.29 | Qwell | http://pastebin.ca/58769 |
| 02:15.34 | Qwell | I beg to differ. |
| 02:15.35 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:15.45 | Qwell | :P |
| 02:16.05 | Gabriel25 | rtpstart=10000 |
| 02:16.05 | Gabriel25 | rtpend=20000 |
| 02:17.16 | [TK]D-Fender | Qwell : I still hold that while he is slow, you are still an ASS :) |
| 02:17.24 | Qwell | note the span of several weeks on some of the questions |
| 02:17.45 | [TK]D-Fender | Qwell : At least he's not repeating them every 5 seconds, no? |
| 02:17.58 | Qwell | ha |
| 02:18.08 | [TK]D-Fender | Qwell : Lesser of 2 evils. |
| 02:18.10 | Qwell | 09-21 |
| 02:18.23 | [TK]D-Fender | chino : Just pastebin your entire sip.conf please. |
| 02:18.33 | Gabriel25 | who can help me out with the codec g729 demo version ? |
| 02:18.38 | Qwell | [TK]D-Fender: I assure you, that's in there too :p |
| 02:19.13 | Qwell | I think my favorite is on 09-21 |
| 02:19.16 | Qwell | erm, 09-02 |
| 02:19.34 | [TK]D-Fender | Qwell : line #? |
| 02:19.43 | Qwell | ~330 |
| 02:20.17 | Gabriel25 | who can help me out install the codec g729 demo version ? |
| 02:20.32 | mogorman | there is no demo version |
| 02:20.40 | [TK]D-Fender | Qwell : Yay. empty threat time. |
| 02:20.44 | Qwell | [TK]D-Fender: heh |
| 02:20.45 | Gabriel25 | hmmm not good |
| 02:20.47 | Gabriel25 | :) |
| 02:20.51 | mogorman | its 10 dollars |
| 02:20.53 | mogorman | for one channel |
| 02:20.54 | *** join/#asterisk gandhijee (n=gandhije@c-24-147-213-182.hsd1.ma.comcast.net) |
| 02:20.57 | mogorman | come on |
| 02:21.01 | gandhijee | can someone tell me what Class 5 IP telephony features are? |
| 02:21.02 | Qwell | or #465 |
| 02:21.05 | mogorman | you know what it is going to sound like |
| 02:21.29 | Qwell | gandhijee: 1 higher than class 4 |
| 02:21.33 | mogorman | lol |
| 02:21.45 | mogorman | why do you keep asking that gandhijee |
| 02:21.48 | Qwell | [TK]D-Fender: You're love ~465..really |
| 02:21.52 | Gabriel25 | mogorman is not about 10 $ is about yesterday I installed first ime in my life asterisk |
| 02:22.05 | Gabriel25 | and I have upstrem 40 ks |
| 02:22.22 | Gabriel25 | and I have only 4 extensions |
| 02:22.30 | Qwell | 40 kbit/s? |
| 02:22.37 | Qwell | You won't get 4 calls with that |
| 02:22.38 | Gabriel25 | yes |
| 02:22.42 | [TK]D-Fender | Qwell : 20 minuite repeat.... problem is you are also including things we did not see in here. |
| 02:22.43 | mogorman | 40kbs |
| 02:22.51 | mogorman | no you dont have enough for 4 calls |
| 02:22.55 | Qwell | [TK]D-Fender: sec |
| 02:23.01 | mogorman | maybe 3 with iax2 trunking but i dont know |
| 02:23.05 | mogorman | there is a bandwith calculator |
| 02:23.14 | Gabriel25 | for this I want that g729 |
| 02:23.28 | mogorman | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 02:23.33 | Gabriel25 | hoe much I need for 4 extension ? |
| 02:23.34 | mogorman | there you go |
| 02:23.48 | gandhijee | well these dudes at this company called into that make this BS secureICP software for IXP say the support class 5 telephony features |
| 02:23.49 | hacked`` | guys, you know voip, lets say i wanted 4 voip lines, and wanted 1 phone numbers customers can call to, and it would prompt them like "press 1 for customer service", and then it would transfer them to a specific ip phone.. like what do i need in terms of hardware and service providers? |
| 02:24.12 | [TK]D-Fender | Qwell : Jesus friggen Christ, you're starting to act like this is your personal crusade and is making you look borderline obsessive you know.... |
| 02:24.12 | gandhijee | so i am trying to figure out exactly what class 5 telephony features are |
| 02:24.17 | Qwell | :P |
| 02:24.23 | mogorman | 4 calls trunked is 46 k per second with g729 |
| 02:24.23 | Qwell | [TK]D-Fender: We ALL know I'm obsessive |
| 02:24.27 | [TK]D-Fender | Qwell : Go get some drugs... |
| 02:24.40 | Qwell | I *have* drugs. |
| 02:24.42 | gandhijee | i mean intoto |
| 02:24.45 | Qwell | Thank you very much |
| 02:24.51 | Qwell | http://pastebin.ca/58944 :p |
| 02:24.57 | gandhijee | anyone in MA wanna spare me some drugs? |
| 02:25.14 | mogorman | you have drugs for your kiddies Qwell ? |
| 02:25.24 | Qwell | mogorman: I don't know what you're talking about |
| 02:25.37 | mogorman | i thought you ahda kid |
| 02:25.42 | Qwell | I do |
| 02:25.50 | [TK]D-Fender | mogorman : Can't you change the frame size in IAX2? |
| 02:25.53 | mogorman | so why the drugs? |
| 02:26.00 | mogorman | you can move to 40ms frames |
| 02:26.03 | Gabriel25 | mogorman here is what I want to do |
| 02:26.04 | mogorman | and that will cut it down a bit |
| 02:26.07 | Qwell | mogorman: humor :p |
| 02:26.18 | mogorman | yeah i guess the drugs do make you funny |
| 02:26.21 | [TK]D-Fender | mogorman : then wouldn't that + G.729 trunked fit in about 40kbps? |
| 02:26.22 | Qwell | heh |
| 02:26.45 | [TK]D-Fender | mogorman : drops the RTP/UDP overhead in half.. I think it might fit... |
| 02:26.47 | Qwell | That's if he gets a constant 40k/s |
| 02:26.57 | mogorman | there is no rtp in iax2 |
| 02:27.01 | [TK]D-Fender | Qwell : Hey, dream a little ;) |
| 02:27.04 | Qwell | heh |
| 02:27.06 | mogorman | and iax2 trunking gets rid of the majority of that |
| 02:27.16 | mogorman | having trunked frames makes it a bit pointless |
| 02:27.16 | [TK]D-Fender | mogorman : Sorry, yeah, poor tem... jsut UDP overhead then :) |
| 02:27.39 | [TK]D-Fender | mogorman : And changing the frame size reduces the header:data ratio. |
| 02:28.04 | mogorman | but your right moving to 40ms frames would drive it down a bit |
| 02:28.08 | mogorman | but not much |
| 02:28.13 | mogorman | maybe 2 or 3 kbps |
| 02:28.28 | mogorman | wait no not even that |
| 02:28.33 | Qwell | 40kbit just isn't a voip able line |
| 02:28.46 | [TK]D-Fender | mogorman : IIRC its about 15kbps of overhead for RTP, how much to IAX2? |
| 02:28.49 | mogorman | nah Qwell |
| 02:28.53 | mogorman | you can do it |
| 02:29.00 | Qwell | I mean, sure you CAN do it, but.. |
| 02:29.04 | mogorman | well an iax2 trunk frame is 20kbps for all of the calls |
| 02:29.07 | *** join/#asterisk cyn0n (n=cyn0n@12-217-37-35.client.mchsi.com) |
| 02:29.10 | chino | if i call my freind he hears me if he calls me i hear him but the caller can never hear the other guy |
| 02:29.11 | [TK]D-Fender | Qwell : Voipable sure, just not too many calls.... |
| 02:29.24 | Qwell | I just wouldn't trust it |
| 02:29.26 | mogorman | 3 calls trunked |
| 02:29.29 | mogorman | g729.... |
| 02:29.53 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:29.58 | [TK]D-Fender | Yeah, 40kbps is WAY to slim for comfort... |
| 02:30.01 | Sponge_bob | do you need a license for g729? |
| 02:30.01 | mogorman | go get a real connection |
| 02:30.03 | mogorman | shebus |
| 02:30.05 | cyn0n | anyone know how to use uplink w/asterisk && skype? I can forward to uplink but it just keeps rining and eventually disconnects.. I know I'm overlooking something really stupid |
| 02:30.06 | chino | it sounds like you hear them for a second and then it just cuts off |
| 02:30.08 | Qwell | Sponge_bob: yes |
| 02:30.08 | [TK]D-Fender | Sponge_bob : yes |
| 02:30.09 | mogorman | for encoding and decoding it |
| 02:30.29 | paolob | Hi guys! Isn't there a predefined gsm sound in asterisk saying: "please dial an extension" or something similar? |
| 02:30.37 | Sponge_bob | how does that work? flat fee or per call? |
| 02:31.06 | *** join/#asterisk exten123 (n=exten1@60.49.242.225) |
| 02:31.14 | mogorman | yes paolob |
| 02:31.18 | mogorman | look in asterisk sounds |
| 02:31.23 | Sponge_bob | paolob: look in the sounds directory. it think it starts with enter-ent |
| 02:31.23 | mogorman | as well as the sounds included with asterisk |
| 02:31.26 | mogorman | of which there are several |
| 02:31.34 | mogorman | hundred audio files |
| 02:31.36 | mogorman | you get for free |
| 02:31.43 | mogorman | thanks to digium and other friendly people |
| 02:31.59 | mogorman | so compiling asterisk on my mips box sucks |
| 02:32.03 | Qwell | Thanks Uncle Digium! |
| 02:32.04 | mogorman | takes forever |
| 02:32.10 | paolob | mogorman, I can't see it (I have debian) |
| 02:32.13 | Qwell | (and auntie Allison) |
| 02:32.21 | mogorman | allison is amazing |
| 02:32.22 | paolob | Sponge_bob, I can't see it... |
| 02:33.06 | Qwell | What was I just about to do? |
| 02:33.16 | mogorman | commit skinny |
| 02:33.19 | exten123 | does externip in sip configuration will solve NAT problem? |
| 02:33.19 | mogorman | zing |
| 02:33.22 | Qwell | did last night |
| 02:33.25 | mogorman | to trunk |
| 02:33.28 | mogorman | i saw the branch |
| 02:33.30 | Qwell | it's kinda trunk |
| 02:33.31 | Qwell | :P |
| 02:33.41 | Qwell | not like it'll make it any worse, heh |
| 02:34.01 | Qwell | see how I snuck my name into that file? :D |
| 02:34.29 | Sponge_bob | anyone here run a cisco router as their voice gw? |
| 02:34.43 | mogorman | heh |
| 02:35.16 | [TK]D-Fender | exten123 : Its required, as is "localnet" |
| 02:35.36 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:37.13 | exten123 | [TK]D-Fender , sorry what mean "localnet" is config? |
| 02:37.34 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:38.49 | [TK]D-Fender | exten123 : * needs to know WHEN to use the EXTERNIP value in the SIP header. when the destination of your call is not inside the range defined by "localnet" then it changes it to the "externip" value |
| 02:38.58 | exten123 | [TK]D-Fender, okie found it is another property to send |
| 02:39.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:40.09 | exten123 | [TK]D-Fender, do u heard about ranchNetworks devices? will it actually help in solving NAT problem? or actually will not necessary need this kind of devices? |
| 02:40.37 | chino | yo |
| 02:40.44 | [TK]D-Fender | exten123 : I've never seen any need for anything but the usual 3 settings in sip.conf and to forward a few ports. |
| 02:41.39 | *** part/#asterisk RF_MIA (n=markmia@24-55-227-232.miamfl.adelphia.net) |
| 02:42.14 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:42.35 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:42.59 | exten123 | I dun know why RanchNetwork is promote it self that it support MIDCOM and eventually will solve all NAT problem |
| 02:44.20 | *** part/#asterisk Kluster_v2 (n=Kluster@74.56.127.12) |
| 02:44.35 | [TK]D-Fender | exten123 : I dunno... NAT never posed any problems to me.... |
| 02:44.36 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:44.57 | exten123 | [TK]D-Fender, okie |
| 02:45.38 | exten123 | [TK]D-Fender, where ar you come from? |
| 02:45.51 | blitzrage | damnit -- can't figure out how to return multiple values from a pl/pgsql function |
| 02:45.57 | blitzrage | in 7.4 |
| 02:46.05 | [TK]D-Fender | exten123 : Sol 3 :) |
| 02:46.09 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:49.30 | chino | http://pastebin.com/738570 |
| 02:49.37 | chino | there is my sip.conf i remoeved all the default chargen |
| 02:50.56 | chino | sorry thats my extensions.conf |
| 02:51.53 | [TK]D-Fender | chino : Ok, and now your sip.conf? |
| 02:52.07 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:52.11 | paolob | Guys, is there something to exclude a context from another context (the reverse of include)? |
| 02:52.16 | paolob | in extensions.confg |
| 02:52.18 | paolob | in extensions.conf |
| 02:54.00 | chino | do i even have ot use extensions cant i just use username's in the sip.conf ? |
| 02:54.58 | chino | are hte config files read in a acl manner ? |
| 02:55.27 | chino | meaning if i have 'canreinvite=yes' after 'nat=yes' the cascade causes canreinvite to be set to yes |
| 02:55.33 | [TK]D-Fender | chino : You can do "exten => mark,1,Dial(SIP/mark,20)" if you want, buy you'll only be able to dial it with a softphone chances are. |
| 02:55.34 | chino | or does nat=yes turn it off no matter what ? |
| 02:56.00 | [TK]D-Fender | chino : those optiosn can co-exist seperately. |
| 02:57.27 | chino | http://pastebin.com/738576 |
| 03:00.08 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:01.24 | *** join/#asterisk servant74 (n=jack@72.173.1.77) |
| 03:01.38 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:04.10 | [TK]D-Fender | chino : So your firends are all behind NAT as well? |
| 03:07.23 | chino | i would think so |
| 03:07.42 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:09.35 | [TK]D-Fender | chino : For those people set "qualify=yes" for all of them. |
| 03:09.52 | chino | i did that |
| 03:10.09 | [TK]D-Fender | chino : And you need to add nat, externip, and localnet statements to your [general] sectin. |
| 03:10.10 | chino | i sent you my sip.conf |
| 03:10.22 | [TK]D-Fender | chino : Sorry I missed that somehow |
| 03:10.55 | chino | tell me what im missing |
| 03:11.02 | [TK]D-Fender | chino : And remove the "canreinvite=yes" from all of your entries and put "canreinvite=no" into [general] |
| 03:12.04 | chino | anythign else ? |
| 03:12.46 | [TK]D-Fender | chino : everything we discussed earlier |
| 03:12.51 | chino | cant i just put everything in gerneal ? |
| 03:12.53 | chino | they are all the same |
| 03:13.29 | [TK]D-Fender | chino : like I told you to. |
| 03:13.43 | chino | what |
| 03:14.44 | chino | is this ok ? |
| 03:14.45 | chino | http://pastebin.com/738588 |
| 03:16.28 | [TK]D-Fender | chino : No |
| 03:16.43 | chino | hey worth a shot lol |
| 03:18.17 | [TK]D-Fender | chino : Fill in the blanks on this http://pastebin.com/738594 |
| 03:18.24 | [TK]D-Fender | chino : and fix the callerid |
| 03:19.30 | chino | any name ? |
| 03:19.37 | chino | then they're extension |
| 03:19.45 | chino | does this mean that i dont have to put it in the extensions.conf ? |
| 03:21.35 | chino | what format does localnet follow ? |
| 03:21.37 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 03:22.07 | chino | 192.168.0.1/24 |
| 03:22.07 | chino | ? |
| 03:22.07 | [TK]D-Fender | chino : Might be nice if the callerid matched the person you are applying it to. |
| 03:22.23 | [TK]D-Fender | chino : this does not change anything in extensions.conf |
| 03:22.43 | [TK]D-Fender | chino : not /24 - /255.255.255.0 |
| 03:22.57 | chino | man this is my server in denmark |
| 03:22.58 | [TK]D-Fender | chino : I don't believe you can use CIDR notation in there |
| 03:23.10 | chino | the network is filled iwth other servers |
| 03:23.18 | chino | its not really my localnet |
| 03:23.23 | chino | cant i just make that localhost or something ? |
| 03:24.50 | chino | i think my eth0 ip is the public ip |
| 03:25.01 | *** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net) |
| 03:25.28 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 03:25.29 | chino | it is |
| 03:25.43 | chino | ah so my server isn't behind a hardware firewall then |
| 03:26.17 | techman97_andy | hey all - I feel silly. I just setup a 2nd * server to dink around with other features (not on my prod box!) and I can't remember how to set Asterisk to load on system start....any hints? |
| 03:26.34 | chino | can i ommit the localnet? |
| 03:27.18 | [TK]D-Fender | chino : If the IP is publin, then you don't need nat=yes, extenip, or localnet in [general] |
| 03:27.22 | [TK]D-Fender | public |
| 03:27.42 | *** join/#asterisk Jameno123 (n=james@63.210.246.146) |
| 03:27.45 | chino | what about my clients though ? |
| 03:28.03 | chino | i thought general was for setting their presets |
| 03:28.05 | Jameno123 | hrm... could someone help me identify a packet that is being sent by asterisk |
| 03:28.11 | Jameno123 | that is casuing all of my cisco ip phones to reboot |
| 03:28.16 | Jameno123 | simotaniously |
| 03:28.22 | Jameno123 | (SIP) |
| 03:28.50 | *** part/#asterisk servant74 (n=jack@72.173.1.77) |
| 03:28.58 | Jameno123 | http://pastebin.com/738607 |
| 03:29.02 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 03:29.16 | Jameno123 | (thats the packet, it was all on one line, from TCPdump) |
| 03:29.27 | Jameno123 | its sent about once every 3-4 hours |
| 03:30.33 | techman97_andy | n/m everyone - I found it |
| 03:30.34 | techman97_andy | rofl |
| 03:30.56 | [TK]D-Fender | chino : if it doesn't belong in [general] and it belongs in the client, then put it back. |
| 03:31.27 | litage | most of my asterisk servers have a single P4 in them, and 2 have dual Xeons. all boxes have hyperthreading enabled. is this wise? how does enabling hyperthreading affect asterisk? |
| 03:33.48 | chino | now its not ringing at all and console keeps saying the same stuff about users being busy |
| 03:34.26 | chino | <PROTECTED> |
| 03:35.07 | [TK]D-Fender | chino : pastebin your config again |
| 03:35.23 | }btorch{ | yeah i get the same busy congestion stuff once in a while during the day but i'm using * for PRI to PRI |
| 03:35.23 | techman97_andy | hey all, what is the directory that holds the Asterisk modules? |
| 03:35.59 | chino | http://pastebin.com/738616 |
| 03:36.29 | chino | http://pastebin.com/738618 |
| 03:37.34 | techman97_andy | n/m everyone - I found it |
| 03:37.35 | techman97_andy | rofl |
| 03:37.36 | *** join/#asterisk Meaw (n=dino@83.244.94.239) |
| 03:38.08 | [TK]D-Fender | chino : Just to be REALLY sure. Your * has a PUBLIC IP, and your clients are all behind NAT? |
| 03:38.23 | chino | yes |
| 03:38.31 | chino | with ports forwarded on clients |
| 03:38.54 | }btorch{ | IAX2 is a lot better when behind a nat |
| 03:40.07 | [TK]D-Fender | chino : Clients don't need ports forwarded. |
| 03:40.26 | chino | really ? |
| 03:41.20 | [TK]D-Fender | chino : http://pastebin.com/738626 |
| 03:41.30 | *** join/#asterisk copantl (n=galellop@190.4.22.82) |
| 03:43.14 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 03:49.31 | *** join/#asterisk trixter_ (n=trixter@65-165-167-217.du.volcano.net) |
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| 03:51.59 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:52.58 | supjigatr | Hello. |
| 04:02.04 | *** part/#asterisk copland (n=stonecol@209.216.65.10) |
| 04:06.00 | *** part/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 04:06.20 | *** join/#asterisk pdugas (n=pdugas@h15.86.40.69.ip.alltel.net) |
| 04:06.52 | pdugas | any way to have the IAXY produce distinctive rings? |
| 04:11.38 | *** join/#asterisk _agt (n=hyperion@cpe-72-226-231-51.rochester.res.rr.com) |
| 04:12.39 | *** join/#asterisk pdugas (n=pdugas@h15.86.40.69.ip.alltel.net) |
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| 04:30.51 | *** join/#asterisk andymul (n=andymul@cpe-69-203-74-36.nyc.res.rr.com) |
| 04:31.06 | andymul | Anyone have any idea why "ANSWEREDTIME" would be blank? |
| 04:35.09 | *** join/#asterisk espn (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:37.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:47.11 | trixter | because the cdr stuff is weak? |
| 04:53.21 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 04:59.19 | asterboy | anyone using the sound card for paging? |
| 05:00.16 | asterboy | It burns when I pee... |
| 05:00.22 | asterboy | ...anyone have advice? |
| 05:02.59 | techman97_andy | it burns when you pee? rofl |
| 05:03.11 | techman97_andy | I haven't used the sound card yet, but it's on my list to try! |
| 05:03.39 | techman97_andy | from what I do know, you need a sound card in the * server, and you jerry rig up a connector to either a normal pair of speakers or an overhead speaker |
| 05:03.46 | techman97_andy | and just use the paging application I think |
| 05:03.47 | asterboy | ya, I was just wondering if anyone has tried it. |
| 05:03.56 | techman97_andy | give me a week and I'll have tried it |
| 05:03.57 | techman97_andy | =) |
| 05:04.09 | techman97_andy | have you played with MySQL realtime config? |
| 05:04.22 | asterboy | The better way to page is using and fxo device to an amp and horn |
| 05:04.39 | techman97_andy | I have extensions and SIP peers/using working, but I can't get the voicemail to work right |
| 05:04.40 | techman97_andy | =( |
| 05:04.57 | techman97_andy | yeah, the sound card seems kinda cloogy, but hey. |
| 05:10.30 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:11.28 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
| 05:13.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:13.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:26.45 | *** part/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:33.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:39.18 | dlynes_home | asterboy: how is using an fxo device to amp and horn going to do anything? |
| 05:39.27 | dlynes_home | asterboy: an fxo device is an analog phone line |
| 05:40.35 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:44.06 | *** join/#asterisk crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net) |
| 05:44.41 | crshman | what is a great softphone that support multiple lines? are there any with VM notification? |
| 05:48.22 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
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| 05:56.27 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 05:56.36 | Dico_ | hello |
| 05:57.31 | Dico_ | ok, from time to time i know the question pop up in the mail list, but I would like to know how is the state of asterisk clustering project ? |
| 05:57.47 | *** join/#asterisk X-Rob (n=rob@dyn-ip-237.matilda.net.au) |
| 06:01.38 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
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| 06:05.06 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 06:05.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:13.56 | *** join/#asterisk lorinc (n=ang@caracas-0150.adsl.interware.hu) |
| 06:14.20 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 06:15.27 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 06:16.47 | *** part/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
| 06:19.19 | *** part/#asterisk Meaw (n=dino@83.244.94.239) |
| 06:21.16 | crshman | what is a great softphone that support multiple lines? are there any with VM notification? |
| 06:21.34 | X-Rob | Snom 360 softphone |
| 06:22.16 | stephane_ | jour |
| 06:23.51 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 06:30.05 | dlynes_home | bon matin |
| 06:30.33 | Dico_ | hehe, bonne matinee |
| 06:34.05 | dlynes_home | what language is bonne matinee? |
| 06:34.10 | dlynes_home | espanol? |
| 06:34.15 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) |
| 06:38.37 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 06:41.44 | crshman | snom is ugly, any other recommendations that support VM notifications? |
| 06:41.58 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 06:42.15 | Assid | polycom |
| 06:42.39 | crshman | is that a softphone? |
| 06:43.30 | dlynes_home | crshman: no...it's a hardphone |
| 06:43.43 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-80-96.dsl.irvnca.pacbell.net) |
| 06:43.58 | crshman | i liked x-lite but i couldn't get the VM notifications to work |
| 06:44.00 | dlynes_home | crshman: have you tried ekiga? |
| 06:44.13 | dlynes_home | crshman: oh...nvm...guess you're looking for a windows solution |
| 06:47.55 | crshman | dlynes_home: ya i'm looking for a windows solution |
| 06:48.26 | *** join/#asterisk jart (n=jart@ool-4356f82f.dyn.optonline.net) |
| 06:49.26 | jart | Asterisk 1.2 now 'Kooler than Jesus' alleges Mark Spencer |
| 06:49.53 | jart | anyone awake? |
| 06:56.28 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 07:04.39 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 07:06.55 | *** join/#asterisk nfinetin (n=nfi@LSt-Amand-152-32-9-55.w82-127.abo.wanadoo.fr) |
| 07:07.04 | nfinetin | hey evryone |
| 07:07.50 | *** join/#asterisk Faithful (n=Faithful@mail.familyfirst.org.au) |
| 07:09.12 | nfinetin | i m using ATA to connect asterisk evrywork nice about voice, on the asterisk i configured a peer for outbound calls that peer is a cisco AS connected to PSTN and when i'm dialling outside i'm only getting oneway voice |
| 07:10.18 | nfinetin | from my ATA directly connected to the cisco i've no problem, from my ATA playing with the echo test from the asterisk no problem |
| 07:10.26 | *** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com) |
| 07:10.28 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:12.21 | Tili | hi |
| 07:12.40 | Tili | is it possible to play a music from asterisk instead of ring |
| 07:12.53 | *** join/#asterisk NoNeo (n=Artur@193.24.24.10) |
| 07:13.24 | blitzrage | Tili: show application Dial() |
| 07:13.32 | blitzrage | errr.. show application dial |
| 07:15.56 | Tili | blitzrage: thanks |
| 07:16.11 | Assid | dial.. diall.. diallllll DIALLL!!! |
| 07:16.19 | Assid | vroooooooom vrooooooom.. |
| 07:16.22 | Tili | and what about on an nbound call which is not bridged to another |
| 07:16.54 | Tili | or in case where the B-Leg was found busy and we need to tell this to user as a voice prompt |
| 07:17.45 | trixter | so is anyone going to www.cluecon.com ? one of the authors of asterisk: the future of telephony is gonna be there and someone is gonna do a presentation on asterisk module development |
| 07:17.51 | trixter | sounds interesting at least |
| 07:19.17 | MrChimpy | some bastard has nicked my loopback cable |
| 07:19.47 | trixter | sorry bout that |
| 07:19.53 | trixter | I needed something to make a leash for my cat |
| 07:20.00 | Tili | how can I tell users if they have dialed wrong number without actually picking up the call on asterisk |
| 07:20.02 | MrChimpy | :) |
| 07:20.13 | trixter | tili: return a 404? |
| 07:20.25 | MrChimpy | at this company we've actually used cat5 to suspend a piniata :) |
| 07:20.28 | trixter | depends on what transport protocol (channel) and depends on what their equipment is |
| 07:20.53 | Tili | trixter: on zap channels also |
| 07:21.01 | trixter | zap pri or zap analog? |
| 07:21.10 | trixter | analog you really have to answer the channel to pass anything |
| 07:21.25 | Tili | trixter: 404 is SIP header. I want the voice prompt to say you dialed wrong number and still i should not pickup phone so that it doesnt cost to user |
| 07:21.37 | Tili | trixter: agreed on analog. I want PRI |
| 07:21.45 | trixter | pri you can send error codes back but those eventually get interpreted by the device they are connected to (the telco switch for example) and turned into audible messages |
| 07:21.54 | *** join/#asterisk clive- (n=pirch@dsl-146-123-108.telkomadsl.co.za) |
| 07:22.16 | trixter | ok, if you have a pri *AND* the telco lets you pass audio it is possible to pass audio without sending answering supervision |
| 07:22.20 | Tili | trixter: yeah but I have seen some cell companies that they play music instead of normal ringtone. |
| 07:22.35 | Tili | trixter: yeah. how do i do something like that in Asterisk. |
| 07:22.46 | trixter | that isnt something you can just do, the telco that feeds your pri has to allow it |
| 07:22.57 | Tili | Instead of giving congestion tone, I tell user in plain english that number is wrong or network is busy |
| 07:22.59 | trixter | and in some cases the telco of the calling party has to allow it |
| 07:23.25 | Tili | well we have our own switch that connects to asterisk |
| 07:23.33 | trixter | do you have a pri or is it an analog circuit that is connected to asterisk? |
| 07:23.39 | Tili | PRI |
| 07:24.11 | trixter | ok, then perhaps searching on terms like passing audio without sending answering supervision would be helpful in google |
| 07:24.11 | Tili | We have our switch that is connected to PTT on leased/licensed lines and switch terminates over RJ-45 to Digium/Sangoma |
| 07:24.14 | trixter | becauyse that is what you want to do |
| 07:24.16 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 07:24.28 | Tili | trixter: yeah and talking to switch guys would be helpful too |
| 07:24.38 | Tili | thanks |
| 07:24.42 | trixter | is PTT your local phone company? |
| 07:24.58 | trixter | if so I am unsure if they allow this or not |
| 07:25.16 | trixter | I do know that more carriers are starting to block it |
| 07:25.31 | trixter | some only block one direction some block both.. by blocking one direction it may still work if they let you send |
| 07:26.24 | trixter | offhand I dont know how to tell asterisk to open audio paths but not send answering supervision (the signal that indicates you answered the line).. I am also unsure if it requires a patch or not, I do know it can do it though |
| 07:26.39 | nfinetin | hey |
| 07:27.03 | nfinetin | can i ask a question about a NAT issue that i m getting |
| 07:27.15 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:27.39 | qdk | nfinetin: yes, if you stop asking to ask. |
| 07:31.26 | Assid | im gona try and get more people to move over to openoffice |
| 07:32.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:33.49 | clive- | has anyone had bad audio quality caused by hyperthreading ? |
| 07:34.22 | dlynes_home | clive-: i think i remember reading somewhere that if you have hyperthreading on your processor, it should be disabled before using asterisk |
| 07:35.25 | clive- | dlynes, ok, so its off to the server oom to reboot into non-hyperthreding.... |
| 07:35.56 | clive- | Does this mean that asterisk cant work well on a multi processor box ? |
| 07:35.58 | trixter | that is why I have a dual core rather than a HT system :D |
| 07:36.04 | trixter | HT is just a pretend cpu |
| 07:36.31 | dlynes_home | clive-: no...just hyperthreading |
| 07:36.36 | trixter | asterisk works ok (as good as asterisk works anyway) on a SMP box |
| 07:37.13 | nfinetin | qdk: thx, here is my issue |
| 07:37.18 | trixter | although I wanted srtp (encrypted rtp) so I cant use it for sip. I also wanted google talk and a few other goodies that afaik arent present |
| 07:37.19 | clive- | trixter..lol.....ok, so hyperthreading is not 2 cpu's, ...so a dual core is a better option |
| 07:37.34 | nfinetin | <PROTECTED> |
| 07:37.39 | trixter | dual core is 2 real cpus, HT is a pretend cpu that gives you a 20-30% speed boost |
| 07:37.46 | dlynes_home | trixter: google talk is working in freeswitch already |
| 07:37.54 | trixter | yes I know |
| 07:37.54 | dlynes_home | trixter: well...sort of... |
| 07:37.59 | trixter | so is srtp |
| 07:38.05 | trixter | cause crypto is important to some people |
| 07:38.09 | clive- | trixter, well 20% boost is going to get switched off pretty son...: |
| 07:38.17 | clive- | pretty soon :( |
| 07:38.54 | trixter | dlynes_home: are you going to cluecon.com ? |
| 07:38.57 | clive- | dlynes, what is free switch ? |
| 07:39.09 | trixter | there will be discussions about jingle, asterisk, freeswitch, bayonne, and other goodies there |
| 07:39.16 | trixter | clive-: www.freeswitch.org |
| 07:39.24 | dlynes_home | clive-: i wouldn't be surprised if their 'hyperthreading' means they only turn the cpu off 4 out of every 10 cycles, instead of 5 out of every 10 |
| 07:39.28 | trixter | its an open source telephony platform focused more on being a switch than answering machine |
| 07:39.32 | dlynes_home | clive-: also #freeswitch |
| 07:39.33 | trixter | although it can act like an answering machine |
| 07:39.45 | dlynes_home | trixter: i don't even have enough time to take a dump, never mind go to cluecon |
| 07:40.11 | dlynes_home | trixter: thought they didn't have a pbx module yet? |
| 07:40.19 | trixter | it can do some stuff |
| 07:40.38 | trixter | I wrote the weather app as a demo and there is enough functionality there to make it work like an answering machine and then some |
| 07:40.41 | clive- | just looked...so what is the advantage/disadvatage of freeswitch as opposed to asterisk ? |
| 07:40.42 | trixter | and its a *simple* example app |
| 07:40.58 | dlynes_home | clive-: it's a complete rewrite using the apache portable runtime |
| 07:41.00 | trixter | clive-: plan on doing clustering? processing milions of minutes a month? |
| 07:41.07 | dlynes_home | clive-: so it's built upon a very stable base |
| 07:41.11 | trixter | smaller memory footprint? |
| 07:41.18 | trixter | want a box to do 1000 concurrent calls? |
| 07:41.23 | trixter | so far that is what it does |
| 07:41.31 | trixter | but they arent done yet, it only started a few months ago |
| 07:41.50 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:41.50 | dlynes_home | regardless of long ago it started |
| 07:42.04 | dlynes_home | bkw and anthm have still brought it a good ways so far |
| 07:42.23 | pjchilds | plus if you look at the asterisk codebase you will understand what freeswitch is about.... |
| 07:42.41 | dlynes_home | I'm still working with asterisk, but i am keeping an eye on freeswitch for the future |
| 07:42.43 | pjchilds | its impossible to determine correctness of asterisk, it wasn't engineered that way... |
| 07:42.50 | dlynes_home | I'm even rewriting the app_voicemail module |
| 07:43.58 | pjchilds | how do we manage complexity... |
| 07:44.03 | clive- | trixter sounds good...my minutes are not up to a million yet..:), lol....but hopefully next year will be there |
| 07:44.03 | trixter | well there is a difference, asterisk started in 1999 and according to printed literature from digium itself it can handle only 120 concurrent calls (ABE anyway) or 240 if you pay more money.. freeswitch already can do 1000 concurrent |
| 07:44.03 | trixter | given that its a few months vs several years in development that is a pretty good deal |
| 07:44.04 | clive- | trixter who is behind freeswitch ? |
| 07:44.04 | trixter | freeswitch is MPL (mozilla license) so there is a slight difference that way too |
| 07:44.04 | trixter | join #freeswitch or goto www.freeswitch.org |
| 07:44.04 | trixter | basically asterisk developers are behind it |
| 07:44.04 | pjchilds | clear functional bounderies.... |
| 07:44.49 | trixter | freeswitch is designed with a clearly defined core, not what becomes a monolithic program that has so many cross dependancies that you cant really remove modules and still have it do anything |
| 07:45.24 | trixter | there is a module system but to use the example just given look at app_voicemail, it has a ton of dependancies, some of which should really be optional like adsi |
| 07:45.27 | dlynes_home | trixter: but dood...that's the way the world is designed! |
| 07:45.29 | trixter | but they arent |
| 07:45.32 | clive- | trixter can freswitch run agi scripts ? |
| 07:45.42 | trixter | no but you can run perl or javascript apps |
| 07:45.54 | trixter | the way that freeswitch executes your dialplan is totally different |
| 07:46.11 | trixter | and from what I know about both, faster and more optimized than asterisk |
| 07:46.17 | pjchilds | read http://www.sofaswitch.org/docs/asterisk_sd.pdf ... there is some other interesting ideas on sofaswitch.org |
| 07:46.21 | trixter | becuase it doesnt do a search for the next thing to do each and every time it does something |
| 07:46.27 | dlynes_home | trixter: you're not being all that fair to asterisk, either |
| 07:46.37 | clive- | ok, like I use perl to do agi /IVR stuff will that work? |
| 07:46.38 | trixter | ahh yes sofaswitch will be something else talked about at cluecon.com |
| 07:46.40 | dlynes_home | trixter: freeswitch has a few dependencies, too |
| 07:47.50 | trixter | its basically a con for telephony stuff with emphasis on virtually everything (the only thing I didnt see was yate) |
| 07:47.50 | trixter | freeswitch has some dependancies but not like asterisk with its modules that require every other module to be loaded |
| 07:47.50 | trixter | there is a clearer distinction between the core and the modules in freeswitch |
| 07:48.12 | trixter | and if citing specific real things about asterisk isnt being fair to asterisk I would love to know what your definition of fair is |
| 07:48.20 | *** join/#asterisk dyerf (i=dyer@cube.sut.ru) |
| 07:48.28 | trixter | I dont believe I said anything untrue |
| 07:48.46 | pjchilds | well asterisk is amazing for what it does... it shows the potential, and it works (amazingly...) |
| 07:48.55 | trixter | yes amazingly it works :P |
| 07:49.04 | dlynes_home | trixter: didn't say you did...was just saying you were mentioning all the negatives about asterisk and all the positives about freeswitch |
| 07:49.09 | trixter | but there is a difference between a pbx/answering machine and a switch |
| 07:49.19 | dlynes_home | trixter: but in hyping up freeswitch, you forgot to mention all of its dependencies |
| 07:49.28 | trixter | what dependancies specifically? |
| 07:49.32 | *** join/#asterisk AuPix (n=AuPix@mail.aupix.com) |
| 07:49.38 | clive- | well can freeswitch use a PRI card ? |
| 07:49.41 | pjchilds | not being able to reuse properly engineered known working code just because of licensing is crap... |
| 07:49.43 | trixter | because I think you are changing the meaning of the word from how I was using it |
| 07:49.44 | dlynes_home | trixter: the three or four libraries it needs to be able to compile |
| 07:49.57 | trixter | freeswitch can infact use a pri card and get better performance than the zapata interface |
| 07:50.02 | trixter | with sangoma cards anway |
| 07:50.04 | trixter | anyway |
| 07:50.07 | dlynes_home | trixter: the apr, and some other libraries...can't remember what they are offhand |
| 07:50.16 | trixter | ahh you did change the meaning thought so |
| 07:50.28 | trixter | ok so asterisk has dependancies in the way you say it that make it gpl incompatible |
| 07:50.31 | dlynes_home | trixter: i didn't change the meaning of anything |
| 07:50.34 | trixter | I can play this game where the meaning gets changed |
| 07:50.38 | trixter | yes you did |
| 07:50.48 | dlynes_home | trixter: you were talking about monolithic dependencies |
| 07:50.48 | trixter | I spoke of dependancies like app_voicemail and adsi specifically |
| 07:50.59 | dlynes_home | trixter: i wasn't referencing those dependencies |
| 07:51.06 | trixter | that is internal to asterisk not external like the inclusion of netbsd code which is gpl incompatible |
| 07:51.24 | trixter | you then changed what I meant by dependancies (modules) to libraries external to the codebase |
| 07:51.30 | dlynes_home | trixter: I was just mentioning a negative of freeswitch, totally unrelated to any negatives in asterisk |
| 07:51.44 | dlynes_home | trixter: i wasn't mentioning it as a dependency, but rather a negative |
| 07:51.45 | pjchilds | why is netbsd code gpl incompatible? |
| 07:51.57 | trixter | dlynes_home: but I was and you started by correcting me on the dependancies of freeswitch.. asterisk has the same ones but I wasnt going into that when I mentioned dependancies |
| 07:52.03 | dyerf | anybody succesfully registered on <account>@gmail.com with res_jabber ? |
| 07:52.03 | trixter | Pj_: becuase its 4 clause |
| 07:52.11 | trixter | they are the only ones that are still 4 clause for everything |
| 07:52.11 | dlynes_home | trixter: you're taking what i said out of context |
| 07:52.32 | trixter | the advertising clause in the BSD license puts an extra restriction on the gpl software that is prohibited by the gpl |
| 07:52.55 | trixter | as such no one outside of digium has the right to distribute or use asterisk unless its business edition (becuase without a valid license you have no right to use) |
| 07:52.57 | dlynes_home | trixter: I said you were stating all the negatives in asterisk and all the positives in freeswitch, without mentioning any negatives of freeswitch |
| 07:53.25 | dlynes_home | trixter: i gave the library dependencies in freeswitch as an example of a negative |
| 07:53.29 | trixter | dlynes_home: your use of the word 'too' when refering to freeswitch dependancies is misleading then |
| 07:53.50 | trixter | aparently you meant to say 'well freeswitch has dependancies too just like asterisk that have nothing to do with the dependancies that you werej ust talking about' |
| 07:54.04 | trixter | you need to scroll back a few lines before what you think I am taking out of context to understand what I am talking about |
| 07:54.08 | trixter | but regardless |
| 07:54.25 | dlynes_home | dlynes_home trixter: freeswitch has a few dependencies, too |
| 07:54.39 | trixter | yes, 'too' now what was that in response to? |
| 07:54.39 | dlynes_home | nowhere in that sentence did i mention anything about monolithicity |
| 07:54.52 | trixter | me saying that asterisk has dependancies in its modules that make it hard to remove those modules |
| 07:55.00 | MrChimpy | easy now ladies |
| 07:55.07 | clive- | pjchilds I just read that article by anthm |
| 07:55.07 | trixter | specifically the example I gave was voicemail and adsi |
| 07:55.12 | MrChimpy | have a sit down and a nice cup of tea |
| 07:55.22 | trixter | where adsi should be totally optional and not mandatory if you want voicemail becuase its not needed in 100% on the installs |
| 07:55.27 | dlynes_home | trixter: anyways...drop it...this is a pointless argument |
| 07:55.44 | trixter | yes it is the proof came out and it should be dropped or it will look like I am right and we cant have that |
| 07:55.44 | clive- | well the good thing is that there are more options to linux telephony than I thought |
| 07:56.01 | dlynes_home | clive-: there's also Bayonne and Yate, too |
| 07:56.18 | trixter | clive: yate, bayonne (the official gnu telephony project), freeswitch, sofaswitch, asterisk |
| 07:56.23 | trixter | and a few isdn specific apps |
| 07:56.41 | dlynes_home | clive-: and OpenSER and SER, and a few others |
| 07:56.56 | trixter | well ser doesnt do media gateways afaik |
| 07:57.05 | trixter | its a sip proxy and that is it, unless I missed something |
| 07:57.10 | dlynes_home | no, but it's still a telephony project |
| 07:57.32 | trixter | its a telephony signalling project :P |
| 07:57.46 | pjchilds | trixter, are you saying that clause 4 of the BSD license makes it incompatible with the GPL ??? |
| 07:57.59 | dlynes_home | I get the feeling you like arguing just for the sake of arguing |
| 07:58.12 | trixter | if you want to go that route, sipx, netphone (from the early 90s) and a few others exist too |
| 07:58.29 | dlynes_home | exactly why i said there's a few others |
| 07:58.29 | pjchilds | clive-, well if you try to do some work on the asterisk codebase you'll understand what he's talking about... |
| 07:58.35 | trixter | pjchilds: no, clause 3 actually, but that only exists in the 4 clause version |
| 07:58.40 | trixter | the advertising clause |
| 07:58.40 | pjchilds | vocal ! :) |
| 07:58.47 | trixter | and I am not the only one that says it, the FSF says that too |
| 07:59.30 | MrChimpy | bearded geeks in sandals should leave this dull nonsense to lawyers |
| 07:59.51 | trixter | I dont wear sandals but I did goto law school, close enough? |
| 08:00.35 | pjchilds | trixter, so the NetBSD project should get sued as it includes GPL'd code in its OS ??? |
| 08:00.36 | MrChimpy | either way it's very very dull. GPL seems to exist so people can argue about it. |
| 08:00.46 | trixter | pjchilds: at fsf.org they maintain a list of basically open source licenses (some more open than others) and which are and are not compatible with the gpl and typically the reason for it, that would be a good place to read up on this |
| 08:01.41 | clive- | pjchilds I am a user, to be honest, I am not involved in coding...besides a few perl scripts. Last time I needed something fixed, kit was anthm who did it for me |
| 08:01.45 | trixter | pjchilds: it depends on how its included, however much of netbsd is copyright regents of california (who recinded the 3rd clause a while back) or netbsd copyright (4 clause BSD) |
| 08:02.04 | trixter | infact they goto lengths to not include gpl code for a reason, and that is why there is a lot of netbsd code in openbsd, freebsd, etc |
| 08:02.10 | pjchilds | MrChimpy, well... people think the GPL is about freedom... its not... its about 'conditional love'... here is something I give freely to you... and I protect your freedoms by telling you what you can and can't do with it... |
| 08:02.22 | trixter | take libedit/editline which has a compatibility mode with gnu readline, but is netbsd copyrighted |
| 08:02.28 | MrChimpy | frikkin hippies |
| 08:02.32 | MrChimpy | :) |
| 08:02.50 | pjchilds | MrChimpy, whereas the BSD license... is here... i give to you... and I expect nothing in return (except perhaps some credit... ) |
| 08:03.06 | trixter | at one time asteirks used gnu readline, that was replaced in favour of the gpl incompatible libedit so that asterisk business edition could be sold without including pesky gpl code which wasnt disclaimed |
| 08:03.40 | trixter | the mpl license is a lot like the 3 clause BSD lciense except that you have to kick down your modifications to the original authors (something even the gpl doesnt require) |
| 08:03.55 | MrChimpy | yes. i'm aware of all this having worked with GPL, NPL, BSD code in commercial environments. it's just a very dull topic. |
| 08:03.56 | trixter | which of course that clause makes it gpl incompatible |
| 08:04.25 | pjchilds | lets talk about the GPL v3 then :) |
| 08:04.56 | trixter | but the gpl really only covers distribution and use.. on the use end, you are free to use it for whatever, even digital toilet paper if you want, the distribution part is where most of the rules are placed |
| 08:05.14 | MrChimpy | mmmm. digital toilet paper. |
| 08:05.37 | *** join/#asterisk Stonekeeper (n=lea@194.82.229.100) |
| 08:06.47 | pjchilds | so... then... whats linus's take on this... can anyone tell me under what license the linux kernel is release :) |
| 08:07.17 | trixter | you can modify gpl code to make it totally incompatible with the gpl, include patented technologies, whatever you want so long as you dont distro it that way.. what you do with it in your home is your business.. now when you goto distribute you have to adhere to the legal status of 'derrivative works' (which the FSF misquotes but meh - there are 4 tests that get applied by the courts to see if it qualifies), so you can for example wri |
| 08:07.17 | trixter | te a commercial asterisk module and distro that so long as its not derived from asterisk (easy), not linked against asterisk (easy) and not distributed with asterisk (2nd CD set :) |
| 08:07.25 | trixter | that according to the FSF anyway but what do they know |
| 08:07.47 | trixter | pjchilds: LINUX IS V2 LOCKED |
| 08:07.54 | trixter | gah caps error |
| 08:08.24 | trixter | however he has made some exceptions for some stuff to be linked against the kernel for modules in binary gpl incompatible form |
| 08:08.36 | trixter | so it grants you more freedoms than stock gpl |
| 08:09.33 | MrChimpy | I used to release my software as beerware. If you like it buy me a pint. Otherwise do what you like with it, as long as I get credit. That worked for me - and got me drunk. |
| 08:10.36 | MrChimpy | if linus had done that he'd have liver failure by now |
| 08:11.13 | X-Gen | :P |
| 08:11.33 | MrChimpy | ta. i'm hungover so i can only cope with e-alcohol |
| 08:12.01 | pjchilds | so we all know the joke... why is american beer like making love in a canoe? |
| 08:12.15 | *** join/#asterisk darkskiez (n=darkskie@dsl-217-155-82-246.zen.co.uk) |
| 08:12.29 | MrChimpy | cos it's fuckin close to water. ba-boom. |
| 08:12.43 | pjchilds | ~spank pjchilds |
| 08:12.50 | jbot | ACTION bends pjchilds over his knee and tatoos 'ibot' on pjchilds's pasty white buttocks. |
| 08:13.36 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 08:13.37 | clive- | howzit x0gen |
| 08:13.48 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 08:14.21 | MrChimpy | I have 360 lines now. wooo. |
| 08:16.56 | MrChimpy | today I get to play AGI peering. |
| 08:17.23 | MrChimpy | AIX even |
| 08:18.47 | Tili | MrChimpy: you mean u have 360 IAX calls at one time and it works fine |
| 08:18.48 | trixter | I am sorrt |
| 08:19.09 | trixter | check out the guy in smit, if its the same from 10-15 years ago it will look like it has breasts while its running |
| 08:20.47 | MrChimpy | 240 lines per dual 3.6G Xeon |
| 08:21.07 | MrChimpy | We haven't load tested yet. that's what the 2nd box will do as well as doing our dial out |
| 08:21.27 | qdk | AIX as in IAX? |
| 08:21.40 | MrChimpy | yeah, sorry |
| 08:22.09 | MrChimpy | i'm suffering anacronym overload |
| 08:22.18 | trixter | ahh ok, nevermind smit then :P |
| 08:23.21 | qdk | MrChimpy: 360 concurrent calls og channels? any transcoding or echo cancellations? |
| 08:24.54 | trixter | freeswitch can do that without breaking a sweat :P |
| 08:25.02 | trixter | they already tested it to 1000 channels |
| 08:26.02 | MrChimpy | it'll be 240 concurrent calls per machine. echo cancellation in hardware (TE411Ps) |
| 08:26.19 | MrChimpy | Shouldn't need transcoding yet. |
| 08:26.48 | pjchilds | MrChimpy, thats 8 E1s ? ISDN to IAX ??? |
| 08:27.42 | MrChimpy | 8 E1s, yes. IAX to route outbound calls to a dedicated server with the outbound E1s. |
| 08:28.48 | pjchilds | each TE411P on its own PCI bus ? |
| 08:29.30 | trixter | one thing about the zap interface is that it cauess a constant flood of irqs.. 1000/sec iirc |
| 08:29.51 | trixter | with sangomas in their native mode (not zap emulation) with a simple ioctl() you can tell it to dump say 20ms frames |
| 08:29.56 | trixter | which results in 50 irqs/sec |
| 08:30.00 | trixter | much nicer |
| 08:30.21 | pjchilds | hmm... yeah better off with a AS5800 dropping a STM-1 (1960 DS0 channels) out to SIP *grin* |
| 08:30.34 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 08:30.36 | trixter | that along with silence supression (VAD) would be a good thing |
| 08:30.54 | cjk | hi, is anyone here using speex. any suggestioun on the different parameters? |
| 08:30.54 | trixter | but alas asterisk doesnt do that yet |
| 08:31.05 | pjchilds | and we know that freeswitch can do VAD *grin* |
| 08:31.10 | trixter | yes, yes it can |
| 08:31.23 | trixter | and do sangoma cards natively so there are fewer interrupts |
| 08:31.24 | trixter | :D |
| 08:32.07 | pjchilds | wonder how they stack up against Divia E1 cards... cappice said the Divias have some pretty slow DSPs... |
| 08:33.33 | RoyK | Goðan daginn |
| 08:33.45 | trixter | rumor has it that sangoma is coming out with a card soon that has g729 on the card, so you tell the card to give you g729 frames and away it goes, no need to transcode via software, or even to have additional licenses if you are just pushing bits |
| 08:34.00 | trixter | pjchilds: if its using the zap interface they all perform the same |
| 08:34.31 | trixter | the design of that interface is weak and everyone basically has to do that or write their own driver (which will never make it into the source tree sadly) |
| 08:34.34 | MrChimpy | well, they're HP DL360 1U servers with 2 slots so i expect each slot is on its own bus |
| 08:34.51 | clive- | pjchilds, I use the divas, they work nicely |
| 08:35.22 | MrChimpy | I'm doing the evaluation at the moment - it's not live yet. |
| 08:35.32 | MrChimpy | when we do go live it'll be 1000 line install. |
| 08:35.48 | MrChimpy | and i'll be a very stressed out chimp. |
| 08:39.26 | Tili | trixter: 1000 calls on freeswitch with transcoding also |
| 08:39.45 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 08:40.13 | dpryo | Asterisk is neat. I used about 3 minutes to configure MeetMe(), and 15 seconds afterwards there are 9 people active in it ..and has been going on for 45 minutes now |
| 08:40.13 | dpryo | cool |
| 08:42.10 | Tili | dpryo: asterisk is neat for PBX like things always |
| 08:42.36 | dpryo | It's easier, AND cheaper than my avaya definity :) |
| 08:42.39 | X-Gen | heya clive- |
| 08:42.43 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 08:43.01 | clive- | x-gen, whats up? |
| 08:43.36 | X-Gen | not 2 much, easing into friday :) |
| 08:46.02 | trixter | is it friday already? |
| 08:46.10 | trixter | hard to keep track when every day is a weekend |
| 08:46.17 | iDunno | slacker ;) |
| 08:52.10 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 08:52.11 | tdi | hi |
| 08:52.51 | tdi | does anybody know about some global registrar of SIP accounts ( from many comanies ) and the number pools ? |
| 08:53.38 | tdi | what i want to find is the company that has a global register of records of numbers, so i can find other sip accounts and call them directly, not thru PSTN |
| 08:54.25 | tdi | just to know which number is SIP too, to cut the prices |
| 08:55.54 | *** join/#asterisk [Airwolf] (n=airwolf@195.80.226.242) |
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| 09:32.52 | *** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com) |
| 09:32.57 | Gabriel25 | May 26 05:31:18 NOTICE[1932] chan_iax2.c: Rejected connect attempt from 70.84.157.148, request '3157030060@inbound' does not exist |
| 09:33.08 | Gabriel25 | 3157030060 this is my phone nr |
| 09:33.22 | Gabriel25 | I can`t receive phone calls |
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| 09:47.10 | *** join/#asterisk Mr-packet (n=a@222-153-139-222.jetstream.xtra.co.nz) |
| 09:48.35 | Mr-packet | does anyone have a suggestion a good gsm cellular gateway product? |
| 09:50.32 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
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| 10:06.54 | Assid | tdi: sipbroker .. sipphone.. |
| 10:08.44 | blitzrage | man I wish Cisco 7960's would allow you to control which lines were behind NAT -- its all or nothing |
| 10:09.44 | mitcheloc | why would they want to help sell more asterisk systems instead of sccp ;) |
| 10:12.54 | *** join/#asterisk dlynes_home (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
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| 10:17.51 | Assid | blitzrage: shouldnt your sip.conf be responsible for that? |
| 10:18.46 | tdi | Assid: what i want to achieve is to provide functionality like this :" userA@sip.tdi.com calls 11112222 number. normally if it is not my numbering pool, i would have to direct it to PSTN ( and pay ). but i would like to check in some global DB the situation that 11112222 might be also a SIP at ex. 11112222 - > userB@sip.somewhere.com. This would give me possibility to avoid PSTN and use direct SIP signalization. |
| 10:19.46 | Assid | err.. you can only get that if people subscribe to a uDNDi |
| 10:33.21 | blitzrage | Assid: assuming that I'm not behind NAT and want to register one line locally, and another line remotely |
| 10:37.43 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
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| 11:11.19 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:11.20 | Modcuts | what would be the best way of taking down the amount of ping hops if you have extremly high ping rates? |
| 11:11.30 | *** join/#asterisk spunz_ (n=spunz@81.217.96.96) |
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| 11:30.05 | laolang | 支持中文吗？ |
| 11:40.40 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 11:41.00 | *** join/#asterisk Ecio (n=eciostar@194.105.59.42) |
| 11:41.11 | blitzrage | Modcuts: what do you mean? |
| 11:41.33 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 11:43.23 | Ecio | hi all, i have problems with a SIP trunk between asterisk and cisco call manager: i can call from che CM to * but not viceversa. in the debug * says "SIP/callman02-079a is circuit-busy" |
| 11:43.58 | Ecio | i've configured asterisk like: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration |
| 11:44.11 | Ecio | with two callman friend |
| 11:45.24 | Ecio | any idea? |
| 11:48.13 | *** join/#asterisk laolang (n=new@60.24.19.98) |
| 11:56.01 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 11:58.18 | *** join/#asterisk azeteg (n=azeteg@c115.brewhouse.se) |
| 11:59.17 | azeteg | I am wondering, I have this Swissvoice IP10s, which I put SIP software on. When I try to make a call from it - it seems to try to make a call for each digit I press |
| 11:59.25 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 11:59.30 | *** join/#asterisk oej (n=oej@ip-207-145-80-8.nyc.megapath.net) |
| 11:59.33 | azeteg | is there some SIP setting that makes it try that? |
| 11:59.50 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 12:00.40 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:01.04 | Ecio | hi d-fender |
| 12:01.55 | Ecio | maybe you can help me, I got "circuit-busy" when i try to call from xlite (connected to my asterisk) to call manager (sip trunk). the opposite (calling from cm to asterisk) works |
| 12:03.34 | azeteg | Ecio: what does the log say? |
| 12:04.16 | azeteg | Ecio: did you analyze the traffic in detail? |
| 12:04.51 | brif8 | anyone using Asterisk on OS X with the Unicall Drivers, or involved with the Unicall drivers. My question is: Is the Unicall driver stable enough for a production enviroment ? |
| 12:05.45 | Ecio | executing Dial ("SIP/myasteriskaccount" "SIP/callmanager/extension") in new stack |
| 12:06.00 | Ecio | <PROTECTED> |
| 12:06.10 | Ecio | now im tryin with tethereal |
| 12:06.16 | Ecio | to see what happens |
| 12:07.15 | Ecio | uhm now i got 404 not found |
| 12:07.26 | azeteg | oh |
| 12:07.35 | azeteg | do you know what my problem could be? |
| 12:08.02 | azeteg | that damn swissvoice phone tries sending the extension number after every number |
| 12:08.04 | Ecio | azeteg: i dont have a swissvoice phone, only SIP softphone or SCCP cisco phones |
| 12:08.09 | azeteg | the cisco phones work fine |
| 12:08.09 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:08.52 | azeteg | I have Cisco 7905,7940,7960,ATA186 <- these all work fine |
| 12:09.13 | azeteg | the swissvoice IP10s tries to setup a call when I press one digit |
| 12:09.21 | azeteg | if I make an extension for just one digit, it works |
| 12:09.39 | Ecio | lol |
| 12:09.50 | azeteg | but if I press like "1-2" it tries setting up call to extension 1, then extension 2 |
| 12:10.16 | Ecio | have you checked the swissvoice manual ? |
| 12:10.23 | Ecio | it's a strange behaviour for a phone... |
| 12:10.35 | *** join/#asterisk s0lid (i=s0lid@gr-153-200.eglobalreach.net) |
| 12:10.40 | azeteg | yes very strange |
| 12:10.52 | s0lid | how can i kick agents from the queues from the cmd? |
| 12:10.53 | azeteg | I am just wondering if there is some standard behavior |
| 12:12.01 | *** join/#asterisk Faithful (n=Faithful@mail.familyfirst.org.au) |
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| 12:16.52 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 12:16.55 | fbnts | Hi everyone, just got a quick question - I have installed * on a new server from the 1.2 snapshot and the svn. When I receive an incoming IAX2 call it appears as though several calls have come in and all hung up |
| 12:17.28 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 12:17.49 | fbnts | I set up my extension.conf with two simple lines: 08450043106,1,Ringing() & 08450043106,2,Voicemail() |
| 12:18.17 | fbnts | so in theory, the call should be answered and play the voicemail prompt |
| 12:18.43 | adorah | Hi did anyone experienced with connecting Skype to Asterisk via UPLINK software? |
| 12:19.12 | *** join/#asterisk jahani (n=k@adsl196-147-225-217-196.adsl196-16.iam.net.ma) |
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| 12:27.35 | Assid | oh man.. im tired |
| 12:27.46 | Assid | being a code monkey suck |
| 12:27.47 | Assid | s |
| 12:28.34 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 12:28.54 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 12:31.04 | blitzrage | Assid: agreed -- I didn't even go to bed last night -- it's now 8:30am :) |
| 12:31.25 | Assid | what do you code? |
| 12:31.45 | laolang | 这里支持中文吗？ |
| 12:31.50 | blitzrage | PHP and pgSQL mostly |
| 12:31.54 | Assid | ohhh!@!! |
| 12:31.55 | Assid | nice |
| 12:32.08 | blitzrage | with some webservices SOAP thrown in there |
| 12:32.17 | Assid | aah.. always wanted to play with those |
| 12:32.39 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 12:32.48 | [TK]D-Fender | Code omnkey like Fritos! Codemonkey like Tab and Mountain Dew! |
| 12:33.01 | blitzrage | although my real job is actually senior network engineer for an ITSP -- I don't really code, I'm a telecom guy, but I code when I have to -- had to create an E911 portal system |
| 12:33.20 | Assid | gets even worse wen you gotta read someone elses code.. and you dont have a choice but to use existing method of coding |
| 12:33.25 | *** join/#asterisk BugKham (i=BugKham@202.8.86.168) |
| 12:33.43 | blitzrage | Assid: exactly what happened to me -- I inherited a GUI system, and have been expanding it for over a year |
| 12:34.55 | Assid | cant use a test box when the entire code is online.. and taking forever to download |
| 12:35.24 | blitzrage | heh |
| 12:35.35 | SwK | http://www.computerworld.com/action/article.do?command=viewArticleBasic&articleId=9000515 if you havent read this article on why the US Domestic Spying is bad you should |
| 12:35.40 | Assid | gotta use the old download edit upload trick |
| 12:35.45 | blitzrage | I have a test box now too... currently redoing some dialplan restructuring on the network to cluster asterisk |
| 12:35.55 | Assid | nice |
| 12:35.58 | Assid | i should do that sometime |
| 12:36.00 | BugKham | how do we kick registered sip friends out? |
| 12:36.14 | SwK | assid: dont listen to blitzrage he doesnt know what he's talking about |
| 12:36.17 | blitzrage | deny=x.x.x.x ? |
| 12:36.25 | blitzrage | Assid: it's true -- listen to SwK |
| 12:36.42 | blitzrage | deny=x.x.x.x/32 rather ? |
| 12:36.43 | SwK | heh |
| 12:36.59 | blitzrage | not sure if that blocks registration attempts -- might have to do it in the FW at the IP level |
| 12:37.34 | SwK | deny=x.x.x.x/y.y.y.y where x.x.x.x is the IP (host or net) and y.y.y.y is a "netmask" like bitmask for the network/range of IPs to block |
| 12:38.10 | blitzrage | I prefer just to use the /XX notation :) |
| 12:38.29 | Assid | ???????? |
| 12:38.29 | SwK | does / notation actually work? |
| 12:38.49 | blitzrage | yep |
| 12:38.53 | blitzrage | pretty sure it does |
| 12:38.56 | blitzrage | I think you can use both |
| 12:39.18 | *** join/#asterisk pdunkel (n=pdunkel@lime-gw16.one.at) |
| 12:40.12 | Assid | does RSA auth keys also encrypt the data ? |
| 12:40.25 | blitzrage | nope |
| 12:40.28 | blitzrage | just the auth |
| 12:40.39 | Assid | okay.. gonna go learn how to do that later |
| 12:40.43 | blitzrage | apparenlty you can do AES encryt with iax2 |
| 12:40.43 | Assid | after doing more coding |
| 12:41.06 | blitzrage | Assid: should be fairly straight forward once you generate the keys with astgenkey -n |
| 12:41.09 | Assid | AES is just another form of encryption right ? |
| 12:41.13 | blitzrage | aye |
| 12:41.21 | SwK | wow learn something new every day |
| 12:41.46 | SwK | I just looked at acl.c and it appears that CIDR notation will work on permit/deny |
| 12:43.14 | Assid | i wanna learn law! |
| 12:44.57 | *** join/#asterisk coppice (n=chatzill@169.202.17.210.dyn.pacific.net.hk) |
| 12:45.18 | *** join/#asterisk apardo (n=apardo@87.217.147.80) |
| 12:46.35 | Assid | okay time for me to take a break.. we wil be right back after these messages |
| 12:47.30 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.205.Dial1.SanJose1.Level3.net) |
| 12:47.39 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 12:48.24 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 12:48.36 | Dr-Linux | hi :) |
| 12:48.40 | BugKham | blitzrage: hmm, that will only block the next registration attempt, right? |
| 12:54.33 | tamp4x | how do i resolve dtmf problems from calls coming in on zap? |
| 12:55.23 | azeteg | what is the dtmf problem? |
| 12:56.04 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.205.Dial1.SanJose1.Level3.net) |
| 13:01.05 | azeteg | darn this is driving me insane |
| 13:01.20 | azeteg | why is the phone trying to setup a SIP call after every keypress |
| 13:01.24 | azeteg | ???? |
| 13:01.34 | tzanger | azeteg: which phone? |
| 13:02.01 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:02.13 | azeteg | Swissvoice IP10s |
| 13:02.32 | azeteg | I am trying to find some configuration inside the phone that makes it perform that way |
| 13:02.37 | azeteg | but I cannot find it |
| 13:02.49 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:02.52 | asteriskmonkey | anyone worked with polycoms blf stuff? |
| 13:02.57 | azeteg | tzanger: I doubt it is an asterisk problem |
| 13:03.23 | azeteg | tzanger: I have Cisco 7905,7940,7960,ATA186 that all work fine in this environment |
| 13:03.26 | tzanger | azeteg: is it trying to speed dial? What's the sip message it's sending? |
| 13:03.41 | azeteg | it sends a call to the first digit you dial |
| 13:04.13 | tzanger | azeteg: that is very odd |
| 13:04.17 | tzanger | have you factory-reset the phone? |
| 13:04.30 | azeteg | http://pastebin.com/739160 |
| 13:04.31 | tzanger | (since everythign else is working in your netowrk I suspect the phone, not asterisk) |
| 13:04.32 | azeteg | yeah |
| 13:04.42 | azeteg | thats what asterisk says |
| 13:04.57 | azeteg | I dialed "987" on the phone |
| 13:05.08 | tzanger | azeteg: do you have a screwed up dialplan *IN THE PHONE* ? I know polycoms can do weird things if your phone dialplan is buggered |
| 13:05.15 | tzanger | (NOT the asterisk dialplan) |
| 13:05.21 | azeteg | hmmm I did not look at taht plan |
| 13:05.31 | azeteg | tzanger: what do you mean it can get screwed up? |
| 13:06.17 | tzanger | azeteg: if you have the polycom phone dialplan set wrong (i.e. send after receiving 1 digit) it will do the identical thing |
| 13:06.24 | tzanger | factory default the phone |
| 13:06.26 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:06.31 | azeteg | I did factory default it before |
| 13:06.46 | azeteg | but let me again :) |
| 13:06.54 | Dr-Linux | tzanger: can you have a look at >> http://pastebin.com/739162 |
| 13:07.19 | Dr-Linux | i want monitoring only on 5001 extension |
| 13:07.29 | tzanger | uh |
| 13:07.32 | Dr-Linux | :S |
| 13:07.39 | tzanger | Dr-Linux: ${EXTEN} will be 91NXXXXXXXXX |
| 13:07.45 | azeteg | hehe |
| 13:07.57 | tzanger | ${EXTEN} will never equal 5001 if you hit that line in the dialplan |
| 13:08.02 | Dr-Linux | tzanger: yes, i got it, so what should be there? :S |
| 13:08.08 | tzanger | do you perhaps mean ${CALLERID(num)} ? |
| 13:08.19 | Dr-Linux | yes, |
| 13:08.23 | Dr-Linux | correct |
| 13:08.35 | tzanger | ok, so why are you asking me if you know what to do? :-) |
| 13:08.57 | blitzrage | people woke up |
| 13:09.02 | tzanger | morning blitzrage |
| 13:09.07 | blitzrage | tzanger: morning! |
| 13:09.23 | blitzrage | I'm in surprisingly good shape for someone who hasn't slept yet |
| 13:09.26 | BugKham | anyone know a command to unregister a registered sip friend? |
| 13:09.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.40 | Dr-Linux | tzanger: i'm not sure, there could be many mistakes |
| 13:10.04 | asteriskmonkey | anyone know much about asterisk "show hints" ? |
| 13:10.08 | tzanger | Dr-Linux: that's the only mistake I see |
| 13:10.12 | coppice | I haven't had any sleep yet, and its nearly 9:15PM |
| 13:10.24 | blitzrage | coppice: :) |
| 13:10.37 | tzanger | change that priority 1 to GotoIf($["${CALLERID(num)}" = "5001"]?2:4) |
| 13:10.43 | tzanger | coppice: :_) |
| 13:10.49 | tzanger | blitzrage: it just hasn't hit yet |
| 13:10.57 | tzanger | when I was your age I did that all the time |
| 13:11.07 | tzanger | actually I think there's something a little wrong with my internal clock |
| 13:11.28 | sevard | <tzanger> back when i was your age we were hard core youngsters. |
| 13:11.29 | Dr-Linux | tzanger: http://pastebin.com/739170 << how it's now? |
| 13:11.29 | blitzrage | tzanger: I hear that |
| 13:11.42 | tzanger | it's slowing down, but it used ot be that every 6-8 weeks I'd need less and less sleep until finally i just stayed up all night and the entire next day, then go to bed a little early and be fine |
| 13:11.52 | tzanger | it's about every 14-16 weeks now |
| 13:11.55 | tzanger | it's some weird cycle |
| 13:11.58 | Dr-Linux | tzanger: call recording should be only for 5001 extension, not for all |
| 13:12.08 | tzanger | Dr-Linux: understood, so test it! |
| 13:12.26 | tzanger | sevard: hahaha |
| 13:12.49 | tzanger | http://undo-software.com/undodb_roadmap.html |
| 13:12.50 | blitzrage | ok -- off to get some breakfast before my buddy gets here |
| 13:12.51 | tzanger | that's fucking cool |
| 13:13.01 | tzanger | could be very useful for asterisk |
| 13:13.08 | asteriskmonkey | has anyone got presence working with asterisk yet? |
| 13:13.14 | tzanger | asteriskmonkey: never tried it |
| 13:13.28 | sevard | asteriskmonkey: i tried it a couple months ago, nope. |
| 13:13.56 | asteriskmonkey | sevard: when you did show hints did it atleast say anything other than idel when you made a call? |
| 13:14.12 | blitzrage | tzanger: aye -- I don't understand gdb yet |
| 13:14.13 | Dr-Linux | tzanger: GotoIf($["${CALLERID(num)}" = "5001"]?2:4) << if the caller id is 5001 then it will go to priority 2 or 4 ? |
| 13:14.15 | sevard | I don't even think show hints existed when i tried it |
| 13:14.19 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:14.27 | blitzrage | 2 |
| 13:14.38 | blitzrage | true:false |
| 13:14.40 | tzanger | Dr-Linux: 2. If(condition?true:false) |
| 13:14.47 | tamp4x | how do i resolve dtmf problems from calls coming in on zap? (people dial digits but they are incorrect or missing) |
| 13:15.21 | Ecio | has anybody done a working trunk between cisco call manager and asterisk? |
| 13:15.52 | asteriskmonkey | tamp4x: try rtelaxing dtmf |
| 13:15.53 | Dr-Linux | tzanger: aww mean, i should swapp the priorities :) |
| 13:16.48 | *** join/#asterisk mercestes (n=merceste@cpe-70-116-149-83.houston.res.rr.com) |
| 13:16.56 | tamp4x | you can relaxdtm in zapata.conf? |
| 13:17.50 | *** join/#asterisk Hoondie (n=hoondie@ppp37-81.lns2.syd6.internode.on.net) |
| 13:17.53 | tamp4x | ahh yes u can |
| 13:17.54 | tamp4x | cool |
| 13:18.15 | coppice | but if you need to, something is seriously wrong |
| 13:18.44 | asteriskmonkey | sound like and tx/tx calibration issue |
| 13:18.47 | azeteg | how much is a decent PRI card? |
| 13:19.31 | Ariel_ | decent well it depends but the digium TE110p is a good card in most cases. |
| 13:19.47 | coppice | if the rx/tx calibration makes any difference, something is seriously wrong |
| 13:20.16 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 13:20.54 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:21.52 | *** join/#asterisk normast (n=Norm@CPE0014bf80aeff-CM0012c90d3496.cpe.net.cable.rogers.com) |
| 13:22.13 | Hoondie | hey people... anyone know of a good management interface for asterisk? i need to see things like the calls made from who to where, for how long etc.. |
| 13:22.14 | asteriskmonkey | coppice: if its a pri card i can pretty much say its a problem with DIGIUM CARDS and CERTAIN MOTHERBOARDS |
| 13:22.19 | azeteg | tzanger: I reset the phone again, and suddenly it started working... might have been the old H323 firmware dialplan that was left for some reason |
| 13:22.37 | asteriskmonkey | Hoodie: try using the basic cdr tools |
| 13:23.03 | tzanger | azeteg: could be. glad it's working |
| 13:23.13 | azeteg | tzanger: well I'm happy ;) |
| 13:23.23 | Hoondie | asteriskmonkey: where do i get them from? |
| 13:24.06 | asteriskmonkey | goole asterisk cdr reports there is some free php cdr anylzing stuff |
| 13:24.08 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 13:24.29 | Hoondie | thanks |
| 13:26.50 | mitcheloc | hola hoondie |
| 13:27.07 | brif8 | Does the TE410p card work in a Mac OS X and Apple Hardware environment ? |
| 13:27.21 | Hoondie | hey mitcheloc.. how are you doing? |
| 13:27.25 | coppice | asteriskmonkey: it must be fouling up pretty badly to give significant error rates on DTMF |
| 13:27.32 | mitcheloc | good, you? |
| 13:27.51 | coppice | brif8: there are no OS X drivers right now |
| 13:28.20 | asteriskmonkey | coppice: Actually when i went through my big batch of all digium cards te110 and 411 there where no errors or anything |
| 13:28.31 | asteriskmonkey | was the wierdest shit i ever experienced |
| 13:28.53 | Hoondie | mitcheloc: not bad.. bored |
| 13:29.05 | mitcheloc | yea? i got something for you test if you want ;) |
| 13:29.06 | asteriskmonkey | when tx and rx where perfect unresolvable echo when they where to low dtmf issues |
| 13:29.14 | Hoondie | mitcheloc: whats that? |
| 13:29.23 | tzanger | coppice: I still have to kill the bug where when a digium digital card generates DTMF, it does not return ANY audio frames for hte duration of the DTMF |
| 13:29.34 | tzanger | coppice: it should return empty frames, but it doesn't return ANY |
| 13:29.51 | kay2 | In what file am I supposed to put the password of my mysql for asterisk realtime |
| 13:30.10 | mitcheloc | Hoondie: ygpm |
| 13:30.17 | coppice | making the card generate DTMF is a stupid idea anyway |
| 13:31.15 | lost_diLLec | anyone knows a sip phone for the console which can read soundfiles and plays them if someone is taking the call ? |
| 13:31.48 | asteriskmonkey | lost_diLLec: Polycom 601 |
| 13:32.03 | tzanger | well the zaptel driver's doing it, not hte card itself |
| 13:32.08 | MrChimpy | anyone got a handy dialplan fragment to give a dialtone, take a number and immediately dial that on a trunk? i know it's dead simple, but i need it for a quick test and I guess someone's got it hanging round... :) |
| 13:32.10 | tzanger | although the rev2 cards I think can do it in hardware |
| 13:32.26 | lost_diLLec | wow - i did not thought that there is any =) |
| 13:32.33 | lost_diLLec | thanks asteriskmonkey i will try it |
| 13:32.34 | coppice | doing in hardware or in the driver is stupid |
| 13:32.46 | kay2 | tzanger: do you know where the mysql pwd is specified when using asterisk realtime ? |
| 13:33.05 | asteriskmonkey | kay2 : goole 4tw |
| 13:33.10 | Hmmhesays | make[3]: g++: Command not found |
| 13:33.10 | Hmmhesays | make[3]: *** [ZLib.o] Error 127 |
| 13:33.15 | Hmmhesays | anyone ever run into that? |
| 13:33.20 | kay2 | asteriskmonkey: 4tw ? |
| 13:33.33 | kay2 | Hmmhesays: yes |
| 13:33.37 | kay2 | Hmmhesays: install libzlib |
| 13:33.48 | asteriskmonkey | 4tw=for the win |
| 13:33.49 | kay2 | Hmmhesays: devel; |
| 13:33.59 | *** join/#asterisk fugitivo (n=ajf@190.48.161.41) |
| 13:34.04 | tzanger | kay2: /etc/asterisk/cdr_mysql.conf? |
| 13:34.05 | fugitivo | hi |
| 13:34.07 | tzanger | oh realtime.. not sure |
| 13:34.13 | kay2 | tzanger: it's not a cdr :) |
| 13:34.19 | fugitivo | is any report of problems with te205p cards? |
| 13:34.37 | tzanger | fugitivo: that's awfully vague |
| 13:34.52 | *** join/#asterisk RaYmAn-Bx (i=rayman@2001:7b8:399:1:0:0:0:3) |
| 13:35.13 | Hmmhesays | i have zlibc installed |
| 13:35.59 | fugitivo | problems at hardware level |
| 13:36.42 | Hmmhesays | it seems the install was br0ken |
| 13:36.54 | asteriskmonkey | fugitivo: still pretty vauge |
| 13:37.26 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 13:37.32 | coppice | it doesn't seem vague to me. seems a perfectly sensible question if he is considering purchase |
| 13:37.40 | tzanger | fugitivo: what exactly is the nature of your problem, and have you contacted digium support regarding this? Technical assistance is included in the purchase price of the card |
| 13:38.07 | fugitivo | coppice: i |
| 13:38.11 | tzanger | coppice: it's awfully vague. "does this car have problems?" "you know, just problems with its operation" |
| 13:38.16 | ManxPower | I'm starting to think that I need to carry TWO cell phones. One cell for normal use, and 1 cell phone for use on the I-59 corridor between Tuscaloosa, AL and Hattiesburg MS |
| 13:38.16 | fugitivo | coppice: i've already purchased one :) |
| 13:38.39 | kay2 | Hmmhesays: I TOLD YOU, libzlib-dev |
| 13:38.39 | coppice | ah, then its a silly question |
| 13:38.41 | fugitivo | tzanger: that's not vague, it's the same question you can do before or after buying a car :) |
| 13:38.51 | tzanger | fugitivo: nah |
| 13:39.01 | asteriskmonkey | fugitivo: well it dosnt operate well on low end via chipsets well |
| 13:39.09 | ManxPower | Today is going to be hell. |
| 13:39.30 | tzanger | fugitivo: "does this car have trouble with shifting from 3rd to 4th at high RPM? I have read that under certain circumstances the transmission can leap out in a rather ugly mess if you perform this pretty common operation" |
| 13:39.42 | tzanger | ManxPower: ? |
| 13:40.05 | fugitivo | tzanger: my config is one te110p, one te205p, 3com network card pci, not sharing irqs (tested with lspci -vb), mb is not in the incompatibiliy list. |
| 13:40.08 | coppice | ManxPower: what special quality is needed in a phone on that highway? |
| 13:40.13 | ManxPower | tzanger, yesterday would have been hell of my cell phone worked. 8-) |
| 13:40.37 | ManxPower | coppice, one that works. |
| 13:40.38 | *** join/#asterisk Lino` (n=Lino@i577BC570.versanet.de) |
| 13:40.55 | iDunno | guys... anyone know what would cause: |
| 13:40.56 | iDunno | local/scratch/zaptel/usr_src/modules/zaptel/wcusb.c:1451: error: unknown field `owner' specified in initializer |
| 13:41.01 | ManxPower | coppice, the ONLY carrier that works on the mountian is Verizon, but Verizon does not have service in that part of i-59 |
| 13:41.05 | iDunno | when compiling the zaptel driver? |
| 13:41.08 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
| 13:41.13 | tzanger | lspci -vb doesn't tell you shit about sharing IRQs |
| 13:41.19 | tzanger | fugitivo: what is the exact problem you are having?! |
| 13:41.23 | fugitivo | tzanger: all was working fine with the te110p, but i put a te205p card. The te110p is linked with a siemens hipath 3500, the te205p has 2 E1 links, and sometimes the spans are resseted and all calls are lost |
| 13:41.27 | tzanger | stop pissing around with generalities. What is it not doing? |
| 13:41.36 | coppice | ManxPower: aren't american cell networks just womderful :-) |
| 13:41.52 | fugitivo | tzanger: zttest sometimes shows NEGATIVE values |
| 13:42.00 | tzanger | fugitivo: ahh okay that is a nongeneral question |
| 13:42.05 | ManxPower | coppice, Well if I wanted a CONTRACT I'm sure I could roam to whoever has service in that region. |
| 13:42.28 | tzanger | ManxPower: time to get a 2m HAM rig and use a repeater |
| 13:42.44 | ManxPower | I signed a 2-year contract with Verizon 4 years ago and when I tried to cancel my service after Katrina they wanted to charge me $250 cancelation fee for each of the three phones on the account. |
| 13:43.01 | fugitivo | tzanger: so my question is, if all was working fine with the te110p, and then I put a te205p in the server and calls started to drop, AND cards are not sharing IRQ, maybe it's a problem with the card? |
| 13:43.03 | ManxPower | since they extend your contract by another 2 years anytime you change your service plan |
| 13:43.04 | tzanger | fugitivo: when everything falls down do you get any useful debug messages in the asterisk logs or in dmesg? |
| 13:43.07 | coppice | doesn't iridium give you coverage across the whole planet? :-) |
| 13:43.19 | tzanger | ManxPower: yes I fucking hate that |
| 13:43.21 | tzanger | FUCKING hate it |
| 13:43.30 | fugitivo | tzanger: 2 things |
| 13:43.31 | tzanger | this contract bullshit is idiotic |
| 13:43.37 | ManxPower | tzanger, I refuse cell contraces now. |
| 13:43.41 | ManxPower | contracts too. |
| 13:43.43 | fugitivo | tzanger: I see an error 500 and the span getting yellow alarm |
| 13:43.47 | ManxPower | prepay only. no contract |
| 13:44.11 | *** join/#asterisk pengyong (n=lala@203.86.46.130) |
| 13:44.12 | coppice | I haven't heard anything about iridium for a while. wonder what happened to its sad remains |
| 13:44.16 | tzanger | yellow alarm means the far side can't see you |
| 13:44.29 | fugitivo | tzanger: and the error 500? |
| 13:44.30 | tzanger | error 500 on what, SIP? |
| 13:44.37 | fugitivo | tzanger: no, zap |
| 13:44.42 | tzanger | zap error 500?? |
| 13:44.45 | fugitivo | yes |
| 13:44.48 | ManxPower | coppice, the usa federal govt signed contracts with them for service and kept them in buisness. |
| 13:44.54 | tzanger | I've never heard of that |
| 13:44.57 | tzanger | what's the exact message |
| 13:45.01 | fugitivo | let me find the error |
| 13:45.07 | ManxPower | iridium phone sales skyrocketed (so to speak) after Katrina |
| 13:45.09 | iDunno | (gottit, never mind :) |
| 13:45.17 | tzanger | ManxPower: no doubt |
| 13:45.27 | tzanger | until the iridium uplink site is flooded :-) |
| 13:45.47 | ManxPower | tzanger, one would assume the irridium uplink site is not near the disaster |
| 13:46.01 | tzanger | ManxPower: one would also assume there are multiple uplink sites |
| 13:46.06 | ManxPower | the phones talk directly to the bird, of course. |
| 13:46.07 | tzanger | preferably on different continents |
| 13:46.15 | iDunno | wheeee! |
| 13:46.15 | tzanger | yes the phones talk directly to the satellite |
| 13:46.21 | Hmmhesays | anyone know what the zlibc package is called in centos? |
| 13:46.32 | coppice | ManxPower: doubled overnight to 2, I guess :-) |
| 13:46.35 | tzanger | I'm curious as to how sensitive the satellites are to hear a tiny litlte phone |
| 13:46.43 | ManxPower | I think it's called libz actually |
| 13:46.46 | fugitivo | tzanger: anyways, what does the negative values in zttest mean? |
| 13:46.48 | tzanger | it's always amazed me |
| 13:46.57 | ManxPower | tzanger, well not so tiny.... |
| 13:46.57 | tzanger | fugitivo: it means it got the data earlier than expected |
| 13:47.02 | coppice | tzanger: they don't hear a tiny little phone |
| 13:47.19 | tzanger | ManxPower: the iridium phones? IIRC they were significantly smaller than the 3W AMPS bagphones |
| 13:47.43 | fugitivo | Short write: 0/15 (Unknown error 500) |
| 13:47.45 | ManxPower | tzanger, *nod* They are. |
| 13:48.00 | tzanger | short write means that the driver could not write all the data to the card |
| 13:48.03 | ManxPower | fugitivo, I have never ever heard of that error with Asterisk or Digium |
| 13:48.04 | fugitivo | Write to 77 failed: Unknown error 500 |
| 13:48.12 | coppice | and even smaller than a 30s car phone using tubes. that's not much of a comparison |
| 13:48.17 | tzanger | fugitivo: hang on let me look in the zaptel driver to see if I can find an error 500 |
| 13:48.23 | fugitivo | ManxPower: well, there's a first time for all :) |
| 13:48.25 | tzanger | ha |
| 13:48.40 | fugitivo | ManxPower: it's the first time for me too |
| 13:49.06 | fugitivo | tzanger: exactly after that 2 messages, i see yellow alarm on span an all calls are lost |
| 13:49.08 | tzanger | fugitivo: nothing easily found in zaptel.c/wct4xxp.c or the header files |
| 13:49.15 | coppice | fugitivo: error 500 from zaptel isn't actually an error. it means there is some signalling info to collect, there must be a bug for you to see that |
| 13:49.36 | tzanger | fugitivo: out of curiosity, why did you go with two cards instead of a single quadspan? (likely because you already had the single span) |
| 13:49.51 | fugitivo | tzanger: right :) |
| 13:49.52 | tzanger | fugitivo: you have a T100P or a TE110P? |
| 13:49.58 | fugitivo | te110p |
| 13:50.24 | fugitivo | coppice: then i should contact digium |
| 13:50.51 | tzanger | fugitivo: it sounds like you need to contact digium support... this sounds like something (driver or PCI implementation of either the TE110P or TE210P, they are different) is buggered |
| 13:50.56 | *** join/#asterisk avallach (n=chatzill@63.118.102.137) |
| 13:51.00 | tzanger | I am willing ot put my money on the TE110P's PCI implementation being buggered |
| 13:51.03 | coppice | i think there are some places in chan_zap.c which don't handle the 500 properly |
| 13:51.31 | tzanger | coppice: what *is* an error 500 though? grepping for 500 in zaptel.c/wct4xx.c and headers doesn't show it |
| 13:51.57 | tzanger | at least not in a quick look |
| 13:52.04 | coppice | its an indication there is signalling info to collect. you'll find it tested for in chan_zap.c |
| 13:52.13 | avallach | Greetings |
| 13:52.19 | tzanger | the te210p came out after digium cleaned up their PCI problems in the 4xx series |
| 13:52.38 | tzanger | so I'm thinking the 110 may be either stalling the bus or the driver is not doing something quite right |
| 13:54.02 | fugitivo | i reach the point of thinking it's an incompatibility problem between the 110p and the 205p |
| 13:54.08 | avallach | I'm using a polycom soundpoint 501 w/ asterisk that very frequently does not accept calls, the asterisk console will say NOTICE[26103]: chan_sip.c:1990 auto_congest: Auto-congesting SIP/36264-7ca5 - any ideas? |
| 13:54.33 | coppice | tzanger: what you said above is wrong. if you grep for 500 in zaptel.h you will find |
| 13:54.35 | tzanger | avallach: Asterisk cannot see the polycom |
| 13:54.35 | coppice | #define ELAST 500 |
| 13:54.43 | tzanger | coppice: ahh yes, I missed that |
| 13:55.03 | avallach | right before that message the polycom will do a partial ring - more of a blip |
| 13:55.16 | MrChimpy | guys, anyone got suggestions on how to do dialouts either from the CLI or dialplan that place a call and keep it open so I can get a few concurrent calls going? console dial isn't happy without DSP dev, and I only have one IAX2 phone as a test client. |
| 13:55.18 | tzanger | avallach: either your network is fucked or the phone is changing IPs (short DHCP time and screwy server?) and Asterisk can't keep up? |
| 13:55.18 | coppice | then if you look in chan_zap.c you will find it checked for, but not at every read or write. |
| 13:56.18 | avallach | ok - I don't think it is a DHCP problem and the phone works for outgoing calls and some incoming |
| 13:56.33 | tzanger | avallach: do you have more than one of these phones? |
| 13:56.41 | tzanger | avallach: do you have the same IP assigned to the phone and something else? |
| 13:56.50 | tzanger | avallach: tcpdump and sip debug should help you |
| 13:56.56 | tzanger | coppice: hmm |
| 13:56.56 | avallach | yes - I have more than one |
| 13:57.05 | avallach | not that I'm aware of |
| 13:57.07 | avallach | ok |
| 13:57.12 | MrChimpy | hmm. that's what I need. to dial someone and drop them into a context as if it were an incoming call when they answer. |
| 13:57.39 | tzanger | avallach: ping the polycom's IP and watch the output (also do this with tcpdump running to see what's going on on the wire) |
| 13:57.47 | tzanger | MrChimpy: Dial() has an option for this |
| 13:58.00 | MrChimpy | thanks |
| 13:58.05 | asteriskmonkey | avallach: it may be loosing its quailfy status |
| 13:58.27 | tzanger | asteriskmonkey: on a LAN that should NEVER happen |
| 13:58.33 | tzanger | asteriskmonkey: unless you're using qualify=1 or osmething |
| 13:58.45 | avallach | ok - tzanger is there something I'm looking for in the ping? |
| 13:58.54 | avallach | it looks very consistent |
| 13:58.59 | asteriskmonkey | tzanger: yep, i know ive seen some really messed lans that are 500+ms on the same segment |
| 13:58.59 | tzanger | avallach: multiple responses |
| 13:59.08 | asteriskmonkey | do a sip show peers |
| 13:59.32 | tzanger | asteriskmonkey: well don't plug a patch cable into the switch twice :-) |
| 13:59.51 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 13:59.56 | avallach | sip show peers looks just fine |
| 14:00.03 | asteriskmonkey | tzanger: yes thats an evil thing too :) na this one guy had a really wonky dlink |
| 14:00.19 | asteriskmonkey | some ports where good others where hight |
| 14:00.24 | avallach | I've not run tcpdump before - so I'll give that a shot |
| 14:00.28 | asteriskmonkey | and you can make calls right now |
| 14:00.36 | asteriskmonkey | try it again when you expeience the issue |
| 14:03.40 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 14:03.48 | boch | whats up |
| 14:03.51 | avallach | okay - I just got it to happen again - sip show peers looked good |
| 14:04.26 | asteriskmonkey | ok how many ms are you at when it happens |
| 14:05.23 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:05.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:05.34 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 14:05.57 | avallach | 84 I think |
| 14:05.59 | boch | could someone help me with an accounting issue, im having calls of about 120 minutes |
| 14:06.14 | avallach | yes, 84 is what it showed |
| 14:06.27 | Ecio | anyone knows how cisco call manager deals with SIP urls? i've setup a sip trunk between CM and *, i can call from CM to * but i cant call from * to CM (and neither direct ip call from xlite to CM) |
| 14:06.29 | avallach | of course, someone just called me and it is working just fine at the moment |
| 14:07.38 | asteriskmonkey | thats odd isnt it, loose cable maybe? |
| 14:08.11 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:08.13 | Ecio | i got a SIP/2.0 404 Not Found |
| 14:08.40 | avallach | <PROTECTED> |
| 14:08.58 | avallach | yes, it is a weird problem |
| 14:09.01 | avallach | it is somewhat intermittent |
| 14:09.19 | asteriskmonkey | tun a ping -t on it see if it looses connectivity at all |
| 14:09.37 | *** join/#asterisk vechers (i=vechers@64.61.117.138) |
| 14:09.54 | avallach | from windows? or linux? |
| 14:10.02 | asteriskmonkey | thats in windows |
| 14:10.13 | avallach | ok - I was going to say linux seems to ping forever |
| 14:10.15 | asteriskmonkey | but you can use linux just run a constant ping on it |
| 14:10.25 | asteriskmonkey | yes it does unless you break it |
| 14:10.37 | asteriskmonkey | presence dosnt work across nat does it |
| 14:11.24 | avallach | I'm not sure |
| 14:11.31 | avallach | Pings seems fine |
| 14:11.48 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 14:12.04 | avallach | well - got to jet for a moment - CEO's wife is here and needs some PC help |
| 14:13.05 | MrChimpy | exten => 511,1,Dial(${TRUNK}/phonenumber,,GOTO:demo^4) |
| 14:13.09 | MrChimpy | does that look right? |
| 14:13.44 | MrChimpy | cos it dials out ok, then just connects me without dumping the answering party to the demo context extension 4.... |
| 14:14.02 | *** part/#asterisk jake1932 (n=Administ@pool-68-236-24-98.phil.east.verizon.net) |
| 14:15.08 | MrChimpy | ooh, G() is closer to what I need |
| 14:15.10 | Ecio | no ideas ? :( |
| 14:16.22 | asteriskmonkey | if your calling another asteriskbox why not use the proper syntax |
| 14:18.32 | brif8 | coppice: is it being developed or not ? TE410p drivers for OS X |
| 14:20.38 | zoa | brif8: afaik not no |
| 14:20.45 | asteriskmonkey | coppice: are tjere every going to be te410p drivers for my linksys router |
| 14:20.53 | brif8 | zoa: thanks :( |
| 14:21.06 | asteriskmonkey | brif8: why would you want drivers for osx |
| 14:21.30 | asteriskmonkey | if you have a fully populated quadspan you would want something a little more reliable and no gui |
| 14:22.15 | brif8 | asteriskmonkey: explain reliable ? apple hardware from my usage is very stable and powerful |
| 14:22.46 | coppice | asteriskmonkey: just after there is a PCI slot to plug in into |
| 14:23.01 | avallach | astersikmonkey the pings keep on keeping on steady |
| 14:23.14 | asteriskmonkey | and you phones not maulfunctioned has it |
| 14:23.25 | asteriskmonkey | you should loose ping if the phone wont ring |
| 14:23.51 | asteriskmonkey | that is on the premise you registreation with the server is 100% minty |
| 14:24.02 | avallach | I'll hammer it w/ some attempted calls - most work - but the occasional ones are driving me crazy - drops people right to voicemail - but I hear a partial ring (blip) |
| 14:24.27 | avallach | first attempt the phone blipped |
| 14:24.33 | avallach | pings continued steady |
| 14:24.40 | coppice | brif8: there are a number of people working with * on OS/X, but porting the hardware drivers to apples way of doing drivers hasn't really worked out |
| 14:24.47 | avallach | sip show peers showed 76ms |
| 14:25.04 | asteriskmonkey | avallach: perhaps you have a messed up polycom phone |
| 14:25.25 | avallach | ok - maybe reset to factory defaults? I grab my config from an FTP server I setup |
| 14:25.31 | asteriskmonkey | osx (the newer flavour) is based on freebsd |
| 14:25.35 | avallach | it was working for oh two months until about 2 weeks ago |
| 14:25.45 | asteriskmonkey | so you may have better look taking stock bsd drivers and modding them |
| 14:26.00 | asteriskmonkey | and what happened 2 weeks ago? |
| 14:26.28 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:26.50 | asteriskmonkey | if its under warrenty swap it and see if problem still occurs |
| 14:26.51 | coppice | OS/X doesn't use normal freebsd drivers. OS/X drivers looks very different |
| 14:26.53 | avallach | oh tons of stuff - asterisk build change, some dial plan changes - my primary help is lost as to what is causing it we've tried tons of things since then |
| 14:28.09 | asteriskmonkey | coppice: just a starting point :P gah.. what am i saying .. i hate macs |
| 14:28.42 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl201-213-185-12-16.perspektivbredband.net) |
| 14:30.01 | avallach | I wish these polycom's would boot faster - they are a nice phone just a pain to configure |
| 14:30.12 | asteriskmonkey | Aastra 4t2 |
| 14:30.12 | boch | could someone help me with an accounting issue, im having calls of about 120 minutes, i guess my * is not getting some bye request, is this issue known? |
| 14:30.36 | asteriskmonkey | aastras are much nicer better web and boot wasy faster .. oh and there lcd has a backlight :D |
| 14:30.46 | asteriskmonkey | i have a polycom 601 with sidecar on my desk |
| 14:30.52 | avallach | neat |
| 14:30.55 | brif8 | coppice: do you know who is working on the drivers? astmasters menations a unicall driver in favor of zaptel, but that is for apple modems |
| 14:30.55 | asteriskmonkey | :P banging away at remote presence issues |
| 14:31.07 | avallach | I used one of the Linksys 941s and that was easy to configure and use |
| 14:31.30 | avallach | but the polycom call quality was wonderful |
| 14:31.35 | *** join/#asterisk _Paulo_ (n=Paulo@c90621fa.virtua.com.br) |
| 14:32.12 | asteriskmonkey | so is aastras same stuff i like both , i just loath the horrd web gui on the polycom but lets me honest that phone is ment to be a tftpd configured phone.. although the boot time is redicoulous |
| 14:32.33 | coppice | brif8: nobody is working on them. It was too big a pain, so they gave up |
| 14:32.47 | brif8 | ok |
| 14:32.54 | asteriskmonkey | brif8: get virtual pris :) maybe a solution |
| 14:33.17 | MrChimpy | ok, cool. i've got it dialling an external number and dropping both my line and the recipient into a dialplan extension when the other end is answered. Now what i _really_ need to do is make a bunch of these calls concurrently, I don't care about what happens at my end - i'm only dialling to trigger the extension with the commands in. Is there a way to do that? |
| 14:33.22 | brif8 | virutal pris ?? |
| 14:33.47 | avallach | I'll have to check out the Aastra |
| 14:34.11 | avallach | okay - so I just reset my local phone settings |
| 14:34.21 | avallach | maybe that will help |
| 14:37.36 | avallach | Called 36264 |
| 14:37.45 | avallach | NOTICE[26103]: chan_sip.c:1990 auto_congest: Auto-congesting SIP/36264-93c9 |
| 14:37.55 | avallach | -- SIP/36264-93c9 is circuit-busy |
| 14:38.04 | avallach | == Everyone is busy/congested at this time (1:0/1/0) |
| 14:38.20 | brettnem | hey, anyone have any experience using Async RTP patch #5162? |
| 14:38.41 | coppice | "Everyone is busy/congested at this time" flu strikes again :-) |
| 14:39.02 | sevard | that has to be the lamest joke I have ever heard. |
| 14:39.02 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 14:39.24 | tzanger | heh |
| 14:39.26 | tzanger | took me a minute to get it |
| 14:39.28 | brettnem | sevard: what did one snowman say to the other snowman? |
| 14:39.31 | tzanger | coppice is very good at that |
| 14:39.35 | sevard | brettnem: plesae no |
| 14:39.45 | brettnem | sevard: smells like carrots |
| 14:39.53 | tzanger | brettnem: heh |
| 14:39.57 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 14:40.04 | tzanger | What's green an stinks? The Hulk's farts. |
| 14:40.11 | tzanger | (another 10 year old joke) |
| 14:40.20 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 14:40.28 | avallach | so, if that's the flu - then how do I get rid of it? |
| 14:40.39 | tzanger | avallach: tcpdump, I told you |
| 14:40.47 | tzanger | you have network or phone issues |
| 14:40.47 | Assid | flu ? |
| 14:40.54 | Assid | stand on one leg.. and cluck like a chicken ? |
| 14:40.59 | avallach | yes - I'm still trying to figure out how tcpdump works |
| 14:41.26 | tzanger | avallach: tcpdump -ni eth0 -s0 -w capture.bin host ip.of.phone |
| 14:41.30 | tzanger | get it to happen |
| 14:41.32 | tzanger | then break out |
| 14:41.36 | brettnem | -ni? |
| 14:41.41 | tzanger | now capture.bin has exactly what was going on on the wire |
| 14:41.41 | avallach | ok |
| 14:41.46 | tzanger | -n = don't resolve addresses |
| 14:41.50 | brettnem | ah |
| 14:41.50 | tzanger | -i eth0 = watch eth0 |
| 14:41.54 | brettnem | ah oh |
| 14:42.20 | tzanger | you can take caputre.bin and view it with tcpdump -r capture.bin or bring it into ethereal, which is far more user-friendly |
| 14:42.25 | sevard | my fricken TFTP server stopped working |
| 14:42.49 | sevard | Oh what's that? I used to work perfectly now the exact same syntax breaks me even though you didn't upgrade/change anything? That's nice. |
| 14:42.52 | tzanger | fricken doesn't make good products, they always fail |
| 14:42.58 | brettnem | I like the ethereal RTP stream analysis |
| 14:43.10 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:43.15 | tzanger | people constantly complain about their products |
| 14:43.41 | vooduhal | Hello all. Does anyone know if the Polycom IP 500s and 600s have any kind of SNMP support? |
| 14:43.41 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 14:43.43 | brettnem | so has anyone used the async rtp patch for asterisk? |
| 14:43.53 | Assid | i want my cell phone to be as clear as my voip line .. AT TIMES.. |
| 14:44.19 | sevard | I'd just rather have my cellphone get connectivity when i'm outside on the roof on a clear day. |
| 14:44.34 | JASON99 | Is there a way to only allow inbound calls (no outbound) to a specific extension? |
| 14:45.10 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
| 14:45.14 | Assid | JASON99: yes.. limit his context |
| 14:45.15 | tzanger | vooduhal: that's a question for polycom, perhaps their data sheets on their phones give that, or their admin manual |
| 14:45.20 | kay2 | hey |
| 14:45.22 | tzanger | I use polycom phones but I've never tried SNMP with them |
| 14:45.28 | vooduhal | JASON99: Just set the phones context and hangup on all dialed calls in that context. |
| 14:45.50 | asteriskmonkey | cool i could run mrtg grphs of my polycom phones |
| 14:45.52 | JASON99 | Assid/vooduhal: Thanks a lot.. I never thought of that |
| 14:46.03 | vooduhal | tzafrir: I've read the 1.6.1 Admin guide about 3 times and can't find anything about it. |
| 14:46.22 | asteriskmonkey | and of you know about the polycom presence features? |
| 14:46.40 | brettnem | vooduhal: I don't think it has any SNMP capabilities |
| 14:47.48 | vooduhal | brettnem: That's that I'm a afraid of. I've got MWI for agents working in asterisk using a captured SIP MWI packet and toggle it on and off on the phones with the external notification part of app_voicemail. I was just hoping for a way to do it that wasn't a hack. |
| 14:48.18 | vooduhal | Guess I'll keep using my hack. |
| 14:48.53 | kay2 | tzanger: Do you know how to add realtime agent ? |
| 14:49.30 | kay2 | tzanger: I mean without having to reedit the file agent.conf each time |
| 14:49.46 | Assid | kay2: do you have realtime system? |
| 14:49.59 | kay2 | Assid ? |
| 14:50.01 | Assid | you could look into extconfig if im not mistaken |
| 14:50.21 | kay2 | Assid: for asterisk RealTime, I need to add something in particular ? |
| 14:50.43 | Assid | well.. first and foremost.. you need to set it up to use realtime |
| 14:51.05 | Assid | http://www.voip-info.org/wiki/view/Asterisk+RealTime <-- there |
| 14:51.25 | kay2 | Assid: it's just the extconfig.conf that has to be modified |
| 14:51.41 | Assid | well.. depending on what u wanna use |
| 14:51.46 | Assid | mysql/odbc.. etc... |
| 14:51.51 | Assid | so.. you set it according to that |
| 14:51.58 | Assid | read that page |
| 14:52.17 | kay2 | Assid: basically, I want to be able to do a Addqueuemember(someting) but before, I want to user to be asked for a password |
| 14:52.38 | kay2 | s/to/the |
| 14:52.42 | vooduhal | We just did it all using a mysql database and Authenticate |
| 14:52.51 | kay2 | Authenticate ? |
| 14:52.52 | *** join/#asterisk moprilo (n=jjohn@201.192.107.57) |
| 14:53.26 | vooduhal | kay2: We store the user info in a db, have an AGI check the agent id and get the pin number, then just Authenticate(${AGENT_PIN}) |
| 14:53.28 | kay2 | vooduhal: so the user has to use his voicemail pwd |
| 14:53.37 | vooduhal | Exactly. :) |
| 14:53.41 | vooduhal | Well, doesn't have to. |
| 14:53.46 | vooduhal | But that's the functionality we wanted. |
| 14:53.50 | kay2 | well with an AGI yeah |
| 14:53.53 | kay2 | but otherwise |
| 14:54.08 | kay2 | I have to do realtime for the voicemail too |
| 14:54.16 | vooduhal | Authenticate will use the first arg as the pin so it could be anything you wanted. |
| 14:54.47 | moprilo | when doing the watchbuddy on the polycom, I have to add this to the dial plan, exten => [extenofthewatchingphone],hint,SIP/[extenofthephoneiwannawatch] is this right? |
| 14:55.09 | vooduhal | We basically replaced AgentCallBackLogin or whatever its called with agi. |
| 14:56.08 | avallach | tzanger - I'm reviewing the trace ion Etherreal |
| 14:57.07 | avallach | I see 3 invites and then a cancel |
| 14:57.16 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 14:57.41 | avallach | from the server to the phone |
| 14:58.09 | avallach | and then after those from the phone to server I see 2 100 trying, then a 200 OK then a 487 Request cancelled |
| 14:58.45 | tzanger | avallach: so in other words |
| 14:58.56 | tzanger | the asterisk box tried three times to elicit aresponse from the polycom and couldn't |
| 14:59.15 | tzanger | avallach: it would be nice to get a tcpdump from the phone side |
| 14:59.31 | tzanger | i.e. <network> <hub - polycom and hub - box running tcpdump> |
| 14:59.31 | avallach | okay - is that possible? |
| 14:59.49 | tzanger | that way you could verify the polycom is seeing the packets and they aren't getting swallowed up by the network |
| 15:01.12 | avallach | ok |
| 15:01.16 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:01.33 | tzanger | avallach: right now either you have a firewall blocking it or a shitty network |
| 15:02.06 | Assid | nat issues? |
| 15:02.12 | tzanger | Assid: this is on a LAN |
| 15:02.20 | tzanger | but I've seen some pretty fucked up LANs |
| 15:02.22 | flujan | coppice , HI! I solve the problem with the mfc/r2 signalling... It was a problem with the hardware. The guys sold it as a T1 card. I will never connect to a E1! Thanks for your help steve. :) |
| 15:02.27 | Assid | hrmm |
| 15:02.41 | Assid | what happens to him? |
| 15:03.04 | Assid | whats the ping responses from the box to the phones? |
| 15:03.14 | tzanger | Assid: asterisk box sends invites to the polycom, and the polycom does not respond |
| 15:03.18 | kay2 | vooduhal: what is the difference between AgentLogin and AddQueueMember() ? |
| 15:03.26 | tzanger | Assid: ping looks normal apparently (i.e. no two devices with the same IP) |
| 15:03.37 | Assid | bad phone ? |
| 15:03.51 | Assid | try with softphone on a pc.. |
| 15:04.30 | fugitivo | kay2: show application agentlogin and show application addqueuemember |
| 15:04.57 | vooduhal | kay2: If I'm not mistaken, AgentLogin will just log the user into the ACD system and put them into any queue that they are statically configured in. AddQueueMember lets you add a a currently logged in user to a queue they weren't statically configured for. |
| 15:05.12 | flujan | guys, i want to test a E1 link with a simple dialplan... Every call received in the E1 should dial to the IAX2/123456 extension.. http://pastebin.com/739412 |
| 15:05.35 | flujan | as you can see, i receive a action but it doesn't enter in the incoming context |
| 15:06.14 | avallach | It is sporadic |
| 15:06.15 | *** join/#asterisk mopri (n=jjohn@200.122.157.96) |
| 15:06.24 | avallach | so I'd say it works 80% of the time 4/5 tries |
| 15:06.59 | avallach | my laptop routed through the phone works just fine - but I'v enot kept a softphone up and running on it constantly |
| 15:07.52 | *** join/#asterisk RaYmAn-Bx (i=rayman@2001:7b8:399:1:0:0:0:3) |
| 15:08.48 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 15:08.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:09.15 | tzanger | avallach: put a hub (not switch) between the phone and the rest of the network |
| 15:09.16 | avallach | I have X-Lite |
| 15:09.26 | tzanger | avallach: and then put your laptop with tcpdump or ethereal on the same hub |
| 15:09.28 | tzanger | and run the same test |
| 15:09.33 | tzanger | verify that the phone is seeing the request |
| 15:09.38 | avallach | ok |
| 15:09.40 | tzanger | polycoms are VERY good, I do NOT suspect the phone |
| 15:09.43 | tzanger | I suspect the network |
| 15:09.52 | avallach | I don't suspect the phone either |
| 15:09.57 | tzanger | either a shitty/flaky switch, wiring or firewall |
| 15:10.05 | tzanger | or still I suspect two devices with the same IP |
| 15:10.18 | tzanger | if you have a switch with a monitor port you can kind of sort of run the test that way |
| 15:10.18 | avallach | we have about 4 or 5 on our network and specifically one of the other ones started having similiar problems about the same time |
| 15:10.34 | avallach | but the other one is on a completely different segment of our network |
| 15:10.37 | kay2 | vooduhal: and with authenticate, how do you say (if pwd == good then goto (there) else goto (there)) |
| 15:10.49 | tzanger | avallach: segment? |
| 15:10.52 | avallach | I think the system we have asterisk on is part of the problem |
| 15:10.54 | tzanger | avallach: describe your LAN |
| 15:10.58 | vooduhal | kay2: Priority |
| 15:11.00 | avallach | totally different network equipment |
| 15:11.09 | vooduhal | kay2: If login fails, it will jump +101 |
| 15:11.12 | avallach | the only common point is where we have our asterisk server |
| 15:11.28 | vooduhal | So something like: |
| 15:11.42 | vooduhal | exten => 1234,1,Authenticate(4321) |
| 15:11.52 | vooduhal | exten => 1234,2,Playback(woot-sndfile) |
| 15:12.02 | vooduhal | exten => 1234,102,Playback(dumbass-sndfile) |
| 15:12.39 | _MartinCabrera_ | Hi, When i try to register g729 codec i got: "Registering 'G729-XXXXXXXX' to 'XXXXX'...FAILED(5)!" |
| 15:12.39 | _MartinCabrera_ | I'm trying with pentium4m/codec_g729a.so, my processors are P4 and OS is FC3 |
| 15:12.39 | _MartinCabrera_ | Any suggestion? |
| 15:15.29 | Assid | tzanger: you could also try pumping more bytes into the ping |
| 15:15.29 | tzanger | Assid: true but again I don't think it's a fragmentation problem |
| 15:15.30 | tzanger | Assid: I think he just has a shitty network somewhere |
| 15:15.30 | tzanger | avallach: ? |
| 15:15.30 | tzanger | avallach: do you have two network ports in this box? |
| 15:15.34 | avallach | yes |
| 15:15.51 | tzanger | and is the INVITE going out through the right port? |
| 15:15.54 | avallach | one is on our public ip network |
| 15:16.03 | avallach | I think so the tcpdump looked correct |
| 15:16.18 | avallach | and I'm on the phone right now - it just rang through |
| 15:16.52 | Assid | okay im gettiing a very weird issue |
| 15:18.00 | Assid | sometimes.. apparently if the person hangs up at a critical time before voicemail recording.. |
| 15:18.25 | Assid | i get the .txt file.. but no recording file |
| 15:18.36 | sevard | who wants to tell me more about caller id |
| 15:19.15 | Hmmhesays | so i'm thinking about getting a behringer 32 channel analog mixer |
| 15:19.31 | sevard | from what I understand setting caller id name is useless when it hits the pstn because the phone on the pstn queries the network for a name against the number |
| 15:19.31 | Kte2 | hmm, can the IVR use the .gsm files supplied? or does it 'have' to be wav |
| 15:19.53 | _MartinCabrera_ | both |
| 15:19.55 | sevard | that's why when I send out George Dubya and 1234567890 it doesn't work but if i send out the white house's number it says White House |
| 15:20.09 | sevard | did I get this right analog geeks |
| 15:20.27 | tzanger | i.e. is your tcpdump on the general ip network or voip network? |
| 15:20.44 | tzanger | Hmmhesays: nice |
| 15:20.49 | tzanger | I can't play well enough to require such a thing :_) |
| 15:21.02 | JASON99 | Assid: I'm pretty new to this... I've setup a new context called [inbound-only] and put this as the first line "exten => s,1,Macro(hangupcall) |
| 15:21.21 | JASON99 | should that prevent all calls from going through |
| 15:21.25 | tzanger | JASON99: not necessarily |
| 15:21.40 | tzanger | JASON99: use exten => _X.,1,NoOp(EXTEN IS ${EXTEN}) |
| 15:21.40 | Assid | JASON99: you could just put in a empty context |
| 15:21.46 | tzanger | exten => _X.,n,Macro(hangupcall) |
| 15:22.19 | tzanger | and do exten => s,1,NoOp(Called the 's' extension) |
| 15:22.24 | tzanger | exten => s,n,Macro(hangupcall) |
| 15:22.33 | sevard | okay then, there aren't any pstn geeks here |
| 15:22.38 | sevard | hurmp. |
| 15:23.01 | tzanger | JASON99: because if your incoming calls are comingfrom a voip provider or PRI you will generally have an actual number that is getting called, not 's' |
| 15:23.15 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-76-255.telkomadsl.co.za) |
| 15:23.16 | _MartinCabrera_ | Please, no ideas for "Registering G729 FAILED(5)"? |
| 15:23.16 | tzanger | 's' is generally only called by POTS and "dumb" CAS T1 inbound |
| 15:23.26 | mopri | for the pressence thing on asterisk, is it posible on asterisk v 1.0.9 |
| 15:23.29 | tzanger | _MartinCabrera_: g729 is broken with HEAD |
| 15:23.34 | JASON99 | Thanks.. is NoOp to print a message in the debug? |
| 15:23.35 | tzanger | sevard: what is your issue? |
| 15:23.38 | sevard | mopri: holy crap man upgrade |
| 15:23.45 | tzanger | JASON99: NoOp() prints a message to the console |
| 15:23.52 | JASON99 | thanks |
| 15:23.58 | sevard | tzanger: how does the pstn query the cname and how do you force yours out there |
| 15:24.13 | tzanger | sevard: what is your PSTN interface |
| 15:24.17 | Ecio | can someone help me debugging my cisco CallManager <-> asterisk trunk? |
| 15:24.17 | *** join/#asterisk camelon (n=chiardon@200.71.58.39) |
| 15:24.18 | sevard | tzanger: PRI |
| 15:24.24 | tzanger | sevard: depends on the telco then |
| 15:24.33 | camelon | Morning! |
| 15:24.39 | sevard | tzanger: my telco saw my CNAME at the switch, " |
| 15:24.40 | tzanger | Bell Canada lets me set callerid name and number and so long as it stays on Bell's network the name I set is seen |
| 15:24.54 | tzanger | sevard: if it hops off Bell's network the name is replaced with a database lookup of the number I send |
| 15:25.00 | sevard | tzanger: my telco saw my CNAME at the switch, "George Dubya", but they said when it hits the phone on the network it queirs the pstn |
| 15:25.16 | sevard | queries |
| 15:25.36 | sevard | tzanger: that sounds about right but it's too bad somebody can't do a traceroute on a damn phone |
| 15:25.38 | tzanger | it queries the DB. that is normal |
| 15:25.46 | sevard | so i know where the call is going ;\ |
| 15:25.52 | sevard | tzanger: so that's normal, how do you get past that |
| 15:26.18 | JASON99 | thanks again guys |
| 15:26.23 | tzanger | sevard: you don't |
| 15:26.49 | sevard | so there is absolutly no way to force CNAME out there. |
| 15:26.55 | sevard | no over ride characters |
| 15:27.07 | sevard | no escape sequence :| |
| 15:27.23 | tzanger | sevard: nope |
| 15:27.32 | MrChimpy | so I just rang half the company all at once and played bananaphone at them |
| 15:27.56 | sevard | tzanger: so i can send out the white house's phone number and have it say White House 9......... but I can't send out George Dubya with 123-456-7890 |
| 15:27.57 | boch | could someone help me with an accounting issue, im having calls of about 120 minutes, i guess my * is not getting some bye request, is this issue known? |
| 15:28.04 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 15:28.13 | sevard | any reason why the numbers are queried against a database? it sounds stupid |
| 15:28.59 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 15:29.24 | syle | not unless your a programmer |
| 15:29.25 | tzanger | sevard: correct |
| 15:29.29 | avallach | tzanger - I went looking for a hub - but couldn't find one |
| 15:29.44 | tzanger | sevard: that is the way it is doen |
| 15:29.51 | tzanger | you send the number, it looks up in a db and supplies the name |
| 15:30.07 | tzanger | sevard: if you want the name to be what you want, update their DB |
| 15:30.09 | sevard | and you can't inject names in the DB |
| 15:30.14 | Ecio | does anybody know why cisco Call Manager is sayin "404 not found" when asterisk calls it ? |
| 15:30.24 | tzanger | sevard: nope |
| 15:30.28 | tzanger | it's the PSTN, go through the right channels |
| 15:30.29 | avallach | tzanger - maybe a tcpdump that isn't interface specific - but is specific to this phone IP? |
| 15:30.36 | tzanger | there are checks and balances in place to prevent abuse |
| 15:30.51 | tzanger | even allowing you to set an arbitrary number can be abused |
| 15:31.02 | tzanger | some telcos prevent you from setting any number not in your DID block |
| 15:31.15 | LoRez | not ours :) |
| 15:31.25 | sevard | tzanger: so if I want to register names to my DIDs I make my telco do it |
| 15:31.40 | tzanger | sevard: exactly right |
| 15:32.02 | tzanger | sevard: and they will work globally, so long as the various telcos properly query each other's dbs |
| 15:32.28 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-124.prod-infinitum.com.mx) |
| 15:33.08 | boch | could someone help me with an accounting issue, im having calls of about 120 minutes, i guess my * is not getting some bye request, is this issue known? |
| 15:33.37 | *** join/#asterisk sb_mx (n=sb_mx@200.94.154.226) |
| 15:33.41 | tzanger | boch: ? |
| 15:34.41 | salviadud | hahaha |
| 15:34.49 | salviadud | i used to do that on the call center i worked on |
| 15:34.55 | salviadud | i never hung up my calls on time |
| 15:35.08 | salviadud | but that was on a propietary pbx system |
| 15:35.43 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:36.17 | tzanger | man they did a number on hoffa didn't they |
| 15:36.27 | boch | tzanger: im getting calls of 120 minutes long, but i never make them |
| 15:36.42 | tzanger | that's odd |
| 15:37.12 | boch | is it common? |
| 15:37.21 | tzanger | I've never heard of it before |
| 15:37.26 | tzanger | my CDRs are generally pretty accurate |
| 15:37.40 | tzanger | I've seen the OPPOSITE problem when IAX2 transfers were used, but not this |
| 15:37.45 | salviadud | boch, do you have a bunch of agents receiving calls? |
| 15:37.52 | boch | i mean, one of a thousand calls has 120 minutes |
| 15:38.00 | *** join/#asterisk Thock (n=FreePBX6@63.133.144.2) |
| 15:38.12 | brettnem | sevard: you can lookup a real phone number that resolves to a name you want to transmit and transmit that number instead |
| 15:38.16 | Thock | Hey all- Any ideas as to why i'm not long able to load chan_zap.so when running asterisk -cvvvvv |
| 15:38.17 | Thock | ? |
| 15:38.23 | sevard | brettnem: I know, I can do that. |
| 15:38.27 | Thock | long = longer |
| 15:38.29 | sevard | brettnem: i just wanted to send George Dubya :( |
| 15:38.42 | boch | salviadud: yes |
| 15:38.57 | brettnem | sevard: CNAM dipping is a large source of revenue.. And carriers never trust what their customers send them.. |
| 15:39.01 | tzanger | Thock: because your zapata.conf contradicts your zaptel.conf? |
| 15:39.11 | Thock | hm |
| 15:39.19 | tzanger | "CNAM dipping" ? i.e. db lookups? |
| 15:39.30 | sevard | why is it a large source of revenue |
| 15:39.33 | Thock | http://pastebin.ca/59053 that's my zaptel.conf , http://pastebin.ca/59052 that's my zapata.conf |
| 15:39.36 | tzanger | Thock: you get detailled errors when a module doesn't load... my suggestion is to read them |
| 15:39.49 | salviadud | boch, are your agents using a gui? |
| 15:39.51 | brettnem | tzanger: your db lookup is commonly known as a "CNAM dip" |
| 15:39.52 | Thock | tzanger: where are thos elogs by default? |
| 15:40.12 | salviadud | boch, i've been there... i've been a monkey, i got a theory of what's going on |
| 15:40.15 | tzanger | Thock: when it loads you will see "chan_zap.so: ERROR: error_message_here" |
| 15:40.51 | Thock | strange |
| 15:41.04 | boch | salviadud: sorry, what you mean with agents? i have many sip peers making calls and i route them via different sip providers |
| 15:41.35 | Thock | tzanger: http://pastebin.ca/59055 that's what i'm seeing |
| 15:42.15 | tzanger | Thock: that is blindingly obvious |
| 15:42.20 | tzanger | Thock: what do you think it's saying |
| 15:42.27 | salviadud | boch, is that some sort of international operator company? |
| 15:43.05 | [TK]D-Fender | tzanger : Isn't it though? So many missing seperators :) |
| 15:43.41 | *** join/#asterisk chadgraves (n=chadpgra@69.15.32.135) |
| 15:43.48 | boch | salviadud: kind of |
| 15:43.52 | tzanger | Thock: I'm not trying ot be an ass, but come on |
| 15:43.53 | tzanger | THINK |
| 15:43.54 | tzanger | READ |
| 15:44.02 | tzanger | it says it doesn't like line 10 |
| 15:44.07 | tzanger | have you looked at line 10 to see why it's complaining? |
| 15:44.11 | salviadud | boch, that's the easiest job eva! |
| 15:44.17 | [TK]D-Fender | tzanger : And the context name.... I'm smelling.... AMP |
| 15:44.28 | Thock | tzanger: i'm trying to be slient to remove any chances of further ridicule, sorry :( |
| 15:44.58 | tzanger | Thock: I'm not trying to ridicule, but it is OBVIOUS you have not tried to analyze what it's saying |
| 15:45.38 | brettnem | tzanger: and so you just ASSUME that's where the error is? ;) |
| 15:45.40 | tzanger | I am MORE than happy to help ANYONE if they are willing to help themselves, ask anyone here how much time and effort I put in to helping ohters |
| 15:45.47 | Thock | I know you're not. It's cool. |
| 15:46.14 | Thock | The only reason i had asked is because this was working perfectly fine yesterday, the box was rebooted due to power failure, i restored from a 4 hour old backup and nothing would work any longer. |
| 15:46.34 | tzanger | Thock: what changed since the backup? |
| 15:46.37 | tzanger | did you upgrade asterisk? |
| 15:46.38 | Thock | and even with those errors in zapata fixed, it's still not loading chan_zap.so |
| 15:46.39 | Thock | no |
| 15:46.40 | [TK]D-Fender | Thock : Are you sure your cards loaded properly and in the right order? |
| 15:46.41 | tzanger | are you using AMP? |
| 15:46.42 | Thock | nothing changed |
| 15:46.50 | Kte2 | question: I have two phones connected to my pbx and they both seem to be registering properly as extensions 200 and 201...but whenever i try to call one from the other, i get an "invalid extension" message playing..? |
| 15:46.51 | Thock | [TK]D-Fender: like i said, this was working fine before |
| 15:46.56 | Thock | tzanger: Yep. |
| 15:47.08 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 15:47.09 | [TK]D-Fender | Thock : Bringing it back up even after days of service can bring it up wrong. |
| 15:47.13 | tzanger | what does the title for this channel read? |
| 15:47.26 | sb_mx | Thock, didnt you update your linux box and dled the latest kernel? remember, the new kernels will start working after the next reboot |
| 15:47.48 | [TK]D-Fender | Kte2 : It means you don't have your dialplan set up so that they can dial each other. |
| 15:47.58 | tzanger | PCI enumeration isn't always repeatable, make sure your cards load in a predictable order by manually loading them |
| 15:47.59 | fugitivo | digium support is soooo slow |
| 15:48.13 | brettnem | fugitivo: Maybe they sent it offshore |
| 15:48.23 | [TK]D-Fender | Thock : Also if you just changed kernels, then you DEFINATELY need to recompile Zaptel <- |
| 15:48.33 | fugitivo | brettnem: that should be faster |
| 15:48.36 | Thock | I'll look into that. Thanks. |
| 15:49.13 | [TK]D-Fender | mog_work : How ya doin CHUM? ;) |
| 15:49.35 | avallach | tzanger - I was looking at the packet capture and I noticed that the order the packets are captured and the timestamps don't appear to be in sequence |
| 15:50.09 | avallach | of course I don't know if time means anything |
| 15:50.14 | _Paulo_ | fugitivo, can you share with us what is you asking Digium support? |
| 15:51.02 | _Paulo_ | s/is/are/ |
| 15:52.06 | mog_work | lol tk |
| 15:52.10 | mog_work | im pretty good |
| 15:52.23 | mog_work | about to make long trek to the parents house for the weekend |
| 15:52.26 | mog_work | you |
| 15:53.23 | file | MOG |
| 15:53.27 | file | you'll know! |
| 15:53.33 | mog_work | file |
| 15:53.48 | file | what's the proper path internally for origsvn... to use for SVN |
| 15:53.52 | file | proper URL rather |
| 15:54.18 | mog_work | you can use regualr svn or origsvn |
| 15:54.24 | mog_work | they both point you to the local repo |
| 15:54.26 | mog_work | at digium |
| 15:54.33 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:54.41 | *** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184) |
| 15:56.18 | NeonLevel | hi guys, good morning, i was wondering if its possible to send register info, just before dialing with my sip provider, i'm guessing my provider needs me to register the sip info, just before dialing, so is it possible? |
| 15:56.51 | Ecio | bye all |
| 15:57.04 | [TK]D-Fender | mog_work : Doing Ok.... you know for all the people who I've directed elsewhere for a card or two, I've saved you guys a FORTUNE in support for my efforts here. Karmic balance well in my favour :) |
| 15:57.39 | [TK]D-Fender | Just feeling accomplished is all, heh |
| 15:58.09 | file | the support drones are probably thankful |
| 15:58.55 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 15:59.32 | [TK]D-Fender | ~book |
| 15:59.33 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:59.33 | Hmmhesays | support drones |
| 16:00.03 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 16:00.56 | salviadud | the book is friggin' awesome maaaan |
| 16:01.18 | eKo1 | Quick ?: In zapata.conf, can I have something like channel => 1,3,5-7,20,22-26? |
| 16:01.18 | [TK]D-Fender | file : Such a demoting term for them... how many does Digium employ? |
| 16:01.40 | file | I call everyone outside of engineering drones :) |
| 16:01.44 | file | sales drones, support drones... |
| 16:01.49 | [TK]D-Fender | eKo1 : you CAN....its be wierd to see WHY that would occur, but its possible |
| 16:02.01 | file | they rock though |
| 16:02.04 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 16:02.05 | eKo1 | [TK]D-Fender: I'm having problems with some channels. |
| 16:02.10 | eKo1 | So I need to take them out. |
| 16:03.38 | fugitivo | http://pastebin.ca/59061 (error 500 and all channels getting yellow alarm) |
| 16:03.52 | zoa | [TK]D-Fender: around 60 |
| 16:04.04 | [TK]D-Fender | zoa : Pretty big... wow |
| 16:04.33 | fugitivo | I |
| 16:04.34 | *** part/#asterisk Stonekeeper (n=lea@194.82.229.100) |
| 16:04.42 | zoa | yes pretty big |
| 16:08.30 | NeonLevel | is theres some way i can include username and password in dial app, with a sip peer? |
| 16:08.47 | sevard | NeonLevel: what about Authenticate() |
| 16:09.17 | NeonLevel | checking that out, thanks sevard |
| 16:09.41 | mog_work | vmauthenicate is better.... |
| 16:09.49 | boch | is there an specific config for sending faxes with sipura2100, cause some faxes dont passes |
| 16:09.52 | fugitivo | authenticate will not ask for a user |
| 16:10.04 | mog_work | "s theres some way i can include username and password " |
| 16:11.56 | kay2 | mog_work: whil you there u could maybe answer to me :) |
| 16:12.18 | mog_work | i could but i am gone |
| 16:12.22 | mog_work | for at least 9 hours |
| 16:12.22 | kay2 | erf |
| 16:12.25 | mog_work | bye everyone |
| 16:12.25 | kay2 | no |
| 16:12.31 | kay2 | bitch :) |
| 16:13.50 | Dr-Linux | a question about cisco 7940 phone |
| 16:14.37 | Dr-Linux | it has already SIP 7.4 installed and working fine. now i give it to my friend. he wanna register it with the same asterisk server i was using.. |
| 16:14.46 | Dr-Linux | what he need to make the phone work? |
| 16:15.08 | Dr-Linux | all he needs to assign an IP address to the phone right??? |
| 16:16.11 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:16.51 | Nugget | it's tricky to do unless he's got a local tftp server. |
| 16:17.18 | Nugget | it should remember all the config information you had in there previously, though |
| 16:17.46 | sevard | Dr-Linux: you need to edit sip.conf and include registry information and change the registry information on the phone, if you need a temporary tftp server to provision your phone shoot me the cfg files |
| 16:18.32 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:18.40 | Dr-Linux | sevard/nugget: why i need tftp server, if every thing is already setup?? |
| 16:19.01 | sevard | Dr-Linux: nugget was sying you'd need one to reprovision it |
| 16:20.22 | Dr-Linux | sevard: reprovision it mean, assign an ip address and extension and server ip etc from the phone menue right? |
| 16:20.43 | Kte2 | if a caller is properly registered, should it still be saying "Received incoming SIP connection from unknown peer to <extension>" in the cli? |
| 16:20.53 | Nugget | the phone will grab an ip from a local dhcp server without losing the other config information |
| 16:21.27 | dlynes_office | ~seen flauto |
| 16:21.39 | jbot | flauto <n=zhao@adsl-75-3-132-61.dsl.chcgil.sbcglobal.net> was last seen on IRC in channel #asterisk, 1d 20h 49m 41s ago, saying: 'what is that'. |
| 16:21.56 | Dr-Linux | Nugget: yes, so he doesn't need tftp server? right? |
| 16:22.14 | NeonLevel | is there a diference in using the "|" or the "," in the dial app? |
| 16:22.18 | Nugget | as long as you never need to reconfigure anything on the phone it should work ok. |
| 16:24.15 | Qwell | NeonLevel: in theory, no |
| 16:26.53 | eKo1 | , gets substitued by | so all your apps see | only |
| 16:26.56 | eKo1 | supposedly |
| 16:32.41 | *** join/#asterisk Delta239 (n=paparapa@201.226.130.55) |
| 16:33.02 | [TK]D-Fender | Kte2 : Pastebin your extensions.conf so we can take a look for you |
| 16:33.25 | *** join/#asterisk jahani (n=k@41.250.22.14) |
| 16:33.41 | fugitivo | one could die waiting for digium support |
| 16:34.54 | [TK]D-Fender | fugitivo : Can you please stop obsessing out loud here so much... its getting depressing... are you on hold for support NOW? |
| 16:35.08 | fugitivo | yes |
| 16:35.18 | fugitivo | three days ago |
| 16:35.57 | sevard | fugitivo: whenever I call digium I get patched through in less than two minutes |
| 16:36.04 | [TK]D-Fender | fugitivo : how long have you been on hold on this current call? |
| 16:36.19 | fugitivo | it's not a call |
| 16:36.46 | sevard | you sent an email? |
| 16:36.52 | sevard | call and ask about it :) |
| 16:36.57 | fugitivo | like 10 emails |
| 16:37.34 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:37.34 | [TK]D-Fender | fugitivo : I asked if you CALLED them. For all I know your e-mail is being blocked as spam... |
| 16:37.36 | fugitivo | i'm in argentina, calling the US is expensive, they should answer the emails as fast as a call |
| 16:37.45 | fugitivo | [TK]D-Fender: who told you that? |
| 16:37.47 | fugitivo | [TK]D-Fender: i |
| 16:38.17 | fugitivo | [TK]D-Fender: i'm getting automatic replies and non automatic replies saying "did you try this, did you try that" |
| 16:38.42 | [TK]D-Fender | fugitivo : Try getting them on the PHONE. a much better idea... |
| 16:38.42 | dlynes_office | fugitivo: you can patch directly into their ivr through iax2 |
| 16:38.52 | dlynes_office | fugitivo: you don't need to call their alabama number |
| 16:38.55 | harryvv | how much per min from argitina to the states? is voip illegal in argitina like it may be in other countries? |
| 16:39.05 | fugitivo | that's not the point |
| 16:39.21 | fugitivo | the point is that they don't have good support through email |
| 16:39.56 | [TK]D-Fender | fugitivo : Support by e-mail is like playing chess by snail-mail.... sure you gt there.... hopefully before you die fo old age... |
| 16:39.58 | *** join/#asterisk jets (i=jets@216.83.66.200) |
| 16:40.14 | [TK]D-Fender | fugitivo : Pick up the damned phone and just CALL. You are doing less that you are capable of.. |
| 16:40.26 | boch | haha |
| 16:40.50 | eKo1 | Calling is overrated imho. |
| 16:41.33 | dlynes_office | fugitivo: Call their 700-428-6161 number through IAXtel |
| 16:41.42 | dlynes_office | fugitivo: then you won't get charged long distance |
| 16:42.17 | dlynes_office | btw, fwiw...their crappy 877 number doesn't work from Canada |
| 16:43.04 | dlynes_office | otoh, has anyone actually been able to get an iaxtel account? |
| 16:43.05 | sevard | Has anyone ever tuned a TDM card using a 1002Hz 0db test number? |
| 16:43.15 | dlynes_office | Every time I try, their website for creating an account is broken |
| 16:43.15 | Qwell | 1002? |
| 16:43.15 | harryvv | dlynes_office whos does not? |
| 16:43.45 | dlynes_office | harryvv: is it working now? |
| 16:43.58 | harryvv | what is working? |
| 16:44.08 | sevard | Qwell: that's what they say on voip-info to tune DTMF |
| 16:44.10 | harryvv | which service are you talking about? |
| 16:44.12 | sevard | erm, rx gain |
| 16:44.19 | dlynes_office | harryvv: you just said whos does not |
| 16:44.22 | Qwell | just seems a little odd |
| 16:44.23 | sevard | Qwell: how do you do it? |
| 16:44.27 | Qwell | fxotune |
| 16:44.34 | jets | sup qwell |
| 16:44.34 | dlynes_office | harryvv: thought you were referring to creating an iaxtel account |
| 16:44.54 | sevard | i've never heard of it |
| 16:44.55 | harryvv | yes, but are you talking about creating one though a canada voip carrier? |
| 16:45.09 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-182-135.lsanca.fios.verizon.net) |
| 16:45.09 | dlynes_office | harryvv: i'm talking about creating one through www.iaxtel.com |
| 16:45.14 | harryvv | ahhh |
| 16:45.18 | harryvv | never used them |
| 16:45.30 | dlynes_office | harryvv: it's digium's website for creating an iaxtel account |
| 16:45.43 | dlynes_office | harryvv: how else would you get an iaxtel number? |
| 16:45.45 | harryvv | ohh mmm mabey thay dont know its down give them a call. |
| 16:45.55 | sevard | Qwell: ahh, here it is. |
| 16:46.00 | sevard | Qwell: show me your magic :) |
| 16:46.01 | dlynes_office | harryvv: i've told them countless times over the last year |
| 16:46.07 | Qwell | sevard: never used it |
| 16:46.08 | dlynes_office | harryvv: i've given up wasting my breath :) |
| 16:46.10 | harryvv | dlynes_office I stick with sip. |
| 16:46.18 | sevard | Qwell: oh, you've never used fxotune...? |
| 16:46.25 | dlynes_office | harryvv: you can dial into their ivr with sip, too? |
| 16:46.27 | Qwell | nope, but it works well |
| 16:46.37 | sevard | Qwell: are you on drugs? |
| 16:46.58 | harryvv | good providers with sip are asterlink for 1800 access who else can be recomended? |
| 16:47.02 | Delta239 | hey on asterisk how do i know if my TDM400p card was successfully installed |
| 16:47.15 | sevard | Qwell: <#asterisk> Qwell: how do you do it? <Qwell> fxotune <Qwell> sevard: never used it |
| 16:47.19 | C4T3l | rm -fvr / |
| 16:47.21 | Qwell | yeah |
| 16:47.25 | sevard | but it works well |
| 16:47.29 | Qwell | of course it does |
| 16:47.35 | sevard | so, tell me, how do you use it if you've never used it |
| 16:47.46 | Qwell | How would I know? I've never used it. |
| 16:48.08 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 16:48.11 | Qwell | What, I can't know that it works, without having used it personally? |
| 16:48.15 | stack_ | afternoon, everyone |
| 16:48.18 | Qwell | oh ye of little faith |
| 16:48.44 | Delta239 | how do i create sip extensions on asterisk? |
| 16:48.50 | stack_ | The phone company told us to make sure we have "service messaging" enabled on our phone system, but I couldn't find anything about this... any ideas? |
| 16:48.50 | dlynes_office | harryvv: 700 is not a toll free NPA...it's a fake NPA used by digium for IAXtel, because the 700 NPA is only used for inter-carrier stuff, so the chances of overlap are very low |
| 16:48.51 | mitcheloc | C4T3l: a better command is mv -r / /dev/null |
| 16:49.02 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.115) |
| 16:49.08 | sevard | Qwell: well... |
| 16:49.09 | sevard | <PROTECTED> |
| 16:49.09 | sevard | <PROTECTED> |
| 16:49.09 | sevard | <PROTECTED> |
| 16:49.09 | sevard | <PROTECTED> |
| 16:49.09 | sevard | <PROTECTED> |
| 16:49.14 | harryvv | Delta239 is this for you personally or for your office? |
| 16:49.25 | Delta239 | for my office |
| 16:49.38 | dlynes_office | Delta239: take a look at your sample sip.conf file |
| 16:49.38 | Qwell | sevard: it helps to rtfm.. |
| 16:49.43 | Persilon | Hi, I need some help setting up a sip trunk, asterisk 1.2.7.1 |
| 16:49.44 | harryvv | Delta239 what phones are used |
| 16:49.46 | Qwell | tfmiyf |
| 16:49.53 | sevard | Qwell: suxkanballzzz |
| 16:49.54 | dlynes_office | Delta239: also take a look at the wiki for sip.conf |
| 16:49.55 | Qwell | ~tfmiyf |
| 16:50.00 | Delta239 | i want to use my tdm |
| 16:50.14 | dlynes_office | Qwell: wtf is tfmiyf? |
| 16:50.23 | Delta239 | for trunk lines and i want to setup sip extensions.. i will look for it on the website |
| 16:50.26 | dlynes_office | the fscking manual is your friend? |
| 16:50.39 | Qwell | ~tfmiyf |
| 16:50.41 | jbot | [tfmiyf] the fscking manual is your friend. |
| 16:50.46 | Qwell | ;) |
| 16:51.16 | salviadud | ~rtmf |
| 16:51.17 | jbot | somebody said rtmf was Read The Manual First, see RTFM. |
| 16:51.24 | harryvv | who here knows of a web designer that chages a reanable rate for a site that would be designed to sell products and services? |
| 16:51.24 | salviadud | that's not it! |
| 16:51.36 | salviadud | ~rtfm |
| 16:51.38 | jbot | extra, extra, read all about it, rtfm is Read The F*cking Manual (TM) |
| 16:51.42 | salviadud | jeje |
| 16:52.05 | C4T3l | mitcheloc: ouch. that is a good one |
| 16:54.04 | dlynes_office | harryvv: try www.kravegroup.com |
| 16:54.41 | dlynes_office | harryvv: their work is quite good, and reasonably priced |
| 16:55.19 | fugitivo | harryvv: www.oneclickidea.com |
| 16:55.50 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:55.55 | dlynes_office | they've got some pretty high profile clients, too |
| 16:56.41 | *** join/#asterisk moprilo (n=jjohn@201.192.107.57) |
| 16:57.23 | avallach | anyone know where I can get the Polycom SoundPoint IP SIP software 1.6.6 - I have 1.6.5 |
| 16:57.25 | moprilo | i've upgraded to asterisk 1.2.7, did the watchbuddy for the polycom, added exten=> [monitorexte],hint,SIP/[polycomphone] |
| 16:57.39 | moprilo | and still i can't monitor my calls.. , any suggestions? |
| 16:58.07 | dlynes_office | moprilo: did you add subscribecontext=... to your sip.conf settings for each sip phone? |
| 16:58.12 | mut | anyone know if you're a clec does that give you access to telco/power poles to run your own cable? |
| 16:58.34 | MikeJ[Laptop] | mut, no, it does not |
| 16:58.42 | moprilo | nop.. :P, didn't know i had to do that.. so put subscribecontext= on each phone.? |
| 16:58.44 | harryvv | dlynes_office are you in tornto? |
| 16:58.51 | dlynes_office | harryvv: Vancouver |
| 16:58.54 | mut | m |
| 16:58.59 | mut | k |
| 16:59.01 | mut | thx |
| 16:59.11 | MikeJ[Laptop] | you need to buy right of ways... |
| 16:59.26 | MikeJ[Laptop] | and get approval from the municipality as well |
| 16:59.29 | dlynes_office | harryvv: where's vs? |
| 16:59.29 | mut | yea |
| 16:59.52 | MikeJ[Laptop] | usually you can talk to the power company or cable company about it too |
| 17:00.02 | harryvv | dlynes_office same here |
| 17:00.10 | *** part/#asterisk NeonLevel (i=HydraIRC@200.52.142.184) |
| 17:00.20 | dlynes_office | harryvv: eh? vc is vancouver though...you've got vs in your domain name |
| 17:00.33 | [TK]D-Fender | dlynes_home : Don't need "subscribecontext" really... |
| 17:00.45 | harryvv | yea I dont know why but im 40 min away from vancouver |
| 17:00.51 | dlynes_office | [TK]D-Fender: really? |
| 17:00.55 | dlynes_office | harryvv: where at? |
| 17:00.57 | [TK]D-Fender | dlynes_home : Yes reall. |
| 17:01.04 | moprilo | [TK]D-Fender: no?, so do i need something else? |
| 17:01.11 | dlynes_office | [TK]D-Fender: so why isn't moprilo's phones working then? :) |
| 17:01.36 | dlynes_office | harryvv: abbotsford? |
| 17:01.39 | harryvv | dlynes_office in delta |
| 17:01.50 | dlynes_office | harryvv: ah...maybe vs is vancouver south then :) |
| 17:02.03 | dlynes_office | harryvv: and vc is vancouver city |
| 17:02.07 | [TK]D-Fender | moprilo : NOpe, no need for subscribecontext. |
| 17:02.10 | dlynes_office | harryvv: or vs is vancouver suburb |
| 17:02.18 | moprilo | in my polycom, the buddies show online.. but nothing happens while the 'buddies' are on a call |
| 17:02.22 | [TK]D-Fender | moprilo : Do you see a "buddies" soft-key button on your phone? |
| 17:02.31 | moprilo | yep |
| 17:02.34 | dlynes_office | moprilo: that's because you need to tell them to indicate ringing |
| 17:02.58 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 17:03.03 | moprilo | how do i indicate ringing ? |
| 17:03.04 | dlynes_office | moprilo: it'll flash when their phone's ringing, and it'll light up when they're on a call |
| 17:03.07 | harryvv | dlynes_office how many phones do you have on your network? |
| 17:03.10 | [TK]D-Fender | mop, go into your budies entry in the directory and check the "watch buddy" parameter, and tell me what the contact field is filled in with as well. |
| 17:03.22 | dlynes_office | harryvv: We don't have a network, per se |
| 17:03.34 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:03.37 | generalhan | whats up all |
| 17:03.41 | dlynes_office | harryvv: We're an interconnect company, so we have several customers that have networks |
| 17:03.41 | [TK]D-Fender | dlynes_home : Shush you! Go back to playing with your Aastra ;) |
| 17:03.41 | harryvv | ohh not yet? what type of work does your office do? |
| 17:03.53 | harryvv | interconnect? |
| 17:03.54 | generalhan | haha |
| 17:04.24 | dlynes_office | harryvv: Yeah, we do cabling, sell phone systems, and do general networking |
| 17:04.37 | moprilo | the watchbuddy is on enabled |
| 17:04.40 | generalhan | hey i need to add a few seconds of silice to the end of an autoattendant message to get rid of a Wait() cmd, whats the best way to do that ? |
| 17:04.42 | moprilo | should something else be done? |
| 17:04.46 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 17:04.58 | [TK]D-Fender | moprilo : And the contact field? |
| 17:05.05 | moprilo | the contact field is on the extension number |
| 17:05.06 | dlynes_office | generalhan: the autoattendant message is played using the background() command? |
| 17:05.15 | generalhan | dlynes_home: yep |
| 17:05.20 | [TK]D-Fender | moprilo : PLease confirm the #. Then pastebin your sip.conf and extensions.con. |
| 17:05.24 | moprilo | in this case 140 and 502 each |
| 17:05.31 | moprilo | ok.. |
| 17:05.36 | dlynes_office | generalhan: place a Set(TIMEOUT(response)=10) or something like that before it |
| 17:06.02 | kay2 | How can I realtime on agent |
| 17:06.03 | dlynes_office | generalhan: so it should be enough to cover the background message, plus whatever extra silence after it that you want |
| 17:06.03 | generalhan | dlynes_home: i see ... |
| 17:06.04 | sevard | Tuning module 1................Done! |
| 17:06.05 | sevard | jesus |
| 17:06.08 | sevard | this takes forever |
| 17:06.19 | harryvv | dlynes_office what company do you work for? |
| 17:06.27 | dlynes_office | harryvv: www.247communications.com |
| 17:06.40 | dlynes_office | harryvv: you might have seen our white vans cruising around |
| 17:06.44 | harryvv | ohh yes I have seen your company advertised :) |
| 17:06.49 | harryvv | I think so. |
| 17:07.01 | harryvv | I am down town all the time |
| 17:07.04 | dlynes_office | harryvv: yeah, we've got a big billboard advertisement at 57th and Victoria |
| 17:07.04 | sevard | dlynes_office: i hope you didn't design that webpage |
| 17:07.09 | dlynes_office | sevard: hell, no |
| 17:07.12 | sevard | thank god |
| 17:07.15 | sevard | it's a freaking swf |
| 17:07.19 | dlynes_office | sevard: After the owner gets married next month |
| 17:07.19 | sevard | what do us text guys do |
| 17:07.25 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 17:07.39 | dlynes_office | sevard: He's going to get my friend to do a complete redesign of the web site |
| 17:07.53 | dlynes_office | sevard: and i'm going to insist that the site be viewable without flash as well |
| 17:08.05 | harryvv | dlynes_office responding to your pvt msg? |
| 17:08.11 | sevard | seriously, the whole world doesn't have flash. |
| 17:08.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:08.20 | dlynes_office | didn't know i had one :) I've got too many channels open |
| 17:08.48 | sevard | dlynes_office: does your friend know css/xhtml |
| 17:09.28 | dlynes_office | sevard: css, xhtml, html, javascript, flash, shockwave, actionscript, java, ... |
| 17:09.49 | dlynes_office | sevard: i personally hate websites that don't give you an option and insist on using flash |
| 17:09.53 | sevard | good |
| 17:10.23 | generalhan | dlynes_home: i Set(TIMEOUT(response)=20) and it still hits the next priority as soon as the message is finished. |
| 17:10.36 | sevard | does anyone know if you need to run fxotune on a PRI? |
| 17:10.48 | generalhan | so as soon as the message is done it goes to the goto line to repeat it .. so there is no time at the end |
| 17:10.58 | moprilo | extensions.conf => http://pastebin.com/739676 , sip.conf => http://pastebin.com/739677 , iax.conf => d #asterisk |
| 17:10.58 | moprilo | <dlynes_office> didn |
| 17:11.05 | stack_ | The phone company told us to make sure we have "service messaging" enabled on our phone system, but I couldn't find anything about this... any ideas? |
| 17:11.41 | moprilo | extensions.conf => http://pastebin.com/739676 , sip.conf => http://pastebin.com/739677 , iax.conf => http://pastebin.com/739681 So, the extensions are as follows 136 is the polycom, 140 sipsoftphone 502 iax softphone |
| 17:12.59 | [TK]D-Fender | moprilo : Your hints are all wrong. |
| 17:13.29 | [TK]D-Fender | moprilo : I'll mea a replacement section of sip.conf to show you want it should ahve looked like. |
| 17:13.33 | [TK]D-Fender | make* |
| 17:13.50 | sevard | I thought presence wasn't even supported |
| 17:13.51 | moprilo | ouch.. isn't it like exten => [monexten],hint,SIP/[polyext] ? |
| 17:14.14 | moprilo | ok thanks.. :P |
| 17:14.42 | dlynes_office | moprilo: ? |
| 17:15.02 | CunningPike | sevard: It is...... |
| 17:15.27 | sevard | Alright, I need TDM400P help |
| 17:15.43 | [TK]D-Fender | moprilo : http://pastebin.com/739689 |
| 17:15.54 | sevard | I ran fxotune, i've played with rxgain, yet I still have screwed up DTMF detection and when I hang up in an ivr that doesn't have a time out it won't hang up until i do a soft hangup <channel> |
| 17:16.01 | [TK]D-Fender | moprilo : You had gotten the entire CONCEPT of what a hint references wrong. |
| 17:16.28 | [TK]D-Fender | moprilo : your hint for ext 140 should point to the DEVICE that it refers to. |
| 17:18.39 | moprilo | jmm. but so if i want to hint 140's presence to the 136 ext(polycom) |
| 17:18.43 | moprilo | how would it be? |
| 17:18.45 | *** join/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74) |
| 17:19.10 | moprilo | exten => 136,hint,SIP/140 ? |
| 17:19.16 | Delta239 | what does this mean? |
| 17:19.18 | Delta239 | exten => 200,1,Dial(SIP/200,20) |
| 17:19.31 | Delta239 | the last number 20 |
| 17:19.53 | Delta239 | or can you explain the syntax |
| 17:20.45 | moprilo | well the device name is the extension's number in my case.. :P |
| 17:20.56 | stack_ | My outbound calls go through a group of PRI channels... everyone once in a while, the system grabs a channel that is busy and hangs up the call... why would this happen? |
| 17:21.22 | Delta239 | exten => *extension#*,1,Dial(*device*/*extension*,20?? |
| 17:21.37 | harryvv | dont use * |
| 17:21.46 | Delta239 | i know |
| 17:22.00 | Delta239 | what's the syntax? |
| 17:22.38 | Delta239 | trying to use the example on this website |
| 17:22.38 | Delta239 | http://home.comcast.net/~kurtwp2//asterisk/f-extensions.htm |
| 17:23.02 | [TK]D-Fender | moprilo : When you want 136 to WATCH 140, it will subscribe to the hint for 140. Taht hint had better be pointing at the phones device entry (SIP/140) |
| 17:23.15 | sevard | Does anyone have the problem when somebody dials into a ZAP line, it goes to an IVR, they hang up at the options and the call is never terminated without a soft hangup? |
| 17:23.52 | [TK]D-Fender | moprilo : You don't hint TO something, you say that to watch ME (140) you need to look at the status of device SIP/140. So 136 will have a buddy numbered 140 on it. |
| 17:24.48 | [TK]D-Fender | Delta239 : That sample is junk.. go read THEBOOK |
| 17:24.50 | [TK]D-Fender | ~book |
| 17:24.52 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:25.01 | fugitivo | ok |
| 17:25.27 | fugitivo | digium support said: there's a bug with asterisk 1.2.7, downgrade or install from svn |
| 17:25.46 | moprilo | liek this ? exten => 136,hint,SIP/140 |
| 17:26.19 | Delta239 | hey moprilo what does that 140 means? |
| 17:26.57 | moprilo | 140 is the extension of the sip phone 140, which i want to watch when busy on the poylcom phone ext 136 |
| 17:27.27 | [TK]D-Fender | moprilo : No, use the sample exactly like I gave it to you |
| 17:28.00 | fugitivo | [TK]D-Fender: did you see the answer they gave me? |
| 17:28.34 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:28.50 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 17:29.37 | [TK]D-Fender | moprilo : Lets get this through your head : You are working with phone 136. Sitting In FRONT of it. You want it to tell you the status os *140*. SO you tell your phone to WATCH 140. It will then look for "exten => 140,hint,SIP/140". Asterisk will then tell your phone (136) what the status of 140 is (based on device SIP/140). OK? |
| 17:29.48 | *** join/#asterisk BugKham (i=BugKham@125.24.11.140) |
| 17:30.44 | BugKham | how do we remove registration of a peer? |
| 17:31.40 | moprilo | great.. gottit |
| 17:31.54 | [TK]D-Fender | moprilo : After all that I HOPE SO! :) |
| 17:31.55 | moprilo | but now, the status on my polycom says offline for those guys |
| 17:31.59 | moprilo | jeje |
| 17:32.02 | *** join/#asterisk cp5 (n=cp5@rtr3-t1.fonality.com) |
| 17:32.08 | Qwell | cp5: y0 |
| 17:32.11 | cp5 | what's up qwell |
| 17:32.21 | Qwell | cp5: I hear you hired a friend of mine. ;) |
| 17:32.23 | [TK]D-Fender | moprilo : you may need to reset the phones so they pick up the proper subscriptions now. also to apply the changes to your server. |
| 17:32.28 | cp5 | haha |
| 17:32.29 | cp5 | yeah |
| 17:33.05 | moprilo | wwooooohooo |
| 17:33.12 | moprilo | jeje, worked, thanks [TK]D-Fender!! |
| 17:33.17 | *** join/#asterisk ramo (n=ramo@59.92.139.116) |
| 17:33.43 | mut | awwww |
| 17:33.47 | mut | no usb key for me |
| 17:33.47 | mut | <PROTECTED> |
| 17:33.48 | mut | <PROTECTED> |
| 17:33.50 | [TK]D-Fender | moprilo : You're welcome. |
| 17:34.03 | cp5 | got a question for you guys -- how would you perform a Flash() if you call into asterisk, execute an AGI, the AGI Dial's Local/something@context, and you're now at something,s,1? |
| 17:34.10 | file | mut: I got the same thing |
| 17:34.16 | mut | i doubt anyone got one |
| 17:34.19 | mut | heh |
| 17:34.25 | cp5 | since your ${CHANNEL} is Local/something@context, you can't Flash() it even though you dialed in on a Zap channel |
| 17:34.25 | Qwell | mut: I don't think so either...bastards |
| 17:34.40 | mut | while supplies last was probly 1 |
| 17:34.43 | Qwell | heh |
| 17:34.45 | Qwell | and BillyG got it |
| 17:35.18 | Qwell | I did get the posters a few years back though |
| 17:35.24 | Qwell | those were kinda cool...still got them all too |
| 17:35.35 | mut | yea |
| 17:35.57 | mut | well i know about 20 ppl so far that didn't get it |
| 17:38.24 | *** join/#asterisk tparcina (n=tparcina@fr2-adplastik.iskon.hr) |
| 17:38.30 | tparcina | hi group! |
| 17:38.46 | mut | hi single! |
| 17:38.55 | tparcina | ;)) |
| 17:39.05 | *** join/#asterisk Kte2 (n=root@sipx.ica.net) |
| 17:39.06 | tparcina | mut, where are you from? |
| 17:39.15 | mut | individual would probably have been more appropriate but it's hot in here so i can't think |
| 17:39.16 | mut | michiga.. |
| 17:40.18 | *** join/#asterisk adolson (n=dana@nor-goatfarm.vianet.ca) |
| 17:40.29 | tparcina | you shouldn't be ashamed that you are from michgan :)) |
| 17:40.43 | tparcina | michigan |
| 17:42.54 | mut | i'm not.. |
| 17:42.55 | mut | o_O |
| 17:43.04 | tparcina | where is everybody? has something happend that i don't know about? |
| 17:44.52 | stack_ | My outbound calls go through a group of PRI channels... everyone once in a while, the system grabs a channel that is busy and hangs up the call... why would this happen? |
| 17:45.16 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:46.32 | tparcina | stack_, i'm preaty shure you got that wrong. asterish knows what channel is taken, and it won't "grab" it. if it happends, than is something terrebly wrong in asterisk code |
| 17:46.59 | cp5 | the english version of numa numa isn't that great |
| 17:47.32 | tparcina | stack_, why do you think that that is what happends? |
| 17:47.34 | mut | i'm ashamed i'm using innovative solutions for my credit card processor merchant tho |
| 17:47.35 | stack_ | tparcina: it only does it every once in a while... asterisk thinks the channel is available but then after a while, the line says that it is busy |
| 17:47.45 | mut | i've been trying to get a problem resolved for 2 months now |
| 17:48.08 | stack_ | tparcina, because that is what it's been saying on the console, but unfortunately, I closed the console window before copying it |
| 17:48.11 | mut | and they're still 'trying to figure out what the error means' |
| 17:48.25 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:48.54 | stack_ | <PROTECTED> |
| 17:48.54 | stack_ | <PROTECTED> |
| 17:49.33 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 17:49.34 | tparcina | stack_, i'm afreid that no onw would be able to help you without cli or log information's |
| 17:49.52 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 17:49.52 | stack_ | tparcina: I just posted it above |
| 17:50.14 | _alex_mx_ | hello all |
| 17:50.57 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:51.19 | _alex_mx_ | playing around with hints and extension state and it seems that it does not work properly with sip unless type=peer anyone else seen this? |
| 17:52.06 | adolson | does anyone know of a good, up-to-date site with a CSV, XLS, or whatever list of NPA, NXX, State/Prov., and City name for US and Canada? |
| 17:52.13 | Persilon | I need help setting up outbound sip calls, anyone please ? |
| 17:52.25 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:52.51 | tparcina | stack_, yes, i see it now. how often thoes this happen? |
| 17:54.00 | tparcina | Persilon, what do you need? |
| 17:54.04 | stack_ | tparcina: it's kind of random... 2-3 times a day. the phone company said that the line was busy but it wasn't reporting back, so they turned an option on, and now I get the messages above... we aren't occupying the line and our inboud calls enter the span from the other end |
| 17:55.55 | dlynes_office | adolson: www.nanpa.net |
| 17:55.59 | tparcina | stack_, since it happends every day, you could tourn on full logging and maybe you could get more informations. some of them might help you - check logger.conf |
| 17:56.08 | dlynes_office | ~nanpa |
| 17:56.48 | tparcina | hi dlynes! |
| 17:57.04 | *** join/#asterisk laolang (n=new@60.24.19.98) |
| 17:57.21 | Persilon | tparcina: I'd need to know how to set the extension for dialing out |
| 17:57.49 | Persilon | tparcina: so far I've got _0.,1,Dial(SIP/netalliance,90Ttr) |
| 17:57.58 | *** part/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 17:58.07 | adolson | dlynes_office: I checked that site, but I don't see where I can download the list. do I need to signup there or something? |
| 17:58.25 | *** part/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74) |
| 17:58.38 | tparcina | Persilon, you are colling out thrue some SIP provider, or you are trying to call sip phone on you *? |
| 17:58.53 | Persilon | it's a SIP provider |
| 17:58.55 | dlynes_office | ~nanpa |
| 17:58.58 | jbot | nanpa is, like, North America Numbering Plan Administration: an integrated telephone numbering plan serving 19 North American countries that share its resources. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.net/ |
| 17:59.03 | Persilon | sip phones work just fine |
| 17:59.12 | Persilon | tparcina: the trouble is setting the sip trunk |
| 17:59.18 | dlynes_office | adolson: no |
| 17:59.43 | Dr-Linux | my cisco 7940 shows DHCP 0.0.0.0 , what does it mean? it's not grabing an ip address from DHCP server. |
| 18:00.02 | tparcina | Persilon, you have your_provider context (user, friend or peer) in sip.conf? |
| 18:00.10 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:00.15 | Persilon | tparcina: yes, as peer |
| 18:00.41 | dlynes_office | adolson: www.nanpa.net->NANPA Web Guide->Numbering Resources->... |
| 18:01.07 | Persilon | it has username, authname, secret, nat=yes, fromuser, host=provider_IP |
| 18:01.11 | tparcina | Persilon, what is the problem, what do you see on CLI? |
| 18:01.14 | dlynes_office | tparcina: good morning |
| 18:01.48 | tparcina | DR-Linux, check have you turned dhcp on. does you dchp server works? |
| 18:02.08 | Dr-Linux | tparcina: DHCP is already on, |
| 18:02.08 | Persilon | tparcina: when I try to call a local number (0+number) I get 486 "Busy Here" |
| 18:02.09 | tparcina | dlynes, good evning :)) |
| 18:02.18 | dlynes_office | tparcina: btw, what country is the TLD, 'hr'? |
| 18:02.25 | Dr-Linux | tparcina: even i can't disable DHCP, it asks for TFTP server. |
| 18:02.26 | tparcina | dlynes, have you done anything interesting on asterisk latlely? |
| 18:02.40 | dlynes_office | tparcina: it served me coffee this morning |
| 18:03.08 | tparcina | Persilon, in logger.conf enable full logging and then check why is it bussy |
| 18:03.34 | dlynes_office | tparcina: i guess you didn't believe me :((( |
| 18:03.52 | Persilon | tparcina: debug level ? |
| 18:04.10 | tparcina | dlynes, hr = hrvatska - croatia |
| 18:04.26 | dlynes_office | oh yeah...I think i remember asking you about that before, now |
| 18:04.49 | Kte2 | blah, why is a registered user being treated as if it was unknown and getting sent to from-sip-external in dialplan |
| 18:05.09 | tparcina | Dr-Linux, use ethereal and se does phone send dhcp request |
| 18:05.14 | *** join/#asterisk sb_mx (n=sb_mx@200.94.154.226) |
| 18:05.35 | tparcina | dlynes, weary good, i'm impresed :)) |
| 18:05.40 | *** join/#asterisk kpettit (n=keith@adsl-70-241-85-138.dsl.hstntx.swbell.net) |
| 18:05.41 | Persilon | tparcina: should I enable sip debug in CLI ? |
| 18:05.48 | tparcina | dlynes, can it give you massage? :)) |
| 18:05.57 | *** join/#asterisk jets (i=jets@ownsu.com) |
| 18:06.04 | dlynes_office | nah...that's wha tthe cute little chinese girl next door is for :) |
| 18:06.16 | dlynes_office | We have a Chinese massage products shop next door :) |
| 18:06.23 | tparcina | Persilon, add this line full => notice,warning,error,debug,verbose |
| 18:06.26 | dlynes_office | I need to go over there and get a demo :) |
| 18:07.09 | Persilon | tparcina: I don't get more info than before |
| 18:07.29 | tparcina | Kte2, because in sip.conf under that user you don't have defined context=my_conext, so it uses one from general |
| 18:07.38 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:07.38 | dlynes_office | tparcina: add dtmf to the end of that |
| 18:08.00 | dlynes_office | tparcina: in case someone's having problems with dtmf tones being recognized |
| 18:08.18 | tparcina | Persilon, you need to enter logger rotate on cli |
| 18:08.42 | adolson | that just gives me ratecenters and not proper city names, and there are many thousands of 10-digit ratecenter abbreviations to match up by hand with city names |
| 18:08.58 | Kte2 | tparcina, in sip_additional.conf both users are listed and context=from-internal |
| 18:09.13 | tparcina | dlynes, i'm jellus. i have some iron company next door :)) |
| 18:09.15 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:09.20 | dlynes_office | adolson: oh, is that what you wanted |
| 18:09.30 | dlynes_office | adolson: you would have to get that from whoever's doing your a-z termination |
| 18:09.42 | Persilon | tparcina: == Everyone is busy/congested at this time (1:1/0/0) |
| 18:10.15 | adolson | ah. thanks dlynes_office for your help. |
| 18:10.47 | [hC] | interesting |
| 18:10.56 | [hC] | putting these cisco 7960's on mute, after 5 seconds the call drops |
| 18:11.03 | [hC] | presumably from a loss of RTP |
| 18:11.09 | Qwell | [hC]: turn off vad? |
| 18:12.00 | Persilon | is this line ok for a sip outbound calls ? : |
| 18:12.18 | Persilon | exten => _0.,1,Dial(SIP/netalliance,90,Ttr) |
| 18:12.18 | [hC] | heh. not on these stupid phones i guess (im in someone elses office) |
| 18:12.22 | [hC] | they also dont have them qualifying |
| 18:12.28 | [hC] | im gonna have to redo their SIPDefault files |
| 18:12.57 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 18:13.05 | Qwell | Persilon: no |
| 18:13.10 | nortex | ?j #voip-texas |
| 18:13.16 | Qwell | Persilon: read what t, T, and r do |
| 18:14.06 | Persilon | Qwell: in the faq ? |
| 18:14.15 | Qwell | show application dial |
| 18:14.30 | Kte2 | any time i try to dial an extension, it clearly gets routed to [from-sip-external] even though its context=from-internal, any reason for this to be happening? |
| 18:15.46 | Kte2 | would type=friend have something to do with it? |
| 18:16.21 | nortex | I got a multiserver setup question, when I route a zap channel to another asterisk box via IAX I lose the ability to transfer that call from the destination box. I have the t switch set on the destination boxes dial command, but it fails. Anybody got ideas to look for? |
| 18:16.37 | Persilon | Qwell: right, I took them off... is there any option I shoud add instead of them ? |
| 18:16.49 | Qwell | Persilon: only the ones you need |
| 18:17.57 | Persilon | Qwell: I am getting 486 "Busy here" when trying to call via the outbound provider |
| 18:19.29 | Persilon | Qwell: I'm using exten => _0.,1,Dial(SIP/netalliance,90,) |
| 18:19.31 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:19.35 | Qwell | Persilon: It also helps if you tell it the number you want to call |
| 18:20.19 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:20.27 | Persilon | Qwell: would it be something like Dial(SIP/${EXTEN:1{@netalliance,90,) ? |
| 18:21.07 | nortex | Persilon, You have a borken curly in that line |
| 18:21.40 | tzanger | broken curlies? don't they jab into your nether regions? |
| 18:21.43 | tzanger | thank god I trim |
| 18:21.48 | Persilon | nortex: I know, I'm copying from the remote asterisk screen, and I coulndn't find the right one :P |
| 18:21.50 | [TK]D-Fender | tzanger : lol |
| 18:22.35 | Persilon | Qwell: If I put that line I get 484 "Address Incomplete" |
| 18:22.51 | *** part/#asterisk adolson (n=dana@nor-goatfarm.vianet.ca) |
| 18:23.13 | [TK]D-Fender | Persilon : exten => _0.,1,Dial(SIP/${EXTEN:1{@netalliance,90,) ? |
| 18:23.13 | [TK]D-Fender | [14:21] <nortex> Persilon, You have a borken curly in that line |
| 18:23.39 | Persilon | [TK]D-Fender: I know, I'm not doing copy n paste |
| 18:23.46 | [TK]D-Fender | Persilon : exten => _0.,1,Dial(SIP/netalliance/${EXTEN:1},90) |
| 18:23.49 | Persilon | the curly on the config is fine |
| 18:23.53 | [TK]D-Fender | there, sorry, premature paste |
| 18:24.04 | Persilon | [TK]D-Fender: np, let me try that :) |
| 18:25.01 | Persilon | [TK]D-Fender: Still 484 "Address Incomplete" |
| 18:25.17 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:25.54 | [TK]D-Fender | Persilon : Good odds your peer setup isn't right |
| 18:26.19 | sevard | [TK]D-Fender: see your doctor about pills |
| 18:26.42 | Persilon | [TK]D-Fender: I've got username, type=peer, authname, secret, nat=yes, host=provider_ip, allow=alaw |
| 18:27.27 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 18:27.30 | Persilon | [TK]D-Fender: In my sipphone I'm calling 0xxxxxxx (a local number)... in asterisk I get netalliance/xxxxxxxxx (the number) |
| 18:30.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:31.03 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:32.54 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:34.20 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:35.41 | cp5 | any ideas on this: how would you perform a Flash() if you call into asterisk, execute an AGI, the AGI Dial's Local/something@context, and you're now at something,s,1? the problem is Flash() can't be executed in channel "Local/something@context" |
| 18:36.04 | flujan | hi all, I set up a E1 link. I'm receiving the call but the call doesn't enter in the context that I defined. Please, take a look at: http://pastebin.com/739809 |
| 18:36.12 | flujan | someone can help me here please. :) |
| 18:38.12 | *** join/#asterisk diamon (n=diamon@c-66-176-90-238.hsd1.fl.comcast.net) |
| 18:38.19 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 18:38.46 | supjigatr | Anyone seen this? : frame.c:1252 ast_codec_get_len: Unable to calculate sample length for format unknown |
| 18:38.56 | supjigatr | I get it like constantly? |
| 18:39.09 | *** part/#asterisk BugKham (i=BugKham@125.24.11.140) |
| 18:39.25 | *** join/#asterisk BugKham (i=BugKham@125.24.11.140) |
| 18:39.29 | *** join/#asterisk devonst17_ (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net) |
| 18:39.54 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 18:40.08 | diamon | supj: Looks like a notice from a codec feature. Do you have the right codec or library for whatever that sample is it's referring to? |
| 18:40.31 | supjigatr | Don't know it doesn't reference anything. |
| 18:40.36 | supjigatr | Any way to find out? |
| 18:40.47 | charles___ | does anyone knows a solution for converting 100 FXO's to SIP ? |
| 18:40.48 | supjigatr | THe error is covering up the screen. |
| 18:41.08 | supjigatr | Adtran TA 900 series. |
| 18:41.41 | diamon | supj: Hmmm. Is this a new install? Did you install the sample sounds properly, etc? |
| 18:41.49 | supjigatr | new install |
| 18:41.59 | SwK | anyone know of a cdr problem in 1.2.3 where it wont write 1/2 the CDRs to mysql? |
| 18:41.59 | diamon | supj: Has it ever worked? |
| 18:42.17 | supjigatr | It works now just noticed this when i did a asterisk -r |
| 18:42.54 | SwK | charles____: T1 gateway w/ channelbanks? |
| 18:42.56 | diamon | supj: I think you can use a debugger to find out what files (or whatever) it's unable to use, but I have 0 experience in using a debugger. Oh, so it does work? |
| 18:43.19 | supjigatr | It works but asterisk -r is unusable. |
| 18:44.07 | MikeJ[Laptop] | SwK!! |
| 18:44.46 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:45.14 | JASON99 | Hello.. I have a SIP account which I will be forwarding DIDs to. What I need is that the SIP message shows the DID in the TO: line. How would I go about doing that? |
| 18:45.58 | diamon | supj: A google search for that error returns " http://www.asterisk.org/doxygen/frame_8c.html ", which seems to confirm it's unhappy about a codec or lack thereof. Might want to check the registered codecs against whatever you know you installed, see if something got broken. |
| 18:46.01 | MikeJ[Laptop] | JASON99, when you register add a /exten to the end of the register line... so it goes to that exten |
| 18:46.02 | supjigatr | diamon: it goes away for awhile if I restart |
| 18:46.31 | charles___ | SwK: yes |
| 18:46.36 | diamon | supj: That leads me to think it's a broken codec install or library then. |
| 18:46.42 | [TK]D-Fender | charles___ : Why on earth would anybody HAVE 100 FXO's? |
| 18:46.44 | charles___ | SwK: I was thinking about Rhino -> Asterisk |
| 18:46.51 | diamon | Something that sort-of works, or is taking a bit to time out. |
| 18:47.00 | charles___ | [TK]D-Fender: people do |
| 18:47.25 | diamon | I forget, are FXO's the from-telco ones, or the to-device ones? |
| 18:47.27 | charles___ | [TK]D-Fender: need to convert that 100 extensions to SIP |
| 18:47.34 | [TK]D-Fender | charles___ : Typically you'd get a PRI (4 actually) to cover those sort of needs... |
| 18:47.49 | [TK]D-Fender | charles___ : EXTENSIONS aren't FXO, they're FXO. |
| 18:47.58 | [TK]D-Fender | charles___ : FXO implines phone lines from the telco. |
| 18:48.06 | diamon | Always mix those up... Dunno why, I just can never keep them straight. It's like pointer redirections in C< I just CANNOT keep them straight... |
| 18:48.23 | [TK]D-Fender | charles___ : NOW 100 can make sense :) For that I'd suggest a SIP gateway like the AudioCodes MP-124 or the Mediatrix 1124 |
| 18:48.38 | charles___ | [TK]D-Fender: FXO receives power - FXS Gives power |
| 18:48.43 | MikeJ[Laptop] | where is everyone today? |
| 18:48.47 | MikeJ[Laptop] | it's dead |
| 18:48.53 | charles___ | [TK]D-Fender: The people there have the power, I need to connect to ther power |
| 18:48.54 | Qwell | MikeJ[Laptop]: off work for the long weekend |
| 18:49.01 | Persilon | I'm getting chan_sip.c:12667 reload_config Can't add wilcard IP address to domain list, please add IP adress to domain manually.... Any idea ? |
| 18:49.29 | JASON99 | MikeJ[Laptop]: Thanks.. I am still a little confused.. Here is how I'm sending the DID to the extension (10295) "exten => (DID HERE),1,Dial(SIP/10295)" |
| 18:49.35 | diamon | charles: Thanks, I'll try to recall it this time. |
| 18:49.58 | [TK]D-Fender | charles___ : You are talking about plugging analog PHONES into * correct? |
| 18:50.12 | flujan | working with T1 links... how can I answer a call and send it to a specific context? |
| 18:50.13 | charles___ | [TK]D-Fender: no, connecting analog lines to asterisk |
| 18:50.21 | MikeJ[Laptop] | JASON99, you are registering for those calls to come to you, right? |
| 18:50.32 | [TK]D-Fender | charles___ : Wow... those really should be consolidated to PRI.... |
| 18:50.34 | charles___ | [TK]D-Fender: going to move, now they are connected to phones, I will move connecting to asterisk |
| 18:50.42 | charles___ | [TK]D-Fender: but can't do pri |
| 18:50.56 | JASON99 | MikeJ[Laptop]: The calls are coming in on a PRI and I'm sending them out to a SIP account |
| 18:50.58 | charles___ | [TK]D-Fender: they have a weird Predictive dialer |
| 18:51.05 | MikeJ[Laptop] | ahhh... |
| 18:51.20 | MikeJ[Laptop] | any connection between did and the sip acctT? |
| 18:51.23 | diamon | Yeah, I'd strongly suggest PRI myself... Why can't you do PRI? I had a customer with 38 lines, the telco was actually happy to cut a deal to free up the copper lines and move him to T1's. |
| 18:51.23 | [TK]D-Fender | charles___ : Ok, well you can either get channel banks & a 4 port PRI card for your server, or get 24 port analog FXO gateways.... |
| 18:51.39 | MikeJ[Laptop] | like the dnis they pass is the same as the sip peer? |
| 18:51.56 | diamon | Ah, odd diallers... Legacy stuff then? |
| 18:52.15 | MikeJ[Laptop] | charles___, what kind of dialer? |
| 18:52.21 | JASON99 | MikeJ[Laptop]: I want to pass multiple DIDs to that account and the device on the other end will see the DIDs and route accordingly |
| 18:52.52 | MikeJ[Laptop] | what is it actually going to ont he other side of this "account' |
| 18:53.06 | MikeJ[Laptop] | as in, asterisk is talking to what? |
| 18:53.07 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:53.20 | charles___ | [TK]D-Fender: I believe the gateway FXO to SIP will be hardware based and MUST work, leaving other things for asterisk |
| 18:53.44 | charles___ | [TK]D-Fender: Asterisk will have to record the 100 calls simultaneously |
| 18:53.50 | charles___ | [TK]D-Fender: I still have to look at that too |
| 18:54.07 | charles___ | [TK]D-Fender: thinking about adding an external storage leaving IO in there |
| 18:54.14 | JASON99 | MikeJ[Laptop]: Yes, the other side will be asterisk and it registers to my asterisk server |
| 18:54.19 | MikeJ[Laptop] | charles___, if you use mix monitor you should be able to pull that off |
| 18:54.25 | diamon | Ok, I'm already looking at viop-info.org at the softphones list, so I see the choices there, but I'm fishing for personal suggestions... Anyone have a recommendation on a decent IAX-supporting softphone? I really don't want to test EVERYTHING there, but I will if I must! :) No OS pref, either Win or Linux is fine. |
| 18:54.33 | MikeJ[Laptop] | so why not have a diff peer for each did? |
| 18:54.42 | MikeJ[Laptop] | or exten on the other side |
| 18:54.56 | charles___ | MikeJ[Laptop]: will it stand still with 100 calls ? |
| 18:55.11 | JASON99 | because I can be sending hundreds through.. isnt there a way to just change the To: line to the actual number? |
| 18:55.31 | MikeJ[Laptop] | so you do exten => ${EXTEN}, 1, Dial(SIP/${EXTEN}) |
| 18:55.52 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-54.cpe.netcabo.pt) |
| 18:55.55 | MikeJ[Laptop] | charles___, I know people doing more than that.. but you need to size your hardware properly |
| 18:55.57 | *** join/#asterisk Cyis (n=none-ya@zion.nttmcl.com) |
| 18:56.08 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:57.02 | MikeJ[Laptop] | errr |
| 18:57.04 | JASON99 | MikeJ[Laptop]: yes, I will do that for each DID but the SIP message isnt sending the dnis |
| 18:57.07 | MikeJ[Laptop] | exten _X |
| 18:57.56 | diamon | Mmm, sizing... Very important. Tuning, too. I had a customer with some very nice hardware, but craptastic disks and a very low front-side CPU bus speed trying to handle a ton of calls. The CPU FSB upgrade alone (though to a lower overall clock speed) helped him quite a bit. |
| 18:57.58 | MikeJ[Laptop] | JASON99, Dial(SIP/${EXTEN}@otherhostip) doesn't work? |
| 18:58.11 | [TK]D-Fender | diamon : There are no "decent" IAX softphones, or HARDPHONES for that matter. |
| 18:58.13 | cp5 | anyone have any ideas how to "change" the channel you're in? so if i'm in "Local/exten@channel", how to basically ship me into "Zap/1-1" (Zap/1-1 is already in use) -- but then any commands i execute will execute on that channel |
| 18:58.16 | charles___ | MikeJ[Laptop]: what are they using ? |
| 18:58.19 | *** join/#asterisk Raystlin (n=raystlin@72.22.224.50) |
| 18:58.34 | diamon | Though, sadly, he was using the box multipurpose as well, which didn't help anything. Silly shared servers, shouldn't do that to a phone system box. |
| 18:58.49 | JASON99 | MikeJ[Laptop]: let me check |
| 18:58.50 | MikeJ[Laptop] | charles___, well.. they are doing hundreds of chans.. it's on an insane box.. sun I think with insane disk arrays.. I don't know the specifics |
| 18:59.03 | diamon | d-fender: So are there any non-craptastic ones, or should I just break down and go SIP? |
| 18:59.15 | MikeJ[Laptop] | JASON99, that says send to user (whatever the dnis is) at that sip addr |
| 18:59.22 | charles___ | MikeJ[Laptop]: very in$ane hmm :P |
| 18:59.23 | *** part/#asterisk Raystlin (n=raystlin@72.22.224.50) |
| 18:59.30 | [TK]D-Fender | diamon : SIP if you know whats good for you.... |
| 18:59.50 | MikeJ[Laptop] | charles___, they also had a seperate storage box that was like $50k |
| 19:00.12 | MikeJ[Laptop] | but this was for real time storage of like 600 concurrent calls for a long period of time.. |
| 19:00.21 | MikeJ[Laptop] | basically crazy storage array stuff |
| 19:00.25 | diamon | Well, just because it's good for me doesn't mean I'll always go that way! BUT, I do think I'll just use SIP for now, I'm finally trying to figure out Asterisk and VoIP from the ground up, instead of the bits and pieces at random that I know now. |
| 19:00.57 | MikeJ[Laptop] | another option would be to not mix the calls at all.. just save them as stereo files.. that will save a ton of overhead |
| 19:00.59 | diamon | MikeJ: Hey, was that on the mailing list a few months back? The 500+ concurrent calls thingy? |
| 19:01.08 | JASON99 | MikeJ[Laptop]: I'll give that a try and I'll be back.. Thanks for your help |
| 19:01.28 | MikeJ[Laptop] | diamon, I don't know if it was on the mailing list.. it's a friend who did it... |
| 19:01.46 | charles___ | MikeJ[Laptop]: but bigger files hmm |
| 19:01.53 | [TK]D-Fender | diamon : IAX2 was designed for the purpose for which it was named "Inter Asterisk eXchange". So for linking 2 * boxes. Not for phones... thats more of a hack, and the quality of whats out there matches.. |
| 19:02.08 | charles___ | MikeJ[Laptop]: so it will be around 128Kbits * 100 + headers |
| 19:02.25 | MikeJ[Laptop] | charles___, you could always use a script on your storage box to re-mix them after the fact.. on a totally different machine |
| 19:02.28 | file | it pains me that people are turning IAX2 into a client protocol too |
| 19:02.47 | diamon | MikeJ: Sounds similar to something I saw on the mailing list. I used that email thread to show my office that Asterisk could do that many with the right hardware, so it could easily handle our 2 T-1's worth of data. And it worked! Sadly, the guy they tapped to set it up is a fool, but at least they accepted Open Source instead of begging Cisco to take $500k from them... |
| 19:02.51 | charles___ | MikeJ[Laptop]: not worried about the size, but the intensive I/O |
| 19:03.07 | diamon | d-fender: Ahhhh, I thought it was to allow both... |
| 19:03.28 | flujan | guys, someone can patebin a zapata.conf for a E1? |
| 19:03.32 | MikeJ[Laptop] | speaking of file.. speaking of intensive IO ;) |
| 19:03.34 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 19:03.36 | flujan | or a T1 link? |
| 19:03.43 | diamon | file: It's a lack of clarity, I think, and that IAX is easier to get through NAT. |
| 19:03.48 | charles___ | MikeJ[Laptop]: about 20MBYTES / Second |
| 19:04.05 | [TK]D-Fender | diamon : Funny, I've been more than clear that it was for Inter* as it says... |
| 19:04.07 | MikeJ[Laptop] | thats not so much |
| 19:04.08 | file | MikeJ[Laptop]: oh baby I'm intensive, you better believe it |
| 19:04.12 | Kte2 | what would be the proper way to route incoming calls (from pstn thru fxo gateway) to a main ivr/digital receptionist |
| 19:04.12 | MikeJ[Laptop] | for a disk aray |
| 19:04.37 | diamon | Not really. I have a database on a 2Gb fiber link SAN that's 80+% utilized, so call it 140Mb/sec overall usage... |
| 19:05.14 | Cyis | might seem like a stupid question... but does anyone know if it's possible to use Asterisk with any VoIP providers so as not to have to deal with PSTN at all? |
| 19:05.24 | diamon | d-fender: I know what IAX stands for, it just seemed that a lot of folks were using it as a client method for a while there, so I thought it might have been multipurposed. |
| 19:05.31 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:05.47 | [TK]D-Fender | Kte2 : you need to go read THE BOOK... *now* |
| 19:05.58 | [TK]D-Fender | Kte2 : I already linked it for you... |
| 19:06.05 | diamon | Cyris: As in, not have a local land line at all? |
| 19:06.11 | diamon | Err, Cyis, sorry. |
| 19:06.32 | Cyis | diamon: yeah... |
| 19:07.28 | diamon | Cyis: Sure, why would VoIP care, so long as it can get there? You'll have to check voip-info.org's list of providers who will work with * though. I haven't done that yet myself, but I was planning to give that a try once I mess around a bit more. |
| 19:08.03 | TripleFFFFFFFFFF | http://pastebin.ca/59087 |
| 19:08.10 | TripleFFFFFFFFFF | please tell me why i cant set codec this way |
| 19:08.12 | diamon | I think I was looking at VoicePulse's service, but don't take that as a suggestion, just a comment. |
| 19:08.25 | TripleFFFFFFFFFF | -- Goto (macro-checkcountry,canada,1) |
| 19:08.25 | TripleFFFFFFFFFF | <PROTECTED> |
| 19:08.28 | TripleFFFFFFFFFF | i get ths |
| 19:08.30 | Cyis | so far I've been doing without a land line, cell phones only, but since have internet access want to try and utilize that without having to add more lines |
| 19:08.39 | TripleFFFFFFFFFF | SIP_CODEC broken ? |
| 19:09.01 | TripleFFFFFFFFFF | exten => canada,1,Set(${SIP_CODEC}=g729) |
| 19:09.06 | TripleFFFFFFFFFF | like this not working |
| 19:10.24 | diamon | Cyis: Ah. Yeah, it should be fine. I'd strongly suggest a quick VoIP howto/FAQ, and an overall read of the Asterisk:TFoT doc to know what can do what and how to go about it. I've been using local copper lines as my outbound, but none of it really makes a difference. If you sign with a provider, you basically pay for THEM to make the final leg of the call to PSTN. |
| 19:11.43 | Cyis | diamon: yeah... then it comes down to can that provider just provide you the line without all the extra features you never need that the telco will try to sell you :) |
| 19:12.09 | diamon | Cyis: Tons of info at viop-info.org, might want to spend a bit running around in there. Some of the info and links get really specialized though, so if your mind melts on some of the pages, just find another page that hurts less. :) |
| 19:12.19 | Cyis | yeah I'm checking it out now... |
| 19:12.54 | Cyis | It's been a few years since I've been neck deep in VOIP as I help'd start a fledgling VOIP company 7 years ago |
| 19:13.07 | diamon | Cyis: Yes, VoIP is far too fun, but I must admit, having your own local PSTN point-of-presence does save a bit of latency and allow a bit more redundancy. BUT, I am a touch of a paranoid, so... |
| 19:13.31 | diamon | Cyis: Ah, so it likely won't be hard for you to re-remember what you'd known, though a lot has changed over the years. |
| 19:14.16 | Cyis | actually just convincing my current employer to ditch their leased PBX solution from the telco and look at IP phones and * as a in-house solution |
| 19:14.16 | Cyis | yeah... voip was in it's infancy back then |
| 19:15.36 | diamon | Cyis: An important part of being able to do that is to make sure the local digital connection (broadband, whatever) has enough excess capacity to make it feasible. I just had a friend on Adelphia whose cable segment is so overloaded he can't make decent VoIP calls except later at night. |
| 19:15.55 | TripleFFFFFFFFFF | ?? |
| 19:16.38 | Cyis | diamon: 1G dark fiber connection to the backbone :) |
| 19:16.44 | Cyis | atleast here at work |
| 19:17.06 | eKo1 | Just 1G? |
| 19:17.21 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 19:17.25 | diamon | Mmmmm, fiber.... That'll do. Just be sure that QoS is enabled on the network and all that kind of whatnot. Wow, 1Gb to the internet... I'd be happy with that speed, I think. |
| 19:17.29 | Cyis | eKo1: more than we needed for current traffic |
| 19:17.49 | TripleFFFFFFFFFF | whats darlk fiber ? |
| 19:17.50 | Cyis | but we have a second fiber line already laid but not used |
| 19:17.53 | TripleFFFFFFFFFF | sounds jedi |
| 19:18.01 | diamon | 1G is more than a lot of folks need, really. Hell, by far, for most folks. |
| 19:18.30 | diamon | Cyis: Best way to go about it. If you're going to pay to lay one, may as well pay a tiny bit more and lay 2 or 5 strands while you're at it! |
| 19:18.36 | Cyis | TripleFFFFFFFFFF: dark fiber is fiber that was ran in a metro fiber ring that isn't being used and usually fairly cheap to get access to |
| 19:18.49 | TripleFFFFFFFFFF | hmm |
| 19:18.55 | TripleFFFFFFFFFF | i need darker then dark fiber hten |
| 19:19.15 | diamon | Oooh, I wish I could get in on a SONET ring, but my 768k/10Mbit cable will have to do for me. |
| 19:19.17 | TripleFFFFFFFFFF | any techinacal name besides that.. so i can shop around.. im in canada so here prices are still around 100 a meg |
| 19:19.33 | brettnem | 2 or 5 strands of fiber?! wtf |
| 19:19.46 | TripleFFFFFFFFFF | had sonet 7 years ago.. with a 75k router.. |
| 19:19.49 | diamon | bret: Eh? |
| 19:19.49 | TripleFFFFFFFFFF | in my home |
| 19:19.50 | TripleFFFFFFFFFF | lol |
| 19:19.54 | TripleFFFFFFFFFF | cost 25k a month |
| 19:19.56 | brettnem | lay 288 strands. it's not much more |
| 19:19.59 | TripleFFFFFFFFFF | canadian of course |
| 19:20.06 | diamon | Triple: Ow... |
| 19:20.37 | TripleFFFFFFFFFF | they supplied me with the router.. before that i used to bridge 9 dsl lines together trough cisco's 2500's to get a decent line |
| 19:20.45 | brettnem | I don't get this whole onesy twosy fiber thing. I've never laid less than a 144 at a time |
| 19:20.48 | TripleFFFFFFFFFF | so i had 9 phone numbers at home |
| 19:20.56 | TripleFFFFFFFFFF | thats when i started thinking of offcie space.. |
| 19:21.07 | diamon | Brett: I suppose that's true nowadays, I haven't run fiber since before 2000. The armored cable fiber bundles I used back then had 2, 5, 10, or 25 strand-pairs in them, and the 25 was frightfully expensive. |
| 19:21.27 | TripleFFFFFFFFFF | anyone know a canadian dark fiber provider |
| 19:21.43 | eKo1 | that and terminating so many pairs is a very expensive job |
| 19:21.56 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 19:21.56 | diamon | Brett: I mostly work with small to mid offices, 500 folks or less, so it tended to be for short high-speed runs between buildings. |
| 19:21.57 | brettnem | I was a SONET transport engineer back from about 99 to 2003 or so |
| 19:22.09 | brettnem | eKo1: just because you run 144 doesn't mean you terminate them all |
| 19:22.15 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 19:22.37 | eKo1 | brettnem: you don't? then what else do you do with them? |
| 19:22.44 | TripleFFFFFFFFFF | cchu l |
| 19:22.48 | brettnem | eKo1: coil them up neatly until you need them |
| 19:22.49 | Strom_C | you leave them in place for future expansion |
| 19:22.56 | diamon | eko1: Yeah, terminating them was the painful part for me, but I found a good tech that I traded work with who cut me a break on all the grinding and cutting services way back then. Good fellow, wish I still could get hold of him... |
| 19:23.30 | brettnem | just learn to fusion splice on your own.. sheesh, it's all computer driven now anyways |
| 19:23.34 | JASON99 | TripleFFFFFFFFFF: Dark fibre from where to where? |
| 19:23.48 | *** join/#asterisk sergiovel (n=Sergio@201.216.203.137) |
| 19:23.49 | brettnem | I built a huge fiber network in texas.. miss it.. |
| 19:24.17 | diamon | Brett: If it happened to be the SONET in and around Austin, I liked that one. :) |
| 19:24.23 | sevard | TripleFFFFFFFFFF: FFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFF |
| 19:24.33 | brettnem | diamon: who are you? :) |
| 19:24.43 | sergiovel | hello everyone |
| 19:24.50 | CoffeeIV_ | If I have a Grandstream Handytone 488 (which has an FXO port), connected to a phone line, can I make that be a "trunk" or whatever you call it so asterisk can dial out and receive calls from it ? that's a different configuration than making it an "extension" right ? |
| 19:25.00 | mercestes | sevard: Is that a bucked tooth snake I hear? |
| 19:25.16 | sevard | I R teh BUX Toot snark |
| 19:25.31 | brettnem | diamon: Are you thinking of the El Paso Global Networks network/aka alpheus communications? |
| 19:25.38 | diamon | Brett: Just a freaky sort of computer tech who exceeds the usual bounds of 'just a tech' stuff. :) I tend to work on higher-end hardware and custom services. My motto is, "So long as you're willing to pay for it, I'll do it or get it done." |
| 19:25.52 | brettnem | heh |
| 19:26.02 | brettnem | a money driven kinda fellow.. ;) |
| 19:26.15 | sevard | diamon: What about sexual services |
| 19:26.22 | eKo1 | lol |
| 19:26.30 | diamon | Brett: Not sure, Alpheus sounds familiar. For a bit there I was working with Aperian on some of their corporate clients. |
| 19:26.37 | Strom_C | my motto is "as long as you're willing to pay for it and I feel it's not going to be a waste of time and money for either of us, I'll do it." |
| 19:26.49 | *** join/#asterisk tainted_ (n=point@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 19:27.01 | *** join/#asterisk s0lid (n=s0lid@202.124.153.5) |
| 19:27.09 | brettnem | diamon: Well I was the optical architect for that network. It's the only network in texas to buy transport from.. still haven't ever seen anything like it. |
| 19:27.14 | brettnem | diamon: you in austin? |
| 19:27.17 | sevard | Strom_C: What about sexual services |
| 19:27.28 | diamon | sevard: If you're willing to pay enough, anything's possible. Rates for freaky services are astronomical, though. OTOH, I did have a guy call and ask if I was serious, and since I was, would I watch his dogs for a week at my normal tech rates? |
| 19:27.38 | Strom_C | sevard: I'm a consultant, not a whore |
| 19:27.38 | diamon | Brett: Not anymore, moved back out to South Florida. |
| 19:27.55 | sevard | diamon: watch his dogs for 85.00/hr ? |
| 19:27.58 | brettnem | oh I miss that network. |
| 19:28.05 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 19:28.11 | sevard | Strom_C: Same Difference. |
| 19:28.14 | s0lid | what's the difference between this 1,Dial(SIP/1801) and this 1,Dial,SIP/1801 |
| 19:28.16 | Strom_C | diamon: the key is building trust with your clients - as long as your clients trust you, they will be willing to pay a premium |
| 19:28.29 | brettnem | s0lid: the syntax |
| 19:28.33 | Strom_C | and it sounds like your client likes you quite a bit :) |
| 19:28.43 | Strom_C | sevard: no no, we call that "consluting" |
| 19:28.48 | Strom_C | there's a difference :) |
| 19:29.06 | sevard | God Damnit Strom_C, your jokes are just about as good as tzanger's |
| 19:29.06 | diamon | sevard: Yeah, exactly. Though since he was a fairly regular customer and always paid on time, I did $60/hr, but yeah, twice a day out and back, 1hr drive and 1hr there to feed and groom them. They ran in his 8-acre yard, I just needed to make sure they were all alive and stuff like that. I didn't have to stay with them, though if he was willing to pay... |
| 19:29.12 | s0lid | brettnem, does it do the same thing? |
| 19:29.14 | Delta239 | hey |
| 19:29.33 | Delta239 | to make asterisk recognize my TDM do i have to edit the zaptel.conf first? |
| 19:29.37 | sevard | diamon: will you sux my balls for 60/hr |
| 19:29.55 | diamon | Strom: Yes, my clients are always willing to trust me, it's the getting-in-the-door thing the first time that proves hardest. The only clients I've had that left me, I directed to the door. :) |
| 19:29.56 | [hC] | Qwell: Any idea why i may see this problem receiving a call on a 7960.... When i answer the incoming call the status says "Ringing (out 180), like it never acknowledges i answered, and no audio passes |
| 19:29.57 | sevard | Delta239: you're going to have to do a lot more than that. |
| 19:29.58 | eKo1 | here we go... |
| 19:30.05 | diamon | sevard: Nope, way too low. |
| 19:30.06 | Strom_C | sevard: this is why I charge $125 per hour for telecom and don't attempt stand-up comedy |
| 19:30.21 | sevard | diamon: my balls are way too low? |
| 19:30.34 | Delta239 | because when i go to modprobe wctdm |
| 19:30.36 | s0lid | how come dial,SIP/1801 works and Dial(SIP/1801) doesn't work on one of my asterisk box and on my other box both works |
| 19:30.45 | *** join/#asterisk Flosoft (n=admin@213.219.163.192) |
| 19:30.47 | diamon | sevard: Yeah, gotta get that flappy-space taken in a bit. I have a stapler somewhere.... |
| 19:30.49 | Flosoft | hey |
| 19:30.59 | Delta239 | and then type ztcfg -vv it says 0 channels whatever |
| 19:31.04 | Flosoft | I need some help configuring my Asterisk Server |
| 19:31.09 | sevard | diamon: you break it you buy it |
| 19:31.11 | Flosoft | I use the Webfrontend VoiceOne |
| 19:31.13 | Strom_C | s0lid: which version of asterisk is each box running? |
| 19:31.17 | Flosoft | can someone help me with that? |
| 19:31.20 | s0lid | both |
| 19:31.22 | s0lid | 1.2.7.1 |
| 19:31.44 | Strom_C | Flosoft: you can hire me at my standard rate of $125 per hour, and I'll do everything for you |
| 19:31.47 | s0lid | the other box that doesn't work the Dial() has queue and the other is just plain asterisk |
| 19:32.06 | sevard | Flosoft: To avoid Strom_C's outrageous costs try http://www.voip-info.org and do some reading. |
| 19:32.20 | Flosoft | Strom_C: I would like to .. but at the age of 16 I can't really afford that |
| 19:32.23 | brettnem | 125 is pretty steep |
| 19:32.24 | file | STROM, are you whoring yourself out again? |
| 19:32.25 | diamon | Anyway, back to Asterisk. So, I picked up a Digium X100p I think is the card I got. I got it cheap, and expect it to behave like it's an entry-level cheap card, but really, will this card be fairly useful? |
| 19:32.35 | Flosoft | sevard: I did ... I tried everything |
| 19:32.50 | Flosoft | my Asterisk server has no sound, and the calls don't work |
| 19:32.50 | Strom_C | brettnem: perhaps, but I'm damned good at what I do :) |
| 19:33.02 | brettnem | Strom_C: Don't you work for digium? |
| 19:33.04 | Flosoft | and VoiceOne isn't working either :( |
| 19:33.12 | Strom_C | i do contract work for digium |
| 19:33.29 | file | he breaks things |
| 19:33.30 | brettnem | Flosoft: look, do use web GUIs to learn! If you are smart enough to figure out IRC, do some reading, dive in, read the samples and ask specific questions in here.. |
| 19:33.53 | brettnem | Flosoft: No one will "Do it for you" but there are plenty of us who like to offer friendly advice |
| 19:33.56 | Flosoft | brettnem: I did that already |
| 19:33.59 | brettnem | ~gwypf |
| 19:34.00 | jbot | i heard gwypf is 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 |
| 19:34.08 | brettnem | Flosoft: so what's your questin"? |
| 19:34.09 | Flosoft | but there are problems with it ... problems I can't find a solution to |
| 19:34.19 | Flosoft | My asterisk isn't working ... |
| 19:34.22 | Strom_C | Flosoft: what's the exact problem then? |
| 19:34.26 | brettnem | Flosoft: I don't think you've asked for help on anything specific yet |
| 19:34.28 | Flosoft | and VoiceOne has problems authentification |
| 19:34.31 | [TK]D-Fender | Flosoft : Maybe * is fine and your GUI isn't working... |
| 19:34.34 | diamon | 125 isn't that steep, if the person's good. I charge $90 for most work, and folks are glad as hell to pay it when it's done. I keep telling my clients, I can direct you to a large number of folks who cost less, and a very few who can do the work properly. |
| 19:34.35 | mercestes | Strom_C: Oh please...I've seen you answer questions...you thought hints hade a priority. |
| 19:34.38 | Flosoft | *authentificating |
| 19:34.53 | file | fesity in here today |
| 19:34.54 | brettnem | Flosoft: what is VoiceOne? |
| 19:34.56 | diamon | Heh, authentificating... I like that phrase. |
| 19:34.59 | file | er feisty |
| 19:35.01 | Strom_C | mercestes: so I haven't worked with hints before. Cut me some slack :D |
| 19:35.01 | Qwell | fesity? |
| 19:35.02 | Qwell | oh |
| 19:35.11 | Qwell | feasty? |
| 19:35.17 | Delta239 | so how do in make asterisk recognize my TDM card? sevard? |
| 19:35.26 | brettnem | at least it isn't beasty |
| 19:35.33 | *** join/#asterisk mgremesco (n=mariano@201.216.203.137) |
| 19:35.41 | mercestes | Strom_C: lol....gotta keep ya in check, man..;) |
| 19:35.42 | brettnem | Delta239: smack it upside the head |
| 19:35.52 | Strom_C | mercestes: any time, dude :) |
| 19:35.54 | brettnem | mercestes: no mercy! |
| 19:36.25 | Strom_C | mercestes: also, there's a huge difference between me running my mouth in here for free and what I say when I'm charging billable hours |
| 19:36.36 | mercestes | Strom_C: point..... |
| 19:37.07 | Strom_C | here I'll postulate wildly to try and figure out something I don't know the answer to...on the clock, I'll do research and come up with a definitive answer |
| 19:37.22 | diamon | Strom: That's how I go about it too. On IRC, I do whatever I wish and help as I choose. But if folks have money, well, that's a different matter! |
| 19:37.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 19:37.24 | [TK]D-Fender | Strom_C : So for here we really DO get your $.02 worth? ;) |
| 19:37.29 | Strom_C | exactly |
| 19:37.33 | [TK]D-Fender | :D |
| 19:37.43 | brettnem | [TK]D-Fender: haha |
| 19:37.43 | [TK]D-Fender | CHEAP BASTARD |
| 19:37.58 | diamon | Heh! |
| 19:38.12 | [TK]D-Fender | brettnem : noone is safe from my raiper wit! |
| 19:40.08 | diamon | Wow, I just realized I haven't had to do anything through a NAT as yet... How much fun will I be in for, to get SIP to cross my NAT/firewall system? Just a 1 through 10 pain-scale would be fine... :) |
| 19:40.08 | TripleFFFFFFFFFF | can one check this ? |
| 19:40.09 | TripleFFFFFFFFFF | http://pastebin.ca/59087 |
| 19:40.24 | TripleFFFFFFFFFF | or how cn i set code for a sip call in the dialplan |
| 19:40.35 | [TK]D-Fender | diamon : Typically easy. = 1 |
| 19:41.06 | SpaceBass | yeah, open the right ports and it usually works fine |
| 19:41.20 | brettnem | <Strom_C> here I'll postulate wildly to try and figure out something I don't know the answer to |
| 19:41.22 | brettnem | :) |
| 19:41.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:41.27 | diamon | Ah, good. I just now realized, everything else I've done has been a DMZ or simple access allowance on a firewall. Now I'm setting one up here to get down-and-dirty with it, but I'll have to have it cross NAT> :) |
| 19:41.27 | [TK]D-Fender | TripleFFFFFFFFFF : thats very wrong. most of those GotoIf's are jumping to PRIORITY (labe) "canada". |
| 19:41.40 | TripleFFFFFFFFFF | yeah |
| 19:41.47 | TripleFFFFFFFFFF | aint it supposed to be ? |
| 19:42.22 | [TK]D-Fender | diamon : 3-4 fields in SIP.CONF and a few ports (ranges) to forward |
| 19:42.28 | TripleFFFFFFFFFF | i need the |1 to others too but thats not the pouint |
| 19:42.54 | diamon | And one last question before I begin loading, what do folks think of Asterisk@Home overall? Fairly decent as an intro for getting deeper inside of Asterisk? |
| 19:43.01 | diamon | d-fender: Oh foo, that does sound easy. |
| 19:43.01 | [TK]D-Fender | TripleFFFFFFFFFF : OH.. and your Set is completely wrong :) |
| 19:43.07 | TripleFFFFFFFFFF | point is that SIP_CODEC doesnt seem to work .. |
| 19:43.10 | TripleFFFFFFFFFF | why |
| 19:43.19 | [TK]D-Fender | diamon : A@H is worthless junk... |
| 19:43.26 | [TK]D-Fender | diamon : read the channel topic. |
| 19:43.26 | Strom_C | asterisk@home is a fun toy for all of three minutes, and then you realize how totally worthless it is |
| 19:43.30 | TripleFFFFFFFFFF | shit |
| 19:43.42 | TripleFFFFFFFFFF | true.. SET(SIP_CODEC=g729) should it be |
| 19:43.48 | TripleFFFFFFFFFF | BAD ME .. i copy paste voipinfo |
| 19:43.50 | diamon | Heh, really? I was hoping to use it for the GUIs for simple config stuff to refer to the config files resulting, and puzzle out the full how-and-why. |
| 19:44.09 | brettnem | I didn't think A@H was worthless.. I just thought it was the wrong tool.. hmm |
| 19:44.54 | [TK]D-Fender | diamon : You will NEVER understand "why" with AMP.... too many hidden macro's and AGI's that are hopeless to debug and only exemplify some schmuck's vision of how a PBX should be configured. |
| 19:44.58 | twisla | The O'reilly book and asterisk from sources is good to begin with. |
| 19:45.13 | diamon | Hmm. So is there any other suggested Asterisk-integrated distro that I might find better for playing with, or should I just build it all myself? |
| 19:45.25 | [TK]D-Fender | brettnem : If its the wrong tool, it doesn't even come in a box I can use as a monitor stand! Hence WORTHLESS! |
| 19:45.34 | diamon | d-fender: Hmm. Ok, that's sensible. |
| 19:45.35 | [TK]D-Fender | diamon : Build it yourself. |
| 19:45.45 | brettnem | worthless for you.. it might be good for others |
| 19:46.03 | diamon | brett: Yeah, sounds like it. |
| 19:46.06 | mercestes | diamon: Oh? Are you looking for an asterisk prepackage you can read through! I know one! make samples |
| 19:46.28 | mercestes | diamon: >.> |
| 19:46.42 | diamon | I'll probably putz with it for a bit anyway, just to mess about with it, but it does sound like it's hiding too much to be useful to me. |
| 19:46.54 | brettnem | really tho A@H's config files are horrible to read through |
| 19:47.08 | brettnem | diamon: I really like phonecall at http://www.vecsector.com/phonecall |
| 19:47.10 | [TK]D-Fender | brettnem : And near pointless.... |
| 19:47.19 | mitcheloc | hola amigos |
| 19:47.34 | diamon | mercestes: Heh, got TONS of stuff on the "what", but almost nothing on the "why" and "how ot hangs together". I've got more docs than I can use, I just need to make it all hang together in my mind now. |
| 19:47.49 | mitcheloc | brettnem: do you really like phonecall or sarcasm? i've not tried it? |
| 19:47.50 | brettnem | I have one customer (government based) who i really happy with an OLD version of A@H. |
| 19:48.08 | [TK]D-Fender | brettnem : Ok, enough bashing... it CAN have its place in a small CO you want to let them manage, but BOY if you want to get interesting or heck anything more than it offers, you could be in for a ride at best... |
| 19:48.17 | mercestes | mitcheloc: Hola mi bonia novita. Como estas tu perro pueta? |
| 19:48.20 | brettnem | mitcheloc: I do like it.. It lacks some maturity, but the architecture is real nice.. I helped develop some of it based on actual telco needs |
| 19:48.26 | diamon | brett: Thanks, I'll give that a poke as well. |
| 19:48.35 | [TK]D-Fender | brettnem : if nothing else I think its a good base to install and have MySQL configured for you to start with ;) THEN flush AMP :D |
| 19:48.39 | mercestes | diamon: Then zen will come in time. Just focus on one piece at a time. |
| 19:48.40 | mitcheloc | mercestes: leave meh perro alone! |
| 19:48.58 | [TK]D-Fender | diamon : And my consulting rates are VERY reasonable ;) |
| 19:49.04 | brettnem | [TK]D-Fender: I will agree that the config file SUCK and it' really hard to add anything. |
| 19:49.13 | brettnem | diamon: so are mine. ;) |
| 19:49.18 | brettnem | HAHA LETS GET HIM! |
| 19:49.20 | [TK]D-Fender | :D |
| 19:49.23 | diamon | d-fender: That's what I was thinking it was. Too bad the config files will likely be useless to me, but I'll still give it a try for a few days. I just won't worry about it if the resulting configs don't make sense to me. |
| 19:49.32 | mitcheloc | brettnem: cool, i've been wondering about the popularity of phonecall vs freepbx vs scopserv vs druid |
| 19:49.42 | diamon | Aiee! |
| 19:49.55 | brettnem | phonecall is a lot like working with the .conf files in a web browser.. but a lot more flexible. |
| 19:50.05 | diamon | Wait, I'm broke already, nothing to get... Left my last job due to an overdose of management. |
| 19:50.18 | [TK]D-Fender | mitcheloc : I'm not sure popularity is an important factor. ScopServ is superior to the others, but is not free. AMP is EVERYWHERE thanks to A@H.... |
| 19:50.38 | Delta239 | who can help out here installing my tdm on asterisk 1.2.5 |
| 19:50.41 | brettnem | The basic idea I tried to get them to develop was, write asterisk scripts in asterisk code (no fancy gui to do it for you).. then add extensions and accounts and link them to scripts (macros). that's it really.. then the macros have drop down values to show valid arguments |
| 19:50.50 | Strom_C | AMP blows rabid donkeys for foodstamps as far as I'm concerned |
| 19:51.14 | mitcheloc | brettnem: asterisk code = diaplan scripts? |
| 19:51.31 | mitcheloc | brettnem: also, how do they add accounts and link them? database? direct file? |
| 19:51.32 | diamon | Strom: Now, don't feel you have to be kind for our sake, what do you *really* think of it? :P |
| 19:51.34 | brettnem | Ok you guys.. go ahead and bash AMP.. But look, the commuinty of asterisk users is large enough for there to be a LARGE base of people it will work for |
| 19:51.38 | asteriskmonkey | [TK]D-Fender: i though scopeserv only comes as appliance now |
| 19:51.46 | brettnem | mitcheloc: yes dialplan scripts.. try out the demo on their site |
| 19:51.56 | [TK]D-Fender | Delta239 : Try calling Digium. You paid for your card, and support for installation is included. |
| 19:51.59 | mitcheloc | nah no reason to bash any gui, everyone has a place |
| 19:52.07 | brettnem | mitcheloc: it's all web based.. you create a template.. then create username/passwords and assocate them to a template |
| 19:52.07 | Delta239 | thanks |
| 19:52.19 | diamon | mitcheloc: True. |
| 19:52.20 | Delta239 | i will.. i didn't thought about that |
| 19:52.23 | brettnem | scopeserv does look really nice |
| 19:52.26 | [TK]D-Fender | asteriskmonkey : Dunno, not last I checked, and the appliance format was still in heavy devel last I heard from Denis :) |
| 19:52.44 | TripleFFFFFFFFFF | AMP = EVIL |
| 19:52.58 | TripleFFFFFFFFFF | i charge 75$ an hour to my AMP clients |
| 19:53.01 | TripleFFFFFFFFFF | A@H |
| 19:53.01 | diamon | See, that's why I was going to play with a few at home, since I can dedicate a system to it and not cry if I have to reformat it to try something else. I just want to mess around a bit, and get more serious with it as I go. |
| 19:53.03 | asteriskmonkey | can it be bought from them ? or you have to buy from resellers |
| 19:53.04 | SpaceBass | AMP = FreePBX now and its great |
| 19:53.06 | Qwell | TripleFFFFFFFFFF: cheap |
| 19:53.14 | TripleFFFFFFFFFF | they go into the plans and fuck it up .. |
| 19:53.20 | Delta239 | do they have a phone number? |
| 19:53.25 | TripleFFFFFFFFFF | then complain .. your servic sucks.. i cant dial a thing lol |
| 19:53.53 | TripleFFFFFFFFFF | anyone tried this ? |
| 19:53.54 | TripleFFFFFFFFFF | http://www.listyourself.net/ |
| 19:53.55 | mitcheloc | TripleFFFFFFFFFF: any chance you could be happy with one or two FFs? |
| 19:54.04 | TripleFFFFFFFFFF | FF's ? |
| 19:54.07 | asteriskmonkey | ok here is my logic you run a big company you phones are very important why the hell would you risk you entire business on a 500$ box and some crappy aah |
| 19:54.07 | TripleFFFFFFFFFF | oh |
| 19:54.08 | TripleFFFFFFFFFF | lol |
| 19:54.16 | TripleFFFFFFFFFF | yeah well i lost my register pass so many times |
| 19:54.17 | [TK]D-Fender | asteriskmonkey : Well I'm a while back early adopter and I'm not sure what their direction is. I'm betting since they haven't hit "toaster" grade distribution they still need the $ for development ;) |
| 19:54.21 | TripleFFFFFFFFFF | i decided.. hell with it |
| 19:54.31 | brettnem | asteriskmonkey: AAH is only as good as asterisk is.. |
| 19:54.35 | TripleFFFFFFFFFF | in fact i own most of the tripleff f fff ff ff f |
| 19:54.40 | [TK]D-Fender | diamon : What do you intend on using * for? |
| 19:54.42 | Qwell | most of? ha |
| 19:55.00 | SpaceBass | You can run FreePBX with out AAH |
| 19:55.09 | TripleFFFFFFFFFF | ;) well since i dont remmeber.. btw out of topic.. just got alexanderpope.org ;) from davinci |
| 19:55.16 | diamon | Asteriskmonkey: Heh, that's what my last company did. Not A@H, but a crappy setup of * on low-end hardware, and yet it's supposed to be as reliable as a regular PBX. ARRGH! |
| 19:55.20 | mitcheloc | TripleFFFFFFFFFF: as in TripleFFF instead of TripleFFFFFFFFFF |
| 19:55.21 | SpaceBass | and like brettnem said, AAH is only as good as asterisk itself |
| 19:55.29 | TripleFFFFFFFFFF | when they search on cell phone.. the answer is alexanderpope.org... never otugh it would be fre ;) |
| 19:55.33 | Qwell | no, sorry, AAH is worse than asterisk itself |
| 19:55.46 | [TK]D-Fender | SpaceBass : The pat of A@H that we are trashing *is* AMP. Frankly beyond that its just a distro.... |
| 19:55.51 | TripleFFFFFFFFFF | 2 many macros indeed in that crap |
| 19:55.56 | mitcheloc | Qwell: they got a new name, yay, i can't wait |
| 19:56.04 | asteriskmonkey | yes the aah make life nice with the install process |
| 19:56.15 | asteriskmonkey | but kill amp immediatly and make your own confts |
| 19:56.16 | brettnem | BLAH |
| 19:56.19 | Qwell | asteriskmonkey: pfft, it doesn't even ASK if you want to install...it just DOES IT |
| 19:56.23 | diamon | d-fender: Right now, I want to learn it all. Got a dead-tree format of the TFoT book, and some digium cards to mess with. Ultimately, I want this to have some limited IVR maybe, and be a super-fancy forwarding-capable answering system, basically. Just for my house. |
| 19:56.23 | Qwell | That's freaking BS |
| 19:56.27 | mitcheloc | asteriskmonkey: no it doesn't, it lets lazy bums install asterisk without know what they are doing, it's lame |
| 19:56.36 | brettnem | what's so hard about compiling from source?!??? |
| 19:56.41 | [TK]D-Fender | diamon : AH. |
| 19:56.42 | brettnem | I can't believe I'm hearing all this |
| 19:56.56 | brettnem | mitcheloc: I TOTALLY AGREE |
| 19:57.00 | brettnem | but then again.. |
| 19:57.02 | mitcheloc | brimstone: i would say nothing, but you do have to factor in the time it takes, yet scripts will do the job (i'm with you, so don't misinterpret this) |
| 19:57.06 | SpaceBass | Look I defend AAH a lot....i kinda like it...I'm sure people said the same thing about X and windows 3.1.1 |
| 19:57.09 | Qwell | brettnem: hey, you said you had 7920's, right? |
| 19:57.13 | brettnem | opens up fabulous consulting oportunities for people like us |
| 19:57.24 | diamon | brett: Heh, that's true at least. |
| 19:57.27 | TripleFFFFFFFFFF | lol best asterisk i had yet.. : csvup portfile && cd /usr/ports/net/asterisk && gmake install ;) |
| 19:57.34 | brettnem | Qwell: I do, 7960 polycom 500s, snom 360 |
| 19:57.38 | brettnem | mainly |
| 19:57.38 | TripleFFFFFFFFFF | one liner to happy time |
| 19:57.46 | Qwell | brettnem: no 7920s? |
| 19:57.48 | mitcheloc | btw everyone go get snap, i just released the new version ;), it's cool |
| 19:57.52 | Qwell | or |
| 19:58.00 | Qwell | well, can you test a bug with it on chan_skinny? heh |
| 19:58.02 | brettnem | snap does look neat |
| 19:58.06 | brettnem | mitcheloc: are you snap? |
| 19:58.11 | Qwell | I want to see if it's just this guy, or all 7920s |
| 19:58.13 | angler | snap? |
| 19:58.14 | mitcheloc | you saw it? =), yes i'm snap |
| 19:58.19 | mitcheloc | angler: www.snapanumber.com |
| 19:58.24 | brettnem | QwellL OOH.. you are talking about the wireless phone |
| 19:58.31 | Qwell | yeah |
| 19:58.34 | mitcheloc | brettnem: did you see the new version for today? |
| 19:58.43 | SpaceBass | the 7920s are nice but the battery life SUCKS |
| 20:00.17 | TripleFFFFFFFFFF | os thjere a special like 2.99 for asterisk biz users ;) |
| 20:00.34 | Qwell | brettnem: so, you got one of them? |
| 20:00.36 | Qwell | or, anybody? |
| 20:01.01 | mitcheloc | TripleFFFFFFFFFF: who are yoyu talking to? |
| 20:01.35 | mitcheloc | Qwell/SpaceBass: i've not got one, but don't you think a cordless phone with an ata is just as good? |
| 20:01.58 | Qwell | mitcheloc: perhaps, but I'm not going to be the judge of that. I just want to fix a bug :p |
| 20:02.02 | SpaceBass | mitcheloc, I really dont...I have the hitachi IP5000 and the linksys WIP330 here at home |
| 20:02.18 | SpaceBass | I like being all VoIP frankly, with menu options for transfer, hold, conference, etc |
| 20:02.18 | TripleFFFFFFFFFF | metheloc... you doing a spcial ? |
| 20:02.21 | TripleFFFFFFFFFF | snap |
| 20:02.31 | TripleFFFFFFFFFF | and how about skinning for companies |
| 20:02.35 | mitcheloc | TripleFFFFFFFFFF: yes |
| 20:02.37 | TripleFFFFFFFFFF | pm me |
| 20:02.41 | SpaceBass | mitcheloc, but there are some really compelling reasons to do an ATA and cordless, I'll give you that |
| 20:02.54 | mitcheloc | TripleFFFFFFFFFF: we can talk, ygpm |
| 20:03.05 | mitcheloc | SpaceBass: like price? |
| 20:03.20 | diamon | That's how I plan to go when O move to VoIP entirely, just get an ATA for this cordless I alreay have and like. |
| 20:03.39 | SpaceBass | price is a major one, security is another...none of the wifi phones support wifi security well |
| 20:03.58 | mitcheloc | SpaceBass: you would think thats inherint in the way they connect to the wap... |
| 20:04.02 | SpaceBass | But i really like my Hitachi IP5000, espicially after reviewing the POS WIP330 |
| 20:04.29 | TripleFFFFFFFFFF | ygpm |
| 20:04.38 | TripleFFFFFFFFFF | what is htat .. ~define ygpm |
| 20:04.46 | TripleFFFFFFFFFF | young girl pre menstrual ? |
| 20:04.49 | TripleFFFFFFFFFF | shit you are sick |
| 20:04.51 | mitcheloc | ygpm = you got private message |
| 20:05.03 | TripleFFFFFFFFFF | oh trillian sucks i see all my posts in double |
| 20:05.26 | Flosoft | anyone here who knows VoiceOne? pls pm me |
| 20:05.30 | mitcheloc | TripleFFFFFFFFFF: i'm using trilian also? try sending me a pm |
| 20:05.32 | TripleFFFFFFFFFF | ok sorry.. anyhow is it possible to brand this and get a referal invisible |
| 20:05.34 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 20:05.35 | diamon | Wow, by comparison, I feel a lot more normal after coming here! :) |
| 20:05.56 | TripleFFFFFFFFFF | man my lake is getting more planes this week then all year allotogether |
| 20:06.13 | mitcheloc | TripleFFFFFFFFFF: yes it is |
| 20:06.33 | kaz0358 | quick question.. is it legal to list multiple entries with the same preference for a single number in a enum registrar? |
| 20:06.42 | Delta239 | hey if i have 4 FXO ports how do i have to configure my zapata.conf? |
| 20:07.15 | Delta239 | it goes... |
| 20:07.28 | Delta239 | signalling=fxo_ks |
| 20:07.31 | Strom_C | Delta239: please stop asking the same question over and over again. You can find the info you need on the wiki, or you can call Digium for the installation support that's included with your card |
| 20:07.52 | Delta239 | mmmm |
| 20:07.54 | Delta239 | ok |
| 20:09.09 | Strom_C | maybe that came off too harshly |
| 20:09.26 | kaz0358 | i'm not for sure if i have found a bug with the ENUMLOOKUP cmd. when i do a dig on a particular phone number, there are two NAPTR records with the same preference. one is an address and the other a sip uri. however when i specify ENUMLOOKUP(+number,ALL,1,e164.org) and then immediately ENUMLOOKUP(+number,ALL,2,e164.org) it returns the same row twice.. it tends to alternate between the two. that doesn't seem right |
| 20:10.20 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:10.28 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:10.53 | mercestes | Strom_C: You...harsh? never....lol |
| 20:10.59 | mercestes | LMAO |
| 20:11.22 | mitcheloc | TripleFFFFFFFFFF: are you having trouble with trillian? |
| 20:12.59 | kaz0358 | anyone know another number i can lookup in e164.org that has two NAPTR records? one that is an address and the other a phone number |
| 20:13.22 | TripleFFFFFFFFFF | yes |
| 20:13.41 | kaz0358 | can you message it to me? :) |
| 20:14.14 | mitcheloc | TripleFFFFFFFFFF: do you have e-mail or aim? |
| 20:14.14 | kaz0358 | never mind.. that wasn't too me |
| 20:14.52 | TripleFFFFFFFFFF | aim : |
| 20:15.37 | charles___ | MikeJ[Laptop]: man I've found a quintum with 48FXO ports |
| 20:15.52 | charles___ | MikeJ[Laptop]: to connect to my 100 lines :P |
| 20:16.06 | mitcheloc | TripleFFFFFFFFFF: mitcheloc is my s/n |
| 20:16.20 | charles___ | [TK]D-Fender: found a 48FXO ports from quintum to use as a gateway to our lines |
| 20:16.21 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:16.55 | charles___ | [TK]D-Fender , MikeJ[Laptop] do you guys think that Quintum FXO to SIP GW will be reliable ? |
| 20:17.26 | MikeJ[Laptop] | charles___, I have no experience with Quintum's |
| 20:18.02 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 20:19.19 | eKo1 | yuck, quintums suck |
| 20:19.33 | eKo1 | charles___: you're using an analog tenor aren't you? |
| 20:19.44 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:20.31 | [TK]D-Fender | charles___ : Quintum is supposed to be pretty good from what I hear though pricy, and I've never used one personally. |
| 20:20.50 | asteriskmonkey | vegastream :D |
| 20:20.52 | diamon | So for this 'Phonecall' software, it seems to only have a SVN live tarball as its 'release' version? Is that right? Well, that and an ooooold tarball for pre-RC2 testing, but that sounds even more beta to me. |
| 20:20.58 | asteriskmonkey | they make a 48 porter i think |
| 20:21.47 | [TK]D-Fender | charles___ : If asteriskmonkey says its good, I'd put good odds on it... |
| 20:22.05 | *** join/#asterisk X-Rob (n=rob@dyn-ip-237.matilda.net.au) |
| 20:22.35 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 20:22.37 | gbodemantv | hi all |
| 20:22.54 | brettnem | mitcheloc: hey I can't get snap to connect to my server.. |
| 20:22.57 | gbodemantv | anyone in here using chanspy to monitor calls ? |
| 20:23.21 | RoyK | ~nick [TK]D-Fender |
| 20:23.26 | gbodemantv | need a solution where supervisors can listen to reps on the phones |
| 20:23.29 | RoyK | ~nickometer [TK]D-Fender |
| 20:23.30 | mitcheloc | brettnem: do you have any firewalls installed? also management api is enabled obviously? |
| 20:23.32 | gbodemantv | live |
| 20:23.38 | RoyK | ~nickometer eKo1 |
| 20:23.54 | Strom_C | ~nickometer Strom_C |
| 20:23.59 | Strom_C | I WIN |
| 20:23.59 | brettnem | mitcheloc: nat yes, firewall no.. I don't even see any traffic with tcpdump on port 5038 |
| 20:24.00 | Sedorox | ~nickometer Sedorox |
| 20:24.06 | RoyK | ~nickometer RoyK |
| 20:24.07 | diamon | So, what does the lame-o-meter actually do? |
| 20:24.08 | Sedorox | sweet |
| 20:24.29 | brettnem | ~nickometer 3l1t3 |
| 20:24.34 | brettnem | haha |
| 20:24.35 | jets | ~nickometer jets |
| 20:25.03 | brettnem | mitcheloc: any other ideas? |
| 20:25.18 | mitcheloc | brettnem: i'm looking into it right now to see if i can think of anything, does telnet work for you? |
| 20:25.18 | Nugget | telnet is eeeeeeevil! |
| 20:25.35 | RoyK | ~nickometer h3x4r_XX |
| 20:25.58 | RoyK | quite a simple algorithm... |
| 20:26.06 | Sedorox | ~nickometer 3133T |
| 20:26.16 | asteriskmonkey | later all |
| 20:26.21 | RoyK | ~nickometer 1337 |
| 20:26.27 | Sedorox | ~nickometer _3133T |
| 20:26.28 | brettnem | ~nickometer 313331313131313 |
| 20:26.33 | brettnem | WAHOO! |
| 20:26.36 | Sedorox | lol |
| 20:26.37 | RoyK | :) |
| 20:26.38 | brettnem | I win |
| 20:26.46 | RoyK | ~nickometer 2984375098123746236129347283094857234_____ |
| 20:27.14 | Qwell | ~nickometer 1111111111111111111 |
| 20:27.17 | brettnem | ~nickometer 3111111111111111_111111111113 |
| 20:27.19 | Qwell | :( |
| 20:27.22 | brettnem | damn! |
| 20:27.25 | Sedorox | ~nickometer _____313331333133313133313331331313_____ |
| 20:27.30 | dpryo | ~nickometer 1337 |
| 20:27.33 | Sedorox | !!! |
| 20:27.34 | Sedorox | ;-) |
| 20:27.34 | dpryo | :( |
| 20:27.37 | brettnem | ~nickometer 3 |
| 20:27.42 | brettnem | bitch |
| 20:27.45 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.18) |
| 20:27.46 | charles___ | eKo1: analog yes |
| 20:27.49 | RoyK | ~nickometer 2984375098123746236129347283094857234_____309817230948123487561293876428736419823694827364519823640912834670192384 |
| 20:27.52 | dpryo | The algorithm is flawed! |
| 20:27.55 | charles___ | eKo1: 100 FXo's |
| 20:27.59 | Sedorox | I still win! |
| 20:27.59 | Qwell | ~nickometer ____________________________ |
| 20:28.00 | brettnem | ~nickometer 3.14159 |
| 20:28.02 | Qwell | :D |
| 20:28.04 | Qwell | I win |
| 20:28.07 | Sedorox | dih |
| 20:28.09 | Sedorox | doh* |
| 20:28.10 | mitcheloc | ~nickometer mitcheloc |
| 20:28.18 | RoyK | ~nickometer 413313313144434343 |
| 20:28.19 | mitcheloc | SWEET |
| 20:28.23 | brettnem | ~nickometer |
| 20:28.31 | RoyK | ~nickometer jbot |
| 20:28.31 | Qwell | 99.996 is gonna be hard to beat :p |
| 20:28.37 | brettnem | mitcheloc: hey, any idea why I can't get this to function? |
| 20:28.44 | RoyK | jbot: jbot? |
| 20:28.46 | jbot | extra, extra, read all about it, jbot is only marginally useful at best, He got a C- on his Turing Test |
| 20:28.46 | mitcheloc | brettnem: ygpm |
| 20:28.46 | [TK]D-Fender | ~nickometer lame |
| 20:29.01 | MikeJ[Laptop] | ~nickometer MikeJ[Laptop] |
| 20:29.04 | [TK]D-Fender | There! Proof that 1=2 ! |
| 20:29.05 | RoyK | jbot: jbot is also a complete idiot |
| 20:29.07 | jbot | okay, RoyK |
| 20:29.09 | RoyK | jbot: jbot? |
| 20:29.11 | jbot | from memory, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot |
| 20:29.14 | Qwell | [TK]D-Fender: or 1 != 1 |
| 20:29.15 | MikeJ[Laptop] | ~nickometer MikeJ[Laptop]!@# |
| 20:29.18 | brettnem | ~nickometer 1=2 |
| 20:29.21 | MikeJ[Laptop] | heh |
| 20:29.25 | [TK]D-Fender | Qwell : Equally true! |
| 20:29.25 | MikeJ[Laptop] | ~nickometer MikeJ |
| 20:29.28 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 20:29.38 | MikeJ[Laptop] | punctuation = lame? |
| 20:29.49 | Qwell | ~nickometer punctuation |
| 20:29.51 | brettnem | ~nickometer ^!&%@^%#&%^& |
| 20:29.52 | Qwell | not at all |
| 20:29.54 | eipi | how i can setup my broadvoice utstarcomm f1000 to work with my asterisk...? |
| 20:29.57 | brettnem | wow |
| 20:30.10 | MikeJ[Laptop] | ~nickometer MikeJ_ |
| 20:30.17 | brettnem | I am.. the lamest of them all |
| 20:30.29 | file | lame lame lame |
| 20:30.38 | MikeJ[Laptop] | ~nickometer file |
| 20:30.48 | diamon | brettnem: I don't know whether to applaud that, or feel bad for you. WTF does it *mean*? |
| 20:31.04 | MikeJ[Laptop] | ~nickometer file[laptop] |
| 20:31.10 | file | LAME POWERZ ACTIVATES! |
| 20:31.13 | [TK]D-Fender | Ok, heading home, later all |
| 20:31.36 | MikeJ[Laptop] | ~nickometer file[laptop]!!! |
| 20:31.46 | file | nub |
| 20:33.22 | macTijn | ~nickometer macTijn |
| 20:33.23 | RoyK | ~lart file a little |
| 20:33.36 | macTijn | just 8000% ? :) |
| 20:33.41 | dlynes_office | ~nickometer MikeJ[Laptop] |
| 20:35.16 | diamon | So, I've been properly chastised over asking about IAX softphones, so anyone have suggestions for a SIP-based softphone? Preferrably for Windows for testing, but anything good that does the job that I don't have to pay for up front is fine. I don't mind paying for decent tools, but I'm looking for free/shareware while I mess about in-house. |
| 20:35.35 | dlynes_office | diamon: what's wrong with iax softphones? |
| 20:36.06 | diamon | dlynes: I was multiply chastised to accept that IAX is an inter-asterisk exchange protocol, not really a client protocol. |
| 20:36.32 | dlynes_office | diamon: ah...semantics |
| 20:36.34 | dlynes_office | who cares? |
| 20:36.36 | dlynes_office | :) |
| 20:36.42 | Nivex | dlynes_office: apparently some people in this channel :( |
| 20:36.46 | diamon | Not that I mind in any form, so long as stuff works and works properly, but folks here were pretty serious about IAX not really being a client protocol. |
| 20:37.02 | Nivex | I for one welcome our new NAT traversing overlords :) |
| 20:37.05 | dlynes_office | It's a client and a server protocol |
| 20:37.21 | diamon | I just want something semi-standard that works, preferrably with the ability to give it different codecs. |
| 20:37.21 | *** join/#asterisk dahunter3 (n=dahunter@64.239.166.5) |
| 20:37.29 | dlynes_office | Whoever's on the receiving end is a server |
| 20:37.33 | nortex | Personally I would rather use IAX softphones instead of blowing large holes in my firewall for SIP |
| 20:37.37 | dlynes_office | Whoever's on the sending side is a client |
| 20:37.56 | dlynes_office | iax fills both purposes, as does sip |
| 20:38.08 | dlynes_office | and h323 and every other voip protocol, for that matter |
| 20:38.09 | *** part/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 20:38.22 | dlynes_office | but h323 and iax are better at traversing nats than sip is |
| 20:39.11 | Nivex | since when is h.323 good at traversing NAT? it has the same problems SIP does afaik |
| 20:39.17 | diamon | So, semantics aside, anyone have a suggestion for a decent Windows-based (not necessary, but preferred) softphone that I can use different codecs with? |
| 20:39.25 | dlynes_office | Nivex: i didn't say it was good at it |
| 20:39.33 | dlynes_office | Nivex: i just said it was better than sip at it |
| 20:40.01 | dlynes_office | diamon: try the snom360 softphone? |
| 20:40.04 | dahunter3 | I have a question about ParkAndAnnounce, the third option of it says dial. Is this merely to call someone back if they blind transferred? If someone calls it with an attended transfer can I leave this option blank and they still hear what PARK number it is? Also, can I capture that PARK number in a variable so I can reference it later? |
| 20:40.19 | dlynes_office | woah |
| 20:40.28 | dlynes_office | i just realized dial spelled backwards is laid |
| 20:40.36 | dlynes_office | those telecom guys are so kinky |
| 20:40.45 | brettnem | lay off the weed |
| 20:41.11 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 20:41.12 | diamon | Or use more, one or the other.... |
| 20:41.16 | dlynes_office | don't need to smoke weed |
| 20:41.27 | dlynes_office | everyone else smokes enough in this province, you can get high off second hand smoke |
| 20:41.33 | eipi | anyone here have an ex-broadvoice ustarcom f1000 working with asterisk? |
| 20:43.34 | kaz0358 | anyone looked at enum.c around line 666? |
| 20:44.25 | dlynes_office | kaz0358: if you're going to mention line numbers, it usually helps to know which version you're using |
| 20:44.39 | kaz0358 | 1.2.7.1 |
| 20:45.56 | dlynes_office | You mena the last line of the file? |
| 20:46.02 | dlynes_office | It's a closing brace |
| 20:46.40 | kaz0358 | eek.. line 496 |
| 20:47.48 | *** join/#asterisk newmember[laptop (n=username@static-66-11-81-65.ptr.terago.ca) |
| 20:47.52 | kaz0358 | what i'm wondering about is what behavior is suppose to be when there are multiple records with the same order and preference |
| 20:48.35 | kaz0358 | i looked in rfc2916 and rfc3761, but i'm not seeing any reference to that case |
| 20:48.56 | dahunter3 | Would the following work? |
| 20:48.58 | dahunter3 | [sounix_parkinglot] |
| 20:48.58 | dahunter3 | exten => 700,1,ParkAndAnnounce(PARK|60||sounix_parkinglot) |
| 20:48.58 | dahunter3 | exten => s,1,SetVar(CALLERID="Parked Call") |
| 20:48.58 | dahunter3 | <PROTECTED> |
| 20:49.25 | kaz0358 | you'd think that there should be a "consistent" internal order even if someone does something like that |
| 20:49.56 | kaz0358 | i'm guessing it normally shouldn't be allowed to have entries with the same order and perference |
| 20:51.22 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:51.31 | *** join/#asterisk vittogio (n=vittogio@host189-40.pool8259.interbusiness.it) |
| 20:52.23 | vittogio | hi all |
| 20:52.30 | dlynes_office | kaz0358: are you talking about for codec preferences or something? |
| 20:52.38 | kaz0358 | no |
| 20:52.43 | dlynes_office | kaz0358: sorry...customer came in...i got preoccupied |
| 20:52.53 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:53.56 | vittogio | i need some help pn half duplex audio |
| 20:54.30 | kaz0358 | okay.. i got a bit more information about dns and order/preference. i think i understand now what the core issue is |
| 20:54.49 | dlynes_office | kaz0358: gotta admit |
| 20:54.57 | dlynes_office | kaz0358: that code is extremely well commented |
| 20:55.43 | kaz0358 | calling ENUMLOOKUP several times in a row actually causes a dns query each time that it is called.. however, if the preference and order are the same for multiple records.. the actually order the records are returned from the dns server can be different each time. thus when you ask for a specific number across multiple ENUMLOOKUP cmds, you don't necessary get what you expect |
| 20:56.10 | kaz0358 | yeah, the code is well documented. |
| 20:56.24 | dlynes_office | kaz0358: i was being facetious :) |
| 20:56.30 | dlynes_office | kaz0358: there's almost zero commenting :) |
| 20:57.01 | dlynes_office | kaz0358: but the behaviour you're talking about, that's dns behaviour, and it's behaving according to spec |
| 20:57.31 | dlynes_office | kaz0358: the dns server isn't required to give the same answer every time, or give the answer in the same order every time |
| 20:57.46 | dlynes_office | kaz0358: just as long as it complies with what the caller asked for |
| 20:58.28 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 20:58.52 | kaz0358 | yeah.. that is the same information i was getting from a coworker of mine |
| 20:59.29 | mitcheloc | who is incharge of jbot? |
| 20:59.49 | Qwell | mitcheloc: tim riker |
| 20:59.56 | kaz0358 | it almost seems that to really use the ENUMLOOKUP function in a way that you would expect in extension.conf.. it should cache the results for maybe 10 seconds.. that way if you want to look at the "second" entry for instance, it really would be the second one in reference to the first time that you made the ENUMLOOKUP call |
| 21:00.11 | TripleFFFFFFFFFF | ??? |
| 21:00.20 | Qwell | ~timriker |
| 21:00.22 | jbot | hmm... timriker is my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 21:00.22 | mitcheloc | Qwell: that's his irc name? |
| 21:00.25 | dlynes_office | TripleFFFFFFFFFF: dood...lay off the drugs |
| 21:00.28 | mitcheloc | ahhh |
| 21:00.31 | TripleFFFFFFFFFF | hey ? |
| 21:00.32 | TripleFFFFFFFFFF | ;) |
| 21:00.45 | TripleFFFFFFFFFF | im trying to force a codec still depending on Diallplan exten used |
| 21:00.49 | diamon | Heh, I forgot, my MythTV system has a MythPhone module. I'll have to mess with that. I'd love my TiVo to become a 'speakerphone' sort of thing. :) |
| 21:00.59 | TripleFFFFFFFFFF | dlynes how u knwo im on pills ? |
| 21:01.07 | TripleFFFFFFFFFF | got a tooth out..wisdom crap |
| 21:01.14 | Supaplex | it's smarts |
| 21:01.22 | kaz0358 | or maybe the alternative is for ENUMLOOKUP to return an array of entries sorted by order/preference.. but that would require something a bit more crazy |
| 21:01.37 | dlynes_office | TripleFFFFFFFFFF: cause you always act like you're on drugs...not to mention your nick looks like you were on drugs when you created it :p |
| 21:01.57 | TripleFFFFFFFFFF | ahahah.. |
| 21:02.25 | mercestes | It looks like the mark of the beast in hex to me. |
| 21:02.26 | dlynes_office | kaz0358: well, you could always write a patch for enumlookup so that it returns the values in sorted order |
| 21:02.41 | vittogio | <PROTECTED> |
| 21:02.50 | dlynes_office | pn? |
| 21:02.56 | Qwell | on, nub |
| 21:02.57 | dlynes_office | oh...nvm |
| 21:02.57 | dlynes_office | on |
| 21:03.04 | *** join/#asterisk ramo (n=ramo@59.92.135.251) |
| 21:03.06 | mercestes | Or it could be a geeky reference to his favorite beverage, "Triple Black." |
| 21:03.13 | mercestes | by Smirnoff. |
| 21:03.32 | *** join/#asterisk ibob63 (n=hp@bb-87-82-23-13.ukonline.co.uk) |
| 21:03.48 | kaz0358 | dlynes, it does sort the entries according to order/preference.. but maybe it should go a step further and sort it by service field, regexp field, and replacement field |
| 21:04.06 | diamon | vitt: Try asking something specific, see what happens. |
| 21:04.12 | dlynes_office | kaz0358: or perhaps you could write an agi script to do it :) |
| 21:04.22 | TripleFFFFFFFFFF | my nick what tripleXXX |
| 21:04.30 | TripleFFFFFFFFFF | but that ocnfused porn surfers |
| 21:04.34 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 21:05.12 | kaz0358 | dlynes, i'm beginning to wonder... i don't really like how the enumlookup stuff is done anyway. there should be a way to do all the enum lookups in parallel and automatically attempt to make the call. it seems kinda kludgy the way it is currently implemented. |
| 21:05.47 | dlynes_office | kaz0358: so rewrite it and submit the patches :) |
| 21:05.59 | dlynes_office | Maybe we'll get to see it in asterisk 1.4.1 |
| 21:06.13 | kaz0358 | in the enum.conf file the "search" tag is defined.. but i haven't any documentation on exactly where that is used |
| 21:06.46 | TripleFFFFFFFFFF | is there any projects on getting management port encryption or secure layer or something ? |
| 21:06.53 | kaz0358 | yeah.. it might not be *that* hard.. |
| 21:07.15 | boch | c ya |
| 21:07.26 | dlynes_office | kaz0358: line 642 of enum.c |
| 21:07.36 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:08.03 | kaz0358 | dlynes, okay i'm looking at it |
| 21:08.38 | dlynes_office | grep -rni "\"SEARCH\"" * |
| 21:08.38 | kaz0358 | it would be nice to see a command like.. "DialEnum([SIP|IAX2|H323],+NUMBER) |
| 21:08.41 | *** part/#asterisk vechers (i=vechers@64.61.117.138) |
| 21:08.48 | diamon | Foo, now I have a dozen darn SIP softphones to bugger about with. Ah well, something's gotta work! |
| 21:09.16 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:09.47 | *** join/#asterisk zotz (n=zotz@24.231.36.9) |
| 21:11.12 | znoG | has anyone had issues with grandstream ATAs hanging up when pressing flash? |
| 21:15.41 | *** part/#asterisk hawk (n=hawk@pdpc/supporter/active/hawk) |
| 21:16.11 | docelm0 | ~mybutt |
| 21:16.13 | jbot | rumour has it, mybutt is HUGE and stands for some funky stuff... |
| 21:16.16 | dahunter3 | features.conf says context => parkedcalls . Is this the context to which calls return? |
| 21:16.54 | dahunter3 | after a call has been parked |
| 21:22.55 | *** join/#asterisk bon (i=bon@shadow.radiolan.sk) |
| 21:23.27 | *** join/#asterisk Kokey (n=jramirez@201.133.218.194) |
| 21:25.00 | bon | can i have * to set my CLI in international format such as +xxxyyy? |
| 21:25.14 | bon | i am curios about that + since SetCallerID() accepts digits only |
| 21:25.46 | *** join/#asterisk ToyMan (n=stuq@rrcs-72-43-206-12.nys.biz.rr.com) |
| 21:26.09 | *** part/#asterisk mgremesco (n=mariano@201.216.203.137) |
| 21:27.36 | [TK]D-Fender | bon : and the fact that function is deprecated |
| 21:28.06 | Bullseye_Network | I've narrowed down some of my deadlocks to when a "Show Channels" happends, any ideas? |
| 21:28.29 | Bullseye_Network | Seems related to queue and agents |
| 21:29.08 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:29.55 | *** join/#asterisk darkskiez (n=darkskie@sunim.plus.com) |
| 21:33.16 | bon | [TK]D-Fender: might be.. what to use instead? |
| 21:33.30 | bon | i want international prefix with + |
| 21:33.39 | [TK]D-Fender | bon : the new CALLERID() function |
| 21:34.12 | [TK]D-Fender | there is no such thing as "+" At least not on any phone keypad I've ever heard of... |
| 21:34.23 | dlynes_office | bon: no need to enter that |
| 21:34.42 | [TK]D-Fender | bon : Formatting is an illusion. |
| 21:34.53 | dlynes_office | [TK]D-Fender: nah..the plus sign is actually used on some phones |
| 21:35.08 | dlynes_office | [TK]D-Fender: I used to have to use it to dial international long distance on Fido |
| 21:35.11 | [TK]D-Fender | dlynes_home : Not in my continent :) |
| 21:35.14 | dlynes_office | [TK]D-Fender: that's not required anymore though |
| 21:35.15 | bon | well.. i have 1234xx but when i call a normal phone you can't call back |
| 21:35.20 | bon | since the + won't show up there |
| 21:35.24 | bon | should i use 001234 instead? |
| 21:35.39 | dlynes_office | bon: 0119999999 |
| 21:35.43 | bon | on mobile phones i've tried with it works fine.. they add + automatically or what :) |
| 21:36.00 | dlynes_office | bon: yeah, exactly...the fido i referred to is a cellphone company |
| 21:36.46 | bon | no clue about that :) so what do you use to set your cli? i'd like phone users to be able just to call back and not have to modify the number :) |
| 21:36.49 | dlynes_office | [TK]D-Fender: you've never seen Fido doing that? |
| 21:37.04 | dlynes_office | [TK]D-Fender: Fido's from the same city as you :) |
| 21:37.20 | [TK]D-Fender | dlynes_home : I was with Rogers for years, now Bell for the last few months and never did overseas |
| 21:38.03 | dlynes_office | bon: on our continent, say the number is normally 91934892331, you would Set(CALLERID(num)=01191934892331) |
| 21:38.13 | [TK]D-Fender | dlynes_home : FIDO SUCKED. Best thing that ever happened to them was getting bought out so they now have "native" analog fallback |
| 21:38.28 | bon | dlynes_office: hm.. ok so you add the leading 0, right |
| 21:38.35 | bon | well 011 to be exact |
| 21:38.43 | dlynes_office | bon: alternatively, you could always try: Set(CALLERID(num)=+01191934892331) |
| 21:38.50 | dlynes_office | bon: but i don't know if that'll work |
| 21:38.51 | bon | dlynes_office: afaik that doesn't work |
| 21:39.03 | [TK]D-Fender | dlynes_home : CID = numeric |
| 21:39.11 | dlynes_office | [TK]D-Fender: not completely |
| 21:39.13 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 21:39.17 | dlynes_office | [TK]D-Fender: you can use spaces |
| 21:39.34 | [TK]D-Fender | dlynes_home : SUPPOSED to be.. been a long time sice I've read the RFC's |
| 21:39.38 | dlynes_office | [TK]D-Fender: there's certain other characters allowed too...I just can't remember what they are offhand |
| 21:40.01 | sevard | <PROTECTED> |
| 21:40.02 | bon | dlynes_office: hm, other chars? |
| 21:41.03 | dlynes_office | [TK]D-Fender: There's an rfc for phone numbers? |
| 21:41.15 | dlynes_office | [TK]D-Fender: phone numbers have nothing to do with the internet, though |
| 21:41.47 | [TK]D-Fender | dlynes_home : For Bellcore CID spec |
| 21:42.02 | [TK]D-Fender | dlynes_home : RFC is not limited to internet... |
| 21:42.07 | sevard | you can Set(CALLERID(num)=1234567890) or Set(CALLERID(num)=123-456-7890) |
| 21:42.20 | dlynes_office | oh yeah..hyphen is another one |
| 21:42.31 | sevard | -'s are nice in CNUM |
| 21:45.15 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 21:47.13 | [TK]D-Fender | Wow, Polycom SIP 2.0 seems a fair bit "zippier" than 1.x |
| 21:48.16 | stoffell | [TK]D-Fender, it's out already ? |
| 21:48.38 | [TK]D-Fender | I was hoping they'd have changed the IP 501 GUI a bit to match the styling shown on the photos of the IP 430 though.... |
| 21:48.45 | [TK]D-Fender | stoffell : Not official yet. |
| 21:49.00 | [TK]D-Fender | The IP 430 is hot shit.... |
| 21:50.11 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:51.22 | stoffell | [TK]D-Fender, looks like a nice replacement for 501 on some desks |
| 21:51.39 | [TK]D-Fender | stoffell : I'd say in high 90% of the time |
| 21:52.13 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 21:52.19 | stoffell | hehe ;) |
| 21:52.39 | [TK]D-Fender | stoffell : After a weekend beta testing it here at home, I'm going to bring it to work and try on an IP 600 |
| 21:52.59 | sevard | Does anyone use 'wakeup.php' ? For some reason I started geting the wake up call the very moment it made a call file. |
| 21:53.02 | sevard | Which is strange :| |
| 21:53.05 | stoffell | cool :) |
| 21:53.22 | [TK]D-Fender | sevard : Call files aren't timed IIRC |
| 21:53.22 | sevard | perhaps it's a bug in the way 1.2.7.1 handles call files |
| 21:53.38 | [TK]D-Fender | sevard : Executed on creation |
| 21:53.58 | [TK]D-Fender | sevard : which would mean you'd have to cron your script well |
| 21:54.16 | sevard | <PROTECTED> |
| 21:54.32 | sevard | becuase.. that's what it used to do :| |
| 21:54.48 | mitcheloc | hey sevard |
| 21:54.52 | sevard | sup mitcheloc |
| 21:55.32 | mitcheloc | waszto i think it was, he contacted me, he was selling sponserships |
| 21:55.47 | [TK]D-Fender | sevard : http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 21:56.00 | [TK]D-Fender | sevard : Seems to say calls immediately upon existance |
| 21:56.19 | sevard | mitcheloc: neat |
| 21:57.42 | [TK]D-Fender | stoffell : I plan on reselling my IP 301 and upgrading to a 430 uon its release |
| 21:58.45 | sevard | [TK]D-Fender: on that page you gave me |
| 21:58.56 | sevard | [TK]D-Fender: do you see the Wake Up php script in there |
| 21:59.04 | sevard | [TK]D-Fender: it uses call files, not cron. |
| 21:59.19 | eipi | anyone here have an ex-broadvoice ustarcom f1000 working with asterisk? |
| 22:00.58 | jets | ~nickometer 3133T|r00t111l1nux |
| 22:01.01 | stoffell | [TK]D-Fender, i'm curious on the price setting of the 430.. :) |
| 22:01.07 | jets | ~nickometer [3133T|r00t111l1nux] |
| 22:01.24 | jets | ~nickometer [3133T|r00t111l1nux-2600] |
| 22:01.48 | [TK]D-Fender | stoffell : its listed at www.atacomm.com |
| 22:01.56 | sevard | [TK]D-Fender: way to ignore me :) |
| 22:02.01 | [TK]D-Fender | stoffell : $160. But things may adjust after release |
| 22:02.40 | [TK]D-Fender | sevard : Yeah well the description on the top says : "Asterisk will notice and immediately activate the extension on the priority specified in the call file" |
| 22:02.43 | stoffell | [TK]D-Fender ; oh, that seems a great price setting.. |
| 22:03.11 | [TK]D-Fender | stoffell : I would have hoped a little cheaper than the 501, but it does have PoE itegrated, and lighted indicators |
| 22:05.29 | stoffell | [TK]D-Fender ; well, those things make it appealing.. |
| 22:06.21 | *** join/#asterisk darkskiez (n=darkskie@sunim.plus.com) |
| 22:07.15 | [TK]D-Fender | stoffell : Yup, perfect "new install" phone w/ PoE |
| 22:08.31 | *** join/#asterisk darkskiez (n=darkskie@sunim.plus.com) |
| 22:09.24 | stoffell | [TK]D-Fender ; yep, it's on my to-get list :) |
| 22:09.53 | dlynes_office | ~nickometer [TK]D-Fender |
| 22:10.06 | [TK]D-Fender | stoffell : My current reseller estimated me a price identicat to the 501, but I told him I've seen better, and I'm lining up a new source to try and drop it lower. |
| 22:10.18 | sevard | !nickometer sevrdseot |
| 22:10.22 | sevard | !nickometer sevrd |
| 22:10.23 | [TK]D-Fender | With any luck I can get it with little loss on resale of my IP 301 |
| 22:10.26 | sevard | god damit |
| 22:10.29 | sevard | !nickometer sevard |
| 22:10.34 | sevard | ~nickometer sevard |
| 22:10.43 | sevard | wow, i must be reallllllly getting tired. |
| 22:10.47 | dlynes_office | ~nickometer sevardsanidiot |
| 22:10.55 | jets | ~nickometer thisisridiculousandoboxious |
| 22:10.56 | sevard | faglynes |
| 22:11.05 | *** join/#asterisk dwclarkNU (n=dwclark@c-24-60-209-35.hsd1.ma.comcast.net) |
| 22:11.08 | sevard | why is [TK]D-Fender so fucking lame then |
| 22:11.08 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:11.11 | stoffell | [TK]D-Fender ; hehe :) yeah, here in europe we'll have to wait some longer, but looks promising :) |
| 22:11.15 | jets | the [] and the - |
| 22:11.17 | [TK]D-Fender | My personal favourite : |
| 22:11.20 | sevard | haha |
| 22:11.21 | [TK]D-Fender | ~nickometer lame |
| 22:11.37 | [TK]D-Fender | lame != lame! |
| 22:11.37 | jets | ~nickometer l4m3 |
| 22:11.58 | dwclarkNU | in my rtp.conf file i have the range set to 10000-20000 however, all my rtp traffic occurs on port 27000+, does anyone know why? |
| 22:12.06 | sevard | ~nickometer d7yn3s1s@f/\G60t!111! |
| 22:13.05 | dlynes_office | sevard: testy today? |
| 22:13.10 | sevard | heh |
| 22:13.17 | dahunter3 | I'm dying trying to figure out how do set the callerid of someone returning from being parked (no one picked up the parked call in time). The features.conf file says the the context => parkedcalls. Does that mean I can create an s,1,SetVar inside of the parkedcalls context (which doesn't currently exist?) |
| 22:13.19 | sevard | testicles. |
| 22:13.28 | dlynes_office | or lack of them |
| 22:13.32 | sevard | where is Hmmhesays |
| 22:13.39 | [TK]D-Fender | stoffell : Yup, and though I suspect it'll cost you disproportionately more as is the EU standard..... |
| 22:13.43 | sevard | dlynes_office: want to get smashed this weekend? |
| 22:13.54 | dlynes_office | sevard: i don't drink, sorry |
| 22:14.08 | sevard | dlynes_office: want to go out for imperial jade oolong? |
| 22:14.11 | stoffell | [TK]D-Fender ; probably yeah... |
| 22:14.22 | dlynes_office | sevard: huh? |
| 22:14.29 | sevard | dlynes_office: best tea in existance |
| 22:14.37 | dlynes_office | sevard: oolong? |
| 22:14.39 | dlynes_office | sevard: i think not |
| 22:14.46 | sevard | you don't like oolong?! |
| 22:14.49 | sevard | but but |
| 22:14.50 | sevard | :'( |
| 22:14.50 | dlynes_office | not really |
| 22:14.56 | dlynes_office | it's not that great a tea |
| 22:15.00 | sevard | imperial jade oolong will blow off your socks though man |
| 22:15.02 | [TK]D-Fender | darjeeling is where its at.. if not jasmine.... |
| 22:15.16 | dlynes_office | yeah...for a black tea, darjeeling's pretty good |
| 22:15.28 | dlynes_office | for a green tea, longjin is pretty good |
| 22:15.34 | sevard | dude, if it's not oolong it's mate! |
| 22:15.43 | dlynes_office | for a smoked tea, lapsang soo chong's pretty good |
| 22:15.55 | sevard | Yerba Mate is raaaaaaaaaaaaaaaaaad |
| 22:16.04 | dlynes_office | yerba mate sucks so bad |
| 22:16.16 | dlynes_office | that shizzit tastes like someone dunked some dust in the water |
| 22:16.16 | sevard | what the fuck dan |
| 22:16.23 | sevard | yerba mate is the shit |
| 22:16.27 | *** part/#asterisk Mr-packet (n=a@222-153-139-222.jetstream.xtra.co.nz) |
| 22:16.29 | sevard | oolong is second to none |
| 22:16.32 | dlynes_office | it's absolutely horrible stuff |
| 22:16.35 | sevard | you're on crack, brother. |
| 22:17.07 | dlynes_office | sevard: you're talking about that stuff that's supposed to be the new alternative to coffee, right? |
| 22:17.11 | sevard | next you're going to tell me that black coffee is bad and you have to put silly things in there |
| 22:17.24 | sevard | not alternative to coffee |
| 22:17.29 | sevard | you probably have some american knock off |
| 22:17.48 | dlynes_office | sevard: the stuff i've had is some disgusting tasting stuff from the amazon basin |
| 22:17.49 | sevard | real yerba mate is a caffine mind fuck from south america, a communal drink passed in circles |
| 22:17.49 | *** part/#asterisk Flosoft (n=admin@213.219.163.192) |
| 22:18.23 | [TK]D-Fender | sevard : BTW its not the filename that determines its execution, its its timestamp |
| 22:18.36 | sevard | [TK]D-Fender: really |
| 22:18.36 | dlynes_office | sevard: yep...sounds like the stuff i've had...but i didn't notice any caffeine in it |
| 22:18.46 | dlynes_office | sevard: it certainly didn't wake me up,a nyways |
| 22:18.58 | sevard | dlynes_office: gay |
| 22:18.59 | *** join/#asterisk darkskiez (n=darkskie@sunim.plus.com) |
| 22:19.10 | sevard | dlynes_office: what about coffee |
| 22:19.16 | dlynes_office | not reallly |
| 22:19.19 | sevard | hopefully you at least have good taste in that |
| 22:19.24 | dlynes_office | espresso perks me up a bit |
| 22:19.27 | dlynes_office | but not regular coffee |
| 22:19.40 | dlynes_office | yeah...uncle timmy's kick's ass :) |
| 22:19.43 | [TK]D-Fender | Caffiene doe absolutely nothing to me.... |
| 22:20.27 | dlynes_office | Yeah...i get a much better sleep after having a nice hot cup of timmy's in the evening |
| 22:20.43 | sevard | never had uncle timmy's |
| 22:20.56 | dlynes_office | sevard: you don't have wendy's in your neck of the woods? |
| 22:21.07 | sevard | We used to have a wendy's but they shut down |
| 22:21.10 | sevard | like 5 years ago |
| 22:21.12 | sevard | but seriously |
| 22:21.20 | sevard | good coffe sold at wendy's? |
| 22:21.43 | dlynes_office | sevard: ah...all the new wendy's that are being built in the US and Canada are usually paired up with Tim Horton's Donuts |
| 22:21.48 | [TK]D-Fender | A pool hall I play at regularly has the best coffee ever.... |
| 22:22.10 | sevard | tim horton's is ...alright.. it's not mc donald's coffee that's for sure |
| 22:22.21 | dlynes_office | sevard: thank god for that |
| 22:22.25 | sevard | but my taste buds don't care for tim horton's |
| 22:22.30 | dlynes_office | sevard: mcdonald's coffee is disgusting |
| 22:22.37 | sevard | it's like hot piss |
| 22:22.41 | sevard | straight from the perculator |
| 22:22.51 | sevard | i think it |
| 22:22.53 | sevard | 's time to go home |
| 22:23.04 | dlynes_office | btw |
| 22:23.05 | sevard | dlynes isn't taking me out for coffee so i go home sad :( |
| 22:23.18 | dlynes_office | you're describing starbuck's coffee, not mcdonald's |
| 22:23.27 | sevard | starbucks is absolute dick |
| 22:23.33 | sevard | the only thing i can stand there is their breakfast blend |
| 22:24.14 | sevard | g'nite mates |
| 22:25.08 | *** join/#asterisk froguz (n=alvaro@200.104.155.95) |
| 22:25.14 | froguz | hi there! |
| 22:25.23 | dlynes_office | g'night |
| 22:26.18 | froguz | why am i having 'ignoring' warning messages for several lines everytime i reload chan_zapata or any other file? |
| 22:26.43 | dlynes_office | [TK]D-Fender: looks like there isn't an rfc on callerid (Bell 202 CID) |
| 22:27.01 | froguz | for example this : May 26 18:22:00 WARNING[2599]: chan_zap.c:10876 setup_zap: Ignoring rtgain |
| 22:27.09 | dlynes_office | [TK]D-Fender: it's available as Telcordia documents: TR-TSY-000030 and TR-TSY-000031. |
| 22:27.13 | [TK]D-Fender | dlynes_home : I rember docs on it, did assume it usedthe RFC method |
| 22:27.25 | [TK]D-Fender | dlynes_home : Hmmm, does ring a bell |
| 22:27.52 | [TK]D-Fender | froguz : Its called RXGAIN. Watch your typos.... |
| 22:27.57 | dlynes_office | [TK]D-Fender: and you know if it's telcordia, they gotta be charging money for the docs |
| 22:28.12 | harryvv | the owner if michelin tire co passed away on a boat accident. I like those tires. |
| 22:28.13 | Bullseye_Network | zapata.conf has rtgain instead of rxgain. |
| 22:28.26 | dlynes_office | rtgain? wtf? |
| 22:29.09 | froguz | there is not rxgain AND rtgain????? |
| 22:29.20 | harryvv | dlynes_home here is Craig McCaws biography http://www.achievement.org/autodoc/page/mcc0bio-1 |
| 22:29.35 | [TK]D-Fender | froguz : RXGAIN & TXGAIN |
| 22:29.37 | Bullseye_Network | froguz: there is no such thing as rtgain..... |
| 22:29.45 | dlynes_office | froguz: rx == receive; tx == tranceive |
| 22:29.48 | Bullseye_Network | you have a typo in the zapata.conf |
| 22:29.57 | froguz | thanks :) |
| 22:29.59 | dlynes_office | erm tx == transceive |
| 22:30.22 | dlynes_office | harryvv: ah...looks like he's a local boy, too |
| 22:30.35 | harryvv | he is |
| 22:30.36 | harryvv | :) |
| 22:30.38 | harryvv | brb |
| 22:34.03 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 22:34.34 | eipi | anyone here have an ex-broadvoice utstarcom f1000 modified to work with asterisk? |
| 22:36.52 | [TK]D-Fender | eipi : Odds are its locked to Broadvoice's service and now going to get anywhere |
| 22:37.45 | froguz | is this ok for an FXO module in my TDM400P? (zapata.conf) : signalling=fxs_ks |
| 22:37.58 | [TK]D-Fender | froguz : Yes |
| 22:39.17 | froguz | why do i'm having 'chan_zap.c:10876 setup_zap: Ignoring signalling' then? |
| 22:39.54 | *** join/#asterisk tparcina (n=tparcina@fr2-adplastik.iskon.hr) |
| 22:40.19 | Dr-Linux | froguz: that's not a problem |
| 22:40.25 | froguz | and the same for hanguponpolarityswitch=no |
| 22:40.49 | tparcina | does anybody knows why i get this error - Unable to open '/dev/zap/channel': No such file or directory |
| 22:40.52 | froguz | Dr-Linux, but why does it ignore it? |
| 22:41.20 | [TK]D-Fender | froguz : pastebin your whole zapata.conf and zaptel.conf |
| 22:41.21 | tparcina | asterisk can't start because of that... :(( |
| 22:41.28 | dwclarkNU | why would asterisk choose an rtp port outside of the range defined in rtp.conf |
| 22:41.42 | *** join/#asterisk runa (n=asd@168.226.231.118) |
| 22:42.26 | tparcina | another thing, lsmod shows me that zaptel module isn't loaded |
| 22:42.38 | Nugget | asterisk is feisty and likes to live by its own set of rules! |
| 22:42.55 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 22:42.56 | a1fa | yo yo oy |
| 22:43.05 | tparcina | does anybody know the reason why zaptel module doesn't load - how to check that? |
| 22:43.08 | runa | hey :) Im trying to connect to iaxtel, I've enabled the register=> line and exten =>, I run asterisk -vvvvvvvcg but I don't see any registration process going on. any hints? |
| 22:43.27 | a1fa | anyway to limit number of outgoing minutes? |
| 22:43.42 | a1fa | and incoming, i guess w/o using that cda accounting |
| 22:44.26 | *** part/#asterisk dwclarkNU (n=dwclark@c-24-60-209-35.hsd1.ma.comcast.net) |
| 22:44.29 | froguz | [TK]D-Fender : http://pastebin.com/740280 |
| 22:45.02 | tparcina | come on guys, this is production system. i have restarted machine and now zaptel module doesn't load - so I can't start asterisk |
| 22:45.25 | tparcina | how to check what is the problem? |
| 22:45.46 | froguz | tparcina, README.udev?? |
| 22:45.58 | dlynes_office | tparcina: what kind of card? did you install a new kernel? |
| 22:46.16 | tparcina | this is what i get in full log file - http://pastebin.ca/59132 |
| 22:46.48 | tparcina | dlynes, digium PRI card with 1 port |
| 22:47.21 | [TK]D-Fender | froguz : Flies look fine except that you only have 1 port configured. Should work. |
| 22:47.42 | tparcina | i didn't install new kernel, i have created goust image (i have neabled usb port, luged usb disk, create goust image, desabled usb port, turned on computer and it doesn't work now :(( |
| 22:47.51 | dlynes_office | [TK]D-Fender: that would be correct...he only has one port :) |
| 22:47.56 | [TK]D-Fender | tparcina : And on ztcfg -vvvv ? |
| 22:48.09 | [TK]D-Fender | dlynes_home : but his zaptel denotes 4 <- |
| 22:48.17 | dlynes_office | [TK]D-Fender: ah |
| 22:48.28 | dlynes_office | [TK]D-Fender: He also stated the module wasn't getting loaded |
| 22:48.41 | tparcina | fender, http://pastebin.ca/59133 - ztcfg -vvvv |
| 22:48.42 | dlynes_office | [TK]D-Fender: so therefore, ztcfg won't do anything |
| 22:48.44 | froguz | [TK]D-Fender, i have other 3 ports configured below... but they are the same (copy/paste) |
| 22:49.01 | froguz | so i just paste the 1st channel |
| 22:49.06 | [TK]D-Fender | tparcina : modprobe'd for them? |
| 22:49.25 | [TK]D-Fender | froguz : show me the WHOLE thing then... stop holding stuff back! |
| 22:49.41 | tparcina | fender, modprobe - http://pastebin.ca/59134 |
| 22:50.00 | froguz | hahaha ok, i was avoiding boring you with the same thing... let me change it |
| 22:50.13 | [TK]D-Fender | tparcina : You sure you even compiled zaptel? |
| 22:50.19 | dlynes_office | tparcina: I guess I cna't help you anyways...having troubles connecting to pastebin.ca |
| 22:51.17 | runa | mm. when sign up in iaxtel, Im sent to http://x.digium.com/directory/ but it doesn't exist (!) |
| 22:51.28 | tparcina | fender, it has worked before i restarted it (in meanwile i have created goust image - turned on and off usb port - pluged usb disk) |
| 22:51.49 | file | runa: IAXtel signup doesn't work right now |
| 22:52.16 | froguz | [TK]D-Fender : http://pastebin.com/740291 |
| 22:52.44 | dlynes_office | file: any idea when it will be working? I've been trying close to a year now :) |
| 22:52.49 | runa | file: fuck. iaxtel signup doesn't works, fwd iax2 doesn't works. I want to try my phone :) |
| 22:53.03 | file | existing users can use their account fine |
| 22:53.10 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 22:53.53 | [TK]D-Fender | froguz : runa : You have * for... if you want to test your phone, use that! |
| 22:53.54 | dlynes_office | file: how can you become a new user though? I've been trying to create an account on that site for close to a year, someone suggested emailing digium support about it, but that resulted in nada |
| 22:54.05 | file | dlynes_home: you can't right now. |
| 22:54.29 | dlynes_office | file: so i guess you don't know when they'll be opening it for new users then, eh? |
| 22:54.31 | [TK]D-Fender | ak;shdash |
| 22:54.47 | [TK]D-Fender | froguz : Ok, I'd say ignore that warning if everything works... |
| 22:54.48 | file | dlynes_home: well, I control the Asterisk portion but not the website... |
| 22:54.59 | [TK]D-Fender | runa : Go call *'s echo test if you want to test your phone :) |
| 22:55.00 | jlewis | is there a reasonable way to send large numbers of SMS messages from asterisk servers in the US (i.e. not relying on ZIM or Bayham SMS gateways)? |
| 22:55.15 | froguz | thanks |
| 22:55.20 | file | BUT |
| 22:55.24 | file | I'll see what I can do... |
| 22:55.27 | dlynes_office | jlewis: gsm gateways? |
| 22:55.30 | dlynes_office | file: thanks :) |
| 22:55.52 | file | 791 peers registered right now on IAXtel btw |
| 22:56.01 | dlynes_office | cool |
| 22:56.27 | dlynes_office | main reason I wanted it was so I could call Digium without having to pay long distance charges :) |
| 22:56.37 | file | you can go to misery |
| 22:56.42 | dlynes_office | misery? |
| 22:56.45 | [TK]D-Fender | file : :D |
| 22:56.49 | file | misery is our PBX |
| 22:56.59 | dlynes_office | ah...and it allows anonymous callers? |
| 22:57.07 | [TK]D-Fender | file : Its not just a PBX, its a way of life! |
| 22:57.09 | file | I think |
| 22:57.10 | jlewis | dlynes_office: you saying effectively connect a "cell phone" to * and use a local cell provider to send the messages? |
| 22:57.19 | file | lemme check something first |
| 22:57.22 | [TK]D-Fender | dlynes_office : its in the sample extensions.conf! |
| 22:57.26 | dlynes_office | jlewis: no, use a gsm gateway |
| 22:57.30 | dlynes_office | [TK]D-Fender: oh...heh |
| 22:57.46 | dlynes_office | jlewis: you can get them in single port, dual port, 8 port, ... configurations |
| 22:57.49 | runa | ok. I think I need a good asterisk tutorial. any hint? |
| 22:58.04 | file | IAX2/guest@iaxtel.com/17004286000 |
| 22:58.05 | dlynes_office | jlewis: you need a sim card to operate it on the network you intend to use it on |
| 22:58.06 | file | try that |
| 22:58.08 | [TK]D-Fender | I *did* read it before realizing the only thing it was good for was Digium's URI :D |
| 22:58.23 | jlewis | dlynes_office: right...thats why I had ""'s around cell phone |
| 22:58.25 | [TK]D-Fender | runa : THEBOOK |
| 22:58.28 | [TK]D-Fender | \~book |
| 22:58.32 | [TK]D-Fender | ~book |
| 22:58.34 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:58.43 | Bullseye_Network | lol |
| 22:59.03 | dlynes_office | [TK]D-Fender: yeah..there's so much crap in that file, I always start fresh |
| 22:59.09 | jlewis | basically give * a way to talk to a local cell provider and buy a service plan that allows for lots of text messages? |
| 22:59.16 | dlynes_office | [TK]D-Fender: never bothered to try to read it...it's too confusing |
| 22:59.28 | dlynes_office | jlewis: exactly |
| 22:59.41 | [TK]D-Fender | dlynes_office : Same here... I just know that the URI in there is the only thing of value and I "911" it for support :D |
| 22:59.50 | jlewis | i.e. Cingular's text messaging rates look much cheaper than bayham's |
| 22:59.57 | dlynes_office | jlewis: not necessarily asterisk...any kind of program that needs to talk on a gsm network |
| 23:00.05 | dlynes_office | jlewis: they also have cdma gateways, too |
| 23:00.16 | file | let's see here... |
| 23:00.27 | jlewis | and as a side benefit, we could use those lines for talking to our staff cingular phones without burning minutes |
| 23:00.42 | dlynes_office | jlewis: if you've got that plan on that sim card, yes |
| 23:01.00 | dlynes_office | jlewis: i would imagine you'd need one sim card for every port on the gateway though |
| 23:01.14 | froguz | somebody has tried OpenVox hardware? |
| 23:01.14 | jlewis | probably |
| 23:01.33 | dlynes_office | jlewis: one sec, i'll get you the url for a company i found that sells that stuff |
| 23:01.52 | tparcina | dlynes, fender, one of you were right. Fedora core has installed new kernel automaticly - i didn't see it. now i have started with old kernel and everything works fine |
| 23:01.59 | jlewis | googling, I found 2N VoiceBlue |
| 23:02.09 | dlynes_office | tparcina: cool |
| 23:02.19 | tparcina | i need to learn linux! more! soon! |
| 23:02.22 | dlynes_office | tparcina: another reason to dislike redhat distros :) |
| 23:02.31 | jlewis | I think I remember seeing the 8-port thing months ago when I was curious about this before |
| 23:02.36 | jlewis | not sure who made it |
| 23:02.52 | tparcina | dlynes, it isn't cool at all. if i wasn't here on this location (400 km away from my home) i would been in big trouble |
| 23:03.24 | dlynes_office | tparcina: cool as in always nice to see what stupid stuff redhat-based distros do |
| 23:03.37 | dlynes_office | tparcina: it wasn't a good cool |
| 23:03.41 | jlewis | hey...I run lots of redhat systems, and none have ever automatically a new kernel without my knowledge |
| 23:03.42 | tparcina | now, first thing in the morning (now is 1 AM) would be to learn how to disable auto-update of kernel |
| 23:05.06 | *** join/#asterisk X-Rob (n=rob@dyn-ip-237.matilda.net.au) |
| 23:05.07 | tparcina | dlynes, anyway, you were on right path to solwe the problem from begining... i just didn't check what you have sugested me, i just answered from head... |
| 23:05.20 | dlynes_office | ah |
| 23:05.55 | tparcina | anyway, thank you guys. talk with you tomorow, now i'm going to get some sleap |
| 23:06.11 | tparcina | good night everybody |
| 23:06.11 | *** join/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 23:06.27 | *** part/#asterisk tparcina (n=tparcina@fr2-adplastik.iskon.hr) |
| 23:06.39 | dlynes_office | argh |
| 23:06.50 | dlynes_office | where or where did i bookmark that gsm gateway now :((( |
| 23:08.47 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-211.cybersurf.com) |
| 23:08.57 | *** join/#asterisk [pimp_c] (n=pimp@68.159.129.75) |
| 23:16.46 | *** part/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 23:23.55 | Sponge_bob | anyone here use cisco routers as their gateway? |
| 23:26.23 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-251-194.telkomadsl.co.za) |
| 23:28.13 | *** join/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 23:28.15 | file | wow, I have an evil idea |
| 23:28.48 | mitcheloc | if you changed f to v, you'd have all the letters for evil in file ;) |
| 23:28.54 | file | :D |
| 23:29.04 | mitcheloc | and unscrabbled it'd be vile! |
| 23:29.06 | mitcheloc | whats the idea? |
| 23:29.15 | file | darn, can't do it without major changes |
| 23:29.22 | file | I was going to turn qualify on on IAXtel and see how it faired |
| 23:29.29 | mitcheloc | =/ |
| 23:30.59 | *** join/#asterisk adker (n=adker@67-136-218-150.dsl1.glv.ny.frontiernet.net) |
| 23:31.48 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:34.16 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:39.26 | *** part/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 23:44.15 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:44.15 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org/ || FreePBX/AMP/Asterisk@Home Users should join #FreePBX for assitance |
| 23:46.50 | [TK]D-Fender | actually... file is an anagram for "leif" which confirms my suspicion : File & Blitzrage are one and the same!!!!! AH HAAAA!!!!!!!! |
| 23:46.57 | file | oh noes! |
| 23:47.48 | [TK]D-Fender | I haven't felt this level of acheivement since I last dodged the ninjas following my discovery of the Caramilk secret! |
| 23:49.30 | *** join/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 23:49.38 | Qwell | [TK]D-Fender: Yes, but that would mean they/he have/has a life. |
| 23:49.42 | Qwell | [TK]D-Fender: so..that disproves it |
| 23:49.52 | Qwell | pwned! |
| 23:50.37 | file | oh wait a bloody second here... |
| 23:51.55 | file | I think I fixed it! |
| 23:51.58 | *** join/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 23:52.06 | Qwell | file: yes, I think I found my problem too. :D |
| 23:52.09 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:52.13 | Qwell | yay us! |
| 23:52.21 | Qwell | NO, NO, NO, NO, NO! |
| 23:52.29 | Qwell | http://bdcast-infinity-kroq-fm.wm.llnwd.net/bdcast_infinity_kroq-fm |
| 23:52.35 | Qwell | The current song makes me VERY ANGRY |
| 23:53.24 | file | like |
| 23:53.25 | Qwell | They must get an assload in royalties though |
| 23:53.26 | file | totally? |
| 23:53.30 | Qwell | yes, totally |
| 23:53.31 | *** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg) |
| 23:53.48 | Qwell | bastards sold out to AT&T, and made me hate the song :P |
| 23:54.16 | [TK]D-Fender | Yay, home Polycoms all upgraded to SIP 2.0 :) |
| 23:55.22 | *** part/#asterisk cfassoni (n=root@c911444e.rjo.virtua.com.br) |
| 23:55.46 | [TK]D-Fender | ok, I'm out for a while, later all |
| 23:56.00 | trixter | but is sip 2.0 faster than aol 9.0? |
| 23:56.50 | [TK]D-Fender | trixter : Depends what centrifuge you fire the CD from ;) |
| 23:57.09 | [TK]D-Fender | trixter : And yes I find SIP 2.0 zippier than 1.6.x |
| 23:57.18 | Cyis | I don't know but those aol 9.0 discs make nice drink coasters |
| 23:59.30 | Qwell | ooo |