irclog2html for #asterisk on 20070212

`00:00.02`	`mavior`	`mafkees: the user section in the called server must be called as the user=xxx in the peer that place the call?`
`00:00.07`	`Dovid`	`exten => pissed,1,Get(${MY_FAVORITE_BOOZE})`
`00:00.36`	`mafkees`	`exten => wife,1,get(laid)`
`00:00.45`	`Dovid`	`haha`
`00:00.52`	`mafkees`	`exten => wife,2,make(dinner)`
`00:00.53`	`*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)`
`00:00.56`	`Dovid`	`lol`
`00:01.14`	`mafkees`	`BYE !`
`00:01.22`	`*** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk)`
`00:01.23`	`mafkees`	`going to start at priority 1 now`
`00:01.50`	`Dovid`	`exten => wife,3,goto(wife,1)`
`00:02.38`	`mafkees`	`lol`
`00:03.42`	`tzafrir_laptop`	`mafkees, you still here?`
`00:07.52`	`Dovid`	`Exten => working_hard,1,Set(feeling=strested)`
`00:07.58`	`Dovid`	`Exten => working_hard,2,Feels(${feeling})`
`00:08.03`	`Dovid`	`exten => working_hard,3,Goto(pissed_off,1)`
`00:08.11`	`Dovid`	`exten => pissed_off,1,Get(${FAVORITE_BOOZE})`
`00:08.15`	`Dovid`	`nm`
`00:08.18`	`Dovid`	`wont clog the room`
`00:08.58`	`Qwell`	`You can easily fix that...`
`00:09.12`	`Dovid`	`haha`
`00:09.21`	`Dovid`	`http://dovid.net/asterisk_feeling.txt`
`00:09.46`	`Qwell`	`exten => s,1,While(${alive})`
`00:09.49`	`Qwell`	`exten => s,2,Drink()`
`00:09.50`	`Dovid`	`lol`
`00:09.50`	`Qwell`	`exten => s,3,EndWhile()`
`00:09.58`	`Dovid`	`i need to make some corrections`
`00:10.03`	`Dovid`	`thanks qwell`
`00:11.05`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:11.05`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:15.18`	`mavior`	`can somebody help me out with this error : chan_iax2.c:6754 socket_process: Rejected connect attempt from 88.158.12.54, who was trying to reach '123@' ? i am trying to place a call beetween two asterisk servers thorugh iax2`
`00:17.42`	`Dovid`	`mavior: look above at what others have said - u most likely have a dial plan issue`
`00:17.51`	`Dovid`	`u cant make calls in either direction ?`
`00:19.29`	`mavior`	`Dovid i am pastebinning something that i hope will clear the situation`
`00:19.29`	`Dovid`	`i dont know IAX to well but I will give it a shot`
`00:19.29`	`*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es)`
`00:19.42`	`kink0`	`hello, anyone ussing txfax ?`
`00:20.12`	`Dovid`	`not me. sorry :(`
`00:20.21`	`kink0`	`I pretend to call asterisk and then forward the fax call to some fax machine, or as easy as send a fax.`
`00:20.41`	`kink0`	`my txfax appears to run, but is confused documentation about how to dial out to a fax machine`
`00:21.40`	`mavior`	`Dovid do you know if it is necessary to register => even with iax, if you want to place calls beetween two nat'd servers?`
`00:22.32`	`Dovid`	`mavior: i think so`
`00:22.56`	`Dovid`	`unless it works like SIP. where asterisk is a whore and takes anything`
`00:23.17`	`heh_v_water`	`So tonight i am ordering an fxs/fxo device for my home.. I'd like to hear some opinions between the sipura and the grandstream if possible`
`00:24.10`	`Dovid`	`mavior: r u following the wiki ?`
`00:24.15`	`Dovid`	`and have u read the book`
`00:24.18`	`Dovid`	`~book`
`00:24.20`	`jbot`	`somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`00:24.37`	`Dovid`	`heh_v_water: how about linksys ?`
`00:24.49`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`00:25.05`	`Dovid`	`i never used te sipura and or grandstream, have a look at the biz archive and see what people say`
`00:25.07`	`Dovid`	`~lists`
`00:25.09`	`*** join/#asterisk yassine (n=yassine@dsl.voicint.com)`
`00:25.20`	`heh_v_water`	`Dovid, thats what the sipura is maybe I should be more elaborate Linksys Sipura and Grandstream GS-488`
`00:25.27`	`kink0`	`heh_v_water, I used Polycom`
`00:25.48`	`yassine`	`are there any known mini pci cards from diguim ?`
`00:25.59`	`JT`	`nup`
`00:26.17`	`Dovid`	`<----------------- likes linksys`
`00:26.21`	`mog`	`nope`
`00:26.37`	`Dovid`	`people have been happy with grandstream but i never used it so i cant comment on it`
`00:27.18`	`Dovid`	`mog: does iax work like sip in the sense that will accept any incmoming call, even from a server that isnt registed to it ?`
`00:27.34`	`*** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com)`
`00:28.00`	`mog`	`you have the ability to have guest accounts, and you only need to register to send calls to a dynamic ip`
`00:28.22`	`mog`	`otherwise you can take calls from people who hwave usernames or guest accounts`
`00:28.25`	`Dovid`	`what do u mean by guest accounts ?`
`00:28.45`	`mog`	`there is a special account`
`00:28.49`	`mog`	`well not really special`
`00:28.55`	`mog`	`just a standard one`
`00:28.59`	`mog`	`guest`
`00:29.35`	`Dovid`	`for instance with sip if i dial sip:1234@server then it will try 1234 in the default context`
`00:30.00`	`mog`	`no`
`00:30.01`	`Dovid`	`my question is if that can be done in iax ? i just looked at my iax.conf and didnt see any default context settings`
`00:30.03`	`Dovid`	`ok`
`00:30.07`	`mog`	`and thats not the way sip works either`
`00:30.17`	`kink0`	`I read from documentation this: "make your Asterisk call the far FAX machine, and when it answers do..." but how to do some gotoif while Dial ?`
`00:30.20`	`Dovid`	`so what am i missing ?`
`00:30.24`	`mog`	`i dont know`
`00:30.38`	`mog`	`you could have iax2/guest@server/EXTEN`
`00:30.45`	`Dovid`	`ah ok`
`00:30.46`	`mog`	`and route that to extens in default context`
`00:31.18`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`00:31.18`	`*** mode/#asterisk [+o russellb] by ChanServ`
`00:31.36`	`Dovid`	`so iax2/guest@server/${EXTEN} is the same as SIP/${EXTEN}@server`
`00:31.37`	`Dovid`	`?`
`00:31.44`	`mog`	`umm no`
`00:32.02`	`mog`	`as sip/${exten}@server doesnt do that`
`00:32.02`	`mavior`	`I making progress the call get the extension but now i get [Feb 12 01:30:05] WARNING[29941]: channel.c:627 ast_best_codec: Don't know any of 0xe000 formats`
`00:32.09`	`mog`	`sip/guest@server/exten`
`00:32.10`	`mog`	`does`
`00:32.34`	`Dovid`	`well wont the end reslt be the same`
`00:32.45`	`mavior`	`codec problem?`
`00:32.49`	`mog`	`well the first one wouldnt work`
`00:32.50`	`Dovid`	`both calls will go to ${EXTEn} @ DEFAULT`
`00:32.52`	`mog`	`the second does`
`00:33.40`	`Dovid`	`NM. Gona try to roll over and think fresh in the morning. 2:33 am here`
`00:33.59`	`Dovid`	`mog: all of digium is out in alabama ?`
`00:34.03`	`mog`	`peace`
`00:34.05`	`kink0`	`Dovid: here too , too late`
`00:34.05`	`mog`	`nope`
`00:34.13`	`Dovid`	`where u guys @ ?`
`00:34.21`	`kink0`	`me, Spain`
`00:34.37`	`Qwell`	`LoRez: it didn't work :P`
`00:34.42`	`Dovid`	`i was takin bout digium guys`
`00:34.48`	`*** mode/#asterisk [-b !@freenode/staff/bhall] by Qwell`
`00:34.57`	`russellb`	`Digium HQ is in Huntsville, AL`
`00:35.08`	`russellb`	`though there are a few remote workers scattered throughout the world`
`00:35.14`	`*** join/#asterisk florz (n=florz@2002:58c6:2592:1:0:0:0:2)`
`00:35.15`	`Dovid`	`ah`
`00:35.17`	`Qwell`	`Madison is the new Huntsville`
`00:35.19`	`Qwell`	`:p`
`00:35.24`	`kink0`	`well, time for sleep now, I will continue tomorrow with the fax issue research`
`00:35.27`	`Dovid`	`madison ?`
`00:35.27`	`russellb`	`Qwell: ;)`
`00:35.32`	`kink0`	`see later`
`00:35.35`	`mavior`	`anyone can tell me what is this : channel.c:627 ast_best_codec: Don't know any of 0xe000 formats ?`
`00:35.36`	`russellb`	`it's a suburb of Huntsville...`
`00:35.40`	`Dovid`	`night kink0`
`00:35.42`	`russellb`	`where the engineering folks have a temp office`
`00:35.46`	`Qwell`	`russellb: technically, it`
`00:35.47`	`Dovid`	`ah`
`00:35.51`	`Qwell`	`s the other way around`
`00:35.56`	`Dovid`	`i like that digium is growin`
`00:35.56`	`Dovid`	`lol`
`00:36.02`	`russellb`	`Qwell: liar`
`00:36.12`	`Qwell`	`Huntsville is in Madison county :p`
`00:36.12`	`mavior`	`i get it when the incomming calls come in!`
`00:36.14`	`Dovid`	`a dream job would be working in the asterisk lab`
`00:36.21`	`russellb`	`Qwell: oh sheesh :-p`
`00:36.33`	`russellb`	`Dovid: feel free to submit a resume :)`
`00:36.37`	`Dovid`	`if qwell and russel can stop fightin then maybe they can help`
`00:36.53`	`Qwell`	`fighting? this is nothing :p`
`00:37.08`	`Dovid`	`russellb: i only know what i needed to know to build. i learn as i go. dont think i have enough`
`00:37.22`	`Dovid`	`and i cant think about relaoctin to alabama ?`
`00:37.41`	`Dovid`	`isnt there is a state req. that u can have no more than 10 teeth ?`
`00:37.41`	`Dovid`	`hehe`
`00:37.42`	`Dovid`	`;)`
`00:37.52`	`russellb`	`no, there is not.`
`00:38.02`	`russellb`	`I have all of mine.`
`00:38.10`	`Dovid`	`thats the image us people from NY have of AL`
`00:38.18`	`russellb`	`yeah yeah ...`
`00:38.25`	`Dovid`	`wow - how u pull that off ? u must not of been there all ur life`
`00:38.34`	`russellb`	`well, I'm from south carolina`
`00:38.42`	`Qwell`	`Madison county doesn't have that requirement`
`00:38.43`	`Dovid`	`NYC and LA are the only 2 important places in the US`
`00:38.55`	`Dovid`	`DC is just a technicality`
`00:39.48`	`Dovid`	`digium opening any offices out of AL ?`
`00:40.10`	`russellb`	`not in the near future, no`
`00:40.13`	`Dovid`	`real reason is cause i only eat kosher and some other things and i dont think i can do it out in AL`
`00:40.13`	`Dovid`	`:(`
`00:40.23`	`Dovid`	`im kinda jewish`
`00:40.27`	`russellb`	`excuses!`
`00:40.31`	`russellb`	`how are you kinda jewish?`
`00:40.49`	`Dovid`	`lol`
`00:40.51`	`Dovid`	`i am`
`00:40.52`	`Dovid`	`and proud`
`00:40.56`	`Dovid`	`:):)`
`00:41.52`	`Dovid`	`any jews out there ?`
`00:42.30`	`Dovid`	`on serious note - whats the zip for huntsville ?`
`00:42.35`	`russellb`	`90210`
`00:42.35`	`Qwell`	`35806`
`00:42.43`	`Qwell`	`~lart russellb`
`00:42.49`	`russellb`	`eep`
`00:42.51`	`Qwell`	`jbot: too much`
`00:42.54`	`Dovid`	`hahahahahahahahah`
`00:43.15`	`Dovid`	`so i guess that its true what alison said in an interview`
`00:43.23`	`Qwell`	`Dovid: she said we're nuts?`
`00:43.25`	`Dovid`	`u guys do drink way tooo much red bull`
`00:43.28`	`Qwell`	`because we aren't - really`
`00:43.31`	`Dovid`	`thats a given`
`00:43.39`	`russellb`	`it's a requirement to work at digium, actually`
`00:43.51`	`Dovid`	`i think i fit that`
`00:43.57`	`Qwell`	`being nuts, or drinking too much redbull?`
`00:43.58`	`russellb`	`or at least to work in development`
`00:44.01`	`Dovid`	`so u guys get it whole sale i asume`
`00:44.03`	`Dovid`	`all 3`
`00:44.03`	`Qwell`	`or both?`
`00:44.12`	`russellb`	`just nuts`
`00:44.17`	`Qwell`	`we need somebody to get us a redbull contract`
`00:44.21`	`russellb`	`and perhaps a general requirement on caffiene abuse`
`00:44.22`	`Dovid`	`buts, red bull, coffe and "smokin"....`
`00:44.31`	`Dovid`	`hehe`
`00:44.54`	`russellb`	`red bull gets expensive`
`00:45.58`	`Dovid`	`lol`
`00:45.59`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`00:46.10`	`Dovid`	`its still runnin in my vains from 2 years ago`
`00:46.27`	`Dovid`	`my second day on the job as head of IT I invested in a coffe machine.....`
`00:46.32`	`Dovid`	`lots of long nights....`
`00:47.09`	`Dovid`	`company had all machines with win2k no sp's (when sp3 was out) all with public IP's directly connected on a T1 with no firewall at all.....`
`00:47.11`	`Dovid`	`fun fun`
`00:48.40`	`J4k3`	`.... no SPs? no patches either?`
`00:48.42`	`J4k3`	`iirc, that was fatal.`
`00:48.43`	`Dovid`	`nm that they didnt allow the word linux to be mentiond..... i spoke about asterisk and the response was linux sucks, were gona invest in a 50k IP PBX`
`00:48.44`	`Dovid`	`yup`
`00:48.52`	`Dovid`	`hehe`
`00:48.55`	`Dovid`	`took down the company`
`00:49.03`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`00:49.03`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`00:49.14`	`Dovid`	`sorry`
`00:49.22`	`Dovid`	`40k they spent on a NEC IP PBX`
`00:49.30`	`J4k3`	`ouch`
`00:49.40`	`Dovid`	`and this was WITHOUT any hardphones`
`00:49.42`	`J4k3`	`insanity`
`00:49.46`	`J4k3`	`total insanity`
`00:49.51`	`Dovid`	`i still laugh`
`00:51.26`	`fetcher`	`Have dates been chosen yet for AstriCon (USA) 2007 in LA?`
`00:52.05`	`Dovid`	`fetcher: seems not - i want to sign up already`
`00:52.35`	`*** join/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe)`
`00:52.39`	`Dovid`	`atleast not from the site`
`00:52.48`	`fetcher`	`yeah, nothing on the official site so far...`
`00:53.10`	`Dovid`	`i wish.... as soon as i saw pics from TX i was ready to sign up`
`00:53.16`	`Dovid`	`anyone know the cost of it ?`
`00:53.41`	`*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`00:54.37`	`JT`	`Dovid: "took the company down"?`
`00:55.14`	`Dovid`	`JT: computers got infected`
`00:55.29`	`Dovid`	`we had to shut everything down - all servers and clean em one by one`
`00:55.44`	`Dovid`	`no sp's are fix's they were all hacked`
`00:55.58`	`bkruse_home`	`Dovid: infected? windows?`
`00:56.18`	`Dovid`	`haha`
`00:56.28`	`Dovid`	`linux was a word that was not to be mentioned`
`00:56.44`	`Dovid`	`as per my boss "all real good programers are at M$. linux is for rejects"`
`00:56.48`	`Dovid`	`he is not jobless`
`00:57.02`	`bkruse_home`	`hahahaha`
`00:57.10`	`bkruse_home`	`im not a windows v linux flamer, thats fine`
`00:57.36`	`Dovid`	`i am not so much either but he woulnt listen to anything i said`
`00:57.39`	`Dovid`	`in fact`
`00:58.13`	`J4k3`	`one word: kickbacks`
`00:58.19`	`J4k3`	`how much of that $40k PBX`
`00:58.21`	`J4k3`	`ended up in his pocket?`
`00:58.27`	`J4k3`	`$5k? $10k?`
`00:58.30`	`Dovid`	`he didnt know the diffrence between KB and kb`
`00:58.30`	`Dovid`	`none`
`00:58.31`	`J4k3`	`theres no kickbacks in open source`
`00:58.32`	`Dovid`	`he was an idiot that was to full of himself`
`00:58.54`	`Dovid`	`yup`
`00:59.04`	`Dovid`	`but u can fight ur boss`
`00:59.46`	`fetcher`	`What's the best way to auto-delete old voicemail? Anything cleaner than a cron-executed shell script?`
`01:00.00`	`Dovid`	`hmm`
`01:00.07`	`Dovid`	`u can rm -rf all files in the folder`
`01:00.42`	`Dovid`	`i created a nacro that would do it`
`01:01.14`	`Dovid`	`exten _*1XXX,1,System(rm -rf /var/..../${EXTEN:2})`
`01:01.52`	`J4k3`	`or set the whole pbx up on asterisk`
`01:02.07`	`Dovid`	`thats simple, my macro actually asked the user if they wanted to delete and it worked off CID or u can make em enter thier exten and press 1 to do it or 2 to not`
`01:02.16`	`J4k3`	`take the inventory tags off the nec gear, stick it on some crap rackmount boxes`
`01:02.19`	`J4k3`	`sell the pbx`
`01:02.23`	`J4k3`	`and pretend like it never happened.`
`01:02.29`	`Dovid`	`lol`
`01:02.37`	`Dovid`	`the company is no longer`
`01:02.43`	`Dovid`	`the idiots ran it in to the ground`
`01:03.12`	`J4k3`	`if thye had $40k to piss on a PBX that could have easily been ran on asterisk and an admin willing to work on it, they were definetly idiots`
`01:03.24`	`Dovid`	`yup`
`01:03.29`	`Dovid`	`i have stories.....`
`01:03.58`	`fetcher`	`Dovid: yeah, that's simple enough, but the customer wants to delete only VM older than a week... which means having to renumber messages that remain`
`01:04.30`	`Dovid`	`hmm`
`01:04.47`	`Dovid`	`then u would need somethign more complicated that deleted only files that are more than a week old`
`01:08.10`	`*** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com)`
`01:09.48`	`*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)`
`01:12.21`	`Qwell`	`"fast access DSL extreme six dot oh"`
`01:12.22`	`Qwell`	`ha`
`01:12.28`	`Qwell`	`wtg at&t`
`01:13.29`	`bkruse_home`	`Qwell: haha, i want verizon FIOS`
`01:13.55`	`bkruse_home`	`no service in huntsville though, maybe eventually.`
`01:14.07`	`Qwell`	`verizon isn't even in huntsville :p`
`01:14.31`	`J4k3`	`fios is hype`
`01:14.37`	`Qwell`	`J4k3: no it isn't`
`01:14.40`	`J4k3`	`its nothing the cable companies can't do now.`
`01:14.50`	`Qwell`	`can't != won't`
`01:15.01`	`J4k3`	`its also not all that impressive in Dallas, TX`
`01:15.09`	`bkruse_home`	`J4k3: your cable company gives you 50meg down and 8 up?`
`01:15.30`	`J4k3`	`bkruse_home: haha, theres no cable internet offering within 45 miles of here. ;)`
`01:15.42`	`bkruse_home`	`ouch`
`01:15.45`	`bkruse_home`	`satallite :X`
`01:16.06`	`J4k3`	`multiple T1s til I can afford a frac-T3`
`01:16.27`	`J4k3`	`or pull fiber/cable/T3 somewhere cheaper, and haul it over via wireless.`
`01:21.21`	`*** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net)`
`01:21.28`	`fetcher`	`bkruse_home: VoIP over satellite is nearly useless, with 2+ second latency`
`01:22.12`	`mog`	`bah`
`01:25.07`	`orkid`	`lol, not if you're on the middle of the ocean`
`01:26.04`	`fx0`	`2+ sseconds ? that must be the worst satellite isp ever.`
`01:27.30`	`bkruse_home`	`fetcher: i agree, I never recommended it...`
`01:28.36`	`mog`	`no a lot of them around that fx0`
`01:29.14`	`Stp1800`	`On the sat links I've worked on the delay was like between 540 and 700 or 800 at most.`
`01:29.37`	`*** part/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`01:30.22`	`mog`	`i set up a machine in iraq with 3 seconds delay`
`01:30.44`	`Qwell`	`eh?`
`01:31.12`	`Stp1800`	`Even at 3 sec, did the latency bounce all over the place?`
`01:31.36`	`Stp1800`	`I've noticed that on ku band satellite the latency isn't always stable, it changes from time to time.`
`01:31.39`	`mog`	`never worse than 3`
`01:31.43`	`mog`	`never better than 11`
`01:31.45`	`mog`	`er 1`
`01:33.17`	`file`	`eep people`
`01:40.01`	`k-man`	`who is the cheapest vsp that gives DID in australia? cheapest in terms of monthly rental`
`01:40.13`	`Qwell`	`~cheap`
`01:40.15`	`jbot`	`somebody said cheap was when microsoft designs softhardware, or nasty`
`01:40.21`	`k-man`	`yep`
`01:40.24`	`k-man`	`thats what i want`
`01:40.26`	`k-man`	`cheap and nasty`
`01:40.27`	`Qwell`	`~ygwypf`
`01:40.28`	`jbot`	`hmm... ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.`
`01:40.30`	`k-man`	`for testing only`
`01:41.06`	`fetcher`	`Stp1800: the only one I've worked with personally was a "Starband" residential satellite station in the US. It may have been sharing an overloaded transponder, but latency bounced around in the 1.5-2 second range during the evenings`
`01:41.27`	`*** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net)`
`01:42.10`	`fetcher`	`this was a couple of years ago. Hopefully they've improved since then`
`01:42.21`	`*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)`
`01:42.27`	`*** kick/#asterisk [Bhaal!i=lorez@freenode/staff/lorez] by LoRez (LoRez)`
`01:44.08`	`k-man`	`i remember seeing a site somehwere that lists all thee providers of voip in sydney`
`01:44.13`	`k-man`	`anyone know the url of it?`
`01:45.23`	`Qwell`	`LoRez: That's no less spammy, fyi ;p`
`01:49.03`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`01:49.47`	`*** join/#asterisk topping (n=topping@adsl-67-127-190-173.dsl.pltn13.pacbell.net)`
`01:51.43`	`*** join/#asterisk chronos_ (n=chronos@adsl-68-252-255-250.dsl.chcgil.ameritech.net)`
`01:51.51`	`*** join/#asterisk Eil (n=eil@69.89.103.248)`
`01:52.45`	`*** part/#asterisk Eil (n=eil@69.89.103.248)`
`01:57.14`	`LoRez`	`Qwell: who f'n removed the ban I put on him?`
`01:57.27`	`*** mode/#asterisk [+b !@freenode/staff/bhaal] by LoRez`
`01:57.53`	`*** kick/#asterisk [Bhaal!i=lorez@freenode/staff/lorez] by LoRez (LoRez)`
`01:57.53`	`Qwell`	`LoRez: I did - 1 because it was wrong, and 2 because he's immune :)`
`01:57.53`	`LoRez`	`Qwell: he's not presently immune.`
`01:57.57`	`LoRez`	`notice he didn't rejoin?`
`01:58.15`	`Qwell`	`he did rejoin - because the ban was wrong :P`
`01:58.30`	`Qwell`	`You had freenode/staff/bhall`
`01:58.36`	`LoRez`	`fine, it's not wrong now, why didn't you fix it instead of removing it?`
`01:58.52`	`Qwell`	`didn't realize you had "fixed" something`
`02:00.18`	`*** join/#asterisk ManxPower (n=manxpowe@103.sub-70-216-115.myvzw.com)`
`02:02.48`	`*** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`02:03.55`	`etfonhomey`	`Anyone here who can answer some questions?`
`02:04.00`	`Qwell`	`sure`
`02:04.02`	`Stp1800`	`Sure, I can try.`
`02:04.41`	`etfonhomey`	`I have Asterisk open source with a digium card with 2 FXO interfaces.`
`02:05.14`	`etfonhomey`	`Taking two analog lines from the PSTN to the digium card.`
`02:05.35`	`etfonhomey`	`Local phones are Polycom IP 430's.`
`02:05.58`	`etfonhomey`	`Here's my problem.`
`02:06.21`	`etfonhomey`	`When someone calls from outside to check the voicemails, the volume is very faint.`
`02:07.17`	`etfonhomey`	`I've tried messing with rxgain and txgain, but that, I think, messes up my echo cancelling and the coefficients set by fxotune.`
`02:07.49`	`etfonhomey`	`Any thoughts?`
`02:08.07`	`tzafrir_laptop`	`re-run fxotune?`
`02:08.23`	`etfonhomey`	`re-run fxotune after messing with rx/txgain?`
`02:08.35`	`tzafrir_laptop`	`You can keep a copy of your old /etc/fxotune.conf`
`02:09.30`	`tzafrir_laptop`	`for starters: if you're not in the US, have you set up opermode to a proper value?`
`02:09.36`	`etfonhomey`	`Does fxotune have any relation to rx/txgain? I've read in a post somewhere that it does.`
`02:09.58`	`etfonhomey`	`I'm in the US.`
`02:11.11`	`Corydon76-home`	`It has a relation to that setting in the same way that all settings are related to one another`
`02:11.43`	`tzafrir_laptop`	`no. fxotune sets a different set of registers than opermode`
`02:11.43`	`*** join/#asterisk syscon (n=joseph@S01060050da7ae68c.ed.shawcable.net)`
`02:11.49`	`etfonhomey`	`I've read that you should not put rxgain and txgain in your zapata.conf if you're using fxotune. Is that right?>`
`02:13.03`	`tzafrir_laptop`	`hmmm... rxgain/txgain is done by Asterisk, so surely it doesn't affect fxotune. Silly me.`
`02:13.31`	`Corydon76-home`	`No, you shouldn't alter rxgain and txgain after running fxotune unless you run fxotune afterwards`
`02:14.33`	`etfonhomey`	`So, it's OK to mess with tx/rxgain as long as you run fxotune afterwards?`
`02:14.49`	`etfonhomey`	`BTW, what is opermode?`
`02:14.53`	`Opperior`	`Why is that? I always thought they were independant of each other.`
`02:14.57`	`Corydon76-home`	`etfonhomey: pretty much`
`02:15.25`	`Corydon76-home`	`opermode is a kernel module flag to alter the behavior of the drivers according to the locale`
`02:15.27`	`J4k3`	`is there any disadvantage to using asterisk on freebsd?`
`02:15.32`	`Stp1800`	`I was going to suggest to adjust the rxgain andtxgain values so if those don't work for you I don't know what to suggest.`
`02:15.42`	`Corydon76-home`	`J4k3: unsupported kernel drivers`
`02:16.01`	`Stp1800`	`Also when a caller talks to an extension is the audio volume low or is it just low in voice mail?`
`02:16.01`	`J4k3`	`is that an issue if I don't use any interface cards (pure sip)?`
`02:16.14`	`syscon`	`Does anybody have an idea how often the Digium adapter "S101I" registers with asterisk server? Is the registration timing programed into the adapter units or controlled by Asterisk server?`
`02:16.16`	`Corydon76-home`	`J4k3: nope`
`02:16.31`	`J4k3`	`hmm, neat`
`02:16.53`	`Corydon76-home`	`syscon: it's programmed into the S101I unit`
`02:17.12`	`etfonhomey`	`The volume when talking to an extension is acceptable, but I believe it is less than the volume between the SIP phones.`
`02:17.45`	`Corydon76-home`	`syscon: see the iaxyprov utility`
`02:19.06`	`*** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com)`
`02:19.27`	`syscon`	`Thanks Corydon76-home: so it is programable, I'm planning on buying one or two units`
`02:19.28`	`etfonhomey`	`From what I've ready there is a signal loss when calling in using an analog line. And when you're calling in on an analog line to check a voicemail left by someone who called in on an analog line that the signal loss is doubled, hence the very low volume.`
`02:19.56`	`Corydon76-home`	`syscon: yep, the re-registration interval is usually around 60 seconds`
`02:20.11`	`syscon`	`If I'll have two of these units registered to my Asterisk server`
`02:20.12`	`syscon`	`(both units in different places). When I make a call between these two`
`02:20.12`	`syscon`	`units S101I, does the connection (the bandwidth that is being utilized)`
`02:20.12`	`syscon`	`goes through my server or directly between these two adapters?`
`02:20.16`	`Corydon76-home`	`syscon: just low enough to keep most NAT traversals open`
`02:20.53`	`Corydon76-home`	`syscon: it depends upon whether the server is configured to allow them to directly bridge or not`
`02:21.04`	`Corydon76-home`	`syscon: it will also depends upon network topography`
`02:23.13`	`etfonhomey`	`So, for my volume problem, you guys think my best solution is to alter the rx/txgain level? (most likely just the rxgain) Then rerun fxotune?`
`02:23.40`	`Corydon76-home`	`etfonhomey: try different settings until you get the desired level`
`02:24.12`	`syscon`	`I'll be running the server in Canada but the units will be located in the Philippines; I think the bridging option is configured in iax.conf isn't it?`
`02:24.12`	`etfonhomey`	`Other settings that just the gain levels?`
`02:27.02`	`*** join/#asterisk [shodan] (n=shodan@ip097.96-113-216.pppoe1.joliette.intermonde.net)`
`02:28.43`	`rue_mohr`	`ok, gonna try to get my T100P card up`
`02:28.56`	`rue_mohr`	`zapta.conf`
`02:29.42`	`rue_mohr`	`hmm no wait, I need to install the card driver first`
`02:29.48`	`rue_mohr`	`wct1xxp`
`02:30.11`	`rue_mohr`	`no errors, sweet`
`02:30.28`	`rue_mohr`	`also no little red light on the card saying its in trouble`
`02:30.33`	`rue_mohr`	`not sweet`
`02:30.59`	`rue_mohr`	`Feb 11 11:51:09 localhost kernel: Found a Wildcard: Digium Wildcard T100P T1/PRI`
`02:31.00`	`rue_mohr`	`sweet`
`02:32.24`	`rue_mohr`	`oh, zaptel.conf needs to match the card types in the channelbank dosn't it?`
`02:33.54`	`rue_mohr`	`ok, what I need to know to write this: clock master: needs to be the channelbank active channels: no clue what each channel is: no clue`
`02:40.57`	`rue_mohr`	`oh look there are two of them...`
`02:41.36`	`JT`	`ok, you're really not making enough sense`
`02:41.46`	`[TK]D-Fender`	`load chan_monologue.so`
`02:42.15`	`rue_mohr`	`http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf`
`02:42.21`	`rue_mohr`	`ok wait, step 1 is messed`
`02:42.29`	`*** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)`
`02:42.32`	`rue_mohr`	`if I have a T100P, I dont have a TDM do I?`
`02:42.41`	`*** join/#asterisk atlantia (n=scott@64.20.155.56)`
`02:42.49`	`tzafrir_laptop`	`you sould probably provide clock to the channel bank`
`02:43.06`	`rue_mohr`	`I should, but I supposably have this card cause its clock might be flakey`
`02:43.15`	`tzafrir_laptop`	`look at the parameters of "span" in zaptel.conf`
`02:43.33`	`rue_mohr`	`yea, I need a span line dosnt I?`
`02:43.45`	`tzafrir_laptop`	`yeah`
`02:43.46`	`[TK]D-Fender`	`sigh`
`02:43.56`	`rue_mohr`	`sorry, this is my first pbx`
`02:44.03`	`Ryanw`	`Which poe phone for business is the best choice at the moment?`
`02:44.07`	`[TK]D-Fender`	`rue_mohr : Got read the book for a bit....`
`02:44.21`	`[TK]D-Fender`	`Ryanw : General corp user?`
`02:44.22`	`rue_mohr`	`too many books to know where to start`
`02:44.35`	`Ryanw`	`yeah nothing special, using GXP2000's at the moment.`
`02:44.36`	`[TK]D-Fender`	`~book`
`02:44.47`	`jbot`	`[book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`02:44.47`	`rue_mohr`	`I'm really happy right now that I have it down to getting the T100P working`
`02:44.47`	`[TK]D-Fender`	`Ryanw : Polycom IP 430`
`02:44.58`	`[TK]D-Fender`	`THE book.`
`02:45.09`	`rue_mohr`	`oh dear`
`02:45.11`	`rue_mohr`	`a pdf`
`02:45.14`	`rue_mohr`	`that IS serious`
`02:45.17`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`02:45.35`	`*** join/#asterisk etfonhomey (n=root@74-140-206-4.dhcp.insightbb.com)`
`02:45.44`	`[TK]D-Fender`	`rue_mohr : You've been here for some time... you should long since have gotten this already...`
`02:46.21`	`rue_mohr`	`no, I dont beleive anyone pointed me to that yet`
`02:46.32`	`tzafrir_laptop`	`~docs`
`02:46.57`	`jbot`	`i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com`
`02:46.58`	`JT`	`and you didn't notice it being pointed out a dozen times a day, rue_mohr ?`
`02:47.19`	`rue_mohr`	`wow, the channelbank doc is thick enough on its own`
`02:47.26`	`rue_mohr`	`please dont burry me`
`02:47.57`	`rue_mohr`	`I need to narrow my focus`
`02:48.02`	`JT`	`what channel bank?`
`02:48.12`	`rue_mohr`	`newbridge 3624`
`02:48.22`	`rue_mohr`	`the pdf for that is really really big`
`02:48.33`	`rue_mohr`	`pages, not mb`
`02:48.45`	`JT`	`i bet only a dozen pages actually need reading to set it up`
`02:48.58`	`*** part/#asterisk syscon (n=joseph@S01060050da7ae68c.ed.shawcable.net)`
`02:49.32`	`rue_mohr`	`and I cant find the one that tells me if the LGS cards or the LGE cards are for the phones or the pstn connections`
`02:50.10`	`k-man`	`is there any word on when nodephone will do direct indial?`
`02:50.29`	`rue_mohr`	`my simpel goal right now, is to successfully test a T1 loopback`
`02:50.53`	`JT`	`k-man: you'd be hard pressed to find any other users here`
`02:51.04`	`rue_mohr`	`I have the card driver installed, not I need to install zaptel driver, which needs to be configed`
`02:51.19`	`Ryanw`	`D-Fender, what does the polycom 430 offer over the 301 ?`
`02:51.30`	`k-man`	`jt, i just thought i'd ask`
`02:51.58`	`rue_mohr`	`man, so many acronyms, d4 or esf...hmmm`
`02:52.30`	`k-man`	`jt, internode does do DID from nodephone to nodephone, but not from pstn yet, so i can't test DID myself yet`
`02:52.48`	`[TK]D-Fender`	`Ryanw : Built in PoE, Lighted indicators for the line-keys, Built-in PoE, Pixel based Display, XHTML MicroBrowser, Speakerphone.`
`02:53.00`	`[TK]D-Fender`	`rue_mohr : ESF`
`02:53.15`	`rue_mohr`	`:) ok`
`02:53.19`	`[TK]D-Fender`	`rue_mohr : B8ZS`
`02:53.31`	`rue_mohr`	`!?`
`02:53.42`	`rue_mohr`	`oh`
`02:53.47`	`[TK]D-Fender`	`2 settings you should be using.`
`02:53.47`	`Ryanw`	`the 301 has similar features but is cheaper, have you bench tested the 301 ?`
`02:53.52`	`rue_mohr`	`coding`
`02:54.01`	`[TK]D-Fender`	`Ryanw : I've used every phone they produced`
`02:54.49`	`[TK]D-Fender`	`Ryanw : Both excellent phones, but you mentioned PoE. The IP 301 does not do PoE natively. By the time you add the cost of the PoE cable they come too close.`
`02:55.21`	`Ryanw`	`cheers.`
`02:55.30`	`[TK]D-Fender`	`Ryanw : IP 301 = $115. IP 301 + PoE = $135. IP 430 = $150. for 15$ more you get all those bonus'`
`02:55.51`	`[TK]D-Fender`	`And the PoE cable is bulky.....`
`02:56.11`	`Ryanw`	`if you were restricted by $ and wanted something around half the price what would consider?`
`02:56.27`	`[TK]D-Fender`	`Ryanw : Asking for a bigger budget :)`
`02:56.33`	`hads`	`heh`
`02:56.37`	`[TK]D-Fender`	`~ygwypf`
`02:57.44`	`jbot`	`i guess ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.`
`02:57.48`	`Opperior`	`we mustn't abuse the bots, now`
`02:58.39`	`[TK]D-Fender`	`Ryanw : for what you asked, thats the price you pay. GrandSuck is to be avoided with extreme prejudice.`
`02:59.43`	`[TK]D-Fender`	`Ryanw : if you don't need PoE or Speakerphone, then a base IP 301 will do just fine @ $115. I would not consider anything lower for business use.`
`03:01.21`	`Ryanw`	`D-Fender, thanks for your time, i'll go purchase a 430 to evaluate.`
`03:01.23`	`JT`	`k-man: so they don't even come with their own pstn did?`
`03:01.44`	`k-man`	`jt, nodephone?`
`03:02.13`	`JT`	`yeah`
`03:02.39`	`k-man`	`no, its not available yet`
`03:02.42`	`k-man`	`oh.. i mean`
`03:02.43`	`k-man`	`yes`
`03:02.51`	`etfonhomey`	`D-Fender here's my setup if you didn't see it earlier: 3 Polycom Soundpoint IP 430's, 1 Asterisk OpenSource box w/ TDM400 w/ 2 fxo ports`
`03:03.03`	`*** part/#asterisk chronos_ (n=chronos@adsl-68-252-255-250.dsl.chcgil.ameritech.net)`
`03:03.03`	`k-man`	`yes DID is still unavailable on nodephone`
`03:04.02`	`rue_mohr`	`T100P <-> newbridge cahnnel bank, so I dont need dynamic range, just span, and the starting protocol`
`03:04.10`	`etfonhomey`	`I would like to configure the phones so that the two analong lines coming into the TDM40 are connected to each line button.`
`03:04.14`	`rue_mohr`	`and hte starting protocol must depend on the card`
`03:04.41`	`JT`	`k-man: you mean even a standard incoming number, like not an extra did?`
`03:04.53`	`[TK]D-Fender`	`etfonhomey : SIP based PBX's don't work that way.`
`03:05.03`	`k-man`	`jt, they give you a number, but you cannot call it from DID`
`03:05.13`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell)`
`03:05.13`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`03:05.42`	`etfonhomey`	`D-Fender, really?`
`03:05.44`	`k-man`	`jt, but you can call it from other nodephone numbers and maybe from other sip phone providers (not sure about that part)`
`03:05.47`	`[TK]D-Fender`	`etfonhomey : you don't get buttons that once pressed immediately grab a phsical line. That is a key-system PBX's featureset which we can't do.`
`03:06.09`	`JT`	`k-man: it's not a real did then, it's just some number they made up`
`03:06.32`	`[TK]D-Fender`	`etfonhomey : "Line-keys" are used to control independant calls without regard to their origin.`
`03:06.34`	`k-man`	`jt, i guess... unless they got the bank of numbers but have not enabled the DID part yet? i have no idea though`
`03:06.44`	`JT`	`actually you probably can do it, it's just stupid though, [TK]D-Fender :)`
`03:07.02`	`etfonhomey`	`So, can you use the line keys to show if another extension is on a line?`
`03:07.12`	`JT`	`k-man: sounds too shabby if you cant even get incming pstn calls yet, time to ditch nofefone`
`03:07.19`	`etfonhomey`	`In my setup, I have 3 extensions but only 2 outside lines.`
`03:07.43`	`etfonhomey`	`So, if extension 1 and 2 are tying up the outside lines, how to I let extension 3 know so they know that the lines are tied up?`
`03:07.56`	`k-man`	`jt, well... to be fair, it is just an addon service to internodes ADSL plans, its a one time sign up cost of $20 and then no ongoing fees`
`03:08.27`	`JT`	`there should be a $0 cost, no incoming number, no cost for setup or monthlies`
`03:08.35`	`k-man`	`then 18c national calls... not really cheap... but thats ok for testing`
`03:08.43`	`JT`	`there are plenty of voip providers with no fees that do outbound only`
`03:08.50`	`[TK]D-Fender`	`JT : You can TRY to fake it, but it'll come out half-asses at best.`
`03:08.53`	`k-man`	`jt, hmm... good point`
`03:09.24`	`[TK]D-Fender`	`etfonhomey : A congestion tone is highly effective...`
`03:09.27`	`Ryanw`	`with the aastra phones why go for the 9133i over the 9112i ?`
`03:09.29`	`*** join/#asterisk atlantia (n=scott@64.20.155.56)`
`03:09.31`	`Opperior`	`etf: I only have first-hand knowledge of the Snom 360, but with them, I can assign a programable key to an extenion, and the light will turn on when the extension is in use`
`03:09.39`	`JT`	`cool, i'd happily listen to advertising to cal iridium for free, J4k3 :)`
`03:09.40`	`Opperior`	`I assume a Polycom could do the same`
`03:09.53`	`Ryanw`	`i read somewhere asterisk can do indications for zap channels too.`
`03:10.05`	`[TK]D-Fender`	`Ryanw : Again, the 9133 has PoE, supports many more lines, lighted indicators, etc...`
`03:10.39`	`rue_mohr`	`if I'm going to plug in a loopback, what device do I say is on it?`
`03:10.50`	`[TK]D-Fender`	`Opperior : Thats only a LIGHT. He's asking to treat the line key as you would on a key system altogether...`
`03:11.07`	`rue_mohr`	`or does ztdiag just need the card driver?`
`03:11.18`	`Opperior`	`yes, but he also asked if there was a way to monitor the other extensions`
`03:12.13`	`[TK]D-Fender`	`Opperior : Funny, I don't see him asking that...`
`03:12.17`	`etfonhomey`	`Yes, I have three users and I would like the ability to for the users to at least know if an outside line is available by looking at their phone.`
`03:12.48`	`Opperior`	`etfonhomey>So, if extension 1 and 2 are tying up the outside lines, how to I let extension 3 know so they know that the lines are tied up? <-- that's how I interpreted that`
`03:13.16`	`[TK]D-Fender`	`Opperior : not enough indicators on a base.`
`03:13.39`	`etfonhomey`	`How many are on the 430? I don't have one in front of me.`
`03:13.39`	`rue_mohr`	`ok, I'll tell is their all fxs and work it out later`
`03:13.52`	`rue_mohr`	`I dotn need a d channel for hte channelbank?`
`03:14.02`	`Opperior`	`that I wouldn't know about, then. It was just a suggested train of thought`
`03:14.03`	`[TK]D-Fender`	`etfonhomey : You COULD make a microBrowser page and have it poll * through AMI or something to put on the display.... would take a little work, but its do-able.... though kludgy`
`03:14.44`	`JT`	`rue_mohr: usually you use CAS which doesn't have a D channel`
`03:14.50`	`[TK]D-Fender`	`Opperior : IP 430 has 2 line keys. You HAVE to use at least one of them for handling actual calls. The other can be assigned to watch a single device.`
`03:15.11`	`etfonhomey`	`D-Fender, can you point me to any info on how to use the microbrowser and/or create pages for it?`
`03:15.25`	`rue_mohr`	`I'm T1 not E1`
`03:15.26`	`Opperior`	`hmm, I see the problem then. I was thinking of something with a few more keys`
`03:15.32`	`[TK]D-Fender`	`etfonhomey : its in the Admin Guide, and on the WIKI`
`03:15.46`	`etfonhomey`	`Which WIKI? Does Polycom have one?`
`03:15.52`	`atlantia`	`[TK]D-Fender, http://forums.digium.com/viewtopic.php?t=13507`
`03:16.00`	`atlantia`	`[TK]D-Fender, seem sound to you?`
`03:16.08`	`[TK]D-Fender`	`etfonhomey : no, on www.voip-info.org there is a handbook concerning Polycom phones`
`03:16.46`	`JT`	`rue_mohr: actually, it sounded irrelevant :P`
`03:16.48`	`rue_mohr`	`ok... driver loaded`
`03:16.54`	`[TK]D-Fender`	`atlantia : You asked me this the ther day and I told you "no"`
`03:17.17`	`rue_mohr`	`hmm its still not upset about not being plugged into anythink`
`03:17.30`	`atlantia`	`[TK]D-Fender, correct, but this gentleman has obviously got another opinion`
`03:17.33`	`[TK]D-Fender`	`atlantia : But it could be that this is a function I've simply never heard of.`
`03:17.41`	`[TK]D-Fender`	`atlantia : Have you TRIED it?`
`03:18.24`	`JT`	`rue_mohr: of course, you have plugged in a T1 crossover cable between the T100P and the channel bank, haven't you?`
`03:18.48`	`rue_mohr`	`:) its not plugged in yet, I wan tot do a loopback test`
`03:18.59`	`rue_mohr`	`I need to make a serial cable for the channelbank`
`03:19.19`	`JT`	`you mean T1?`
`03:19.27`	`JT`	`or rs-232 management console port`
`03:19.52`	`rue_mohr`	`the managment`
`03:19.57`	`atlantia`	`atlantia, i am right now, man don't get so offended, i figured i'd share and try to get around my issue`
`03:19.59`	`rue_mohr`	`I need to configure the thing`
`03:20.02`	`*** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net)`
`03:20.03`	`atlantia`	`er [TK]D-Fender`
`03:20.08`	`atlantia`	`heh talking to myself again`
`03:20.43`	`[TK]D-Fender`	`atlantia : Not offended at all. Just wondering if you actually tried following what he offered you. I mean if you did and it worked, this would all be a moot point now wouldn't it? :)`
`03:20.57`	`[TK]D-Fender`	`atlantia : "Best conversation in town"`
`03:21.05`	`atlantia`	`lol`
`03:21.06`	`JT`	`rue_mohr: then obviously magic won't work`
`03:21.12`	`atlantia`	`right on hell i hope it works`
`03:21.29`	`rue_mohr`	`no, magic is overrated`
`03:21.38`	`rue_mohr`	`verry little success withthe stuff`
`03:21.59`	`atlantia`	`unfortunately i am outta school on this right now.. i just am beginning to understand the asterisk stuff, have a book on the way, and need to figure out where exactly I should be entering these parms`
`03:22.15`	`[TK]D-Fender`	`~book`
`03:22.20`	`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`
`03:22.20`	`rue_mohr`	`heh, well, I just started at the beggining`
`03:22.23`	`[TK]D-Fender`	`its right there... no need to wait...`
`03:22.24`	`atlantia`	`ordered it`
`03:22.26`	`rue_mohr`	`I have the equip, wanna play?`
`03:22.28`	`atlantia`	`have the pdf`
`03:22.32`	`atlantia`	`hate reading e books`
`03:22.44`	`atlantia`	`no replacement for the real deal`
`03:22.52`	`[TK]D-Fender`	`atlantia : not an e-book, it IS the book. just print it out...`
`03:23.28`	`JT`	`well technically it is an ebook, and a good read whilst the real thing is on the way`
`03:23.36`	`JT`	`it's stupid to print the whole thing out yourself, imho`
`03:23.53`	`*** join/#asterisk infernix (n=nix@spirit.infernix.net)`
`03:23.59`	`rue_mohr`	`I nee to make me a loopback jack`
`03:24.20`	`rue_mohr`	`hmm, should I use blue wire or orange..`
`03:24.29`	`JT`	`you don't need to, i've never needed to, but i guess you can if you want to`
`03:26.08`	`[TK]D-Fender`	`k-man : I could have sworn I gave you the math on this. My lasers run on recycled cartridges at about .009$ / page (CAD). 376 pages = $3.384. 500 Sheet pack of paper = $4 CAD. So thats $1.50 worth of paper. total cost = 5$. So SHUP :)`
`03:26.39`	`[TK]D-Fender`	`k-man : I'll bet most other people payed more on SHIPPING.`
`03:27.23`	`JT`	`i'd rather an offset printed and bound book`
`03:27.41`	`[TK]D-Fender`	`JT : I'd rather invest the extra $50 in HARDWARE :)`
`03:27.50`	`JT`	`$50? exageration`
`03:28.06`	`[TK]D-Fender`	`JT : and I keep mine in a nice large binder with all my Polycom docs and the like...`
`03:28.14`	`JT`	`heh`
`03:28.21`	`[TK]D-Fender`	`JT : How much does it go for shipped?`
`03:28.34`	`JT`	`depends where to`
`03:28.45`	`[TK]D-Fender`	`JT : Montreal, QC.`
`03:28.51`	`JT`	`wouldn't be more than $35 to the us i think`
`03:29.01`	`*** join/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com)`
`03:29.32`	`connecta`	`hey guys, how can i tell whether i have a 586, 686, etc?`
`03:29.44`	`Qwell`	`connecta: Did you buy it within the last 10 years?`
`03:29.46`	`rue_mohr`	`dmesg\|less`
`03:29.54`	`rue_mohr`	`connecta, ^^`
`03:29.56`	`connecta`	`yah it's a dell about 3 years old`
`03:30.06`	`Qwell`	`then it's i686 at least`
`03:30.17`	`k-man`	`[TK]D-Fender, gee, i was joking!`
`03:30.41`	`*** join/#asterisk etfonhomey1 (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`03:30.53`	`etfonhomey1`	`D-Fender I think I lost my connection. You still here?`
`03:30.56`	`[TK]D-Fender`	`k-man : People out there ARE ignorant enough and mathematically challenged.`
`03:31.10`	`connecta`	`Quell: when choosing which codec file to download for g729a, i found the tutorial on digiums site, but i can't get asterisk to start successfully after`
`03:31.13`	`[TK]D-Fender`	`etfonhomey : ummm... no? ;)`
`03:31.29`	`JT`	`i'd also rather let the authors get some royalties`
`03:31.35`	`k-man`	`[TK]D-Fender, err... ok.. anyway... i was joking`
`03:31.49`	`etfonhomey1`	`D-Fender: So, what would you do in my situation?`
`03:31.59`	`rue_mohr`	`ok, plugging my loopback into the channelbank caused hte sync error light to go out, so I suppose it works`
`03:32.02`	`[TK]D-Fender`	`JT : I went out for beer with one of them, and a contributing auther as well :)`
`03:32.10`	`JT`	`heh`
`03:32.15`	`rue_mohr`	`anyone value a photo of a T1 loopback jack?`
`03:32.20`	`[TK]D-Fender`	`JT : And played chauffeur while they were in town.`
`03:32.30`	`Qwell`	`rue_mohr: $4.50`
`03:32.37`	`JT`	`[TK]D-Fender: i spose you've paid your debt to society`
`03:32.43`	`connecta`	`oops, i meant to say Quell`
`03:32.47`	`connecta`	`Ahh, Qwell`
`03:32.49`	`[TK]D-Fender`	`rue_mohr : About $0.30`
`03:32.56`	`rue_mohr`	`no, I mean do you want that I should give you a photo of it`
`03:33.12`	`etfonhomey1`	`LOL`
`03:33.43`	`rue_mohr`	`but I'll take Qwells offer if I can trade it for 2 answers`
`03:34.42`	`k-man`	`in dial plans, why would you want to match any digit from 2-9 as in N?`
`03:34.54`	`rue_mohr`	`if anyone wants a loopback connector, I'll sell at 0.75`
`03:35.20`	`Qwell`	`k-man: NXX`
`03:35.29`	`JT`	`he asked "why"`
`03:35.38`	`JT`	`probably an american thing really`
`03:35.39`	`Qwell`	`that is why`
`03:35.44`	`k-man`	`jt, oh...`
`03:35.58`	`k-man`	`qwell, do you care to elaborate?`
`03:36.30`	`etfonhomey1`	`D-Fender, what do you think about my last question?`
`03:37.00`	`connecta`	`im sure this questions been asked many times, but i really need help choosing the file for g729 codec from ftp.digium.com`
`03:37.02`	`JT`	`it's just a convenience for americans, to match their national dialling format`
`03:37.29`	`k-man`	`JT, OH! i see, N for national`
`03:37.31`	`k-man`	`hmmm`
`03:37.32`	`k-man`	`thanks`
`03:37.32`	`[TK]D-Fender`	`etfonhomey1 : I'd try an actually follow the suggestion they gave you and see what happens. I mean they offer an answer right in your face. What the hell else are you going to do?`
`03:37.42`	`Qwell`	`k-man: what? no..`
`03:38.58`	`etfonhomey1`	`D-Fender, what suggestions are you talking about? I only heard the one about using the indicator light which can't be done because the 430 only has one LED.`
`03:39.26`	`connecta`	`Qwell: can you advise me at all or no?`
`03:39.30`	`k-man`	`jt, do you make a seperate dial plan for national and local numbers?`
`03:39.52`	`JT`	`k-man: seperate extensions in the same context`
`03:40.53`	`*** join/#asterisk gerphimum (i=Trekkie@207.190.58.83)`
`03:41.48`	`k-man`	`can you give me a look at yours please?`
`03:41.55`	`k-man`	`oh.. you showed me something before`
`03:42.03`	`JT`	`i believe so :P`
`03:42.05`	`etfonhomey1`	`k-man where are you located?`
`03:42.41`	`k-man`	`sydney`
`03:42.47`	`rue_mohr`	`http://eds.dyndns.org:81/~ircjunk/images/dscn9333.jpg`
`03:43.09`	`Qwell`	`rue_mohr: why such a thick gauge?`
`03:43.22`	`rue_mohr`	`if anyone makes any comments about my twists/inch I'm not listening :)`
`03:43.25`	`rue_mohr`	`thats 24`
`03:43.41`	`etfonhomey1`	`I can show you an example of how I do local vs. long distance in the US would that help?`
`03:44.01`	`JT`	`it doesn't look that thick, probably the macrophotography that makes it look big`
`03:44.03`	`rue_mohr`	`Qwell, if interested, post it on wikis or whatever you want`
`03:44.14`	`rue_mohr`	`well it is just an rj45`
`03:44.20`	`rue_mohr`	`I mean gee`
`03:45.29`	`etfonhomey1`	`I like this picture myself:`
`03:45.31`	`etfonhomey1`	`http://eds.dyndns.org:81/~ircjunk/images/dscn0549.jpg`
`03:45.39`	`rue_mohr`	`ok, its plugged in, how do I test the card with it?`
`03:45.54`	`rue_mohr`	`heh, yea, we had a ups vent`
`03:46.05`	`rue_mohr`	`when we checked them all... well, it was scarry`
`03:46.12`	`etfonhomey1`	`:)`
`03:46.40`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`03:46.43`	`rue_mohr`	`http://eds.dyndns.org:81/~ircjunk/images/dscn0547.jpg`
`03:46.52`	`rue_mohr`	`you know those ARE melted toghethor`
`03:47.13`	`etfonhomey1`	`Nice.`
`03:47.17`	`Qwell`	`rue_mohr: You misspelled Australia`
`03:47.27`	`rue_mohr`	`:( sorry`
`03:47.38`	`rue_mohr`	`my finger tends to bounce on the l`
`03:47.50`	`JT`	`where did he spell australia?`
`03:48.42`	`rue_mohr`	`if one wants, I can supply photos of the crossover cable to`
`03:49.03`	`hads`	`12:40:01 < k-man> who is the cheapest vsp that gives DID in australia? cheapest in terms of monthly rental`
`03:49.34`	`hads`	`Erm, excuse me.`
`03:50.13`	`rue_mohr`	`kb1 said to run ztdiag with the loopback on, I think he was wrong...`
`03:50.38`	`connecta`	`Can anyone help me install the g729 codec`
`03:54.53`	`rue_mohr`	`ok... I'm going to say it passed the test`
`03:56.29`	`etfonhomey1`	`Any others experienced with Polycom phones?`
`03:57.19`	`*** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net)`
`03:59.42`	`*** part/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net)`
`04:00.16`	`Mavvie`	`since I've added the option Ww to my dial commands the load of the asterisk server has gone up to about 1.2-1.5 (from 0.2-0.4)`
`04:00.18`	`Mavvie`	`any idea why?`
`04:00.29`	`JT`	`rue_mohr: gold flakes?`
`04:01.05`	`rue_mohr`	`no, its special non-conductive pixie dust specifically for computer apllications`
`04:01.22`	`JT`	`ah that's shit, get gold flakes or graphite powder`
`04:02.41`	`rue_mohr`	`what I have here dosn't fit togethor see, I was told" put togethor a loopback connector, and ztdiag to test the T1 card, and make sure the interrupts and stuff are ok" but those things dont seem to go togethor`
`04:03.01`	`rue_mohr`	`well also with the card driver and the zaptel driver`
`04:03.23`	`connecta`	`im surprised nobody here uses G729`
`04:03.37`	`rue_mohr`	`I might, I dont know`
`04:03.41`	`rue_mohr`	`I'm not there yet`
`04:03.51`	`JT`	`connecta: some people use it`
`04:04.00`	`*** join/#asterisk ShadowTech (n=jerespet@c-66-176-202-207.hsd1.fl.comcast.net)`
`04:04.01`	`rue_mohr`	`I'm trying to figur out what control cards my channelbank has`
`04:04.33`	`connecta`	`Yah i figured, I guess nobody just wants to take the time to help me get it installed'`
`04:04.56`	`JT`	`why don't you ask digium support?`
`04:05.24`	`connecta`	`I prefer to seek support this way`
`04:05.27`	`rue_mohr`	`I think were afraid of the price tag`
`04:05.46`	`JT`	`connecta: well, we're not paid here`
`04:05.47`	`rue_mohr`	`connecta, iirc, quite a number of the peopel here ARE digium`
`04:06.06`	`JT`	`connecta: and no-one will answer your question here unless you ask the question, not "does anyone want to help?"`
`04:09.02`	`connecta`	`I've tried putting all the different versions of the g729 codec file from digium in the proper directory, and changed thep ermissions and ownership per the tutorial. no matter which one i use, asterisk won't start`
`04:09.32`	`JT`	`why won't it start?`
`04:10.23`	`connecta`	`JT: i was told that if any of the .so files in the directory are either for the wrong version of asterisk or for the wrong processor type.`
`04:10.32`	`connecta`	`then asterisk won't start properly`
`04:10.43`	`connecta`	`and this seems to be what im experiencing`
`04:11.18`	`*** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`04:15.13`	`JT`	`ok well, do you have any evidence of that?`
`04:15.41`	`k-man`	`jt, i thought i saved your dial plans somewhere but I can't find it. could you send it to me again please?`
`04:15.54`	`k-man`	`jt, i think you pastebinned it`
`04:17.25`	`k-man`	`how does one go about distributing a configuration to sip phones throughout the organisation?`
`04:19.06`	`connecta`	`The most efficient way (assuming you have phones that support it) is to use a dhcp server that advertises dhcp option 66 or 67 to the phones`
`04:20.04`	`connecta`	`When a phone boots, it gets an ip address from the dhcp server, along with instructions to look to a tftp, ftp, or http server for boot and configuration files`
`04:20.21`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`04:20.35`	`*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`04:20.37`	`k-man`	`connecta, ahh... i see`
`04:21.03`	`k-man`	`connecta, what feature do i check for in the phone? whats it usually called?`
`04:21.11`	`rue_mohr`	`darnit, I read the section and I still dont know if LGS or LGE is for FXO or FXS`
`04:21.22`	`etfonhomey`	`Can anyone here provide me with sample Polycom Soundpoint IP 430 config files?`
`04:22.40`	`connecta`	`k-man , it might be referred to as netboot, mass provisioning support, bootp, pxe boot, 'configure phone from network' etc`
`04:22.55`	`k-man`	`connecta, ok, thanks`
`04:22.58`	`connecta`	`etfonhomey, thers a few types of config files for hte phones, do you know which filenames you need`
`04:23.06`	`connecta`	`k-man, what brand of phones do you have`
`04:23.19`	`k-man`	`linksys spa942`
`04:24.21`	`connecta`	`without looking, i gotta believe it's possible`
`04:24.28`	`JT`	`k-man: found it http://www.pastebin.ca/335149`
`04:24.32`	`k-man`	`yeah, i saw something that says it can do tftp`
`04:24.42`	`k-man`	`your a legend jt, thakns`
`04:24.44`	`k-man`	`thanks`
`04:25.27`	`hads`	`Linksys do provisioning, the tools for setting it up aren't usually freely available though.`
`04:25.59`	`connecta`	`i don't think you need tools though just a dhcp server and the config files`
`04:26.10`	`connecta`	`coincidentally, the process is the same for upgrading the firmwares on the phones`
`04:26.15`	`rue_mohr`	`yay, I can modify an old mouse card to make an interface cable`
`04:26.42`	`hads`	`http://spc.pifiu.com/`
`04:27.23`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`04:27.26`	`*** part/#asterisk capt-rogers (n=teddymil@bas14-toronto63-1096770977.dsl.bell.ca)`
`04:29.08`	`hads`	`connecta: When I said tools I was referring to config file guides etc. as much as anything`
`04:31.43`	`k-man`	`jt, i think your dial plan is missing prefix 01 for national dial numbers`
`04:32.47`	`hads`	`Where's 01?`
`04:32.54`	`JT`	`k-man: what's 01?`
`04:33.03`	`k-man`	`they can be called from any state`
`04:33.15`	`JT`	`you mean isp dialin numbers?`
`04:33.32`	`k-man`	`jt, yeah... but afaik they are not limited to isps`
`04:33.44`	`k-man`	`jt, but i'm not expert, i could be wrong`
`04:33.48`	`JT`	`that's all i've ever seen them used for`
`04:33.54`	`hads`	`Don't dial your ISP through Asterisk? :)`
`04:33.58`	`JT`	`and it's not a concern for my setup :)`
`04:34.07`	`k-man`	`jt, ok, just wanted to let you know`
`04:34.19`	`k-man`	`jt, also there is this other number called follow me i think`
`04:34.21`	`JT`	`k-man: there's a lot of weird prefixes too that aren't there`
`04:34.23`	`k-man`	`i forget the prefix`
`04:34.36`	`JT`	`i could build a complete dialplan, but it'd be quite big`
`04:34.38`	`k-man`	`jt, yeah... is there a definitive list of them?`
`04:34.53`	`JT`	`yes, there's the government standard for it`
`04:34.59`	`k-man`	`oh.. interesting`
`04:35.02`	`hads`	`And there's too many silly little gotchas and things`
`04:35.47`	`JT`	`heh`
`04:36.06`	`JT`	`yeah, there's even prefixes for numbers that can only be called from overseas`
`04:36.17`	`k-man`	`really? interesting`
`04:36.32`	`JT`	`no idea if any of that is even used`
`04:36.59`	`hads`	`I entertained the idea of writing one for NZ 'til I got stumped for info about a prefix that covered more than one different area.`
`04:38.27`	`JT`	`don't you guys have number plan documentation?`
`04:38.33`	`connecta`	`how do i check my asterisk version`
`04:38.52`	`JT`	`show version`
`04:38.54`	`JT`	`k-man: ttp://www.comlaw.gov.au/comlaw/legislation/legislativeinstrumentcompilation1.nsf/framelodgmentattachments/708BEF2FBC05FCB1CA25720D002573E9`
`04:39.57`	`hads`	`JT: Yeah, but of course there are weird bits that it doesn't seem to cover :/`
`04:40.04`	`connecta`	`what an ass, i thought i had astersik 1.4 on this box and it was 1.2....`
`04:40.17`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)`
`04:40.31`	`k-man`	`jt, ah, thats cool`
`04:40.32`	`k-man`	`thanks`
`04:40.47`	`joelsolanki`	`JT: u there ?`
`04:40.59`	`k-man`	`jt, weird, it looks corrupt to me, from chapter1 onwards`
`04:41.04`	`JT`	`joelsolanki: yes, for once you're not trying to contact me at 4am :)`
`04:41.15`	`joelsolanki`	`hehe`
`04:41.20`	`hads`	`JT: I'm waiting to see what happens as number portability is (suppost to be) coming in sometime this year.`
`04:41.22`	`JT`	`k-man: hmm weird, maybe your browser can't handle such a large document`
`04:41.34`	`k-man`	`jt, no, its more like an encoding issue`
`04:41.41`	`k-man`	`strange... you don;t have any problem your end?`
`04:41.42`	`JT`	`hads: i guess it doesn't need to be documented without a deregulated telecommunications market :P`
`04:41.44`	`joelsolanki`	`I am trying to PM. u JT`
`04:41.56`	`hads`	`:)`
`04:42.00`	`JT`	`k-man: yes, some weird A characters appearing`
`04:42.06`	`k-man`	`jt, yeah`
`04:42.07`	`k-man`	`thats it`
`04:42.14`	`[TK]D-Fender`	`etfonhomey : you get sample configs with the firmware provisioning packs`
`04:42.18`	`JT`	`didn't happen like that last time`
`04:42.22`	`JT`	`what browser?`
`04:42.28`	`k-man`	`firefox`
`04:42.37`	`JT`	`hads: any word on if it will be opened to competition?`
`04:42.45`	`JT`	`k-man: yeah me too, last time it worked, i was using IE`
`04:43.10`	`k-man`	`jt, yeah, works in IE`
`04:43.11`	`k-man`	`typical`
`04:44.02`	`hads`	`JT: In what way?`
`04:45.04`	`JT`	`hads: the telecommunications industry`
`04:45.35`	`*** part/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com)`
`04:46.06`	`hads`	`OK, I thought you were referring to something specific. Yes, Local loop unbundling has been mandated and is due to take effect soon. I don't really know anything about number portability yet.`
`04:46.56`	`hads`	`I believe it's going to involve fixed to mobile and mobile to fixed which is interesting.`
`04:47.18`	`JT`	`what about anyone just setting up a telco?`
`04:47.25`	*** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net)
`04:47.32`	`rue_mohr`	`ok, mouse cord is now a terminal cord`
`04:47.45`	*** join/#asterisk ez` (n=supermar@c207.134.229-230.clta.globetrotter.net)
`04:48.01`	`rue_mohr`	`hah, no serial drivers eh?`
`04:48.37`	`hads`	`Well, if they have the cash I don't see why not. iHug (recently purchased by Vodafone) have plans to do something.`
`04:48.54`	`k-man`	`jt, i tried to send them feedback about it, but the submit button on their feedback page is broken`
`04:49.01`	`JT`	`ihug was purchased by vodafone?`
`04:49.11`	`JT`	`does that mean ihug australia is a completely seperate entity?`
`04:49.23`	`hads`	`Yeah I think it might be.`
`04:49.57`	`JT`	`hads: i believe there are only 10 licensed telecommunications carriers in NZ, and they're special cases, not a standard process for anyone to apply`
`04:50.19`	`JT`	`well ihug in australia was bought by iiNet, which is now the 2nd or 3rd largest isp in .au`
`04:50.24`	`JT`	`they bought up dozens of isps`
`04:50.36`	`hads`	`They must be seperate then`
`04:51.16`	`JT`	`iinet bought up ozemail, which was australia's biggest isp 10 years ago`
`04:51.57`	`rue_mohr`	`yay, its talking to me`
`04:52.03`	`hads`	`JT: I don't know that much about the industry really.`
`04:52.39`	`JT`	`i think it's still a closed industry in NZ`
`04:52.49`	`JT`	`do you have voip companies yet?`
`04:52.52`	`hads`	`I used to run a 40 person office's mail server over an Ozemail dialup connection :)`
`04:52.54`	`JT`	`offering service to consumers`
`04:52.58`	`JT`	`heh`
`04:53.21`	`hads`	`There's a couple. It's starting to grow now.`
`04:53.49`	`JT`	`are they legal?`
`04:53.56`	`JT`	`did they need to get a special licence?`
`04:54.09`	`hads`	`Yeah they're legal, don't know about the licensing.`
`04:54.33`	`hads`	`http://www.xnet.co.nz/vfx/`
`04:55.53`	`JT`	`hmm ok`
`04:55.59`	`JT`	`still a small industry atm?`
`04:56.36`	`hads`	`Yeah, small but growing`
`04:57.29`	`JT`	`hmm`
`04:58.34`	`hads`	`The main problem (I imagine) for providers over here is all the small/tiny places so they have loads of areas to get DIDs for which aren't going to get a very good return because of the small population of each area.`
`04:59.50`	`JT`	`yeah, i really have no idea how some voip providers do it here`
`05:00.15`	`JT`	`quite a lot have 10c nationwide untimed calls, i don't think even the biggest provider would have DIDs everywhere`
`05:00.28`	`JT`	`i wonder if they just buy wholesale off telstra sometimes`
`05:01.18`	`hads`	`Yeah`
`05:01.21`	`*** join/#asterisk BOLIVIAN (n=klkl@201.222.98.226)`
`05:01.59`	`JT`	`nz needs laws to force competitors to get wholesale rates too :)`
`05:04.24`	`hads`	`Yeah, they are doing something to that effect in the ISP area, don't know about the telecomms side of it though`
`05:04.55`	`JT`	`hmm`
`05:06.06`	`*** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net)`
`05:09.52`	`kuku5`	`Rejected connect attempt from 66.225.202.80, who was trying to reach 's@' I have the ip registered in iax.conf`
`05:11.47`	`kuku5`	`and I have s,1,... in extensions.conf`
`05:12.21`	`JT`	`you should make the destination have a context`
`05:12.31`	`JT`	`asterisk doesn't like iax calls with no context`
`05:12.39`	`*** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net)`
`05:14.24`	`kuku5`	`ok`
`05:14.37`	`kuku5`	`but its a call im getting in, i have no control over it`
`05:16.28`	`kuku5`	`Anything I can do ?`
`05:19.06`	`JT`	`i don't think so, not that i know of`
`05:20.22`	`*** join/#asterisk apardo (n=apardo@87.217.145.181)`
`05:20.52`	`rue_mohr`	`if the LGS has a line reversal option it must be an FXS right?`
`05:21.25`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`05:23.30`	`*** join/#asterisk etfonhomey1 (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`05:23.54`	`etfonhomey1`	`What's the best residential VoIP provider?`
`05:24.25`	`JT`	`on the planet?`
`05:24.40`	`etfonhomey1`	`For the US, I guess.`
`05:25.03`	`JT`	`a good qualifier`
`05:25.26`	`BOLIVIAN`	`let say i will use asteriks to rent my land line to the public, is there a way to put some led visor to the client in order for them to see the call duration so far?`
`05:26.42`	`etfonhomey1`	`Any suggestions?`
`05:35.29`	`rue_mohr`	`if my T100P drivers are working, dispite asterisk not being running, I should get a red alarm shouldn't I?`
`05:35.36`	`*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`05:36.10`	`JT`	`umm probably`
`05:36.35`	`rue_mohr`	`cause I'm not getting a red alarm, and I dont know if it means my drivers aren't working`
`05:37.24`	`mog`	`your card can sync up without asterisk`
`05:37.27`	`*** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net)`
`05:37.42`	`rue_mohr`	`so i should see a red alarm`
`05:37.45`	`rue_mohr`	`:/`
`05:37.59`	`tim0123`	`How do you set MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMESTAMP}-${UNIQUEID}','','0') in a agi script`
`05:38.01`	`mog`	`zttool`
`05:38.05`	`JT`	`if the drivers are up`
`05:40.37`	`bkruse_home`	`<3 zttool`
`05:41.18`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`05:41.44`	`rue_mohr`	`Looping UP span 1...`
`05:41.52`	`rue_mohr`	`after whihc I should see...?`
`05:44.33`	`rue_mohr`	`something is wrong here`
`05:44.48`	`rue_mohr`	`I was supposed to start cooking supper 17 mins ago`
`05:45.01`	`rue_mohr`	`and I'm quite certain I smell nothing burning`
`05:46.37`	`*** join/#asterisk topping (n=topping@adsl-67-127-52-31.dsl.pltn13.pacbell.net)`
`05:47.26`	`rue_mohr`	`maybe I have zaptel.conf wrong`
`05:48.40`	`putzz`	`is it possible to make asterisk announce when I lift the hook on the phone it will notify me?`
`05:48.50`	`k-man`	`jt, why do you not need an _ for your 000 dialplan?`
`05:48.52`	`putzz`	`when I have messages`
`05:49.30`	`JT`	`k-man: it's not a pattern`
`05:49.37`	`JT`	`it is an extension that matches one number`
`05:49.49`	`k-man`	`jt, oh.. i see`
`05:49.52`	`bkruse_home`	`rue_mohr: can you pb your zaptel.conf real quick?`
`05:49.53`	`bkruse_home`	`~pb`
`05:49.55`	`jbot`	`pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`05:50.15`	`k-man`	`so it would be followed by a "dial(000...)"?`
`05:50.56`	`JT`	`it could, ${EXTEN} would also work`
`05:51.22`	`rue_mohr`	`http://channels.debian.net/paste/5359`
`05:51.31`	`rue_mohr`	`I didn't bother with empty lines`
`05:51.42`	`bkruse_home`	`and this is to a telco or what?`
`05:51.52`	`bkruse_home`	`channel bank?`
`05:51.55`	`rue_mohr`	`no a newbridge channelbank`
`05:52.00`	`rue_mohr`	`3624`
`05:52.03`	`bkruse_home`	`gotcha`
`05:52.11`	`bkruse_home`	`and whos providing timing`
`05:52.20`	`rue_mohr`	`I have _NO_ lights on the T100P though`
`05:52.27`	`bkruse_home`	`did you load the driver?`
`05:52.31`	`rue_mohr`	`its prefered for the channelbank t provide timing`
`05:52.40`	`bkruse_home`	`gotcha`
`05:52.45`	`rue_mohr`	`wct1xxp 11616 0`
`05:52.45`	`rue_mohr`	`zaptel 173668 1 wct1xxp`
`05:53.03`	`bkruse_home`	`rmmod wct1xxp && modprobe wct1xxp && ztcfg -vv`
`05:53.11`	`bkruse_home`	`and clip me the end of dmesg`
`05:54.14`	`rue_mohr`	`ooo green light`
`05:54.15`	`bkruse_home`	`:]`
`05:54.15`	`rue_mohr`	`whats new`
`05:54.15`	`rue_mohr`	`oh ztcfg`
`05:54.15`	`mog`	`bkruse_home, !`
`05:54.15`	`rue_mohr`	`whats that`
`05:54.15`	`rue_mohr`	`hmm things are clicking`
`05:54.15`	`bkruse_home`	`ztcfg reads in your /etc/zaptel.conf and configures your device for what you have set there`
`05:54.15`	`bkruse_home`	`thats SOMETIMES good`
`05:54.15`	`bkruse_home`	`mog!`
`05:54.15`	`tim0123`	`anyone no anything about recording using monitor`
`05:54.20`	`rue_mohr`	`oh my`
`05:54.24`	`*** part/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`05:54.33`	`bkruse_home`	`rue_mohr: your welcome!`
`05:54.36`	`rue_mohr`	`thankyou!`
`05:54.44`	`rue_mohr`	`thats just doubled todays progress :)`
`05:54.54`	`bkruse_home`	`rue_mohr: no problem, just remember ztcfg, usually, the modprobe config ztcfg's for you`
`05:54.58`	`bkruse_home`	`but sometimes its TO early`
`05:55.04`	`bkruse_home`	`for grins, what OS you running?`
`05:55.16`	`rue_mohr`	`debina`
`05:55.25`	`rue_mohr`	`its a kind of debian :)`
`05:55.58`	`bkruse_home`	`interesting.......`
`05:56.11`	`rue_mohr`	`now I'm gonna need to work out which of the LGS and LGE modules are for FXO and FXS`
`05:56.13`	`bkruse_home`	`i bet right after you modprobed zaptel and the card, and did ls /dev/zap it wouldnt exist quite yet`
`05:56.19`	`bkruse_home`	`ha`
`05:56.37`	`rue_mohr`	`:)`
`05:56.50`	`bkruse_home`	`so sometimes you gota throw down ztcfg yourself`
`05:57.06`	`bkruse_home`	`alot of people think that this means their card is broke, or asterisk isnt working. Leads to alot of confusion`
`05:57.10`	`bkruse_home`	`spread the word though :D`
`05:57.16`	`rue_mohr`	`yea!`
`05:58.02`	`mog`	`bkruse_home, for the win`
`05:58.28`	`rue_mohr`	`<PROTECTED>`
`05:58.29`	`bkruse_home`	`mog + erlang for the win.....`
`05:58.35`	`bkruse_home`	`rue_mohr: :]`
`05:58.37`	`mog`	`heh`
`05:59.07`	`bkruse_home`	`no wait, i take that back mog, mog + spidaphone + meetme for the win`
`05:59.37`	`mog`	`heh spidaphone and i where hangingout this weekend`
`05:59.49`	`bkruse_home`	`dangit, no one called me`
`06:00.01`	`bkruse_home`	`actually, i shoulda called in, my mistake`
`06:00.25`	`osiris`	`phones suck. jk`
`06:00.27`	`mog`	`exactly`
`06:01.53`	`mog`	`man im bored`
`06:01.57`	`mog`	`maybe i should sleep`
`06:02.01`	`bkruse_home`	`mog: back to work!`
`06:02.21`	`bkruse_home`	`mog: i need to sleep also, i got a pre-cal test tomorrow, GAH`
`06:02.35`	`mog`	`you think thats hard`
`06:02.38`	`rue_mohr`	`oh I know aht build out is now, I shoudl set that to 15`
`06:02.41`	`osiris`	`anyone know the default login to a pap2 or sipura 2100 by chance ?`
`06:02.43`	`mog`	`i have a post-cal test yesterday`
`06:03.02`	`FuriousGeorge`	`i am having a terrible week so far... had an array go degraded, took the box home over the weekend, redid array, reinstalled os and *, take it back two hours ago`
`06:03.09`	`FuriousGeorge`	`the damn thing wont post`
`06:03.19`	`k-man`	`jt, so what number ranges do you use for extensions with that dial plan you suggest?`
`06:03.53`	`bkruse_home`	`FuriousGeorge: </3 post`
`06:04.14`	`JT`	`k-man: not sure, up to you, a lot of offices use 0 to dial an outside line`
`06:04.22`	`osiris`	`or 9`
`06:04.42`	`FuriousGeorge`	`9 is what i always say`
`06:05.11`	`bkruse_home`	`9`
`06:05.27`	`k-man`	`in aus, it is usually 0`
`06:05.45`	`k-man`	`but dialing any prefix to get a line seems a bit redundant to me, at least it is with asterisk?`
`06:06.01`	`bkruse_home`	`you can do whatever you want`
`06:06.02`	`osiris`	`depends on the deployment`
`06:06.08`	`osiris`	`what the needs are`
`06:06.11`	`FuriousGeorge`	`dialing 9 is so 1995`
`06:06.18`	`k-man`	`oh.. maybe you need to to differentiate between say internal extensions and outside lines?`
`06:06.29`	`k-man`	`FuriousGeorge, so what do you suggest?`
`06:06.36`	`bkruse_home`	`pickup the phone and prompt for a lumenvox voice recongnition for outside line. Make another number do an outside line, dont do outside lines and make extensions >5 go to the pstn or voip provider`
`06:06.40`	`osiris`	`or calls for faxing that go to an fxo port`
`06:06.41`	`bkruse_home`	`whatever you want`
`06:07.01`	`bkruse_home`	`usually, you wont have an extension to a phone thats 7 numbers, but who knows, thats up to you to decide!`
`06:07.43`	`FuriousGeorge`	`k-man: i do 1XX for the same building`
`06:07.45`	`osiris`	`i still havent done an asterisk myself, but i know a little about voip`
`06:08.02`	`FuriousGeorge`	`9 or 10 digit dialing`
`06:08.23`	`osiris`	`telephony in general is the newish part`
`06:08.26`	`FuriousGeorge`	`i allow them to dial 7 digits and prefix an area code for them`
`06:08.42`	`bkruse_home`	`FuriousGeorge: yep, good idea`
`06:08.52`	`k-man`	`FuriousGeorge, thats a good idea`
`06:09.39`	`FuriousGeorge`	`k-man: dialing 9 was for when you needed to allow a dumb mechanically switched system to physically connect you to an outside line`
`06:10.06`	`k-man`	`FuriousGeorge, so whats the 21st century option?`
`06:10.17`	`FuriousGeorge`	`k-man: what i said before`
`06:10.20`	`FuriousGeorge`	`just dial`
`06:10.34`	`FuriousGeorge`	`let * decide what to do`
`06:10.37`	`k-man`	`FuriousGeorge, so how to distinguish between internal and external?`
`06:10.45`	`k-man`	`by the number of digits?`
`06:10.49`	`FuriousGeorge`	`internal are 3 to 4 digits`
`06:10.54`	`k-man`	`ok`
`06:10.55`	`k-man`	`i see`
`06:11.44`	`JT`	`yeah 0 is the standard in australia, not 9`
`06:12.40`	`osiris`	`9 seems to be standard in the state`
`06:12.51`	`osiris`	`er states`
`06:13.34`	`*** part/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`06:14.34`	`rue_mohr`	`LGE MODULE (+4/-10DB) 2 FXO circuits per module ahahaha google answers it!`
`06:15.25`	`rue_mohr`	`ok`
`06:15.47`	`FuriousGeorge`	`yeah, im familiar with 9. k-man: i know of people in areas where there is only 1 local exchange who let their users dial only 4 digits. if it doesnt match an internal extension, they prepend 3 digits for the area code and 3 for the exchange`
`06:15.55`	`rue_mohr`	`so I want 1 LGE module and 3 LGS modules`
`06:16.11`	`k-man`	`interesting`
`06:16.15`	`k-man`	`thanks FuriousGeorge`
`06:16.46`	`*** join/#asterisk zeeesh (i=zeeesh@202.38.55.125)`
`06:17.17`	`zeeesh`	`hi`
`06:22.41`	`k-man`	`do i need to reload the voicemail.conf file after modifying it?`
`06:23.07`	`*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`06:23.16`	`rue_mohr`	`I beleive so`
`06:23.22`	`rue_mohr`	`asterisk -r`
`06:23.24`	`rue_mohr`	`reload`
`06:23.24`	`k-man`	`how do i reload it?`
`06:23.25`	`k-man`	`oh`
`06:23.31`	`rue_mohr`	`quit`
`06:23.39`	`rue_mohr`	`but its been a while`
`06:23.52`	`FuriousGeorge`	`you may need to reload res_features or something`
`06:23.56`	`FuriousGeorge`	`use tab complete`
`06:25.23`	`rue_mohr`	`I really wish that system status on the newbridge was green and not red`
`06:25.45`	`rue_mohr`	`653 pages, wheres the info on that damned light`
`06:32.01`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net)`
`06:39.56`	`*** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net)`
`06:44.21`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`06:47.33`	`*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)`
`06:49.08`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`06:54.40`	`joelsolanki`	`anybody using asterisk as large ?`
`06:54.54`	`putzz`	`as large?`
`06:55.24`	`joelsolanki`	`yes means. in large environment`
`06:55.40`	`joelsolanki`	`means i have around 300 concurrent g729 calls.`
`06:55.58`	`J4k3`	`wow`
`06:56.00`	`joelsolanki`	`i want to know how people manage 300 concurrent g729 calls using asterisk`
`06:56.01`	`J4k3`	`transcoding?`
`06:56.33`	`joelsolanki`	`No. i dont have 300 concurrent calls on asterisk. it is on different software.`
`06:56.42`	`J4k3`	`ah`
`06:56.44`	`joelsolanki`	`i want to use asterisk for this calls therefore.`
`06:56.47`	`J4k3`	`ahh`
`06:56.55`	`mog`	`you can do it`
`06:57.02`	`joelsolanki`	`how ?`
`06:58.02`	`joelsolanki`	`i want to remove that software and use asterisk for 300 concurrent calls.`
`06:58.09`	`joelsolanki`	`what would u suggest ?`
`06:58.29`	`mog`	`beefy pc + asterisk`
`06:59.05`	`joelsolanki`	`means ?`
`06:59.14`	`mog`	`means?`
`07:02.56`	`putzz`	`"a decent server"`
`07:02.59`	`putzz`	`lol`
`07:05.15`	`mog`	`gnite`
`07:05.21`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`07:05.46`	`JT`	`joelsolanki: dual quad core xeon machine, running xen, two of those machines, would go well :D`
`07:06.14`	`JT`	`you could run at least 6 seperate asterisk vms on each one`
`07:06.27`	`JT`	`with decent performance, i would think`
`07:06.58`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`07:12.06`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`07:12.06`	`*** mode/#asterisk [+o denon] by ChanServ`
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`07:17.10`	`joelsolanki`	`hmm.`
`07:17.38`	`JT`	`or even dual core xeons`
`07:17.47`	`joelsolanki`	`hmm ok.`
`07:17.50`	`JT`	`quad cores may not be warranted for the price`
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`07:18.04`	`JT`	`well they're actually pretty competitively priced`
`07:18.15`	`JT`	`the problem is lack of existing benchmarking the the VM arena`
`07:18.24`	`joelsolanki`	`ok.`
`07:18.56`	`JT`	`a lot of less parallel applications, the quad core xeons do not quite as well as dual cores due to software being unable to work that many cores well`
`07:19.46`	`joelsolanki`	`hm`
`07:20.06`	`J4k3`	`but with VMs, parallelization of your software isn't as much required.`
`07:20.48`	`JT`	`indeed`
`07:20.59`	`JT`	`but not much testing has been done with VMs + quad cores`
`07:21.13`	`J4k3`	`virtualization doesn't absolutely require complete machine cloning. There are lots of slick little kernel hacks to share a single kernel between several different security systems/file systems/etc.`
`07:21.19`	`J4k3`	`yeah.`
`07:21.29`	`J4k3`	`quad cores also have limited memory I/O, at least in every situation I've seen`
`07:24.21`	`JT`	`yeah, xeons anyway, poor choice in memory architecture by intel`
`07:24.40`	`J4k3`	`does AMD have a quad out yet`
`07:24.40`	`J4k3`	`?`
`07:24.51`	`*** join/#asterisk ModocNet (n=d82e036a@69-12-147-8.dsl.static.sonic.net)`
`07:24.59`	`J4k3`	`I know they've got dual-duals`
`07:25.34`	`JT`	`i don't think so`
`07:25.35`	`JT`	`also`
`07:25.42`	`JT`	`the current "quad core" xeons`
`07:25.45`	`JT`	`aren't truly quad`
`07:25.53`	`JT`	`they're 2 dual core dies next to each other`
`07:26.07`	`J4k3`	`yeah`
`07:26.08`	`ModocNet`	`I know I must be forgetting something pretty simple...GXP-2000 and * 1.2.15 - phone's MWI never comes on....I have Subscribe MWI turned on...and I have called phone and left three messages`
`07:26.33`	`ModocNet`	`I can also press MSG and login to the Phones VM`
`07:27.02`	`ModocNet`	`so extensions.conf and voicemail.conf for configured correct to leaving and retreiving messages`
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`07:43.02`	`rue_mohr`	`well, I seem to have cleared the last of the errors from the newbridge`
`07:43.14`	`rue_mohr`	`little hardware swapping required`
`07:43.22`	`rue_mohr`	`its almost midnight`
`07:43.27`	`rue_mohr`	`have to get up at 6:30`
`07:43.37`	`JT`	`coo`
`07:43.39`	`JT`	`l`
`07:43.41`	`rue_mohr`	`i have to stop playing phone system and go to bed`
`07:43.50`	`JT`	`na it's fun`
`07:43.50`	`rue_mohr`	`I just dont want to`
`07:43.54`	`rue_mohr`	`:)`
`07:44.53`	`rue_mohr`	`well, I suppose next I would configure zapata.conf...`
`07:45.15`	`JT`	`i suppose`
`07:46.31`	`rue_mohr`	`:) it would help if I could see straight...`
`07:46.39`	`rue_mohr`	`possibly bedtime`
`07:46.45`	`JT`	`perhaps`
`07:46.51`	`JT`	`i could give you my zapata.conf`
`07:46.57`	`JT`	`but that'd be too easy`
`07:47.19`	`rue_mohr`	`I'm just looking over whats in it`
`07:47.39`	`JT`	`i have one already setup for a channel bank`
`07:47.46`	`rue_mohr`	`it actually dosn't lokk like theres anything to configure`
`07:48.01`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`07:48.03`	`JT`	`depends if it's already done`
`07:48.18`	`*** join/#asterisk hegars (n=hegars@203.161.78.66.static.amnet.net.au)`
`07:48.23`	`hegars`	`hi everyone`
`07:48.32`	`rue_mohr`	`hi`
`07:48.40`	`dj-fu`	`doh`
`07:48.44`	`dj-fu`	`I just did rm zapata.conf`
`07:48.46`	`dj-fu`	`on ext3`
`07:48.52`	`dj-fu`	`rage`
`07:49.19`	`rue_mohr`	`dosn't ext3 have a rewind?`
`07:50.09`	`rue_mohr`	`I dotn see anything in zapata.conf that I see needing to change`
`07:50.33`	`dj-fu`	`oh bastard. I was supposed to type nano. I hate that`
`07:50.44`	`hegars`	`ive got a 1.4 installation that chewing 100% of the cpu, any ideas what could cause this`
`07:50.55`	`rue_mohr`	`dj-fu, :/ cant help dude`
`07:51.16`	`rue_mohr`	`dj-fu, use the backup you made before you were about to edit it`
`07:51.21`	`rue_mohr`	`?`
`07:51.23`	`dj-fu`	`aha!`
`07:51.27`	`dj-fu`	`I forgot that I backup nightly`
`07:51.37`	`rue_mohr`	`wise man`
`07:52.07`	`dj-fu`	`oh god - what a save`
`07:52.32`	`dj-fu`	`can anyone tell me about channel grouping?`
`07:52.45`	`dj-fu`	`I'm trying to group my two fxo ports so that if one is in use it automaticaly dials out on the other`
`07:52.59`	`rue_mohr`	`signalling=fxo_ls shouldn't that be koolstart?`
`07:53.12`	`dj-fu`	`unless you're in a place that uses loopstart, yeah`
`07:53.14`	`dj-fu`	`usually`
`07:53.35`	`rue_mohr`	`hah, how the heck do I know that`
`07:53.53`	`rue_mohr`	`phone telus and ask?`
`07:53.57`	`rue_mohr`	`haha`
`07:54.00`	`dj-fu`	`try them`
`07:54.03`	`*** join/#asterisk topping (n=topping@204.152.96.238)`
`07:54.19`	`rue_mohr`	`youdont know how many hours it would take to get to a human`
`07:54.30`	`dj-fu`	`no, i mean try all the types of signalling`
`07:54.35`	`dj-fu`	`ks worked on my first try`
`07:54.39`	`rue_mohr`	`ok`
`07:54.56`	`rue_mohr`	`the zaptel.conf instructions said to go with ks`
`07:56.08`	`JT`	`yeah, ks, otherwise ls`
`07:56.15`	`JT`	`ks is like enhanced ls`
`07:56.25`	`JT`	`and remember FXO PORTS USE FXS SIGNALLING! :D`
`07:56.49`	`rue_mohr`	`:)`
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`07:58.02`	`rue_mohr`	`I'm looking at options like threewaycalling=yes and thinking that this isn't a plausable conifg for an T100 card`
`07:58.02`	`dj-fu`	`JT, any ideas about my zap grouping thing? It doesn't wanna work`
`07:58.25`	`JT`	`yes, add a group directive before the relevant channels are listed in zapata.conf`
`07:58.28`	`rue_mohr`	`you want ot do groups for ringing and things?`
`07:59.10`	`rue_mohr`	`what kinda line interfaces are you using?`
`07:59.32`	`dj-fu`	`before the channels. i see`
`07:59.38`	`dj-fu`	`http://rafb.net/p/Sm28hV32.html`
`07:59.39`	`JT`	`dj-fu: zaptel?`
`07:59.41`	`dj-fu`	`I had after`
`07:59.42`	`dj-fu`	`JT, yea`
`08:00.01`	`dj-fu`	`rue_mohr, no, a zap group so that I might dial with Zap/g1/NUMBER and it uses either available line`
`08:00.14`	`JT`	`dj-fu: do not use 2 group directives`
`08:00.15`	`dj-fu`	`i did a cheap pbx by getting two lines and having them cascade for incoming calls`
`08:00.24`	`rue_mohr`	`its midnight, I have to go to sleep`
`08:00.27`	`rue_mohr`	`sorry folks`
`08:00.30`	`JT`	`bye`
`08:00.34`	`dj-fu`	`JT, then how? group=1,2,3?`
`08:00.34`	`*** part/#asterisk rue_mohr (n=rue@h24-207-91-121.cst.dccnet.com)`
`08:00.42`	`JT`	`group=1`
`08:00.47`	`JT`	`channel=1`
`08:00.49`	`JT`	`blah blah`
`08:00.53`	`JT`	`channel=2`
`08:00.54`	`JT`	`blah`
`08:01.29`	`dj-fu`	`i see`
`08:01.31`	`dj-fu`	`great, thanks`
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`08:25.44`	`quelo`	`Hi to all`
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`08:29.29`	`quelo`	`I have a problem: this is my sip.conf http://paste.debian.net/21771`
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`08:31.42`	`quelo`	`this is my sip_additional.conf http://paste.debian.net/21772`
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`08:32.55`	`quelo`	`if I call the extension 100 from another internal extension the voice answear me in Italian but if I call one of the number that the providers gave to me the extension 100 answear to me in english`
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`08:34.34`	`quelo`	`I have a problem: this is my sip.conf http://paste.debian.net/21771`
`08:34.37`	`quelo`	`if I call the extension 100 from another internal extension the voice answear me in Italian but if I call one of the number that the providers gave to me the extension 100 answear to me in english`
`08:34.49`	`quelo`	`this is my sip_additional.conf http://paste.debian.net/21772`
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`08:38.37`	`*** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com)`
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`08:52.00`	`*** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net)`
`08:52.19`	`kippi`	`where is the best place to get support?`
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`09:04.08`	`joelsolanki`	`is there any software to stimulate calls ?`
`09:04.12`	`joelsolanki`	`generate calls ?`
`09:04.27`	`quelo`	`I have a problem: this is my sip.conf http://paste.debian.net/21771`
`09:04.33`	`quelo`	`this is my sip_additional.conf http://paste.debian.net/21772`
`09:05.23`	`quelo`	`if I call the extension 100 from another internal extension the voice answears to me in Italian but if I call from external one of the number that the providers gave to me the extension 100 answear to me in english`
`09:06.36`	`mafkees`	`tzafrir_laptop: no, I wasnt here anymore`
`09:06.39`	`mafkees`	`but now I am :)`
`09:08.44`	`*** join/#asterisk florz (i=nobody@2002:58c6:2592:1:0:0:0:2)`
`09:11.47`	`kippi`	`what is the best way off getting a user to be able to program in there own diverts?`
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`10:23.32`	`yxa`	`can i dial 3 extensions at one time so that all 3 phones ring?`
`10:25.55`	`qdk`	`yxa: yes.`
`10:26.16`	`qdk`	`"asterisk cmd dial"`
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`10:33.29`	`*** join/#asterisk penguinFunk (n=penguin@87.224.86.46)`
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`10:45.28`	`Chris-NB`	`hi`
`10:45.35`	`Chris-NB`	`anyone using flash operator panel?`
`10:47.10`	`Ahrimanes`	`yes`
`10:48.21`	`Chris-NB`	`Ahrimanes, I've stange problems. I've configured about 10 extensions`
`10:48.36`	`Chris-NB`	`Ahrimanes, I can see if there is a call, can see the call history and everything is fine`
`10:48.43`	`yxa`	`<PROTECTED>`
`10:48.57`	`Chris-NB`	`Ahrimanes, but when I try to place a call from fop I get various .. phenomens.`
`10:49.07`	`Chris-NB`	`1. the dialed extension is empty`
`10:49.16`	`Chris-NB`	`2. the dial extension contains SIP/130`
`10:49.25`	`Chris-NB`	`3. the dial extensions contains only 130`
`10:49.48`	`Chris-NB`	`the 3. case is the one which should always happen, but happens not that often`
`10:50.10`	`Chris-NB`	`case 1 and 2 fail and are most likely to occur`
`10:50.24`	`Chris-NB`	`Ahrimanes, you discovered that behavior?`
`10:50.31`	`Chris-NB`	`I can supply configs if you need`
`10:50.47`	`Ahrimanes`	`Chris-NB, hm, well, we dont use it to place calls`
`10:51.01`	`Chris-NB`	`Ahrimanes, ok. : /`
`10:55.31`	`Chris-NB`	`Ahrimanes, ok, I found my ... error. I was my fault : /`
`10:56.01`	`Chris-NB`	`Ahrimanes, there is a difference if you drag the caller to the callee button oder to the mailbox incon of the callee : //`
`10:58.46`	`Ahrimanes`	`hehe`
`11:13.42`	`drako`	`Good morning`
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`11:18.36`	`nfi\|ermes`	`does mISDN need also zaptel ?`
`11:19.35`	`nfi\|ermes`	`i suppose yes`
`11:19.57`	`drako`	`Why Asterisk is creating the voicemail only with user privileges, i need them to be written with user and group`
`11:23.29`	`drako`	`where i can set asterisk umask ?`
`11:26.38`	`*** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za)`
`11:26.42`	`Nobbie`	`heya =)`
`11:27.20`	`drako`	`Nobbie, hi`
`11:28.25`	`Nobbie`	`in a call centre with 5 dynamic agents, how would you indicate to them on their IP phones, wether they're logged in to a queue or not ?`
`11:30.11`	`Nobbie`	`and how many calls are in the queue, how long they've been waiting, etc`
`11:32.22`	`Ahrimanes`	`big screen projector`
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`11:51.47`	`Z-Vi0latR`	`Hey guys, has anyone had any issues with Zaptel 1.4 and TE410P's causing the system to reboot on loading the modulbes?`
`11:51.50`	`Z-Vi0latR`	`modules.`
`11:52.03`	`Z-Vi0latR`	`Its running on a CentOS 4.4 system`
`11:52.07`	`ManxPower`	`Sorry, I don't use 1.4`
`11:52.22`	`ManxPower`	`Z-Vi0latR: this does not happen with 1.2?`
`11:52.43`	`Z-Vi0latR`	`I haven't tried 1.2 yet... I was hoping to go for the latest release on this build.`
`11:52.53`	`Z-Vi0latR`	`Is 1.4 clasified as stable yet?`
`11:53.00`	`ManxPower`	`Z-Vi0latR: I would call 1.4.0 "beta".`
`11:53.09`	`ManxPower`	`Digium does not call it beta, but I do.`
`11:53.26`	`Z-Vi0latR`	`Hmmm... I'll give that a go then :)`
`11:53.41`	`ManxPower`	`Z-Vi0latR: Make sure the card is not sharing IRQs`
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`11:54.23`	`Z-Vi0latR`	`Will that cause a reboot?`
`11:54.48`	`ManxPower`	`Z-Vi0latR: IRQ sharing can cause blackholes and impotence.`
`11:54.55`	`ManxPower`	`So I can imagine where it could cause a reeboot.`
`11:55.15`	`Z-Vi0latR`	`I'll take a look and see what I can test.`
`11:55.23`	`ManxPower`	`"cat /proc/interrupts"`
`11:55.54`	`Z-Vi0latR`	`Will that show the card without the modules loaded?`
`11:56.17`	`ManxPower`	`ah, sorry.`
`11:56.21`	`ManxPower`	`lspci -v`
`12:01.46`	`Z-Vi0latR`	`ManxPower: Its usint IRQ 209 and is the only device listed with that IRQ`
`12:01.50`	`Z-Vi0latR`	`using`
`12:04.10`	`ManxPower`	`that should be fine`
`12:04.28`	`Z-Vi0latR`	`Oh well I guess I'm rolling back a version :)`
`12:05.02`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`12:09.35`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`12:19.35`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`12:22.47`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`12:23.49`	`*** join/#asterisk cratz (n=cratz@adsl-71-159-206-2.dsl.pltn13.sbcglobal.net)`
`12:29.44`	`*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)`
`12:32.38`	`*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)`
`12:35.24`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net)`
`12:39.14`	`*** join/#asterisk apardo (n=apardo@87.217.144.229)`
`12:43.40`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)`
`12:44.25`	`*** join/#asterisk hematitec (n=Tony@adsl-71-159-206-2.dsl.pltn13.sbcglobal.net)`
`12:54.35`	`*** join/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br)`
`12:54.46`	`Shazaum`	`somebody?`
`12:55.09`	`Nobbie`	`hi`
`12:55.19`	`Shazaum`	`hi`
`12:56.14`	`Shazaum`	`<PROTECTED>`
`12:57.54`	`*** join/#asterisk nas_lslsa (n=chatzill@85.75.137.146)`
`12:58.17`	`Z-Vi0latR`	`Hi, does anyone know how to disable APIC from GRUB?`
`12:58.54`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`13:03.35`	`Shazaum`	`Nobbie: http://pastebin.ca/351888`
`13:04.07`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`13:08.34`	`*** join/#asterisk florz (i=nobody@2002:58c6:2592:1:0:0:0:2)`
`13:09.17`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`13:09.31`	`mitcheloc`	`there are some damn nice phones coming out at 3GSM`
`13:09.39`	`mitcheloc`	`almost all of them have voip support, way to go for asterisk :)`
`13:12.38`	`mitcheloc`	`(starting with file and Qwell)`
`13:12.54`	`JT`	`Shazaum: that t1 pri or cas?`
`13:14.08`	`*** join/#asterisk Ahrimanes (n=ma@81.7.159.2)`
`13:14.26`	`Ahrimanes`	`anyone here have a sip_notify.conf for rebooting a cisco 7940 ?`
`13:14.42`	`*** join/#asterisk littleball (n=littleba@bb220-255-93-27.singnet.com.sg)`
`13:15.19`	`littleball`	`hello, i have a box which has zap E1. how to pass the incoming calls to another box which running asterisk SIP?`
`13:19.43`	`Shazaum`	`JT cas`
`13:21.32`	`*** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net)`
`13:21.35`	`*** join/#asterisk Aurs (n=Aurs@81.191.112.104)`
`13:22.53`	`JT`	`littleball: get them to Dial() the other box?`
`13:23.08`	`JT`	`Shazaum: hmm, how's your zaptel accuracy?`
`13:23.29`	`Shazaum`	`moment`
`13:27.34`	`littleball`	`JT, the user call ISDN number of the ZAP E1, the E1 had better not answer the call. instead, it just pass the incoming call to remote SIP`
`13:27.38`	`*** join/#asterisk evisu (i=hIRC@bzq-88-154-4-53.red.bezeqint.net)`
`13:27.40`	`littleball`	`is this possibel?`
`13:28.16`	`hads`	`Yes`
`13:28.32`	`Z-Vi0latR`	`Hi all, I was wondering if anyone has encountered rebooting after running the "ztcfg" program after you've installed your modules?`
`13:28.45`	`Z-Vi0latR`	`Is that related to IRQ?`
`13:28.50`	`Shazaum`	`JT: fxsks=1-4 loadzone=us defaultzone=us`
`13:28.52`	`Shazaum`	`:)`
`13:29.11`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`13:30.19`	`*** join/#asterisk yansolo90 (n=yann@par69-3-82-224-162-203.fbx.proxad.net)`
`13:30.21`	`hads`	`Shazaum: I assume JT meant accuracy as in zttest`
`13:30.54`	`Shazaum`	`ok`
`13:31.40`	`*** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net)`
`13:34.01`	`HarryR`	`Z-Vi0latR, we've had system hangs after running ztcfg (after initializing the modules), but only when using Linux-HA Heartbeat as well :(`
`13:34.45`	`tzafrir`	`Shazaum, what version exactly?`
`13:39.38`	`Shazaum`	`1.2.14`
`13:39.38`	`ManxPower`	`littleball: that is the DEFAULT`
`13:39.38`	`ManxPower`	`littleball: unless you do something that makes asterisk answer the line like playback or background, etc`
`13:39.38`	`tzafrir_laptop`	`Shazaum, I meant: what version of Zaptel?`
`13:39.39`	`tzafrir_laptop`	`Does the system hang? Do you see anything on the console?`
`13:39.39`	`Shazaum`	`ops`
`13:39.39`	`Shazaum`	`nops`
`13:39.39`	`Shazaum`	`zaptel is 1.2.12`
`13:40.03`	`Shazaum`	`" chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 "`
`13:40.34`	`ManxPower`	`Shazaum: this is harmless in my experience`
`13:40.42`	`ManxPower`	`happens all the time on zaptel systems`
`13:41.59`	`*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)`
`13:43.18`	`nas_lslsa`	`hello, I installed asterisk, I added a user at users.conf , and now I am trying to have access from web . I use the link : http://192.168.1.80:5080/as/manager?action=login&username=5000&secret=123 but I can't access ( I added a user at users.conf ) any ideas ?`
`13:43.29`	`littleball`	`after compilation, i cannot find that h323 channel .so file installed. Why?`
`13:44.13`	`ManxPower`	`nas_lslsa: That is because Asterisk does not have a web server included. Perhaps you mean you are using Asterisk GUI or AsteriskNOW.`
`13:44.21`	`*** join/#asterisk smace (n=smace_br@201.18.17.42)`
`13:44.33`	`ManxPower`	`littleball: what did you compile?`
`13:44.38`	`nas_lslsa`	`I use Asterisk Gui ..`
`13:44.54`	`nas_lslsa`	`<PROTECTED>`
`13:44.55`	`ManxPower`	`nas_lslsa: then look at the /topic and notice the AsteriskGUI specific channel.`
`13:45.13`	`nas_lslsa`	`wow , thanks ! :)*`
`13:46.26`	`*** join/#asterisk KnowWhat (n=KnowWhat@host210-2-166-243.isb.dancom.net.pk)`
`13:46.30`	`KnowWhat`	`Hello`
`13:46.56`	`KnowWhat`	`I need some installation notes for fc5`
`13:47.15`	`KnowWhat`	`for installing asterisk 1.4.0`
`13:47.27`	`ManxPower`	`KnowWhat: what specific issue hare you having?`
`13:47.52`	`smace`	`I am getting some errors here, and I've no idea why they are appearing here, see: We could NOT get the channel lock for SIP/testuser \| SIP transaction failed`
`13:48.11`	`*** join/#asterisk Telemac (n=p0369@213.223.113.74)`
`13:48.14`	`Telemac`	`Hello`
`13:49.04`	`Telemac`	`Has anyone ever used iaxclient to develop ?`
`13:49.18`	`*** join/#asterisk kuto (n=kuto@202.164.189.130)`
`13:49.52`	`kuto`	`hi all, can anyone help me, i planning to setup a voip with 100 seat and will be using sip server and g.729 codec, can anyone help how many e1 or t1 do i need, its an inbound and outbound call center`
`13:50.10`	`KnowWhat`	`ManxPower i installed asterisk on fc5`
`13:50.19`	`KnowWhat`	`i started it well`
`13:50.24`	`ManxPower`	`kuto: that depends on how many calls you need to support.`
`13:50.41`	`KnowWhat`	`now i edited sip.conf file to make an extension 1234 for xlite`
`13:50.48`	`ManxPower`	`kuto: your project will fail unless you build test systems to find out what issues you might have`
`13:51.03`	`KnowWhat`	`but when i try to login from xlite, it doesnt log in`
`13:51.23`	`ManxPower`	`KnowWhat: That is not a distro specific issue.`
`13:51.24`	`kuto`	`manpoer: my approximate is 100 agents concurrent usage`
`13:51.27`	`ManxPower`	`~book`
`13:51.38`	`jbot`	`[book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`13:51.38`	`ManxPower`	`kuto: do you need 100 calls at a time?`
`13:52.10`	`KnowWhat`	`ManxPower: i believe that book is for me ManxPower. am i right ?`
`13:52.13`	`ManxPower`	`KnowWhat: try The Book`
`13:52.18`	`kuto`	`ManxPower: yes`
`13:53.43`	`ManxPower`	`kuto: http://www.voip-info.org/wiki/view/Bandwidth+consumption`
`13:54.08`	`*** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan)`
`13:54.34`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`13:57.34`	`jamessan`	`I have a remote phone connecting to my asterisk (* -- firewall -- internet -- firewall -- remote phone). * has been configured to use a specific SIP port for that phone but it is instead trying to signal using the source port of the SIP traffic coming from the phone. any clues on how to fix this?`
`14:06.50`	`*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com)`
`14:07.55`	`ManxPower`	`jamessan: so you have a port=xxx line in the type=peer section for that device in sip.conf?`
`14:08.42`	`*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)`
`14:08.43`	`ManxPower`	`port= specifies the port to send traffic to on the remote phone. It does not specify the port to listen on`
`14:08.43`	`*** join/#asterisk vrm (n=vrm@94.55.101-84.rev.gaoland.net)`
`14:08.45`	`vrm`	`hi`
`14:09.26`	`jamessan`	`ManxPower: yes, that's what I have.`
`14:09.38`	`ManxPower`	`jamessan: paste your Dial line`
`14:09.41`	`jamessan`	`ManxPower: it's not sending SIP traffic to the remote phone on that port though`
`14:11.08`	`KnowWhat`	`how can i uninstall the current version of asterisk??`
`14:11.17`	`KnowWhat`	`i want to make it from the beginning`
`14:11.27`	`ManxPower`	`KnowWhat: did you try "make uninstall"`
`14:11.31`	`nas_lslsa`	`make uninstall ?`
`14:11.33`	`nas_lslsa`	`hahah`
`14:11.38`	`KnowWhat`	`let me try it`
`14:11.48`	`*** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr)`
`14:11.58`	`nas_lslsa`	`sorry again .. asterisk web GUI ... is it included in asterisk .tar ?`
`14:12.04`	`vgster`	`has anyone built 1.2.15 using suse 10.0 yet?`
`14:12.12`	`ManxPower`	`nas_lslsa: no ir is not`
`14:12.16`	`angryuser`	`good day, does anybody have feedback on astra 9133i phones?`
`14:12.22`	`KnowWhat`	`thanks done`
`14:13.32`	`*** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it)`
`14:13.33`	`angryuser`	`i need 9 ip phones not too much costly, with a good sound quality, and BLF(busy lamp field) support`
`14:13.36`	`giasai68`	`hello`
`14:13.43`	`robl^`	`angryuser: 9133i is solid, inexpensive... not a lot of features, a bit of a pain to set up.. but once they are set up, they are rock solid. Decent sound quality`
`14:13.47`	`ManxPower`	`angryuser: we use Polycom`
`14:13.50`	`giasai68`	`any suggest regarding VAD in asterisk?`
`14:14.00`	`Qwell`	`giasai68: disable vad`
`14:14.03`	`giasai68`	`VAD = Voice Activity Support`
`14:14.12`	`ManxPower`	`giasai68: no suggestions since Asterisk neither accepts nor sends vad`
`14:14.15`	`angryuser`	`robl^:Blf working fint with astra?`
`14:14.22`	`angryuser`	`*fine`
`14:14.40`	`giasai68`	`Qwell: can help me how I can disable it?`
`14:14.43`	`robl^`	`angryuser: BLF works if you use new firmware.`
`14:14.49`	`vrm`	`lib zapata is gone, what's the used replacement ?`
`14:14.57`	`Qwell`	`vrm: huh?`
`14:14.58`	`angryuser`	`robl^:ok thank you for info`
`14:15.21`	`robl^`	`angryuser: they are not easy to setup.. and will cause headaches, but they work`
`14:15.46`	`vgster`	`the aastra's can be a pain, but the config is fairly easy i found`
`14:16.01`	`robl^`	`angryuser: I migrated to Polycom from Aastra. Aastra isn't a little more expnsive, more features, and easier to manage and setup`
`14:16.09`	`vrm`	`Qwell, I see lib zapata is gone`
`14:16.13`	`vrm`	`what asterisk use now ?`
`14:16.18`	`Qwell`	`what is "lib zapata"?`
`14:16.21`	`Qwell`	`I've never heard of it`
`14:16.28`	`vgster`	`anyone with asterisk buuild problems on suse10?`
`14:16.47`	`robl^`	`er... Polycom is more expensive, more features, etc. need more coffee`
`14:17.08`	`giasai68`	`I'm using asterisk 1.4 with zapata 1.4`
`14:17.17`	`giasai68`	`on ubuntu server`
`14:17.18`	`jamessan`	`ManxPower: Dial("SIP/203\|30\|")`
`14:17.32`	`ManxPower`	`vgster: the problem you are experiencing has been fixed in SVN and was discussed on the asterisk-mailing lists. The fix is there.`
`14:17.46`	`ManxPower`	`jamessan: deems fine to me`
`14:17.47`	`vgster`	`ah ok`
`14:17.53`	`ManxPower`	`jamessan: Oh! No, that will NOT work!`
`14:17.59`	`ManxPower`	`remove the quotes`
`14:18.57`	`ManxPower`	`It's days like these that I wish I had remembered to get my prescription for my tranqualizers refilled.`
`14:19.08`	`Qwell`	`ManxPower: for you, or for...them?`
`14:19.16`	`ManxPower`	`Qwell: for me.`
`14:19.31`	`Qwell`	`then you're gonna need to share with the rest of us :p`
`14:19.44`	`jamessan`	`ManxPower: sorry, there aren't actually quotes. I was just getting that from Asterisk's logs since I use an AGI script to determine how to call so I can't just look at what the dial string`
`14:19.56`	`ManxPower`	`The telco rep never answers his phone, the pbx guy is not answering his phone, I leave wed and the damn PRI is not up yet.`
`14:20.08`	`ManxPower`	`the pbx guy also has the new did ranges`
`14:20.15`	`ManxPower`	`thank dog I'm paid by the hour.`
`14:20.25`	`angryuser`	`robl^: polycom support BLF feature too?, how many lines for basic version pf phone (301)?`
`14:20.42`	`ManxPower`	`jamessan: First rule of troubleshooting: simplify the problem.`
`14:21.10`	`angryuser`	`robl^: for instance i see that polycom is more expensive compared to astra`
`14:21.10`	`ManxPower`	`jamessan: but I cannot help you further.`
`14:21.26`	`Qwell`	`angryuser: yes, he just said that it was`
`14:21.30`	`ManxPower`	`Qwell: some idiot from #asterisk already private /msg'd me asking for personal help.`
`14:21.49`	`jamessan`	`ManxPower: so that Dial line (minus the quotes) with port set and nat=yes should work?`
`14:22.07`	`ManxPower`	`jamessan: I don't know. I never manually set a port.`
`14:22.20`	`ManxPower`	`asterisk always figures it out based on the source port of the register and the request`
`14:22.21`	`jamessan`	`ManxPower: ok`
`14:22.27`	`Qwell`	`I have a suspicion that using nat=yes will somehow mess with the port`
`14:22.44`	`vgster`	`ManxPower is this fix for 1.2 or 1.4?`
`14:23.01`	`jamessan`	`well, simply using the source port of the register is going to break things because that port isn't guaranteed to stay open. that's why we've configured the port to use for the phone`
`14:23.22`	`ManxPower`	`vgster: I don't know about 1.,4.x, but the fix I saw was for 1.2.15`
`14:23.50`	`vgster`	`ok thanks ill keep looking`
`14:23.53`	`ManxPower`	`jamessan: you need to make sure it stays open using qualify or other methods like short registration times or nat keep alives on the phone`
`14:24.28`	`jamessan`	`ManxPower: ok, I'll look into that. it'd make more sense for Asterisk to use the configured port though...`
`14:25.18`	`ManxPower`	`jamessan: you cannot guarntee the source port when using a NAT router unless you manually portforward on the router and that is a terrible solution`
`14:25.56`	`jamessan`	`ManxPower: the remote phone uses UPnP to setup the ports it needs and then tells Asterisk what ports to use`
`14:26.11`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:28.15`	`ManxPower`	`jamessan: all I know is that MILLIONS of NAT'd VoIP phones do not need the port manually set.`
`14:28.25`	`angryuser`	`Qwell: mee to i need more coffee;)`
`14:28.48`	`jamessan`	`ManxPower: ok. thanks for your help. Hopefully this will get me going in the right direction`
`14:29.25`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)`
`14:30.19`	`*** join/#asterisk benno2 (n=benno2@82.50.92.145)`
`14:31.03`	`*** join/#asterisk ruied (n=ruied@bl7-213-81.dsl.telepac.pt)`
`14:33.18`	`benno2`	`hi, question: I can call for cheap rates my home phone (where I have asterisk installed) but calling from home to mobile is expensive. If someone calls my phonephone (or via SIP-in providers) and I forward the call to my cellphone it's a bit expensive. so one cool thing to save money would to have asterisk call me first on the mobile connecting me with the calling party. then I tell the party to wait a few secs, I put it on hold, hang up, dial`
`14:33.18`	`benno2`	`<PROTECTED>`
`14:37.38`	`KnowWhat`	`where can i find asterisk sound package?`
`14:38.43`	`ManxPower`	`KnowWhat: oddly enough on the asterisk web site`
`14:39.13`	`KnowWhat`	`yeah got them they are in releases folder`
`14:39.47`	`KnowWhat`	`trying to go by the way of the book any way... what does asterisk-addons package contain?`
`14:41.04`	`KnowWhat`	`there are lots of ... i am going for 1.2.1 version the sound`
`14:41.28`	`*** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan)`
`14:43.10`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`14:43.10`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:44.12`	`*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`14:45.56`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`14:45.56`	`*** mode/#asterisk [+o mog] by ChanServ`
`14:48.43`	`Bobthehunter`	`guys have i have weird 30% RTP passing.. only some times..`
`14:48.45`	`Bobthehunter`	`<PROTECTED>`
`14:48.54`	`Bobthehunter`	`my client is on subnet 10.0.0.x`
`14:49.03`	`Bobthehunter`	`and i have that subnet TOO on the server...`
`14:49.15`	`Bobthehunter`	`can this coincidence be the problem ?`
`14:50.23`	`Bobthehunter`	`le rtp use any of the addresses..`
`14:50.26`	`Bobthehunter`	`<PROTECTED>`
`14:50.28`	`Bobthehunter`	`right ?`
`14:50.31`	`Bobthehunter`	`is that a knwon bug`
`14:50.34`	`*** join/#asterisk viperdude (n=jon@195.74.96.120)`
`14:54.17`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`14:55.01`	`*** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com)`
`14:58.12`	`*** join/#asterisk Ahrimanes (n=ma@81.7.159.2)`
`14:58.27`	`*** join/#asterisk rue_mohr (n=rue@h24-207-91-121.cst.dccnet.com)`
`15:03.26`	`KnowWhat`	`i am getting this error while make clean zaptel drivers`
`15:03.27`	`KnowWhat`	`grep: /lib/modules/2.6.15-1.2054_FC5smp/build/include/linux/autoconf.h: No such file or directory`
`15:03.58`	`*** join/#asterisk f_akmal (n=f_akmal@243.34.48.60.klj03-home.tm.net.my)`
`15:03.58`	`Qwell`	`KnowWhat: upgrade`
`15:04.11`	`KnowWhat`	`upgrade what?`
`15:04.15`	`Qwell`	`zaptel`
`15:04.22`	`reber`	`on a 2GHz pc, how many conversations can i have simultaneously with asterisk ?`
`15:04.28`	`Qwell`	`reber: it depends`
`15:05.43`	`KnowWhat`	`Qwell which version should it be?`
`15:05.43`	`Qwell`	`KnowWhat: the latest`
`15:05.43`	`KnowWhat`	`i am using zaptel-1.4.0`
`15:05.43`	`KnowWhat`	`i think they are latest`
`15:05.43`	`Qwell`	`hmm`
`15:05.44`	`*** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`15:05.44`	`file`	`your kernel headers are installed too, right?`
`15:05.44`	`KnowWhat`	`how can i check that?`
`15:05.44`	`reber`	`perfect, exactly what i was asking for`
`15:05.46`	`Qwell`	`file: latest kernels removed autoconf.h - and I'm pretty sure we did too before 1.4`
`15:05.58`	`ruied`	`can I have an UK landline number associated with a voipcheap so people can dial form uk landline to asterisk (placed in Portugal) ?`
`15:06.12`	`KnowWhat`	`Qwell do u think the zaptel patch will work?`
`15:06.17`	`*** join/#asterisk xF\|DarkSC (n=dark@server.vinylkind.de)`
`15:06.22`	`Qwell`	`KnowWhat: what patch?`
`15:06.23`	`reber`	`may i ask of what it depends ?`
`15:06.27`	`xF\|DarkSC`	`hello`
`15:06.28`	`KnowWhat`	`zaptel patch`
`15:06.34`	`Qwell`	`reber: everything you do on your system`
`15:06.37`	`Qwell`	`KnowWhat: what zaptel patch?`
`15:06.46`	`KnowWhat`	`but i can only install if i have zaptel installed`
`15:06.55`	`xF\|DarkSC`	`i want to configure asterisk to use sipgate. can someone help me please?`
`15:07.42`	`f_akmal`	`hi all`
`15:07.46`	`reber`	`i read something on the asterisk official manual (was it oreilly ?), and asterisk was given basicaly with 20 or 80 simultaneous conversations. I can't remember the numer precisely...`
`15:07.46`	`*** part/#asterisk vrm (n=vrm@94.55.101-84.rev.gaoland.net)`
`15:08.13`	`f_akmal`	`i forwarded a call using the Dial command but the sound does not go through`
`15:08.18`	`f_akmal`	`can anyone help?`
`15:08.41`	`reber`	`was it 20 or 80 (on a basic PC, with only asterisk on it) ?`
`15:09.01`	`KnowWhat`	`Qwell so what can i do now?`
`15:10.11`	`KnowWhat`	`file: any comments ?`
`15:10.24`	`Bobthehunter`	`yo?`
`15:10.46`	`xF\|DarkSC`	`i am always getting this errors. http://pastebin.ca/352026`
`15:10.56`	`xF\|DarkSC`	`i cant call in, and i cant call out`
`15:11.02`	`xF\|DarkSC`	`i cant even call the mailboxes`
`15:11.24`	`xF\|DarkSC`	`ok i can call in.. but mh. it gives me the busy error.`
`15:15.28`	`xF\|DarkSC`	`this is what goes on, when i call in http://pastebin.ca/352034`
`15:15.46`	`*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)`
`15:15.46`	`*** mode/#asterisk [+o angler] by ChanServ`
`15:16.33`	`*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)`
`15:17.33`	`*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`15:17.56`	`xF\|DarkSC`	`no one? :(`
`15:18.44`	`*** join/#asterisk saftsack (n=saftsack@pD9E04C9F.dip.t-dialin.net)`
`15:19.41`	`*** join/#asterisk nas_lslsa (n=chatzill@85.75.137.146)`
`15:20.12`	`nas_lslsa`	`hello , ok , I configured my asterisk , I made 2 users , and I am trying to use one of them to connect with sjphone ..`
`15:20.32`	`HarryR`	`and your problem is?`
`15:20.47`	`nas_lslsa`	`how can I found if SIP is runing fine on asterisk ? or what port is it ?`
`15:21.09`	`nas_lslsa`	`I checked 1 - 2 howtos but without any luck`
`15:21.34`	`KnowWhat`	`nas_lslsa type help in asterisk command line interface`
`15:21.44`	`KnowWhat`	`there are some commands related to sip there`
`15:22.59`	`HarryR`	`nas_lslsa, look in $ASTERISK_HOME/etc/sip.conf`
`15:23.24`	`nas_lslsa`	`I vi it .. what I am looking for there ?`
`15:23.29`	`HarryR`	`then check netstat for port 5060 (or whichever ports configured in sip.conf) and see if Asterisk is using it`
`15:23.38`	`nas_lslsa`	`CLI didn't have anything about SIP`
`15:23.51`	`*** part/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br)`
`15:24.36`	`KnowWhat`	`well`
`15:24.38`	`KnowWhat`	`if you do`
`15:24.38`	`nas_lslsa`	`I think IT WORKS ! ! ! !`
`15:24.41`	`KnowWhat`	`sip show registry`
`15:24.47`	`KnowWhat`	`it will show something`
`15:24.58`	`nas_lslsa`	`I'll make some tasts and see ya for thanks / extra questions if it needed ;)`
`15:28.15`	`*** join/#asterisk Dovid (n=Dovid@85.159.160.207)`
`15:28.18`	`Dovid`	`hello all`
`15:28.20`	`Dovid`	`anyone use sjphone ?`
`15:29.43`	`kippi`	`hey`
`15:29.48`	`Moobius`	`Dovid: yes...`
`15:29.58`	`kippi`	`on the voicemail how do you change the from address?`
`15:31.34`	`Dovid`	`Moobius: i was told that u can now have an auto configure file so that if i have clients they can download the phone from my site, insert a file and it will get all the sip settings`
`15:31.43`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`15:32.04`	`Moobius`	`I think that is true, though I've not used it. Sorry.`
`15:32.14`	`Dovid`	`any places for me to look ?`
`15:32.22`	`Dovid`	`or is it for the paid version only ?`
`15:32.35`	`Moobius`	`paid only, I think.`
`15:32.50`	`*** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92)`
`15:32.56`	`KnowWhat`	`hmm`
`15:32.56`	`Moobius`	`Part of their "customization" service.`
`15:33.27`	`KnowWhat`	`any body`
`15:34.05`	`*** join/#asterisk yxa (n=yxa@cm127.gamma228.maxonline.com.sg)`
`15:34.16`	`Dovid`	`hehe`
`15:34.25`	`Dovid`	`so basicly i goto pay per client ?`
`15:36.08`	`*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)`
`15:36.27`	`Ahrimanes`	`hm there should be a profile editor in the free version according to their faq`
`15:38.03`	`Dovid`	`dont see one`
`15:38.07`	`Dovid`	`is it parta the phone or ?`
`15:38.14`	`Dovid`	`or an external program ?`
`15:38.47`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`15:38.49`	`Ahrimanes`	`http://www.sjlabs.com/softphone-faq.html <- q5`
`15:38.50`	`Bobthehunter`	`AMNYWAIT TO FORCE RTP not using the private lan ?`
`15:39.00`	`Bobthehunter`	`im getting 33% sucsses on audio since this monring`
`15:39.03`	`Bobthehunter`	`NOTHING changed`
`15:39.05`	`Bobthehunter`	`on our side`
`15:39.36`	`Moobius`	`or another product...`
`15:40.19`	`Ahrimanes`	`afair it's cleartext files in a special folder`
`15:40.35`	`xF\|DarkSC`	`ok i got asterisk running now... but if i call the voicemail for example asterisk does play the soundfiles, but i cant hear anything`
`15:40.58`	`Moobius`	`Bobthehunter: your asterisk box and clients are on the same lan?`
`15:41.26`	`Dovid`	`Bobthehunter: there u enter it on the phone`
`15:41.32`	`Dovid`	`i want so that i can just upload a file`
`15:41.58`	`Bobthehunter`	`no`
`15:42.03`	`Bobthehunter`	`everyone out of it`
`15:42.05`	`nas_lslsa`	`didn't worked :( :( :(`
`15:42.18`	`Bobthehunter`	`but server has 2 ips.. EXTERNAL 209.X and internal 10.0.0.100`
`15:42.37`	`Bobthehunter`	`and one client has a lan on its own.. with 10.0.0.130 and a natted outbound..`
`15:42.46`	`Bobthehunter`	`im thinking htey conflict..but never did in the past`
`15:42.51`	`Moobius`	`Bobthehunter: double check your external IP and sip.conf's externip and bindaddr`
`15:43.02`	`Moobius`	`also localnet`
`15:43.49`	`Bobthehunter`	`bindaddr=209.xxxx`
`15:43.50`	`Moobius`	`Your asterisk box is also performing routing and NAT functions for you?`
`15:43.58`	`Dovid`	`hmm`
`15:44.04`	`Bobthehunter`	`localnet is set ...`
`15:44.08`	`Dovid`	`seems u have to pay em to create an exe that does the changes`
`15:44.09`	`Bobthehunter`	`externip`
`15:45.49`	`Bobthehunter`	`can default ip field in ARA be NULL`
`15:45.52`	`Bobthehunter`	`in table ?`
`15:46.01`	`Bobthehunter`	`coudl that do anything \? they are registered..`
`15:46.14`	`Ahrimanes`	`Dovid: it's a file placed in a certain folder`
`15:46.26`	`Dovid`	`i know`
`15:46.30`	`Bobthehunter`	`today i got 50% clients getting no audio on 30% of direct calls to asterisk.. like voicemail or echo test.. so no third party involved`
`15:46.32`	`Dovid`	`its encrypted`
`15:46.38`	`Dovid`	`lots of wierd icons`
`15:46.40`	`Ahrimanes`	`Dovid: in preferences in the phone you can make a new profile, that is then saved in a file and it can be copied`
`15:47.18`	`Dovid`	`hmm`
`15:47.21`	`Dovid`	`no copy option`
`15:49.12`	`Dovid`	`there are ini files`
`15:49.19`	`Bobthehunter`	`weird`
`15:49.51`	`Ahrimanes`	`yes, copy the ini file to another computer to the right folder`
`15:50.44`	`*** join/#asterisk marv[work] (n=timr@24.214.206.254)`
`15:51.30`	`*** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) [NETSPLIT VICTIM]`
`15:51.47`	`*** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) [NETSPLIT VICTIM]`
`15:52.11`	`*** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) [NETSPLIT VICTIM]`
`15:52.54`	`nas_lslsa`	`when I try to connect via SIP client to Asterisk , I don't even see any activity in console ..`
`15:53.55`	`*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) [NETSPLIT VICTIM]`
`15:54.02`	`HarryR`	`nas_lslsa, connect to the console with increased verbosity`
`15:54.07`	`HarryR`	`e.g. asterisk -r -vvvvvvvvvvvvv`
`15:54.44`	`HarryR`	`or use commands like "sip show peers"`
`15:56.09`	`*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) [NETSPLIT VICTIM]`
`15:56.10`	`*** join/#asterisk mafkees (i=michiel@vanbaak.xs4all.nl) [NETSPLIT VICTIM]`
`15:56.14`	`*** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) [NETSPLIT VICTIM]`
`15:56.23`	`nas_lslsa`	`id can't find that command .. is there any way that hasn't support for SIP ? I made the default compile / configure ..`
`15:56.26`	`ChicagoBud`	`hey, I upgraded a system from 1.2 to 1.4 and now I'm getting a dozen warnings like:`
`15:56.27`	`ChicagoBud`	`[Feb 12 09:35:23] WARNING[28343] cli.c: Command 'iax2 show cache' already registered (or something close enough)`
`15:56.37`	`ChicagoBud`	`any idea on this?`
`15:58.17`	`ChicagoBud`	`I looked through /usr/lib/asterisk/modules/ and there is only one module with 'iax2 show cache" in it`
`15:58.36`	`*** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl)`
`15:59.16`	`*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) [NETSPLIT VICTIM]`
`15:59.16`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) [NETSPLIT VICTIM]`
`15:59.16`	`*** join/#asterisk [hC] (n=hardcore@66.119.167.162) [NETSPLIT VICTIM]`
`15:59.16`	`*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)`
`15:59.17`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) [NETSPLIT VICTIM]`
`15:59.17`	`ChicagoBud`	`I deleted everything in /usr/lib/asterisk/modules/ before I upgraded`
`15:59.17`	`*** join/#asterisk merbzt (n=banan@136.240.13.217.in-addr.dgcsystems.net) [NETSPLIT VICTIM]`
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`16:00.30`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM]`
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`16:00.31`	`*** mode/#asterisk [+o Qwell] by irc.freenode.net`
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`16:03.27`	`*** part/#asterisk matiasds19 (n=matiasdo@201.253.208.136)`
`16:04.12`	`*** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr)`
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`16:07.24`	`kanelbullar`	`Hi guys, has anybody using chan_unicall for MFC/R2 used any solution to block collect calls in Brazil?`
`16:07.49`	`*** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle)`
`16:07.59`	`*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) [NETSPLIT VICTIM]`
`16:08.39`	`ttuttle`	`I'm having a strange problem with Asterisk and Festival and the Record app. If I synthesize text using Festival before I've recorded anything, it sounds very choppy (as if every other "packet" is missing, somehow), but after I've made a recording, the audio is crystal-clear.`
`16:08.56`	`ttuttle`	`Should I just do a 0-second recording to a temporary file, or is there a way to fix this?`
`16:09.24`	`*** join/#asterisk nachophone (i=boster@ivan.dreamhost.com) [NETSPLIT VICTIM]`
`16:10.17`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`16:11.48`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) [NETSPLIT VICTIM]`
`16:12.27`	`ttuttle`	`Anyone?`
`16:12.48`	`*** join/#asterisk RoyK (n=roy@213.160.242.90)`
`16:12.50`	`Bobthehunter`	`Feb 12 11:12:38 WARNING[14570]: app_voicemail.c:4989 vm_authenticate: Couldn't read username`
`16:12.58`	`Bobthehunter`	`i get this alot when no RTP can be in`
`16:13.01`	`Bobthehunter`	`any idea ?`
`16:13.04`	`Bobthehunter`	`this is driving me nuts`
`16:13.23`	`*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`16:15.32`	`Bobthehunter`	`could it be no udp ports left ?`
`16:16.59`	`ttuttle`	`Are there any ways to debug audio problems with Festival?`
`16:17.28`	`*** join/#asterisk ping2921 (n=marc3234@206-248-128-226.dsl.teksavvy.com)`
`16:17.31`	`ttuttle`	`I get audio, but it's very choppy.`
`16:18.15`	`*** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com)`
`16:21.59`	`yansolo90`	`hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ?`
`16:22.50`	`*** join/#asterisk viperdude (n=jon@195.74.96.120)`
`16:26.17`	`robl^`	`there is none. i uses telnet, not ssh`
`16:26.58`	`nosbig`	`Anyone here get iax trunking to work?`
`16:27.33`	`*** join/#asterisk lorinc (n=ang@caracas-4604.adsl.interware.hu)`
`16:29.32`	`ChicagoBud`	`ttuttle, you might want to look into "flite'`
`16:29.41`	`*** join/#asterisk key2 (n=key2@81.52.138.22)`
`16:30.46`	`*** join/#asterisk yxa (n=yxa@cm127.gamma228.maxonline.com.sg)`
`16:31.04`	`nosbig`	`And since I control both Asterisk servers in the trunk, is it better to make friend entries or peer/user pairs?`
`16:31.19`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`16:31.28`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`16:31.29`	`*** mode/#asterisk [+o russellb] by ChanServ`
`16:31.29`	`ttuttle`	`ChicagoBud: What's the difference, does it work better?`
`16:31.35`	`yxa`	`i'm having problems with DTMF from sip to misdn. can anyone help??`
`16:32.01`	`ChicagoBud`	`ttuttle, yes. -- less resource intensive`
`16:32.07`	`ttuttle`	`ChicagoBud: Ah.`
`16:32.23`	`*** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at)`
`16:32.31`	`ChicagoBud`	`ttuttle, a bit ticky to build but I got it done. I can help you`
`16:32.33`	`ttuttle`	`ChicagoBud: Anything else, or just resource usage?`
`16:32.33`	`*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)`
`16:32.46`	`*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)`
`16:32.46`	`ChicagoBud`	`I think it is very similar`
`16:33.33`	`*** join/#asterisk mkl1525 (n=qwertz@pD9532D64.dip0.t-ipconnect.de)`
`16:33.44`	`ttuttle`	`ChicagoBud: /me can get it in Portage.`
`16:33.55`	`ChicagoBud`	`ttuttle, pretty much a drop in replacement`
`16:33.57`	`ttuttle`	`ChicagoBud: Does it sound better/worse?`
`16:34.34`	`ChicagoBud`	`ttuttle, http://www.speech.cs.cmu.edu/flite/ is the base code then you need the asterisk interface that someone else built`
`16:34.53`	`ChicagoBud`	`ttuttle, http://sourceforge.net/project/showfiles.php?group_id=154235&package_id=181093`
`16:35.33`	`ChicagoBud`	`ttuttle, I suggest you look at http://sourceforge.net/forum/forum.php?thread_id=1542274&forum_id=516333 too`
`16:35.36`	`*** join/#asterisk backblue (n=igor@82.102.1.42)`
`16:35.43`	`ttuttle`	`ChicagoBud: So standard Flite, then asterisk-flite on top of that?`
`16:36.12`	`*** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it)`
`16:36.14`	`backblue`	`hi, anyone knows any tool, to diagnostic why my x100p does not detect my callerid?`
`16:36.16`	`giasai68`	`hello`
`16:36.19`	`mkl1525`	`Hi, is there a way to process further in the dialplan when a caller hangup the phone and was talking in a queue? tried with a noop but no output`
`16:36.26`	`giasai68`	`I have got this warning: Unable to find a codec translation path from g729 to ulaw`
`16:36.33`	`giasai68`	`any suggest for fix it?`
`16:36.38`	`tzafrir`	`backblue, for several countries (uk, Mexico, any other?) a separate patch ("ukcid, asterisk uk caller ID patch) is needed in order to get caller ID with an X100P`
`16:36.45`	`ChicagoBud`	`ttuttle, yes. you need to patch flite's makefile before building to use shared libs. read the forum post I sent`
`16:36.53`	`ttuttle`	`ChicagoBud: ok`
`16:36.58`	`tzafrir`	`I'm not sure about Portugal`
`16:37.07`	`backblue`	`tzafrir: that's kind old, it's needed yet?`
`16:37.52`	`[TK]D-Fender`	`backblue: And some X100P clones CID implementation is just flakey.`
`16:37.53`	`tzafrir`	`backblue, some parts of it were merged long ago. The parts that are only relevant to x100p were left out, also for performance considerations`
`16:38.13`	`[TK]D-Fender`	`backblue: then there is a question as to whether or not you configured your zapata.conf right.`
`16:38.45`	`backblue`	`well, i was trying to look, for a tool, that acts out of the asterisk`
`16:38.56`	`ChicagoBud`	`ttuttle, send me your email and I'll send you a wave sample if you'd like: wwbach at ameritech dot net`
`16:39.25`	`backblue`	`i want to test it, but out of the asterisk`
`16:39.31`	`backblue`	`only directly in zaptel.`
`16:39.47`	`[TK]D-Fender`	`ChicagoBud: shouldn't that att.com now? :)`
`16:40.25`	`ping2921`	`anyone knows where to find 800 number availibity list. I am trying to build a php 800 search.`
`16:40.38`	`tzafrir`	`the caller ID detection is done in Asterisk`
`16:40.51`	`backblue`	`tzafrir: you dont have understood yet.`
`16:40.52`	`ChicagoBud`	`ttuttle, should be as of about 5 years ago but they let us keep the old ones...`
`16:41.01`	`tzafrir`	`Zaptel is basically a pipe (with some basic detection of signalling)`
`16:41.14`	`nosbig`	`And with the IAX trunking, do I need to establish the trunk from both ends?`
`16:41.27`	`backblue`	`tzafrir: you have zaptel running, and after that put on the top, some app, that works over zaptel like asterisk (with chan_zap), and only detects callerid.`
`16:41.38`	`tzafrir`	`have you managed to build clidtest?`
`16:41.47`	`backblue`	`no, i dont, it's old.`
`16:41.59`	`*** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122)`
`16:42.54`	`*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)`
`16:43.08`	`[TK]D-Fender`	`nosbig: As soon as a 2nd calls is passed through the peer, * will combine the RTP on a signgle packet.`
`16:43.10`	`Bobthehunter`	`yansolo90, cisco/cisco`
`16:44.03`	`*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`16:44.25`	`sweeper`	`any word on asterisk <-> polycom attendant sidecards?`
`16:44.45`	`*** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr)`
`16:44.54`	`*** join/#asterisk reivilo (n=olivier@LSt-Amand-152-31-23-31.w82-127.abo.wanadoo.fr)`
`16:44.58`	`backblue`	`sweeper: any problem?`
`16:45.13`	`[TK]D-Fender`	`sweeper: They work. What more is there to say?`
`16:45.22`	`sweeper`	`backblue: I'd heard a few weeks ago they didn't, so I asked now \o`
`16:45.44`	`backblue`	`they are the best on the market, in my opinion`
`16:45.49`	`[TK]D-Fender`	`sweeper: Get better sources. They been fully functional for about a year.`
`16:45.57`	`sweeper`	`[TK]D-Fender: I heard it here :P`
`16:45.59`	`ttuttle`	`ChicagoBud: Thanks for the help, I'll be back later.`
`16:46.05`	`sweeper`	`didn't really dig into it`
`16:46.16`	`[TK]D-Fender`	`sweeper: This is a channel. INDIVIDUALS are sources....`
`16:46.28`	`sweeper`	`lies! I demand unified responsibility!`
`16:46.45`	`[TK]D-Fender`	`sweeper: Prepare for disappointment...`
`16:46.57`	`reivilo`	`Is Somebody experience freeze with snom hard phones and asterisk 1.2 ?`
`16:47.03`	`*** join/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net)`
`16:47.20`	`[TK]D-Fender`	`sweeper: Keeping in mind how many idiots are still out there configuring them via the phone direct, or the web interface.`
`16:47.39`	`websae`	`you are talking about Polycom of course, hehe`
`16:47.56`	`[TK]D-Fender`	`reivilo: They freeze all over the place, not just with *. Snom firmare was coded by Tony The Tiger...`
`16:48.15`	`[TK]D-Fender`	`websae: Correct.`
`16:48.24`	`sweeper`	`what's wrong with the web interface? >.>`
`16:48.38`	`*** join/#asterisk nowork (n=jfu2808@216.254.141.97)`
`16:48.45`	`websae`	`can't configure the right settings`
`16:48.55`	`websae`	`that one typically needs that is`
`16:49.11`	`reivilo`	`[TK]D-Fender: in my office they work fine but in other office 360 and 320 freeze every weekend`
`16:49.14`	`[TK]D-Fender`	`sweeper: It's shit-on-a-stick, causes reboots every time you change anything in a section (2 min avg wait), and you can't doa fraction of what you can through provisioning`
`16:49.29`	`Bobthehunter`	`well said`
`16:49.33`	`*** join/#asterisk nowork (n=jfu2808@216.254.141.97)`
`16:49.42`	`Bobthehunter`	`SOAS`
`16:49.49`	`websae`	`well said...brilliantly said!`
`16:49.51`	`Bobthehunter`	`shit on a stick... let me add to wikipedia`
`16:49.57`	`nowork`	`hi, full file at /var/log/asterisk getting big, can i delete it?`
`16:49.59`	`[TK]D-Fender`	`sweeper: I'd say it should be abolished altogether to make room for things like a MicroBrowser for the IP 501..... but they just did that already, and for the IP 430 as well :)`
`16:50.03`	`Bobthehunter`	`as the main definition of a GUI`
`16:50.18`	`Bobthehunter`	`nosbig, edit logger.conf remove debug`
`16:50.23`	`Bobthehunter`	`then logger reload`
`16:50.23`	`sweeper`	`[TK]D-Fender: provisioning? you mean, aka bootp server?`
`16:51.08`	`[TK]D-Fender`	`sweeper: No, as it FTP, TFTP, HTTP, etc as a warehouse for configuration files.`
`16:51.08`	`[TK]D-Fender`	`sweeper: BOOTP isn't a file repo.`
`16:51.08`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`16:51.09`	`*** join/#asterisk queuetue (n=queuetue@70.54.254.134)`
`16:51.10`	`sweeper`	`well, yea`
`16:51.17`	`sweeper`	`but it starts things off \o`
`16:51.33`	`*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)`
`16:51.33`	`queuetue`	`Hi. I've got a problem authenticating from sipura 2000 to an asterisknow box - both x305 and x306 are set up on it, and when 305 tries to call out, I get the error "WARNING[1103]: chan_sip.c:8023 check_auth: username mismatch, have <306>, digest has <305>", followed by "NOTICE[1103]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user ..." I see a similar problem reported at http://bugs.digium.com/view.php?id=8388 but no resolution.`
`16:51.46`	`[TK]D-Fender`	`sweeper: in more automated environments DHCP would set Opt 66, and the phones would use that IP to try and grab their configs from, so every time they boot they could pick up centrally administered changes.`
`16:52.11`	`sweeper`	`aha`
`16:52.58`	`*** join/#asterisk benno2 (n=benno2@host145-92.pool8250.interbusiness.it)`
`16:53.10`	`benno2`	`i, question: I can call for cheap rates my home phone (where I have asterisk installed) but calling from home to mobile is expensive. If someone calls my phonephone (or via SIP-in providers) and I forward the call to my cellphone it's a bit expensive. so one cool thing to save money would to have asterisk call me first on the mobile connecting me with the calling party.`
`16:53.27`	`benno2`	`<PROTECTED>`
`16:53.27`	`benno2`	`"parking a call"). ? Can asterisk what I described ?`
`16:56.07`	`*** join/#asterisk nfi\|ermes (n=ermes@217.220.121.62)`
`16:56.49`	`[TK]D-Fender`	`benno2: So you want * to call your cell so you can dialout your line after?`
`16:58.17`	`angryuser`	`first par can be done, dont know if you can let people Unpark you frien from remote location;)`
`16:59.27`	`angryuser`	`[TK]D-Fender: i was on the point bu buy 10 snoms and 2 astra when i read you discussion;)`
`16:59.45`	`giasai68`	`I have got this warning: Unable to find a codec translation path from g729 to ulaw`
`16:59.49`	`giasai68`	`any suggest for fix it?`
`16:59.57`	`giasai68`	`I have got this warning: Unable to find a codec translation path from g729 to slin`
`16:59.58`	`giasai68`	`any suggest for fix it?`
`17:00.28`	`giasai68`	`also there is : Asked to transmit frame type 4, while native formats is 256 (read/write = 4/64)`
`17:00.34`	`giasai68`	`pls, help me`
`17:00.36`	`giasai68`	`thanks`
`17:01.26`	`*** join/#asterisk alephant (n=cmd@c-24-3-52-93.hsd1.mn.comcast.net)`
`17:01.39`	`alephant`	`What is the comment syntax for .conf files?`
`17:01.42`	`alephant`	`I see ^;`
`17:01.45`	`alephant`	`which is obvious`
`17:01.52`	`alephant`	`but is ^# valid as well?`
`17:02.22`	`alephant`	`(the confrusion arises from the fact that I see ^#include which looks enough like C that I start second-guessing...)`
`17:02.26`	`alephant`	`what's the story?`
`17:02.33`	`alephant`	`and where is the conf file syntax documented?`
`17:03.19`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)`
`17:03.21`	`Dr-Linux`	`Feb 12 08:58:12 WARNING[25127]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x971a820 (len 774) to 202.125.141.2:-1 returned 0: Invalid argument`
`17:03.29`	`Dr-Linux`	`any idea what is that?`
`17:04.03`	`alephant`	`Anybody? Comment syntax in conf files?`
`17:05.33`	`Dr-Linux`	`;`
`17:05.46`	`alephant`	`so #include?`
`17:05.55`	`alephant`	`is that a C-style syntax for an include directive?`
`17:05.57`	`alephant`	`or is it commented out?`
`17:06.15`	`*** join/#asterisk DJS_2_6 (n=djstillm@cpe-066-057-115-255.nc.res.rr.com)`
`17:09.16`	`giasai68`	`anu suggest for this warning: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/64)`
`17:09.20`	`giasai68`	`please, let me know`
`17:10.50`	`[TK]D-Fender`	`giasai68: Did you buy G.729 licenses for your server from Digium?`
`17:11.01`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`17:11.31`	`*** join/#asterisk evisu (i=hIRC@bzq-88-154-4-53.red.bezeqint.net)`
`17:11.53`	`reivilo`	`giasai68: did you get the license codec and the codec_g729.so file ?`
`17:12.00`	`canapa`	`is there any chance i can get a Account from a SIP Provider to tryout ? is there any thing free ?`
`17:12.46`	`[TK]D-Fender`	`canapa: Yes, the coffee at a Microsoft conference...`
`17:12.53`	`tzafrir`	`sipphone`
`17:13.25`	`drako`	`tzafrir, heya, i managed to see the recorded files from ARI`
`17:13.58`	`tzafrir`	`canapa, basically many would offer you free sip->sip calls in hope that you pay for sip->PSTN calls or other services`
`17:14.00`	`canapa`	`[TK]D-Fender: yer thx`
`17:14.22`	`canapa`	`tzafrir: i would just need a account for tryout realy`
`17:14.29`	`canapa`	`this is my first asteisk`
`17:14.31`	`canapa`	`etc`
`17:14.40`	`canapa`	`*asterisk`
`17:14.41`	`canapa`	`:)`
`17:14.42`	`tzafrir`	`canapa, sip -> sip? sip-> pstn?`
`17:14.46`	`giasai68`	`reivilo: no, how I can do?`
`17:14.46`	`drako`	`tzafrir, but its very unstable, it sometimes have the file available for download and sometimes it does not.. the same file`
`17:14.50`	`*** join/#asterisk jarg (n=jarg@200.56.225.61)`
`17:15.11`	`canapa`	`tzafrir: well, the next step in the book would be sip->pstn`
`17:15.24`	`canapa`	`but i guess i could skip thatone`
`17:15.58`	`nosbig`	`Yeah! IAX trunking seems to be working now!!! Now, I just have ton configure my dialplan properly...`
`17:16.31`	`*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)`
`17:17.24`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`17:17.55`	`nachophone`	`I'm getting a lot of " == Everyone is busy/congested at this time (1:1/0/0)" in my log files. where should I start looking to findout what is causing this?`
`17:18.04`	`angryuser`	`what do you think about Grandstream GXP-2000? what is the sound quality?`
`17:18.22`	`giasai68`	`reivilo: no, how I can do?`
`17:19.02`	`benno2`	`[TK]D-Fender: no, basically I don't need callback. (I got it already working with DISA). I have a cellphone contract where calls from my mobile to my home nr. (where * runs) are cheap. so basically for outgoing calls I simply call * from my cell and then use DISA to dial out. but if I get an incoming call on my homephone and then ring my cellphone I pay more. so the ideal would be to "park" the call, tell the calling party to wait a`
`17:19.03`	`benno2`	`I call back * from my cell and rejoin the call. is this possible ?`
`17:19.46`	`nachophone`	`benno2, dup the call in a meet me conference`
`17:19.56`	`*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`17:19.59`	`BSDTech`	`morning all`
`17:20.44`	`BSDTech`	`I wanted to know how to make a request for sounds that are missing from the sonds dir. I have noticed thata there are no holidays`
`17:21.30`	`BSDTech`	`and the word observance and a few iothers that would be used by people`
`17:21.32`	`benno2`	`nachophone: thanks I will try this. dup = transfer in this case ?`
`17:23.09`	`nachophone`	`sorry, s/dup/drop\|transfer/ it's morning`
`17:23.28`	`*** join/#asterisk atlantia (n=scott@64.20.156.140)`
`17:24.39`	`atlantia`	`hi all, quick newbie q, following the "book" doing the "hello world" scenario on a fresh asterisknow system... i added my extenstion [incoming] in extensions.conf, yet it still plays the stock message when starting... what did i miss?`
`17:24.52`	`atlantia`	`started with a fresh extensions.conf`
`17:25.50`	`reivilo`	`giasai68: go to digium.com in store at g729 licensing and follow the instructions`
`17:26.31`	`angryuser`	`giasai68: or use open g729 IF it is permitted in your country`
`17:26.40`	`atlantia`	`looks like i broke the web GUI as well,.. moved extensions.conf to extensions.conf.sample`
`17:26.41`	`atlantia`	`meh`
`17:26.43`	`atlantia`	`the whole damn server crashed`
`17:26.55`	`*** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com)`
`17:30.49`	`[TK]D-Fender`	`benno2: Yes, quite possible`
`17:31.34`	`nfi\|ermes`	`in version 1.4, is sounds included in asterisk packages ?`
`17:32.29`	`*** join/#asterisk florz (n=florz@2002:58c6:2592:1:0:0:0:2)`
`17:37.11`	`benno2`	`[TK]D-Fender: nachophone thanks, I will try the meetme method :)`
`17:37.16`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`17:37.31`	`*** join/#asterisk CunningPike (n=CunningP@204.239.8.149)`
`17:37.40`	`*** join/#asterisk Cresl1n (i=matt@nat/digium/x-fc964a4a48423abd)`
`17:37.40`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`17:37.48`	`[TK]D-Fender`	`benno2: Call parking is a good way.`
`17:37.57`	`Qwell[]`	`omg, it's a Cresl1n in #asterisk`
`17:38.02`	`queuetue`	`I've got a problem authenticating from sipura 2000 to an asterisknow box - both x305 and x306 are set up on it, and when 305 tries to call out, I get the error "WARNING[1103]: chan_sip.c:8023 check_auth: username mismatch, have <306>, digest has <305>", followed by "NOTICE[1103]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user ..." I see a similar problem reported at http://bugs.digium.com/view.php?id=8388 but no resolution. Can a`
`17:38.15`	`Cresl1n`	`hey Qwell[] !!!!`
`17:38.22`	`*** part/#asterisk xF\|DarkSC (n=dark@server.vinylkind.de)`
`17:39.07`	`*** join/#asterisk thekidrio (n=thekidri@66.107.42.13)`
`17:39.54`	`thekidrio`	`anyone know a place i can walk in an buy digium cards in southern california, near LA would be best`
`17:40.40`	`[TK]D-Fender`	`thekidrio: Good luck. Store wouldn't want to shelf this stuff just to sit around hoping for a person like you to come in.`
`17:40.53`	`[TK]D-Fender`	`thekidrio: Should buy for a normal on-line retailer.`
`17:41.07`	`thekidrio`	`yeah, just in a bit of a hurry`
`17:41.13`	`thekidrio`	`had one burn out and we have a demo later hehe`
`17:41.40`	`thekidrio`	`i can fake the demo with my sip phone but wanted to show him pstn connect`
`17:41.59`	`nachophone`	`signup for an iax provider`
`17:42.08`	`nachophone`	`it'll take 30 minutes`
`17:42.34`	`*** join/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com)`
`17:43.50`	`thekidrio`	`doh, this is why i can't stop drinking coffee in the am`
`17:44.12`	`*** join/#asterisk oon (n=oon@pdpc/supporter/base/oon)`
`17:44.18`	`oon`	`hello !`
`17:44.49`	`oon`	`I'm searching for some detailled documentation about Skinny protocol`
`17:44.54`	`connecta`	`does anyone here use IDEFisk`
`17:45.00`	`connecta`	`with g729`
`17:45.23`	`oon`	`i want to know if developper have good ressource on it ?`
`17:48.41`	`*** join/#asterisk inteliwasp (n=inteliwa@69-168-176-97.clvdoh.adelphia.net)`
`17:49.21`	`angryuser`	`when i press hook-flash button, asterisk tells me "dont know how to indicate condition 9", threewaycalling=yes, transfer=yes`
`17:52.48`	`*** join/#asterisk nextime (n=nextime@unaffiliated/nextime)`
`17:52.52`	`*** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net)`
`17:53.39`	`*** join/#asterisk Assid (i=assid@59.183.3.245)`
`17:55.20`	`sevard`	`Anyone have an extra PCI SATA Controller?`
`17:55.31`	`nextime`	`hello. on voip-info.org i read that intel mb and e1000 nic card aren't so good with * ( if i want to use some digium cards). Anyone know how about broadcom cards? ( i want to bui a dual opteron dell server that mount this card )`
`17:57.10`	`backblue`	`CunningPike: we use 1.2 too, have you used call-limit?`
`17:58.03`	`CunningPike`	`backblue: I'm just checking now..... I don't believe so.....`
`17:58.32`	`backblue`	`hoo you were speaking without knowing :)`
`17:58.45`	`backblue`	`test it, and tell me after that :)`
`17:58.56`	`backblue`	`yes we do have all the polycoms working too.`
`17:59.01`	`backblue`	`with all the features.`
`17:59.18`	`sevard`	`nextime: the only problem i had with the e1000 and the intel motherboard (in a 1u) was that they shared the same IRQ as digium cards, and I couldn't swap IRQs on either.`
`17:59.29`	`*** join/#asterisk florz (i=nobody@2002:58c6:2592:1:0:0:0:2)`
`17:59.35`	`*** join/#asterisk guilherme_jorge (n=guilherm@200-170-201-134.core01.spo.ifx.net.br)`
`17:59.36`	`nachophone`	`are there limits to how many people can be in a queue?`
`17:59.41`	`nachophone`	`not agents, but callers`
`17:59.51`	`Qwell[]`	`nachophone: just based on your server`
`17:59.54`	`sevard`	`nachophone: limits to your hardware, no softlimit iirc`
`18:00.07`	`Qwell[]`	`If you're running asterisk on a 200mhz MMX, you'll get far fewer, obviously`
`18:00.18`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`18:00.36`	`nachophone`	`well, it's a box that ahs sip trunks to other * boxes with 3 T1s each, so it's not cahnnels, and the machine isn't overloaded`
`18:00.44`	`guilherme_jorge`	`hello all, I've some problems to enable Asterisk with Realtime. I've already compile addons, but res_mysql.so didnt created... Any idea?`
`18:00.52`	`nachophone`	`wow, I cannot type today`
`18:01.04`	`canapa`	`tzafrir: well, can u name me a free sip->sip provider ?`
`18:01.29`	`CunningPike`	`backblue: Ah - we're _not_ using call-limit with 1.2. We _are_ using limitpeers in 1.4 - that was the only way we could get hints to work`
`18:02.15`	`backblue`	`CunningPike: hints work nice in 1.2.`
`18:02.33`	`sevard`	`we should use clues rather than hints`
`18:02.37`	`Qwell[]`	`canapa: sip>sip provider?`
`18:02.38`	`backblue`	`CunningPike: so there is a bug?`
`18:03.07`	`Qwell[]`	`you don't need a provider if you're going sip to sip between servers`
`18:03.07`	`CunningPike`	`backblue: yes, they do - we're have a couple of small issues in 1.4 (#8848)`
`18:03.07`	`CunningPike`	`s/have/having/`
`18:03.14`	`canapa`	`Qwell[] its pretty boring talking to myselfe in my LAN`
`18:03.17`	`*** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com)`
`18:03.28`	`backblue`	`ok. but my bug it's working in 1.4`
`18:04.21`	`guilherme_jorge`	`<PROTECTED>`
`18:05.01`	`*** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`18:05.43`	`*** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122)`
`18:06.39`	`codazoda`	`Does anyone know if I can do a single-line-transfer using a TDM2400E? For example, a hook flash then dial 7 digits? Specifically, I'd like to do this from the AGI using "EXEC Flash" and then "EXEC SendDTMF 1234567".`
`18:06.59`	`*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`18:07.04`	`KnowWhat`	`what is spandsp?`
`18:07.59`	`elriah`	`Hi all. Does anyone know how to force a polycom phone to re-register after a preset number of seconds? in this case, 15 seconds. We're trying to find a quick/temporary fix to a NAT issue. I was hoping just setting reg.1.server.1.expires="15" would do the trick.`
`18:10.05`	`codazoda`	`Elriah, I just red a doc somewhere that said they were using that setting (set at 10 seconds) as a sort of "keep-alive" for the NAT. So, according to that doc, that should work.`
`18:10.07`	`danp`	`elriah: it should be basically the same but i use voIpProt.server.1.expires="600"`
`18:10.22`	`codazoda`	`elriah, http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501`
`18:10.27`	`elriah`	`danp: In which section? reg? (thanks)`
`18:10.45`	`codazoda`	`elriah, Search for "heartbeat" to jump to the stuff I was reading.`
`18:10.45`	`danp`	`under <voIpProt>`
`18:11.18`	`danp`	`i keep it in my site config, so it's basically <site><voIpProt><server ... /></voIpProt></site>`
`18:11.31`	`elriah`	`Ok, will try it. Thanks!`
`18:12.56`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`18:13.16`	`wunderkin`	`2.x has a nat keepalive yes`
`18:13.18`	`elriah`	`In a polycom mac.cfg, how does the order work in CONFIG_FILES? right to left or left to right as far as ovverides?`
`18:13.33`	`wunderkin`	`left to right`
`18:13.36`	`connecta`	`canapa, you can use voipdiscount for free outbound calls and stanaphone will give you a free did and free inbound minutes`
`18:13.41`	`connecta`	`together they're pretty bad ass`
`18:13.42`	`elriah`	`Ok, that's what I thought. Thanks.`
`18:15.11`	`CunningPike`	`elriah: It's left to right, with the leftmost file taking precedence`
`18:15.49`	`elriah`	`CunningPike (tnx)`
`18:15.57`	`CunningPike`	`elriah: ytw`
`18:16.08`	`BSDTech`	`when is 1.4 going to be production ready`
`18:17.03`	`myiagy`	`anyone ever used app_loquendo? does it work well?`
`18:17.15`	`guilherme_jorge`	`hello all, I've some problems to enable Asterisk 1.4.0 with Realtime. I've already compile addons, but res_mysql.so didnt created... Any idea?`
`18:17.28`	`file`	`BSDTech: that question can't be answered unless someone can look into the future`
`18:17.55`	`BSDTech`	`ok FIle`
`18:18.06`	`BSDTech`	`so I take it . its going to be some time`
`18:18.26`	`codazoda`	`Anybody here doing single line transfers on POTS?`
`18:18.31`	`BSDTech`	`and who do I email at digium about soundfile that should be but are missing`
`18:18.36`	`BSDTech`	`like holidays`
`18:19.39`	`BSDTech`	`and not just us holidays`
`18:19.44`	`BSDTech`	`but all holidays`
`18:20.42`	`BSDTech`	`states citeis countries`
`18:20.58`	`BSDTech`	`alls things that should be insounds and are not`
`18:22.01`	`file`	`1. It's impossible to say how long and whether it's production ready for your environment right now, I know people who are using it fine - but everyone's setup is different and can expose different issues 2. If you file a bug about the sounds someone can look at it, but I do not know the policy about new sounds that "should" be there... were they there in the old sounds?`
`18:22.31`	`*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)`
`18:23.12`	`angryuser`	`i still got "unable to handla indication 9" when i press flash-hook button;( (R on the phone)`
`18:23.21`	`angryuser`	`handle*`
`18:23.26`	`*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)`
`18:25.14`	`*** join/#asterisk route (n=skarecro@64.62.195.91)`
`18:25.25`	`*** join/#asterisk r2e (n=r2e@user-0ccengp.cable.mindspring.com)`
`18:25.47`	`KnowWhat`	`i want to make 10 local extensions in asterisk... anybody?`
`18:25.56`	`route`	`Would there be any reason my asterisk box would just suddenly stop working? Phones are dead, and trixbox reports asterisks stopped, but I can't restart it.`
`18:26.16`	`connecta`	`does it give an error when you try to start it?`
`18:26.23`	`route`	`nope, says OK`
`18:26.32`	`elriah`	`Anyone know why a polycom phone would display "Could not connect to boot server" on boot? It worked out of the box to get the firmware and config, but now on reboots it won't pull the new mac.cfg... It's definitely phone related... The hostname is right...`
`18:26.35`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`18:26.42`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`18:27.23`	`connecta`	`route, the obvious questions, was anyone working on the server, any changes that you can think of`
`18:27.33`	`connecta`	`elriah , do other phones pull configs ok?`
`18:27.38`	`route`	`I even tried rebooting the box, even though I'm well aware it's Linux, not Windbloze, so rebooting isn't necessary.`
`18:27.38`	`[TK]D-Fender`	`KnowWhat: Ok... you have our permission. Go for it.`
`18:28.00`	`route`	`connecta nope, the phones worked yesterday, but we just noticed about 15 minutes ago that they were out.`
`18:28.02`	`elriah`	`[TK]D-Fender: lol`
`18:28.17`	`connecta`	`are you using a tftp server, ftp server, or http server to host the files?`
`18:28.18`	`[TK]D-Fender`	`route: Would be nice if you tried starting * from the linux CLI and gave and actuall account of where it crashes...`
`18:28.25`	`route`	`Nobody has done anything at all on the server in the past 2 days.`
`18:28.26`	`elriah`	`connecta: ftp`
`18:29.21`	`route`	`[TK]D-Fender I was trynig to start from CLI, and it reported that it started OK, but it's still not running.`
`18:29.26`	`[TK]D-Fender`	`elriah: Either the IP / user/pass its using are wrong ; The files simply aren't there ; or the permissions on them are wrong.`
`18:29.29`	`elriah`	`connecta: They pull the first time out of the box, then won't pull a second time. I'm watching the ftp logs, they successfully "upload" their logs and login, but just don't login on boot to pull mac.cfg`
`18:29.34`	`elriah`	`permission.. hrm`
`18:29.41`	`connecta`	`oh ok`
`18:29.54`	`[TK]D-Fender`	`route elaborate onthe sings that tell you it started OK....`
`18:30.02`	`connecta`	`elriah, whats the username on the polycoms again`
`18:30.06`	`connecta`	`Plcmsim or somehting`
`18:30.09`	`elriah`	`PlcmSpIp`
`18:30.20`	`connecta`	`whats the home directory?`
`18:30.24`	`connecta`	`for that user in linux`
`18:30.29`	`route`	`Shutting down asterisk: [FAILED]`
`18:30.29`	`route`	`Starting asterisk: [ OK ]`
`18:30.37`	`elriah`	`-> /home/PlcmSpIp`
`18:30.45`	`elriah`	`route: debian?`
`18:30.51`	`elriah`	`route: or Ubunut?`
`18:30.52`	`route`	`elriah CentOS`
`18:30.56`	`[TK]D-Fender`	`route : that means nothing to me. Prove it by starting it NOT as a daemon, but LIVE`
`18:30.58`	`Nugget`	`Ubunut. heh.`
`18:31.09`	`route`	`I'd LOVE to run it on a Debian box, but was told it didn't work right.`
`18:31.12`	`elriah`	`Well, I don't think it's permissions because my vsftp log doesn't say 'denied'`
`18:31.15`	`elriah`	`route: Works great.`
`18:31.17`	`connecta`	`route: have you put any new files into the modules folder?`
`18:31.31`	`connecta`	`elriah, are you using trixbox by chance?`
`18:31.32`	`route`	`[TK]D-Fender tell me what to try and I'll try it. I need these phones up 30 minutes ago!`
`18:31.36`	`[TK]D-Fender`	`route: Thats BS. Debian is an OS like the rest if will work if you're competant.`
`18:31.51`	`elriah`	`connecta: Nope. Ubuntu LTS 6.06 w/vsftpd`
`18:31.56`	`Nugget`	`Asterisk doesn't care what Linux you use. Heck, Asterisk doesn't really care if it's Linux at all.`
`18:32.05`	`[TK]D-Fender`	`route: not knowing how to even START * is nearly a capital offense. "asterisk -gvvvvvc"`
`18:32.20`	`connecta`	`try temporarily setting permissions to 777 on the home directory and all the files in it`
`18:32.23`	`[TK]D-Fender`	`elriah: Verify your firewall, etc`
`18:32.37`	`route`	`connecta 2 or 3 days ago we put the hard drive into a different server and tested everything. We had several phones calls going at the same time, and everything worked perfectly. Absolutely nothing whatsoever has been touched or changed since then, and it just stopped working a little while ago.`
`18:33.16`	`connecta`	`elriah, use a ftp app to try to download and upload files to the directory manually`
`18:33.22`	`elriah`	`[TK]D-Fender: It's nothing like that, unfortunately, because I can just ftp from a local command line and it works great. hrm.. And the vsftpd log shows the connect...`
`18:33.33`	`elriah`	`connecta: That works great.`
`18:33.38`	`elriah`	`Odd, eh?`
`18:34.02`	`[TK]D-Fender`	`elriah: I saw a guy who had that problem, and it was firewall related. Look outside your box.`
`18:34.05`	`connecta`	`elriah, yes, the phone says could not connect, yet the log shows that it does connect but doesnt try to download`
`18:34.19`	`connecta`	`i had that problem and i doubt it's firewall`
`18:34.19`	`[TK]D-Fender`	`elriah: and double check your files list in your <mac>.cfg file`
`18:34.34`	`connecta`	`route`
`18:35.06`	`nowork`	`hello i hv two question 1) can i delete /var/log/asterisk/full? 2) how can i check the calls codec acutally using ?`
`18:35.14`	`elriah`	`[TK]D-Fender: It worked for the first pulling of the mac.config files and updated firmware, just not subsequent changes to a file that the mac.cfg poings too, in this case an extension.`
`18:35.19`	`connecta`	`[TK]D-fender , can you give route the command to tail the log`
`18:35.21`	`elriah`	`extension.cfg`
`18:35.38`	`elriah`	`weird...`
`18:35.40`	`connecta`	`route , i think it might be ' tail /var/log/asterisk/full`
`18:35.50`	`elriah`	`tail -f /var/log/whateverlogfile`
`18:36.16`	`[TK]D-Fender`	`connecta: Nope, don't know it personally. he should jsut START it from CLI and see the last thing that appears before it bombs`
`18:36.17`	`connecta`	`nowork 2) sip show channels or iax2 show channels`
`18:36.22`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`18:36.42`	`route`	`Feb 12 13:33:28 WARNING[4501] loader.c: Loading module chan_zap.so failed!`
`18:36.58`	`connecta`	`there you go, it's the module`
`18:37.17`	`route`	`Why would it suddenly fail if nobody changed anything?`
`18:37.24`	`route`	`Feb 12 13:33:28 WARNING[4501] chan_zap.c: Unable to specify channel 1: No such device or address`
`18:37.34`	`connecta`	`route . since some modules are dependet on processor type, by mjoving the Hard drive to another pc, you may have to replace modules. but you're right, it doesnt make sense that it did work at one time`
`18:37.41`	`nowork`	`connecta: thanks..`
`18:37.55`	`codazoda`	`I'm using AsteriskNow, have a digium TDM2400E with 2 FXO ports. I have a soft-phone connected and a phone line connected. Only port 24 is connected on the analog card. I can call the system fine, but it won't make calls out. Any ideas?`
`18:38.35`	`connecta`	`elriah , are you sure the mac.cfg file is pointing to a valid extension.cfg`
`18:38.38`	`[TK]D-Fender`	`route : Guess your card might have flaked out.`
`18:38.46`	`connecta`	`yah could be a card problem`
`18:39.05`	`[TK]D-Fender`	`route: type in "ztcfg -vvvv" and see if it reports any errors. if not, try starting * again manually`
`18:39.08`	`route`	`really? hrm, let me go check something... brb`
`18:39.19`	`elriah`	`connecta: Yea. I'm messing with permissions right now on the ftproot. Don't know where else to look.`
`18:39.27`	`route`	`ZT_CHANCONFIG failed on channel 1: No such device or address (6)`
`18:39.30`	`elriah`	`Maybe in modifying some of the files the perms were reset...`
`18:39.42`	`route`	`^^^ that's what it said when I ran ztcfg -vvvv`
`18:39.43`	`[TK]D-Fender`	`route: what card are you running?`
`18:40.06`	`route`	`[TK]D-Fender X100P`
`18:40.19`	`[TK]D-Fender`	`route : do "modprobe wcfxo" then redo the rest`
`18:40.25`	`elriah`	`Route: I have a TDM400P for sale if you want it. Throw that X100P in the garbage..`
`18:40.28`	`*** join/#asterisk yassine (n=yassine@dsl.voicint.com)`
`18:40.35`	`connecta`	`elriah , how many phyones do you have`
`18:40.38`	`elriah`	`Althought it's probably not your problem with this issue...`
`18:40.42`	`queuetue`	`I've got a problem authenticating from sipura 2000 to an asterisknow box - both x305 and x306 are set up on it, and when 305 tries to call out, I get the error "WARNING[1103]: chan_sip.c:8023 check_auth: username mismatch, have <306>, digest has <305>", followed by "NOTICE[1103]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user ..." I see a similar problem reported at http://bugs.digium.com/view.php?id=8388 but no resolution. Can a`
`18:40.55`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`18:41.01`	`elriah`	`connecta: 70 or so...`
`18:41.14`	`route`	`elriah how much?`
`18:41.21`	`connecta`	`elriah, i think i got it`
`18:41.31`	`PakiPenguin`	`hi`
`18:41.51`	`elriah`	`route: hrm.. retails for abour $450, ebay for around $375-400, say $275+shipping. It has four FXO cards (supports 4 POTS lines)`
`18:41.53`	`connecta`	`it's possible that you ahd working configs, and then you upgraded the bootrom.ld and sip.ld and now your config files are invalid`
`18:41.56`	`[TK]D-Fender`	`queuetue: Just post it on the MB's already..... or at least provide pastebins of your setup`
`18:42.08`	`route`	`elriah and is it full height, or low profile? Cuz the machine it's in can only take low profile.`
`18:42.33`	`elriah`	`connecta: I thought of that, but these are the new config files that came with the latest firmware... hrm...`
`18:42.39`	`elriah`	`connecta: btw, thanks for the help :)`
`18:42.39`	`queuetue`	`[TK]D-Fender: What's an MB? (Sorry if I'm annoying you...)`
`18:42.45`	`[TK]D-Fender`	`route: then get a Sangoma A200.`
`18:42.47`	`elriah`	`route: std height`
`18:42.55`	`*** join/#asterisk cpatry (n=junky@modemcable105.205-56-74.mc.videotron.ca)`
`18:42.56`	`[TK]D-Fender`	`queuetue: * Message boards / mailing lists`
`18:42.59`	`connecta`	`and when you do an ls -l , all the permissions are set properly?`
`18:43.08`	`cpatry`	`some1 has successfully cross zaptel for ppc?`
`18:43.08`	`elriah`	`route: I don't think they make a half height TDM card.`
`18:43.22`	`queuetue`	`Ok, I thought this was an appropriate place to ask questions.`
`18:43.23`	`[TK]D-Fender`	`A200 is LP card.`
`18:43.34`	`elriah`	`connecta: I'm going to go get a new phone and try it again paying close attention to the ftp logs.. be back in a bit..`
`18:43.34`	`route`	`I've got a spare X100P, let me swap it out real quick and see what happens.`
`18:43.35`	`*** join/#asterisk orkid_ (n=orkid@dataq2.utias.utoronto.ca)`
`18:43.57`	`elriah`	`route: If you want that card, PM me with your email address, I'm stepping away from my computer for a few...`
`18:44.14`	`connecta`	`queuetue , this is an appropriate place to ask questions. hang out and someone might be able to help`
`18:44.31`	`route`	`[TK]D-Fender where can I find one of those? I don't see one on eBay.`
`18:44.59`	`[TK]D-Fender`	`route: Try and actual RETAILER. You evidently are stuck in "cheap mode"`
`18:45.17`	`queuetue`	`connecta: I thought I was, but I think I did something to tick off [TK]D-Fender ... I'll try the message boards.`
`18:45.36`	`J4k3`	`unluckily for voip gear, especially anything that isn't a piece of shit, ebay isn't exactly cheap.`
`18:45.44`	`J4k3`	`if its a piece of shit, its cheap and you shouldn't want it.`
`18:45.56`	`*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`18:46.00`	`connecta`	`queuetue , no, you'll frequently get brushed off in IRC rooms . don't let it get to you. just keep asking your questions`
`18:46.06`	`[TK]D-Fender`	`queuetue: You just keep asking the same questio in cut&paste manner and provide nothing to back it up. Show us your configs, don't just ask "why don't they work?!". We can't SEE them.`
`18:46.33`	`[TK]D-Fender`	`PASTEBIN <----------`
`18:47.09`	`connecta`	`[TK]D-Fender, thats kind of a crappy attitude because if you are tyring to get something to work, theres a bunch of different configs that might cause an issue`
`18:47.16`	`connecta`	`theres no way to just paste all the info`
`18:47.39`	`queuetue`	`[TK]D-Fender: What config do you want to see? It's a pretty basic SIP extension, I just don't know what the digest error I was receiving means.`
`18:47.46`	`[TK]D-Fender`	`connecta: Auth is ALL sip.conf and pastebin.ca works for the rest of the known universe`
`18:47.51`	`*** join/#asterisk SycGuy (n=eric_hit@24-38-26-218-st.pittpa.adelphia.net)`
`18:48.31`	`connecta`	`It's great to post info, but if you don't know what info to post, then i guess you deserve to get treated like an ass ?`
`18:48.42`	`*** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca)`
`18:48.46`	`connecta`	`All you can really do is explain the symptoms and if someone can help, they can ask the appropriate questions and ask to see specific files or logs`
`18:49.47`	`J4k3`	`well`
`18:49.50`	`connecta`	`actually this guy gave a bunch of good info and a fairly detailed question`
`18:49.51`	`J4k3`	`you just got advised what to do`
`18:50.04`	`J4k3`	`paste the log entries you have questions about`
`18:50.11`	`J4k3`	`paste the appropriate info from sip.conf`
`18:50.15`	`J4k3`	`into one pastebin`
`18:50.18`	`J4k3`	`post the url`
`18:50.26`	`krondorl`	`A I missing something or is there not a place that one can indicate the longest time a voicemail message can be left to ensure that large files don't bring down the phone system.`
`18:50.30`	`J4k3`	`you'll likely get better feedback`
`18:50.46`	`krondorl`	`A=Am`
`18:50.58`	`J4k3`	`krondorl: bah, disk space is cheap. Nobody likes to get hustled on their voicemail`
`18:51.18`	`J4k3`	`I canceled a cellular provider because of that crap. When I say "Save a voicemail" I don't mean "bother me in 3 days about it again"`
`18:51.38`	`krondorl`	`I understand that but asterisk blows up when a voicemail file is greater that 2 gigs in size.`
`18:51.46`	`connecta`	`i think krondorl means the maximum length of a message`
`18:51.59`	`J4k3`	`2GB @ GSM rates... lets see...`
`18:52.06`	`route`	`[TK]D-Fender yes, unfortunately I'm stuck in "cheap mode". The company is already WAY over budget.`
`18:52.20`	`*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)`
`18:52.25`	`J4k3`	`an 8.5 year voicemail @ GSM quality.`
`18:52.36`	`J4k3`	`err, that can't be right.`
`18:52.49`	`*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`18:53.09`	`[TK]D-Fender`	`connecta: This isn't rocket science. Paste the entire [general] section, all your registere's, and all peers/users/firends involved in the failed attempt.`
`18:53.09`	`J4k3`	`oh, a 74 hour voice mail... thats better`
`18:53.15`	`route`	`Swapped out the X100p card and it's working now.`
`18:53.54`	`connecta`	`[TK]D-Fender, just don't be a dick to people. thats not rocket science either`
`18:54.06`	`mercestes`	`j4k3: Be careful, there could be an imoprtant message in the middle of that.`
`18:54.16`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net)`
`18:54.43`	`[TK]D-Fender`	`sigh`
`18:55.02`	`route`	`[TK]D-Fender looks like the card DID flake out. Where can I buy one of those A200 cards, and what do they usually run?`
`18:55.34`	`[TK]D-Fender`	`route : where are you located?`
`18:56.04`	`[TK]D-Fender`	`route : Try www.atacomm.com or www.telephonydepot.com . Both should be a bit cheaper than VoIP Supply.`
`18:56.27`	`route`	`[TK]D-Fender in New Jersey`
`18:56.46`	`route`	`We already have an account with Voipsupply, which is why I was gonna check there.`
`18:57.22`	`[TK]D-Fender`	`route: Oh, do shop around. In the end its all just commodity equipment, there is no "value added`
`18:57.38`	`connecta`	`yah i find atacomm to be really good and cheap`
`18:59.02`	`krondorl`	`connetca: yes that's what I am looking for.. Max length of a message.. stop it after 30 mins if possible..`
`18:59.02`	`nowork`	`which is the best website to learn asterisk?`
`18:59.26`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:59.29`	`connecta`	`www.voip-info.org www.asteriskguru.com www.asterisktutorials.com`
`18:59.35`	`[TK]D-Fender`	`nowork: ....`
`18:59.37`	`[TK]D-Fender`	`~book`
`18:59.39`	`jbot`	`somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`19:00.00`	`thekidrio`	`thats the one i started with`
`19:00.04`	`thekidrio`	`the oreilly one`
`19:00.18`	`[TK]D-Fender`	`nowork: As you read the book you should be working with * locally. if you want more detailed info on specific points THEN you should try the WIKI`
`19:00.20`	`thekidrio`	`its written with a good sense of humor heh`
`19:00.21`	`[TK]D-Fender`	`~wikis`
`19:00.22`	`jbot`	`somebody said wikis was http://www.voip-info.org`
`19:00.36`	`thekidrio`	`cracks me up sometimes, in a geeky way`
`19:01.05`	`nowork`	`okay..thank you..`
`19:03.10`	`route`	`[TK]D-Fender cheapest I found is $199... That's just not in the budget at the moment. :-(`
`19:04.03`	`[TK]D-Fender`	`route : Ask yourself what your problems cost you and then ack accordingly.`
`19:04.08`	`route`	`How much memory would you guys recommend having to run Asterisk, besides "as much as possible"?`
`19:04.21`	`connecta`	`it uses far less memory than processor`
`19:04.21`	`[TK]D-Fender`	`route: If you can afford a flakey server and down-time, poor call quality etc... hey whatever right?`
`19:04.26`	`J4k3`	`is there any way to convience the echo canceler that screwing up conversation quality isn't a good plan?`
`19:04.33`	`[TK]D-Fender`	`route: Depends how much you're asking of it.`
`19:04.44`	`Cresl1n`	`J4k3: what's wrong?`
`19:04.46`	`*** join/#asterisk s1gny\|wrk (n=s1gny@p54917ED1.dip.t-dialin.net)`
`19:04.46`	`connecta`	`so for 50 calls, 256 megs might be sufficient`
`19:04.59`	`connecta`	`but as always , the more the better`
`19:05.22`	`route`	`[TK]D-Fender the CEO of the company said it's not in the budget, plain and simple.`
`19:05.27`	`*** part/#asterisk s1gny\|wrk (n=s1gny@p54917ED1.dip.t-dialin.net)`
`19:05.32`	`route`	`He's standing over my shoulder watching me type. lol`
`19:05.49`	`[TK]D-Fender`	`route : Ok, tell him to see you about that when he's tired of any problems arise from your current setup.`
`19:06.04`	`codazoda`	`I can't get a TDM2400 to dial out. It answers fine, but I can't get it to call out. I don't see anything in the asterisk CLI that indicates it even tried.`
`19:06.11`	`route`	`connecta oh, that's good. We don't have nearly that many calls going through the system yet. The box has 256MB right now, and we'll be upgrading it to 384 or 512 in a few days.`
`19:06.14`	`[TK]D-Fender`	`route: Means VS ends. Thats all that need be said.`
`19:06.33`	`[TK]D-Fender`	`codazoda: If it didn't even try to dial, its not your card, its your dialplan`
`19:06.38`	`connecta`	`route: Yes, most businesses i deal with rely on their phones heavily and can't afford an ouatage`
`19:07.00`	`codazoda`	`How can I tell for sure if it tried? Would the CLI have said something about it, so I can assume it did not?`
`19:07.01`	`[TK]D-Fender`	`route : I run just fine on 512.`
`19:07.25`	`[TK]D-Fender`	`codazoda: You should have your verbose turned up enough to see a Dial command being called....`
`19:07.53`	`route`	`connecta We don't use the phones much at all, but it IS important that they be available.`
`19:08.23`	`nowork`	`codazoda: maybe u can see /var/log/asterisk/full`
`19:08.40`	`connecta`	`well, a reliable motherboard, processor, and analog cards are all important. i would definately recommend keeping a spare card as you just learned how important that can be`
`19:08.49`	`route`	`In the future they will be used a lot more, especially the remote extentions, as most of our employees work from home and use IP phones connected to our network.`
`19:09.17`	`J4k3`	`hmm`
`19:09.29`	`route`	`connecta you are absolutely right. It's ALWAYS important to have a spare.`
`19:09.46`	`J4k3`	`so... how long does it take digium to respond to support requests concerning purchased codec packs?`
`19:09.51`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`19:09.59`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`19:10.09`	`J4k3`	`backing up and restoring the registration file wasn't good enough for them. Fucking annoying payware bullshit.`
`19:10.21`	`codazoda`	`hm. I'm using "AsteriskNOW". I don't have a /var/log/asterisk/full... I set verbose to 9 and dialed (no info). Set it to 100 and dialed (still no info). No idea how high it goes.`
`19:10.50`	`[TK]D-Fender`	`codazoda: Well so far its sounding like you don't even have a dialplan match to start from....`
`19:11.02`	`*** join/#asterisk b3 (n=xcla@static-b5-251-83.telepac.pt)`
`19:11.06`	`b3`	`Hi all`
`19:11.08`	`[TK]D-Fender`	`codazoda: Go look at your extensions.conf and see whats missing.`
`19:11.36`	`route`	`is there no log rotator for /var/log/asterisk/full ?`
`19:12.09`	`*** join/#asterisk Modcuts (n=Moducts@88-109-10-22.dynamic.dsl.as9105.com)`
`19:12.40`	`nowork`	`route: what shall we do when this " full " file keep getting bigger and bigger?`
`19:12.43`	`b3`	`i have a newbie question: how to configure 2003 server to route voice packets to sip phones?`
`19:13.13`	`connecta`	`can you rephrase your question please`
`19:13.17`	`anonymouz666`	`codazoda: asterisknow uses 1.2 or 1.4?`
`19:13.19`	`b3`	`ok`
`19:13.34`	`[TK]D-Fender`	`b3: Thats a super-general Windows network routing question. nothing SIP specific in there...`
`19:14.17`	`b3`	`i have a trixbox server with a DMZ of the router...`
`19:14.39`	`b3`	`i have 10 sip phones connected to another router and talking`
`19:15.02`	`b3`	`i have 5 sip phones connected to a 2003 server nat and they only register`
`19:15.12`	`b3`	`they don't talk`
`19:15.19`	`connecta`	`That really is a crazy ridiculous layout`
`19:15.33`	`b3`	`?`
`19:15.34`	`connecta`	`for the record.`
`19:16.20`	`connecta`	`if you could maybe draw it real quick in mspaint, i think it would help a lot`
`19:16.40`	`b3`	`the connection trixbox -> Router -> Internet <- Router <- Sip Phones Works ok`
`19:16.41`	`b3`	`but`
`19:17.12`	`b3`	`Trixbox -> router -> internet <- Windows 2003 <- sip phones doesn't worl`
`19:17.24`	`mercestes`	`~trixbox`
`19:17.26`	`jbot`	`extra, extra, read all about it, trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/`
`19:17.29`	`*** join/#asterisk atlantia (n=scott@64.20.151.221)`
`19:18.06`	`atlantia`	`so trying the hello-world part of the "book", on an asterisk-now install.`
`19:18.13`	`connecta`	`b3: well, that helps a lot. in the first working config, does the second router perform nat`
`19:18.18`	`atlantia`	`moved extensions.conf to extensions.conf.sample`
`19:18.38`	`atlantia`	`added [incoming]`
`19:18.39`	`atlantia`	`exten => s,1,Answer( )`
`19:18.39`	`atlantia`	`exten => s,2,Playback(hello-world)`
`19:18.39`	`atlantia`	`exten => s,3,Hangup( )`
`19:18.43`	`giasai68`	`hello`
`19:18.47`	`b3`	`the thing is i have tried the asterisk on debian and the problem is the same`
`19:18.49`	`atlantia`	`to a fresh extensions.conf file`
`19:18.55`	`giasai68`	`pls, give me an suggest`
`19:19.08`	`J4k3`	`hmm... problem solved`
`19:19.21`	`J4k3`	`since I have g729 licenses, I don't have to use digium's codec to use g729 legally.`
`19:19.29`	`connecta`	`b3: if the trixbox is not behind nat, thats good. if the first set of sip phones are behind nat and do work, thats good`
`19:19.40`	`giasai68`	`I I have this warning: src/chan_h323.c:909 ooh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/64)`
`19:19.47`	`giasai68`	`how I can fix it?`
`19:19.52`	`J4k3`	`at least up to the total of concurrent lines installed. Yay. Screw digium's broken auth.`
`19:20.02`	`*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-b3ae5d253d31b3df)`
`19:20.47`	`*** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net)`
`19:20.57`	`kuku5`	`Any sources for cheap 7940's?`
`19:21.16`	`*** join/#asterisk Strom_M (n=strom@m125e36d0.tmodns.net)`
`19:21.58`	`giasai68`	`I I have this warning: src/chan_h323.c:909 ooh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/64)`
`19:21.59`	`giasai68`	`how I can fix it?`
`19:22.19`	`atlantia`	`so my question is what did i break? seems the old extensions file wasn't set to have the [incoming] context.. trying to figure out where that pointer is`
`19:23.36`	`giasai68`	`I'm using asterisk 1.4 and zapata 1.4`
`19:23.52`	`kuku5`	`toresbe: cisco :)`
`19:23.56`	`*** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`19:23.57`	`*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)`
`19:26.18`	`atlantia`	`can anyone here help me understand how this works?`
`19:28.04`	`nowork`	`sip ata device should be setup as extension or sip trunk? i think both will work.not sure which one is better`
`19:28.04`	`*** part/#asterisk Strom_M (n=strom@m125e36d0.tmodns.net)`
`19:28.04`	`*** join/#asterisk Strom_M (n=strom@m125e36d0.tmodns.net)`
`19:28.07`	`Strom_M`	`stupid button`
`19:28.21`	`[TK]D-Fender`	`giasai68: We've already answered this. You do not have any G.729 licenses on your server. Go read about them on Digium.com`
`19:29.37`	`[TK]D-Fender`	`nowork: As what would commonly be called an "extension" by most...`
`19:30.20`	`anonymouz666`	`asterisknow uses 1.2 or 1.4 version?`
`19:31.27`	`tzafrir_laptop`	`[TK]D-Fender, what's the command to show the number of avaible licenses? g729 show codecs?`
`19:31.59`	`tzafrir_laptop`	`anonymouz666, 1.4`
`19:32.34`	`cpatry`	`isnt g729 show licenses?`
`19:33.06`	`*** join/#asterisk Vec (n=Vector@dsl-243-116-81.telkomadsl.co.za)`
`19:33.08`	`giasai68`	`I have installed g729a license`
`19:35.27`	`elriah`	`If I have two codecs in my sip.conf (allow=g729, allow=gsm) and no g729 licenses are available, will the call still complete with gsm?`
`19:36.12`	`giasai68`	`CLI> show g729`
`19:36.12`	`giasai68`	`0/0 encoders/decoders of 1 licensed channels are currently in use`
`19:36.23`	`Corydon-w`	`elriah: no, it will not`
`19:36.27`	`giasai68`	`how I can load it?�`
`19:36.53`	`*** join/#asterisk Renacor (n=vircuser@p54B9D172.dip.t-dialin.net)`
`19:37.02`	`Renacor`	`anybody know why I keep getting app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)`
`19:37.03`	`Renacor`	`<PROTECTED>`
`19:37.19`	`*** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net)`
`19:37.28`	`*** part/#asterisk SycGuy (n=eric_hit@24-38-26-218-st.pittpa.adelphia.net)`
`19:38.59`	`[TK]D-Fender`	`Renacor: pastebin the high-verbose CLI output of your failed call attempt, as your zapata.conf.`
`19:39.04`	`[TK]D-Fender`	`and*`
`19:39.31`	`cpatry`	`if ya do show channeltype zap, theres probably no driver loaded.`
`19:41.34`	`Renacor`	`k one sec`
`19:45.54`	`Renacor`	`http://pastebin.ca/352360`
`19:46.57`	`*** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com)`
`19:47.14`	`*** join/#asterisk atlantiatech (n=scott@64.20.158.177)`
`19:47.29`	`*** join/#asterisk xcla (n=info@static-b5-251-83.telepac.pt)`
`19:47.45`	`atlantiatech`	`ok this is frustrating`
`19:48.45`	`Defend`	`is there a command to close a channel?`
`19:49.35`	`cpatry`	`Defend: read the doc: soft hangup chan.`
`19:49.40`	`mafkees`	`Defend: soft hangup channel`
`19:49.47`	`Defend`	`k thanks fellas`
`19:50.26`	`*** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com)`
`19:50.41`	`riddlebox`	`does the tdm400 cards pass caller id to ananlog extensions?`
`19:50.54`	`Renacor`	`no clue`
`19:53.10`	`[TK]D-Fender`	`Renacor: You do not have a single Zap channel defined in your zapata.conf. there's your problem.`
`19:55.10`	`[TK]D-Fender`	`riddlebox: Yes`
`19:55.38`	`Renacor`	`heh k`
`19:57.48`	`*** join/#asterisk kervel (n=kervel@cable-213-132-140-61.upc.chello.be)`
`19:58.04`	`kervel`	`hello, can anyone help me a bit with the installation of a B410P card in belgium ,`
`19:58.29`	`kervel`	`i think i got the driver part already working (the card gets detected by the kernel, and misdn-init start reports no errors`
`19:58.38`	`nowork`	`TK: so, you mean, most people setup the ATA device on "extension" not sip-trunk..any idea what is the benefit?`
`19:59.10`	`kervel`	`but i'm not sure about what port type to use in belgium, and if it is possible to do some testing on the line (eg detecting rings, dial a number , ..)`
`19:59.12`	`Renacor`	`[TK]D-Fender: k I did signalling=fxo_ks language=de context=ext_germany channel => 1 but i still have the same problem`
`19:59.25`	`kervel`	`so i can verify the line is working before trying to connect to asterisk`
`20:01.00`	`Zaw`	`what's the recommended codec to use for the best quality calls (lowest latency, least jitter) ?`
`20:01.56`	`[TK]D-Fender`	`Renacor: Did you completely stop * and restart it?`
`20:02.14`	`kervel`	`and i have a very simple question: should the light on the back of the B410P card be on when the cable is connected to an isdn nt1 box ?`
`20:02.27`	`Renacor`	`[TK]D-Fender: yes`
`20:03.18`	`dlynes_laptop`	`Anyone know why the new version of zaptel is more than twice the size of the previous version?`
`20:03.18`	`[TK]D-Fender`	`Renacor: ok, completely remove all commented lines permanently from zaptel.conf and zapata.conf , then pastebin both of them. Also pastebin the output of "ztcfg -vvvv" and "cat /proc/interrupts"`
`20:03.45`	`[TK]D-Fender`	`dlynes_laptop: Support for new EC routines, and a large rewite of the PCI interface for reduced load and better inter-op.`
`20:03.50`	`Renacor`	`k`
`20:04.06`	`dlynes_laptop`	`[TK]D-Fender: ah`
`20:04.26`	`dlynes_laptop`	`[TK]D-Fender: Does it affect sangoma cards at all?`
`20:04.51`	`[TK]D-Fender`	`dlynes_laptop: Not one bit I'm sure. Wanpipe just hooks into Zaptel at the point where it passes frames, so it should be unchanged.`
`20:05.09`	`dlynes_laptop`	`ok...didn't figure it would`
`20:05.20`	`dlynes_laptop`	`but i guess it breaks the latest wanpipe, too`
`20:05.44`	`dlynes_laptop`	`because the entry points it expects aren't there anymore`
`20:06.43`	`[TK]D-Fender`	`dlynes_laptop: Dunno, haven't tried`
`20:06.49`	`Renacor`	`http://pastebin.ca/352391`
`20:07.03`	`dlynes_laptop`	`[TK]D-Fender: well, i'm just warning you...it doesn't work :)`
`20:07.56`	`kervel`	`hmm, i installed chan_misdn and now asterisk crashes right after loading misdn`
`20:08.04`	`[TK]D-Fender`	`dlynes_laptop: Then it wouldn't be a GUESS now would it :)`
`20:08.15`	`kervel`	`are the ubuntu packages of asterisk okay ?`
`20:08.43`	`dlynes_laptop`	`[TK]D-Fender: but looks like 1.2.12 doesn't work on linux 2.6.20 either`
`20:10.48`	`[TK]D-Fender`	`Renacor: Congratulations, progress. Now you should realize that your GROUP #'s are all wrong.`
`20:11.40`	`Renacor`	`weee`
`20:12.07`	`*** join/#asterisk voipanywhere (n=pirch@a81-84-60-32.cpe.netcabo.pt)`
`20:12.13`	`Renacor`	`[TK]D-Fender: how can I fix them?`
`20:12.43`	`[TK]D-Fender`	`Renacor: You are dialing "g1" Nowhre have you defined a zapata.conf channel that uses that #.`
`20:12.57`	`Renacor`	`i see`
`20:13.01`	`[TK]D-Fender`	`Renacor: You have also ims-ordered several line in there.`
`20:13.31`	`[TK]D-Fender`	`Renacor: 10-14 should occur BEFORE line 9 (as appeares in your pastebin). repeat this cleanup for others as well.`
`20:13.38`	`[TK]D-Fender`	`mis*`
`20:14.10`	`*** join/#asterisk tuan_modulis (n=chatzill@hvmoduli.enter-net.com)`
`20:14.24`	`voipanywhere`	`has anyone used chan_cellphone? Does anyone knows a way to avoid a call being bridged right after it send the number to the cellphone?`
`20:14.30`	`*** join/#asterisk ionutdavidescu (i=davidf@86.107.82.58)`
`20:15.09`	`atlantiatech`	`can someone help me understand why that when i change extension.conf to reflect the "hello-world" scenario from "the book" it still has the old message on it?`
`20:15.40`	`[TK]D-Fender`	`atlantiatech: what old message?`
`20:16.43`	`atlantiatech`	`[TK]D-Fender, the stock one from the AtseriskNow install`
`20:16.43`	`ionutdavidescu`	`hi everybody..`
`20:16.43`	`[TK]D-Fender`	`atlantiatech: You mean the cctual content of the recording?`
`20:16.43`	`ionutdavidescu`	`i have some problems with the asterisk gui and asteriks`
`20:16.43`	`Renacor`	`[TK]D-Fender: whats ims-ordered ?`
`20:16.43`	`ionutdavidescu`	`asteriks is running fine.`
`20:16.44`	`[TK]D-Fender`	`Renacor: Mis-ordered`
`20:16.48`	`ionutdavidescu`	`ai started to configure users. i added a sip user. everything looks good in the GUI`
`20:16.50`	`Renacor`	`oh in zaptel.conf or zapata.conf?`
`20:16.52`	`ionutdavidescu`	`but in CLI a sip show users does not show any user`
`20:17.00`	`ionutdavidescu`	`i tryned to register that user and it says registration error.`
`20:17.04`	`[TK]D-Fender`	`Renacor: ZApata.conf`
`20:17.05`	`atlantiatech`	`[TK]D-Fender, well, according to the book, it says to basically create an "s" extension, which answers, plays "hello-world" and hangs up`
`20:17.16`	`atlantiatech`	`[TK]D-Fender, i moved extensions.conf from the install to .conf.sample`
`20:17.38`	`atlantiatech`	`and than created the one they mention, with only the hello world extension under incoming`
`20:17.43`	`ionutdavidescu`	`does somebody know why the users created in the GUI does not show with sip show users?`
`20:17.49`	`[TK]D-Fender`	`Renacor: Also your TDM card is sharing interrupts with 2 other devices. NOT good.`
`20:18.22`	`[TK]D-Fender`	`atlantiatech: Contexts matter, but for the divice PLACING the call, as well as where it leads to in extensions.conf.`
`20:19.03`	`[TK]D-Fender`	`atlantiatech: Remove all the junk you aren't actively using, and all comments from it, then pastebin your extensions.conf and the CLI output of a failed call attempt on verbose 10`
`20:19.06`	`atlantiatech`	`[TK]D-Fender, no sip etc here, just a glorified answering machine for now, but yeah, i am guessing context is wrong`
`20:19.07`	`Renacor`	`[TK]D-Fender: but line 9 is commented out no?`
`20:19.09`	`[TK]D-Fender`	`~pb`
`20:19.10`	`jbot`	`extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`20:19.39`	`Renacor`	`[TK]D-Fender: oh you mean in the pastebin`
`20:19.39`	`[TK]D-Fender`	`Renacor: : in your pastebin... the lines are numbered : http://pastebin.ca/352391`
`20:20.21`	`[TK]D-Fender`	`Renacor: Yes, realize that the order matters, and all your parameters have to be set BEFORE the "channel=>" line it is to apply to. Do not leave blank entires like you have in there.`
`20:20.37`	`Corydon-w`	`Anybody know if there's a way to program a Cisco 7960 with a SIP image to do a one-button transfer to 700?`
`20:20.40`	`*** part/#asterisk saftsack (n=saftsack@pD9E04C9F.dip.t-dialin.net)`
`20:21.14`	`Qwell[]`	`Corydon-w: You must be new.`
`20:21.18`	`Qwell[]`	`cisco + sip == suck`
`20:21.32`	`Corydon-w`	`Qwell[]: I must be. I'm coming up with zilch.`
`20:21.41`	`*** join/#asterisk blitz[training] (n=blitzrag@66.135.99.122)`
`20:21.59`	`[TK]D-Fender`	`Corydon-w: Oh... then you've already found the answer :)`
`20:22.32`	`ionutdavidescu`	`does somebody know whay the users added with the GUI are not shown with sip show users, but the users are present in user.conf...?`
`20:22.40`	`Corydon-w`	`[TK]D-Fender: I was hoping someone had some secret documentation that made it possible.`
`20:22.42`	`*** join/#asterisk X-Rob (n=rob-x@124.150.99.11)`
`20:22.46`	`tzanger`	`heh`
`20:22.47`	`tzanger`	`www.willitblend.com`
`20:22.53`	`queuetue`	`[TK]D-Fender: Just so you know, I did listen -- I've posted the problem at http://bugs.digium.com/view.php?id=9044, and stopped spamming irc with my issue. :)`
`20:22.54`	`[TK]D-Fender`	`lol`
`20:23.03`	`ionutdavidescu`	`I am trying to solve thie for about 3 hours and I can;t see the problem here`
`20:23.13`	`Corydon-w`	`tzanger: is that like Letterman's "Will It Float?"`
`20:23.21`	`tzanger`	`dunno`
`20:23.25`	`J4k3`	`or better yet`
`20:23.28`	`J4k3`	`www.willitlend.com`
`20:23.30`	`[TK]D-Fender`	`queuetue: I DID say that if you wanted to ask here, you should at least SHOW us your configs instead of asking why they don't work, and leaving us blind...`
`20:24.19`	`[TK]D-Fender`	`ionutdavidescu: : "sip show peers".`
`20:25.09`	`*** join/#asterisk infernix (n=nix@spirit.infernix.net)`
`20:25.49`	`queuetue`	`Well, I also said I was using standard asterisknow configs, but that's neither here nor there. I'm told by someone at digium that it looks like a bug...`
`20:25.55`	`Renacor`	`[TK]D-Fender: hey how can you tell from /proc/interrupts which interrupt the card is on?`
`20:26.11`	`*** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz)`
`20:26.21`	`dj-fu`	`how can I debug why asterisk isn't starting with the init script?`
`20:26.23`	`[TK]D-Fender`	`Renacor: Look at the line with TDM in it. Should be fairly obviou`
`20:26.32`	`*** join/#asterisk giasai68 (n=administ@ip-3-156.sn2.eutelia.it)`
`20:26.38`	`[TK]D-Fender`	`dj-fu: Try running it by hand and see what happens`
`20:26.40`	`dj-fu`	if I run it by `asterisk -vvvvvc` it runs fine, sits at the CLI. but if I /etc/init.d/asterisk start it doesn't launch
`20:26.43`	`dj-fu`	`^^.`
`20:27.03`	`Renacor`	`ahh i see sorry`
`20:27.10`	`[TK]D-Fender`	`dj-fu: When you try, what exactly do you see?`
`20:27.25`	`Renacor`	`[TK]D-Fender: how can I change what interrupt it's on?`
`20:27.32`	`Renacor`	`with modprobe?`
`20:27.38`	`dj-fu`	`<PROTECTED>`
`20:27.53`	`dj-fu`	`and then ps aux\|grep asterisk returns nothihng`
`20:27.59`	`[TK]D-Fender`	`Renacor: In your BIOS. First disable everything you don't need, see how that works out. Then try seeing if it lets you dedicate on for a slot.`
`20:28.19`	`Renacor`	`k lemme try this again cause it's still not working`
`20:28.19`	`[TK]D-Fender`	`dj-fu: And you ran * manually as user "asterisk"?`
`20:28.26`	`dj-fu`	`[TK]D-Fender, as root`
`20:28.27`	`dj-fu`	`doh`
`20:28.30`	`dj-fu`	`that's wha tit is. can't read config file`
`20:28.31`	`dj-fu`	`thanks`
`20:28.32`	`CunningPike`	`dj-fu: Check your paths in /etc/init.d/asterisk`
`20:28.42`	`CunningPike`	`dj-fu: nm`
`20:29.18`	`dj-fu`	`sweet - I deleted zapata.conf accidently and restored it from a backup as root`
`20:29.23`	`russellb`	`nice.`
`20:29.49`	`*** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at)`
`20:29.53`	`atlantiatech`	`[TK]D-Fender, http://rafb.net/p/8qDRjR68.html`
`20:30.18`	`atlantiatech`	`[TK]D-Fender, keep in mind i am following the book dialplan section, and started from scratch as instructed`
`20:30.22`	`[TK]D-Fender`	`atlantiatech: Ok, thats a start, now show the config of the channel thats supposed to USE that context`
`20:30.49`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`20:32.03`	`atlantiatech`	`[TK]D-Fender, rereading the prvoius chapter, to see where that is supposed to be defined (what file etc)`
`20:33.09`	`[TK]D-Fender`	`atlantiatech: You could name it whatever you wanted and it wouldn't matter so long as you actually referred something to it.`
`20:33.22`	`*** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no)`
`20:33.26`	`[TK]D-Fender`	`atlantiatech: Don't go thinking that [incoming] inherently means anything at all`
`20:34.23`	`atlantiatech`	`[TK]D-Fender, i kinda guessed that was the issue`
`20:35.43`	`[TK]D-Fender`	`atlantiatech: So like I said go look at the channel that is ORIGINATING the call.`
`20:36.15`	`atlantiatech`	`[TK]D-Fender, yes indeed, let me find out what file defines that.. sorry i am learning here`
`20:37.05`	`Renacor`	`[TK]D-Fender: k i just rmod'ded the usb modules so its just`
`20:37.19`	`Renacor`	`225: 19398 215773 IO-APIC-level wctdm`
`20:37.22`	`Renacor`	`oops sorry`
`20:37.28`	`tuan_modulis`	`hello, I'm only a few weeks into asterisk so far... would anyone know how to set an absolute timeout AFTER entering into a queue and having started a conversation in that new channel?`
`20:37.31`	`Renacor`	`anyways still not working`
`20:37.54`	`russellb`	`you shouldn't have any problems sharing interrupts as of zaptel 1.2.13`
`20:37.58`	`[TK]D-Fender`	`Renacor: Well you still didn't fix the group numbers like I told you I'm betting....`
`20:38.09`	`Renacor`	`i think your right :)`
`20:38.16`	`Renacor`	`so the need to be called g1 ?`
`20:38.19`	`Renacor`	`or one of them`
`20:39.34`	`*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)`
`20:40.09`	`atlantiatech`	`[TK]D-Fender, just so i understand it, zapata.conf is where the channels are defined, is this correct?`
`20:40.11`	`[TK]D-Fender`	`Renacor: "g" vs "G" just chooses the ORDER in which they are chosen. "1" is the group # you are eattempting to dial. You have NO channels in that group. You should already know which channels you would want grouped together. Make sure the channels have the # you want in there and that it matches the ones you entered in your channel definition.`
`20:40.21`	`Renacor`	`k lemme pastebin my zapata.conf`
`20:41.13`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`20:41.37`	`Renacor`	`[TK]D-Fender: http://pastebin.ca/352441`
`20:42.28`	`Renacor`	`[TK]D-Fender: k understand that now but its still not working`
`20:42.38`	`Renacor`	`I have channel 1 in that group`
`20:42.39`	`tuan_modulis`	`Lemme rephrase my problem: how do you set an absolute time in a channel created by the Queue function?`
`20:42.46`	`[TK]D-Fender`	`Renacor: Group 1 now has 1 channel, which is an FXS port (should have a PHONE plugged onto it)`
`20:43.20`	`Renacor`	`zaptel.conf says its a fxo port tho`
`20:43.27`	`Renacor`	`fxoks=1`
`20:43.31`	`Renacor`	`no`
`20:43.36`	`Renacor`	`no?`
`20:44.39`	`atlantiatech`	`damn, i am so lost`
`20:44.47`	`atlantiatech`	`there is so much more than i need in these files`
`20:45.56`	`[TK]D-Fender`	`Renacor: No you have a misunderstanding about how signalling works. It the reverse of what you might instintivley think.`
`20:46.02`	`*** join/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net)`
`20:46.31`	`*** part/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net)`
`20:46.37`	`[TK]D-Fender`	`atlantiatech: correct. You only need about 10 line TOPS between zaptel & zapata in most cases`
`20:46.52`	`Renacor`	`so channel has to be 4?`
`20:47.52`	`[TK]D-Fender`	`Renacor: You need to verify the actual order of your modules, and ensure that you are using the right signalling for each port.`
`20:49.08`	`Renacor`	`[TK]D-Fender: as in fxo vs fxs?`
`20:49.20`	`[TK]D-Fender`	`Renacor: correct`
`20:49.31`	`Renacor`	`[TK]D-Fender: I used genzaptelconf`
`20:49.47`	`*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)`
`20:49.58`	`kervel`	`hello, i'm still trying to connect over misdn, and i now get 'Unable to create channel of type 'misdn' (cause 66 - Channel not implemented)'`
`20:50.11`	`kervel`	`i turned on misdn debugging, but i don't get anything`
`20:50.38`	`kervel`	`i tried mISDN/1/$OUTNUM$ and misdn/1/$`
`20:50.44`	`[TK]D-Fender`	`Renacor: As of now. STOP.`
`20:51.02`	`Renacor`	`??`
`20:51.17`	`[TK]D-Fender`	`kervel: Apparently chan_misdn.so isn't even loaded.`
`20:51.57`	`kervel`	`tk, but in the startup log i find [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))`
`20:52.13`	`kervel`	`that's what i see if i do asterisk -vvvv i mean`
`20:52.35`	`[TK]D-Fender`	`Renacor: Your 1st 2 ports are FXS, and shouldn't really be grouped. your port 3 & 4 are your LINES, and should have a group #, and be the same if you want to choose from between the 2 lines.`
`20:54.48`	`kervel`	`[TK]D-Fender could the problem be that i use ubuntu asterisk, but i compiled the misdn channel driver myself ? (the ubuntu supplied one didn't load)`
`20:55.40`	`atlantiatech`	`[TK]D-Fender, ok so i am going to try and make this AsteriskNow install a bare install, i.e. start from the section in the book that deals with setting up zapata and zaptel.. it's funny my goal is to test that forum response (for call forwading) but it's gonna be a long path to get there.. my time is so damn limited.. i kind of wish i could just elegantly hack the asterisknow files to try that forward rule`
`20:56.01`	`cpatry`	`kervel: whats the output of show channeltype misdn ?`
`20:57.31`	`kervel`	`hmm ... 'show channeltype' no such command`
`20:57.45`	`kervel`	`*CLI> show channeltype misdn;`
`20:57.45`	`kervel`	`No such command 'show channeltype' (type 'help' for help)`
`20:57.50`	`kervel`	`i started asterisk with -cvvv`
`20:57.56`	`cpatry`	`which version?`
`20:58.23`	`cpatry`	`ha debian package, and show channeltypes?`
`20:58.24`	`kervel`	`1.2.12.1-dfsg-1ubuntu1`
`20:58.36`	`kervel`	`indeed, ubuntu-branded debian package :)`
`20:58.51`	`kervel`	`should i get rid of it and compile myself?`
`20:58.55`	`atlantiatech`	`whats the extensions.ael file relationship to extensions.conf`
`20:59.00`	`cpatry`	`yep probably.`
`20:59.06`	`kervel`	`awww ...`
`20:59.13`	`kervel`	`i already feared that ...`
`20:59.36`	`cpatry`	`atlantiatech: .ael is for AEL dialplan .conf is for standard dialplan`
`21:00.02`	`atlantiatech`	`urrgh`
`21:00.06`	`atlantiatech`	`whats the diff?`
`21:00.16`	`kervel`	`cpatry i just did 'show channeltypes' and there is no mISDN`
`21:00.27`	`*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)`
`21:00.33`	`kervel`	`altough it reports misdn chan was loaded`
`21:00.35`	`cpatry`	`so ur driver isnt loaded.`
`21:00.48`	`kervel`	`<PROTECTED>`
`21:00.49`	`kervel`	`mISDN_close: fid(17) isize(131072) inbuf(0xb749b008) irp(0xb749b008) iend(0xb749b008)`
`21:01.19`	`kervel`	`going to retry with misdn 0.3.0`
`21:01.30`	`atlantiatech`	`heck anyone wanna make a quick buck ?`
`21:01.52`	`atlantiatech`	`i could use some serious guidance here.. this system is mainly just going to be a call routing box to two cell phones`
`21:01.55`	`[TK]D-Fender`	`atlantiatech: I think you really need to learn the basics... go download THE BOOK.`
`21:01.57`	`[TK]D-Fender`	`~book`
`21:01.58`	`jbot`	`well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`21:02.28`	`atlantiatech`	`[TK]D-Fender, i have it here`
`21:03.51`	`[TK]D-Fender`	`atlantiatech: Time to ditch the GUI, trash your configs, and start clean`
`21:07.59`	`dj-fu`	`Can I specify * in a dialplan?`
`21:08.14`	`*** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`21:08.17`	`dj-fu`	`exten => *97,1,VoiceMailMain()`
`21:08.25`	`dj-fu`	`or does it need to be escaped or something?`
`21:09.49`	`[TK]D-Fender`	`dj-fu: No, that is appropriate`
`21:09.58`	`dj-fu`	`great, thanks`
`21:10.00`	`*** part/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`21:10.46`	`dj-fu`	`if I asterisk -r, how can I detach from it again?`
`21:11.00`	`mercestes`	`dj-fu: exit`
`21:11.06`	`mercestes`	`dj-fu: You should read the book.`
`21:11.10`	`mercestes`	`~book`
`21:11.11`	`jbot`	`book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`21:11.25`	`dj-fu`	`I've read the book. It didn't mention that.`
`21:11.33`	`dj-fu`	`afaicr`
`21:11.53`	`dj-fu`	`Hell. I've got it printed and bound - wouldn't have got this far without it`
`21:12.33`	`kervel`	`i got mISDN working now, but when i try to call out, misdn pauses for some seconds, and then say empty_chan_in_stack:`
`21:12.57`	`kervel`	`anyone knows what could be the issue ?`
`21:13.37`	`*** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`21:13.44`	`*** part/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com)`
`21:14.11`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`21:14.19`	`*** join/#asterisk Dr-Linux\|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk)`
`21:15.07`	`[TK]D-Fender`	`dj-fu: Page 341 begs to differ <--`
`21:18.33`	`Vec`	`I am setting up an asterisk PABX in a production environment for the 1st of March, would anyone recommending using asterisk 1.4 or should I use 1.2 ?`
`21:22.11`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`21:22.46`	`dj-fu`	`lies!`
`21:23.04`	`Renacor`	`anybody here got a tdm400p?`
`21:23.43`	`*** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net)`
`21:23.45`	`Dr-Linux\|home`	`Renacor: ask your queston maybe someone answer you`
`21:24.19`	`*** join/#asterisk kgx (n=kgx@60.234.20.178)`
`21:25.08`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`21:26.27`	`*** join/#asterisk MvG (n=mvg@ppp-88-217-13-17.dynamic.mnet-online.de)`
`21:27.15`	`MvG`	`Hi! I'm looking for someone with a bit of experience with mISDN. Anybode here?`
`21:27.50`	`perd`	`arg, i cant fix my audio quality problem with SIP over the local network`
`21:27.53`	`perd`	`:(`
`21:27.54`	`critch`	`Vec: I have 1.4 in production, you need to think about what interfaces you are using and whether or not you want old and soon to need upgrading code but well known, or relatively new and not going to change too much in the next several months`
`21:29.35`	`perd`	`it seems like the audio is OK if i use GSM sound files, when i use the ULAW ones i get crappy robot-voice and jittery audio`
`21:29.48`	`perd`	`it works fine, then suddenly bad audio for a sec or two, then fine again`
`21:29.52`	`perd`	`arrrg!!`
`21:30.21`	`[TK]D-Fender`	`perd: what phones?`
`21:30.22`	`tuan_modulis`	`__TRANSFER_CONTEXT .... anyone knows what this does?`
`21:30.29`	`critch`	`perd: are you sure you network is sufficiently fast enough to not get hit by file transfers?`
`21:30.35`	`perd`	`fender: both on x-lite and cisco phjones (7912/7960 tested)`
`21:30.43`	`perd`	`yes critch, it's a private network`
`21:30.54`	`perd`	`the only thing on there is the asterisk server and sip clients`
`21:31.08`	`perd`	`system load is 0`
`21:31.19`	`[TK]D-Fender`	`hrm. ok, not much to suggest off-hand.`
`21:31.23`	`critch`	`perd: as in crossover cable or still going through a switch?`
`21:31.25`	`tuan_modulis`	`ah oops, google answered me`
`21:31.35`	`perd`	`i dont suppose hearing the crappy audio would help you figure out what is goign wrong :P`
`21:31.55`	`dj-fu`	`are you in a ulaw country?`
`21:32.02`	`dj-fu`	`actually that won't matter if you're using skype I guess`
`21:32.06`	`critch`	`perd: does it sound like a speakerphone that is set way too loud, and overdriven?`
`21:32.07`	`dj-fu`	`skype, what the fuck`
`21:32.08`	`perd`	`critch the asterisk server has 2 nics, one on the normal office network, the other on the private network which is connected to a fastiron switch`
`21:32.09`	`dj-fu`	`i'm going to shutup now`
`21:32.10`	`dj-fu`	`afk`
`21:32.14`	`perd`	`the server is connected via gigabit ethernet`
`21:32.31`	`perd`	`no critch the sound is fine, then suddenly it will sound robot like`
`21:32.33`	`perd`	`then it will be fine again`
`21:32.41`	`perd`	`i can send you an example`
`21:32.43`	`*** part/#asterisk queuetue (n=queuetue@70.54.254.134)`
`21:33.00`	`perd`	`http://spunknetwork.com/~bill/badaudio.wav`
`21:33.00`	`critch`	`perd: I just wondered if it was the same audio problem I had experienced the other day`
`21:33.14`	`perd`	`listen to htat,`
`21:33.30`	`critch`	`listening`
`21:33.50`	`perd`	`you can hear some popping/clicking and then the part' press one for' is really bad`
`21:34.10`	`perd`	`if i use GSM the sound is much better.. but i'd rather use somethign that is going to be crystal clear`
`21:34.11`	`critch`	`I heard that spot`
`21:34.35`	`perd`	`ulaw should be perfect :(`
`21:34.38`	`critch`	`sounds like maybe a jitter, but I don't know why`
`21:35.20`	`perd`	`i jsut reset the hardware to defaults and reinstalled the os gthinking maybe i did something weird that broke it`
`21:35.29`	`perd`	`but i'm running on a clean os now and still no good :(`
`21:35.36`	`perd`	`maybe i'll try the wav instead of ulaw files`
`21:36.38`	`Moobius`	`do you hear the jitter in the same place each time?`
`21:36.53`	`perd`	`moobius it does seem to be in about the same area yea`
`21:37.04`	`Moobius`	`corrupted sound file?`
`21:37.11`	`perd`	`na`
`21:37.14`	`tuan_modulis`	`before I go waste too much time, does TRANSFER_CONTEXT work with queues? Like if I set __TRANSFER_CONTEXT=somespecialqueue and then perform a Queue(somename\|t\|\|\|\|)`
`21:37.27`	`perd`	`i just reinstalled the os`
`21:37.57`	`perd`	`what prompts do you guys use?`
`21:37.59`	`Moobius`	`low disk cache?`
`21:38.00`	`perd`	`the format i mean`
`21:38.18`	`perd`	`it is default moobius`
`21:38.27`	`perd`	`but it's only running asterisk and it's a raid 5`
`21:38.30`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`21:38.32`	`diclophis-work`	`hello all`
`21:38.34`	`perd`	`lemme check that though`
`21:38.40`	`diclophis-work`	`how often does the "astdb" file get accessed?`
`21:38.58`	`perd`	`depends on what your extensions.conf looks like diclophis`
`21:39.10`	`critch`	`<smartass>As often as needed</smartass>`
`21:40.00`	`critch`	`What all do you have in the astdb file?`
`21:46.05`	`errr`	`when I edit /etc/amportal.conf and change the AUTHTYPE to database I am never able to login. The browser request just times out trying to log me in`
`21:47.02`	`errr`	`if I change it back to none it works just fine though`
`21:47.35`	`Dr-Linux\|home`	`errr: go to #freepbx`
`21:47.40`	`errr`	`ok`
`21:49.27`	`*** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu)`
`21:49.29`	`b11d`	`bonjour chaps`
`21:49.54`	`b11d`	`Does anyone know of any way to increase the audio volume for the handset on a Polycom 501?`
`21:50.05`	`b11d`	`above what you can set using the volume + and - buttons that is`
`21:50.23`	`b11d`	`i've got a few users who are kind of deaf and the polycom on max volume is still too quiet`
`21:51.34`	`mercestes`	`hey b11d`
`21:51.39`	`b11d`	`hey hey`
`21:51.50`	`mercestes`	`b11d: gains.tx.analog.handset=97`
`21:51.57`	`b11d`	`i heart you mang`
`21:52.04`	`mercestes`	`I heart you too, mang.`
`21:52.08`	`kervel`	`hoooray ! i got outgoing calls working over misdn ... and the quality is excellent. i had to try-and-error the misdn-init parameters, and the incoming calls still don't work`
`21:52.23`	`b11d`	`:)`
`21:53.12`	`*** join/#asterisk hegars (n=hegars@203.161.78.66.static.amnet.net.au)`
`21:53.15`	`hegars`	`hi all`
`21:54.12`	`b11d`	`mercestes.. is that real? or are you lying to me?`
`21:54.28`	`b11d`	`i dont see that anywhere as an option`
`21:54.30`	`*** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com)`
`21:54.55`	`mercestes`	`b11d: Well, 97 isn't exactly a good value for it. 24 maybe.`
`21:55.02`	`b11d`	`where does that go? in sip.cfg ?`
`21:55.06`	`mercestes`	`b11d: and it might be gain.tx.analog.handset. It's in sip.cfg`
`21:55.06`	`Renacor`	`k this really blows`
`21:55.18`	`Renacor`	`Im still getting [Feb 12 13:54:40] WARNING[2732]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)`
`21:55.21`	`b11d`	`oh`
`21:55.29`	`b11d`	`there she is`
`21:55.32`	`b11d`	`thanks :)`
`21:55.45`	`Renacor`	`but ztcfg -vvvvv shows channel 01 configured with fxo`
`21:56.34`	`Renacor`	`in zapata.conf do I have to use signalling fxs_ks for an fxo channel or fxo_ks?`
`21:56.48`	`dj-fu`	`fxs signalling for an fxo port`
`21:56.54`	`dj-fu`	`and fxo signalling for fxs port`
`21:57.11`	`Renacor`	`so if I have fxoks in zaptel.conf`
`21:57.17`	`*** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc)`
`21:57.19`	`Renacor`	`in zapata.conf it needs to be fxs_ks ?`
`21:57.47`	`dj-fu`	`sec`
`21:58.10`	`dj-fu`	`Renacor, it should be the same in both`
`21:58.16`	`dj-fu`	`fxo ports should have fxs signalling and vice versa`
`21:58.17`	`dj-fu`	`in both files`
`21:58.51`	`Renacor`	`so fxo is fxo in both files`
`21:59.05`	`*** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net)`
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`22:01.46`	`Renacor`	`can somebody take a look at this and tell me if Im doing something wrong?`
`22:01.47`	`Renacor`	`http://pastebin.ca/352538`
`22:03.56`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
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`22:14.55`	`b11d`	`well since you guys fixed that last one so easily.. how about this: how can I have it so the phone "remembers" what volume setting was last used?`
`22:15.18`	`b11d`	`when someone picks up the handset, ups the volume, and then finsihes the call and they hangup, the volume is back down to its default setting when they pick up the handset next.`
`22:16.30`	`*** join/#asterisk hegars (n=hegars@203.161.78.66.static.amnet.net.au)`
`22:16.49`	`hegars`	`hi`
`22:16.56`	`b11d`	`hi`
`22:17.02`	`Renacor`	`fxo is used for the phone line and fxs for the telephones right? or the other way around?`
`22:17.06`	`hegars`	`im running a 1.4* install and it running at 100% cpu`
`22:17.14`	`hegars`	`what sould i be looking for?`
`22:17.42`	`hematitec`	`FXO is used for the phone line`
`22:17.54`	`b11d`	`Renacor.. FXO is facing the CO.. FXS is facing the telephones`
`22:18.39`	`kervel`	`hmm, now i got misdn incoming calls working too, but asterisk rejects all incoming calls with 'excentions can never mach'`
`22:18.40`	`kervel`	`match`
`22:18.47`	`Renacor`	`is there any way to make the lights blink on the tdm400p to identify which channel is on which physical port?`
`22:18.50`	`kervel`	`anyone knows where i should look at`
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`22:19.57`	`b11d`	`Renacor.. ask Digium`
`22:20.10`	`b11d`	`kervel.. well.. look at your extensions.conf`
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`22:20.25`	`b11d`	`set a high verbose & debug level, and see what exactly is being called in your extensions.conf and why they might not match..`
`22:20.37`	`b11d`	`hegars.. good luck mang.`
`22:22.17`	`hegars`	`b11d: thanks, done that, i can see no issues tho`
`22:22.57`	`dj-fu`	`Renacor, 1 is at the top, 4 is closest to the board`
`22:23.01`	`kervel`	`b11d probably i don't have a 'default' extension or sth`
`22:23.05`	`dj-fu`	`there's a picture in The Book which shows the numbers`
`22:23.19`	`kervel`	`Feb 12 23:24:27 NOTICE[1159] pbx.c: Cannot find extension context 'mISDN'`
`22:23.31`	`kervel`	`b11d that's what it says when i try to call`
`22:23.45`	`kervel`	`maybe i should configure misdn to use the default context ?`
`22:23.49`	`b11d`	`yes`
`22:23.51`	`b11d`	`defaintly`
`22:23.57`	`b11d`	`for now anyway.. change that as you move towards production`
`22:24.02`	`kervel`	`and i guess the name of the default context is just 'default' ?`
`22:24.05`	`b11d`	`aye`
`22:24.10`	`perd`	`hmm so when i use gsm sound files instead of wav i dont get that audio problem`
`22:24.12`	`perd`	`what the hell...`
`22:24.12`	`kervel`	`oki, let's try :)`
`22:24.26`	`perd`	`sip is using ulaw`
`22:24.27`	`b11d`	`hegars.. hrm.. 100% usage from Asterisk eh?`
`22:24.33`	`b11d`	`1.4.. running linux I take it?`
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`22:25.14`	`hegars`	`yeah 2.6.17`
`22:25.36`	`kervel`	`b11d a stupid question .. . a 'context' is just the name of something between [] in extensions.conf ?`
`22:27.22`	`hegars`	`b11d: its back to ~2% running 14 active channels`
`22:27.47`	`kervel`	`b11d you made my day`
`22:27.52`	`hegars`	`then it just blows out to 99.9 with not indications on the cli`
`22:27.53`	`hematitec`	`To be honest, a context start with the tag of "[context-name]" can continues until the next tag`
`22:27.57`	`kervel`	`i can call myself, hooray :)`
`22:28.04`	`thekidrio`	`hehe`
`22:28.05`	`*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)`
`22:29.45`	`hematitec`	`Opps I see that it should have been 'and' instead of 'can'`
`22:30.10`	`b11d`	`:)`
`22:30.19`	`b11d`	`glad i could help`
`22:30.42`	`b11d`	`hematitec is correct about contexts..`
`22:30.58`	`b11d`	`they can be confusing at first but once you actually start using your system, you see how they work`
`22:31.13`	`b11d`	`i had to actually just start using the system before I actually learned how they work and in what way`
`22:31.18`	`b11d`	`thats just me though`
`22:32.11`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`22:33.11`	`hematitec`	`When you realize that a 'context' is a based on XML it is easy to understand. With XML you would have an ending tag`
`22:36.48`	`*** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose)`
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`22:42.23`	`dj-fu`	`is it now? I thought it was more similar to windows .ini than to XML, personally`
`22:42.33`	`dj-fu`	`the syntax of contexts etc is nothing like XML, imho`
`22:43.12`	`[TK]D-Fender`	`Correct, more like .ini`
`22:43.27`	`[TK]D-Fender`	`contexts only matter when something REFERS to them.`
`22:43.59`	`[TK]D-Fender`	`[default] doesn't mean ANYTHING unless something points to it. You can have one called [fred] for all * cares`
`22:44.21`	`[TK]D-Fender`	`hematitec : think httpd.conf then. next closest thing`
`22:44.37`	`hematitec`	`Very true`
`22:44.38`	`dj-fu`	`even httpd.conf has taken a more XML based approach as of late`
`22:44.57`	`dj-fu`	`<VirtualHost *:80> ... </VirtualHost> etc`
`22:45.15`	`tzafrir_laptop`	`dj-fu, as of late?`
`22:45.24`	`hematitec`	`Yes, a start tag and an end tag`
`22:45.25`	`tzafrir_laptop`	`that is clearly not xml`
`22:45.34`	`tzafrir_laptop`	`and this is clearly not as of late`
`22:45.44`	`dj-fu`	`xml != more XML based approach`
`22:45.56`	`dj-fu`	`tzafrir, afaicr, apache1 didn't use that syntax`
`22:46.04`	`tzafrir_laptop`	`You can find the same syntax in the predecessor of Apache (the ncsa httpd) from around 1995 or so`
`22:46.12`	`dj-fu`	`well - forget that idea then ;]`
`22:46.52`	`tzafrir_laptop`	`dj-fu, the PWS (Personal Web Server) that came with windows 98 was also from the same codebase and used config files with similar general syntax`
`22:47.00`	`dj-fu`	`righto`
`22:47.16`	`dj-fu`	`not interested in a debate, was just saying, httpd.conf isn't a good example of a windows .ini based configuration`
`22:49.16`	`*** join/#asterisk genz (n=erdo@im.jobdig.com)`
`22:49.35`	`tzafrir_laptop`	`But this is basically just a header. Lines have a more decent Var Value starture that is nice for editing`
`22:50.00`	`genz`	`Is cdr_mysql 1.4 compatible?`
`22:50.32`	`tzafrir_laptop`	`genz, there are addons for 1.4`
`22:50.45`	`[TK]D-Fender`	`sigh whatever`
`22:50.57`	`tuan_modulis`	`for the dial function, how do i add more than one "option" (recall Dial(type/identifier, timeout, options, URL))`
`22:51.13`	`tuan_modulis`	`like if I want M(xxx) and L(xxx) together`
`22:51.17`	`tuan_modulis`	`as options`
`22:51.27`	`hematitec`	`Not for my window manager, right now I'm using Gnome. But in the past I use to use CDE (Solaris 8)`
`22:51.58`	`hematitec`	`Now I'm running either Debian/Etch or Ubuntu`
`22:54.48`	`*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)`
`22:55.09`	`perd`	`hmm ok so this is odd. my sip audio problems appear to go away when i use GSM sound files instead of ulaw or wav sound files. sip is set to use ulaw only for audio. :(`
`22:55.11`	`perd`	`i dont get it.`
`22:55.15`	`hematitec`	`How about trying just Dial(type/id,timeout,M(xxx)L(xxx),URL)`
`22:55.24`	`perd`	`apparently asterisk is determined not to let me use good quality sound files`
`22:56.12`	`hematitec`	`perd: you have disable gsm in the sip.conf for that context?`
`22:56.23`	`perd`	`i have disallow=all allow=ulaw`
`22:56.33`	`perd`	`that's under the general options`
`22:56.40`	`perd`	`do i have to specify it for each sip entry too?`
`22:57.08`	`hematitec`	`Not unless the context for that entry changes it`
`22:57.18`	`perd`	`yea it doesnt... all sip audio is transmitted via ulaw`
`22:57.38`	`perd`	`apparently the gsm sound files being transcoded works better than stuffing ulaw files right down the pipe`
`22:59.05`	`kervel`	`b11d & hematitec i read your explanation on contexts a bit late, but it was very helpful`
`22:59.46`	`hematitec`	`how did you confirmed this? Did you say do something like PlayBack(file.wav) and then PlayBack(file.gsm)?`
`23:00.01`	`hematitec`	`Kervel: you are welcome`
`23:00.20`	`perd`	`hematitec just by traversing the voicemail menus`
`23:00.31`	`perd`	`that's where i notice the poor audio/jitters the most`
`23:00.49`	`*** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net)`
`23:02.07`	`hematitec`	`Was it on the announcements or the message which was recorded?`
`23:02.14`	`perd`	`on the announcements`
`23:02.24`	`perd`	`http://spunknetwork.com/~bill/badaudio.wav`
`23:02.25`	`kervel`	`does anyone know if it is possible to manage SNOM hardphones centrally ?`
`23:02.36`	`perd`	`thjat is an example of the bad audio, i recorded that using ulaw sound files`
`23:05.09`	`perd`	`man gsm sounds perfect, except for the whole compressed audio quality thing`
`23:05.15`	`perd`	`i just dont understand it...`
`23:05.41`	`perd`	`this is all on separate networks too, the only bandwidth in use is the sip connection i test with`
`23:06.28`	`mercestes`	`perd: Could be server HDD I/O's, memory, or other transcoding issues.`
`23:06.41`	`mercestes`	`perd: try a native ulaw-ulaw call with no transcoding in ulaw and see what it does.`
`23:06.44`	`nowork`	`hi, besides iaxtalk, any other place can i find inernation language voice file for asterisk`
`23:06.57`	`perd`	`that's what the recording is from merc, ulaw-ulaw, jittery and poping`
`23:07.08`	`perd`	`what should i do about disk I/O, i can test with bonnie i guess?`
`23:07.15`	`nowork`	`the file of china voice mesg at iaxtalk doesn't work`
`23:07.35`	`*** join/#asterisk foobar778 (i=johhny@ip68-100-41-120.dc.dc.cox.net)`
`23:07.41`	`mercestes`	`perd: I would suggest some MTR scans then.`
`23:07.58`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net)`
`23:07.59`	`perd`	`forgive my ignorance.. what is an mtr scan?`
`23:08.19`	`mercestes`	`Are you in *nix?`
`23:08.29`	`perd`	`linux`
`23:09.01`	`mercestes`	`perd: It's a "pingsweep" program that ping-tests each hop and measures jitter.`
`23:09.10`	`perd`	`ahhh`
`23:09.22`	`perd`	`is that the name of the package? mtr?`
`23:09.29`	`foobar778`	`Hi I istalled asterisk and I can refister sip through my lan but not the wan my asterisk server is in the dmz Im using Kanotix debian OS my ata is an dvg-1120s any help please`
`23:09.37`	`perd`	`ok slick i have it..`
`23:09.46`	`mercestes`	`the only thing in your network that is of any real concern is the jitter readings. I a call at 350 ms will sound just fine if the lag stays at 350ms steadily.`
`23:09.50`	`perd`	`any recommended settings?`
`23:09.59`	`mercestes`	`perd: yea, show all the jitter settings..:)`
`23:10.08`	`perd`	`i mean for this mtr scanning`
`23:10.10`	`perd`	`hehe`
`23:10.15`	`mercestes`	`there isn't really any settings.`
`23:10.16`	`perd`	`sip doesnt have a jitter buffer does it?`
`23:10.19`	`mercestes`	`just a "fields" key.`
`23:10.30`	`mercestes`	`iirc it does.`
`23:10.36`	`perd`	`ah`
`23:10.46`	`thekidrio`	`anyone able to recommend an iax softphone for linux?`
`23:10.47`	`mercestes`	`I could be wrong...`
`23:10.54`	`mercestes`	`but I tend to fix my network over relying on jitter buffer.`
`23:10.58`	`foobar778`	`Hi I istalled asterisk and I can register sip through my lan but not the wan my asterisk server is in the dmz Im using Kanotix debian OS my ata is an dvg-1120s any help please`
`23:11.00`	`perd`	`haha no doubt merc`
`23:11.13`	`perd`	`lemme see, i'll put ulaw back in place and try this again with mtr going`
`23:12.49`	`*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`23:12.55`	`perd`	`well`
`23:13.02`	`perd`	`this shows no spikes in latency when i get the jitter`
`23:13.28`	`foobar778`	`<thekidrio> kphone`
`23:13.53`	`perd`	`it does appear to uniformly mess up the audio on specific prompts though`
`23:13.57`	`*** part/#asterisk genz (n=erdo@im.jobdig.com)`
`23:13.59`	`perd`	`vm-messages vm-opts are the worst`
`23:14.10`	`perd`	`maybe it is f'd audio files, but i've downlooaded them and redownloaded them several times...`
`23:14.15`	`perd`	`<PROTECTED>`
`23:14.28`	`perd`	`that's 5 mins or so there, during whch i placed calls`
`23:14.40`	`perd`	`159 packets, 60-78ms range`
`23:14.55`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`23:15.39`	`*** part/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
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`23:16.31`	`syzygyBSD`	`is it possible to listen for sip on 2 ips on 2 different networks?`
`23:16.35`	`*** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose)`
`23:17.48`	`foobar778`	`perd can u help me`
`23:17.52`	`perd`	`maybe`
`23:17.53`	`perd`	`what's up`
`23:18.09`	`foobar778`	`Hi I istalled asterisk and I can register sip through my lan but not the wan my asterisk server is in the dmz Im using Kanotix debian OS my ata is an dvg-1120s any help please`
`23:19.16`	`hematitec`	`foobar: have you insured that your allow sip from the WAN side to the LAN side?`
`23:19.50`	`hematitec`	`You may need to punch a hole for ports 5060-5070, 10000-20000`
`23:19.53`	`perd`	`actually i've had the same problem foobar, and i just gave up since i didnt really need registration from the WAN heh`
`23:19.59`	`foobar778`	`how please in sip.conf?`
`23:20.09`	`hematitec`	`On the router it self`
`23:20.17`	`foobar778`	`its in the dmz`
`23:20.22`	`dj-fu`	`:o`
`23:20.42`	`hematitec`	`But you said that you could not get SIP from the WAN`
`23:20.45`	`foobar778`	`running a wrt54g with ddwrt firmware`
`23:20.59`	`foobar778`	`yes I cant get sip through wan`
`23:21.07`	`hematitec`	`Then fix the router`
`23:21.19`	`foobar778`	`how its in the dmz??`
`23:21.29`	`perd`	`ah my issue was registration, not getting connected heh`
`23:21.33`	`dj-fu`	`it's probably no tin the dmz if you can't get SIP from the wanside.`
`23:21.38`	`foobar778`	`yes mine too perd`
`23:21.57`	`perd`	`so if you take out the secret= and have the user connect without authentication credentials it works?`
`23:22.05`	`foobar778`	`it is in the dmz definitley`
`23:22.14`	`mercestes`	`foobar778: Make sure 5060 is open.`
`23:22.22`	`foobar778`	`perd I have no secret set`
`23:22.28`	`dj-fu`	`foobar778, have you configured sip.conf correctly so that you can register to the asterisk machine from the wanside?`
`23:22.30`	`hematitec`	`Your router is in the DMZ?`
`23:22.33`	`mercestes`	`foobar778: And use the external IP of the * server you are regy too, and make sure you have host=dynamic, etc.`
`23:22.34`	`foobar778`	`dmz 5060 has to be open`
`23:22.48`	`mercestes`	`perd: you can trylistening to the fiels offline to see if they work.`
`23:22.57`	`perd`	`yea mercestes they sound fine`
`23:22.58`	`perd`	`i just did that`
`23:23.08`	`perd`	`frustrating :(`
`23:23.13`	`hematitec`	`Really you should open 5060-5070 which are the SIP ports`
`23:23.20`	`*** join/#asterisk madriles (n=cvaldess@28.Red-83-53-45.dynamicIP.rima-tde.net)`
`23:23.24`	`perd`	`i guess i'm stuck using gsm for the time being`
`23:23.26`	`foobar778`	`can someone perhaps do a vnc to my machine to sort this out??`
`23:23.28`	`mercestes`	`perd: Hrm. and these are ulaw encoaded.`
`23:23.33`	`hematitec`	`And don't forget the 10000-20000 for RTP`
`23:23.36`	`perd`	`yea ulaw or wav`
`23:23.37`	`madriles`	`Hi all`
`23:23.57`	`perd`	`and i think the .wav is slinear or something`
`23:24.02`	`madriles`	`any one runing * on gumstix??`
`23:24.04`	`foobar778`	`harmatic can u do a vnc??`
`23:24.06`	`dj-fu`	`lol, you'd be lucky foobar778`
`23:24.08`	`hematitec`	`No`
`23:24.10`	`perd`	`wahtever the case is, anything other than damn GSM jitters and doesnt work :/`
`23:24.34`	`hematitec`	`I'm not allow to`
`23:24.49`	`foobar778`	`hematic if the pc is in the dmz all ports must be open yes`
`23:25.17`	`nowork`	`hello..any website has the step by step instruction of h323 installation on asterisk?`
`23:25.27`	`dj-fu`	`foobar778, unless there is a firewall on the machine inside the dmz`
`23:25.40`	`hematitec`	`If the PC is inside the DMZ then all of the SIP and RPT ports at the DMZ (router) must be open`
`23:25.43`	`foobar778`	`i did an iptable -F`
`23:25.50`	`foobar778`	`iptables -F`
`23:25.54`	`dj-fu`	`dmz forwards all connections to the dmz'd machine. if iptables on the dmz'd machine is blocking sip, then no`
`23:25.58`	`dj-fu`	`is INPUT set to accept?`
`23:25.59`	`perd`	`i even pingflood the damn SIP phone and GSM still sounds fine, wav on the other hand.. :)`
`23:26.16`	`dj-fu`	`lol - ./juno it and see how it handles`
`23:26.21`	`foobar778`	`dj-fu did an iptables -F`
`23:26.31`	`dj-fu`	`that just flushes chains`
`23:26.34`	`dj-fu`	`it doesn't reset the policy`
`23:26.39`	`endre`	`-P ACCEPT`
`23:26.46`	`endre`	`let's make a big hole`
`23:26.49`	`foobar778`	`it open the firewall no?`
`23:26.52`	`dj-fu`	`yes`
`23:26.54`	`hematitec`	`So you are saying the your are routering WAN-ip 5060-5070 to LAN-ip 5060-6070?`
`23:26.57`	`mercestes`	`perd: And ulaw calls are ok??`
`23:27.02`	`perd`	`yeah the calls are fine`
`23:27.03`	`dj-fu`	`foobar778, what is the IP address`
`23:27.10`	`dj-fu`	`I'll try and telnet your wanip 5060`
`23:27.10`	`Nugget`	`telnet is eeeeeeevil!`
`23:27.11`	`mercestes`	`perd: it's just playbacks of ulaw encoded files?`
`23:27.16`	`perd`	`yeah just playback`
`23:27.20`	`foobar778`	`68.100.41.120`
`23:27.20`	`mercestes`	`weird.`
`23:27.23`	`perd`	`i know man`
`23:27.25`	`mercestes`	`nugget: what?`
`23:27.26`	`perd`	`it's driving me friggen NUTS`
`23:27.44`	`dj-fu`	`foobar778, (UNKNOWN) [68.100.41.120] 5060 (?) : Connection refused`
`23:27.46`	`dj-fu`	`fix your router.`
`23:27.54`	`dj-fu`	`you are NOT dmzing to the machine, or the machine is NOT listening on the sip port.`
`23:28.06`	`*** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net)`
`23:28.12`	`foobar778`	`<dj-fu> Im not in linux can i boot in and come back will u still be here`
`23:28.23`	`dj-fu`	`no`
`23:28.26`	`dj-fu`	`I'll be gone, unfortunately`
`23:28.45`	`dj-fu`	`might I suggest correctly configuring your network before embarking on a project such as a software PBX?`
`23:28.46`	`foobar778`	`yes Im not running the asterisk from here`
`23:28.58`	`dj-fu`	`it's probably the correct order of doing things`
`23:29.09`	`dj-fu`	`configure network > secure network > install optional software`
`23:29.20`	`dj-fu`	`DMZ is not secure either, btw`
`23:29.23`	`foobar778`	`so if i go into linux test for open port 5060 and its open then whaty`
`23:29.42`	`foobar778`	`must online piort scan test udp`
`23:29.50`	`foobar778`	`test tcp not udp`
`23:29.53`	`mercestes`	`foobar778: be sure to use --insane`
`23:30.07`	`hematitec`	`It should be noted, that not all routers will work correctly. I have one Netgear router which won't work at all for SIP or IAX. But a friends Linksys did fine and my Cisco is NO problem`
`23:30.08`	`foobar778`	`not using nmap`
`23:30.27`	`dj-fu`	`foobar778,`
`23:30.29`	`dj-fu`	`actually I take that back`
`23:30.32`	`dj-fu`	`I tried to tcp connect().`
`23:30.37`	`dj-fu`	`udp port 5060 on your wanside is open`
`23:30.38`	`dj-fu`	`ip68-100-41-120.dc.dc.cox.net [68.100.41.120] 5060 (?) open`
`23:30.53`	`foobar778`	`yes that me`
`23:31.07`	`foobar778`	`scenario should be ok in windows too`
`23:31.16`	`dj-fu`	`that would imply that your asterisk is configured incorrectly`
`23:31.20`	`foobar778`	`same router setup`
`23:31.29`	`dj-fu`	`unfortunately I don't have a softphone installed otherwise I'd try and connect to it`
`23:31.40`	`foobar778`	`perhaps I need help in astersik config`
`23:31.41`	`perd`	`i have a softphone`
`23:31.45`	`perd`	`what is the username`
`23:32.00`	`dj-fu`	`foobar778, have you read "the book"`
`23:32.02`	`dj-fu`	`~thebook`
`23:32.04`	`jbot`	`from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`23:32.19`	`dj-fu`	`I'd suggest taking a read through that, in particular, the sip section`
`23:32.38`	`foobar778`	`read a lot not the whole book just beginning I set it up so far lan connections works analog phone rings`
`23:33.02`	`dj-fu`	`cool`
`23:33.12`	`dj-fu`	`afk now, wish I could help more`
`23:33.54`	`*** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com)`
`23:34.47`	`*** join/#asterisk JT (n=jon@unaffiliated/jt)`
`23:34.49`	`foobar778`	`Im coming back from linux maybe more help from there thank u all`
`23:35.04`	`perd`	`hey merc which sound files do you use?`
`23:35.19`	`hematitec`	`foobar: One of the things you can do is join FWD (freeworlddialup.com), and register to them with SIP. They can send a test phone call to you`
`23:35.22`	`perd`	`i'm getting this audio jitter issue on two diff systems`
`23:35.29`	`perd`	`i just installed asterisk on another server :/`
`23:35.55`	`mercestes`	`perd: Hm.`
`23:36.05`	`mercestes`	`perd: you said it does it under comedian mail?`
`23:36.12`	`perd`	`yes sir`
`23:36.21`	`hematitec`	`perd: What type of processor (ie: P4-500Mhz, 500 Meg RAM)?`
`23:36.34`	`mercestes`	`perd: =/ doesn't make sense man. What OS you using?`
`23:36.35`	`perd`	`3.0ghz xeon with 2gb ram`
`23:36.39`	`perd`	`centos 4.4`
`23:36.42`	`perd`	`smp kernel`
`23:36.43`	`mercestes`	`hmm.`
`23:36.45`	`mercestes`	`~centosbug`
`23:36.47`	`jbot`	`i heard centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h"`
`23:36.50`	`elriah`	`Hey guys, do any of you have Cisco 7941's? If so, is the firmware ugprade process different from the 7940's? I ask because no matter what I can't get the 7941's to read our OS79XX.TXT and upgrade the firmware to 8.2 (FRom 7.3 SCCP)`
`23:36.54`	`perd`	`2.6.9-42.0.8.ELsmp`
`23:36.54`	`mercestes`	`perd: Know about this?`
`23:36.56`	`hematitec`	`Your processor should not be a problem then`
`23:37.03`	`perd`	`no i didnt know about that`
`23:37.08`	`mercestes`	`try that.`
`23:37.13`	`mercestes`	`have no clue if it's related or not.`
`23:37.19`	`perd`	`im going to, thank you!`
`23:37.22`	`perd`	`crosses fingers....`
`23:37.31`	`perd`	`oh wait the other server i tested it on is debian`
`23:37.34`	`perd`	`but i'll try this anyway`
`23:38.14`	`perd`	`hmmm`
`23:38.35`	`*** part/#asterisk madriles (n=cvaldess@28.Red-83-53-45.dynamicIP.rima-tde.net)`
`23:39.01`	`*** join/#asterisk Waverly360 (n=irc@adsl-070-148-122-203.sip.bna.bellsouth.net)`
`23:39.01`	`perd`	`nah still having the issue :/`
`23:40.13`	`mercestes`	`you recreated on debian?`
`23:40.16`	`perd`	`yep`
`23:40.23`	`mercestes`	`debian has portage, right?`
`23:40.24`	`perd`	`unfortunately`
`23:40.29`	`perd`	`apt`
`23:40.37`	`mercestes`	`oh.`
`23:40.38`	`mercestes`	`hrm.`
`23:40.44`	`riddlebox`	`I am being told that asterisk cant find zapscan.conf, where would I get that from?`
`23:40.54`	`perd`	`i could try to install the debian version`
`23:40.57`	`*** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net)`
`23:40.59`	`perd`	`instead of compiling mine`
`23:41.07`	`perd`	`but i dunno :/`
`23:41.17`	`Waverly360`	`I'm having some trouble understanding how to connect dual asterisk servers together so that I can dial between the two. I'm confused as to where I setup the credentials for the other server to use.`
`23:41.20`	`perd`	`my compile method is pretty minimal.. libpri -> zaptel -> asterisk`
`23:41.27`	`foobar778`	`hi all Im in linux now`
`23:41.33`	`mercestes`	`perd: Could. this debian box is a seperate box??`
`23:42.05`	`perd`	`yes it's a separate box`
`23:42.12`	`*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)`
`23:42.12`	`perd`	`2.4ghz celeron or something with a gig of memory`
`23:42.15`	`perd`	`oh athlon 2200 XP`
`23:42.15`	`perd`	`haha`
`23:42.20`	`Waverly360`	`I have an entry called [mytest] in my sip.conf file, along with type=friend, user=mytest, secret=mypassword`
`23:42.32`	`kervel`	`aaargh , misdn outgoing calls stopped working ... it seems the number is not transferred correctly to the pstn`
`23:42.35`	`perd`	`that one is running asterisk 1.2.12, the othe was 1.2.14 and 1.2.15 (installed 15 today)`
`23:42.42`	`kervel`	`i always get 'phone number not connected'`
`23:42.45`	`perd`	`so debian = 1.2.12, centos = newer vers`
`23:42.49`	`kervel`	`and 1 hour ago it worked ...`
`23:42.57`	`perd`	`which audio file format do you use, merc?`
`23:42.58`	`Waverly360`	`Now...what I'm trying to figure out, is whether that's the credentials that other pbx's connect to me as, or if that's the credentials I use to connect to others`
`23:43.00`	`mercestes`	`perd: this audio sounds slower than normal.`
`23:43.13`	`kuku5`	`Any sources for cheap 7940's?`
`23:43.13`	`mercestes`	`perd: Are you using madplay? mplayer?`
`23:43.17`	`perd`	`no`
`23:43.20`	`perd`	`all internal`
`23:43.28`	`perd`	`i dont even have the app_mp3 stuff compiled in`
`23:43.32`	`perd`	`or whatever it is`
`23:43.44`	`*** part/#asterisk nowork (n=jfu2808@216.254.141.97)`
`23:43.53`	`perd`	`i went bare minimum to try and troubleshoot this thing`
`23:44.16`	`perd`	`i wonder if it could be messed up configuration somewhere`
`23:44.17`	`Waverly360`	`oh crap..I may have just answered my own question...that's what I get for skipping over the howto`
`23:44.20`	`mercestes`	`yea, these audio files sound different.`
`23:44.22`	`perd`	`lemme try putting in the sample files`
`23:44.22`	`foobar778`	`perd can u try to register on my pbx?`
`23:44.41`	`perd`	`merc you're listening to the wav file i put up on spunknetwork?`
`23:44.44`	`mercestes`	`Allison just doesn't sound like her sexy self.`
`23:44.47`	`mercestes`	`Yea.`
`23:44.47`	`perd`	`haha`
`23:44.52`	`perd`	`hrmm`
`23:45.33`	`foobar778`	`hermatic are u here`
`23:45.38`	`hematitec`	`Yes`
`23:45.43`	`foobar778`	`Im in linux`
`23:46.18`	`foobar778`	`have router registerd on one line`
`23:46.26`	`foobar778`	`have another open`
`23:46.41`	`foobar778`	`wanting to have it connect from wan`
`23:46.50`	`foobar778`	`asterisk is running`
`23:46.52`	`foobar778`	`same ip`
`23:47.01`	`foobar778`	`user name 1030`
`23:47.07`	`foobar778`	`no secret`
`23:47.21`	`hematitec`	`What is the IP?`
`23:47.37`	`foobar778`	`68.100.41.120`
`23:48.12`	`mercestes`	`perd: well, I could do an install for you on your hardware and see if it does it again but that's about hte only thing I can think of. =/ it's something to do with audio playback tho.`
`23:48.36`	`perd`	`yeah i dunno man`
`23:48.41`	`mercestes`	`perd: and if you dl those files and they are fine then it can only be either in the playback config or hte hardware involved in playback.`
`23:48.49`	`perd`	`yeah`
`23:48.55`	`mercestes`	`perd: Could be something as dumb as an IRQ conflict with a soundmixer device but , I dunno.`
`23:49.02`	`foobar778`	`Asterisk Ready.`
`23:49.02`	`foobar778`	`*CLI> -- Registered SIP '1020' at 192.168.1.142 port 5064 expires 3600`
`23:49.02`	`foobar778`	`68`
`23:49.05`	`perd`	`two diff servers having the same rproblem`
`23:49.10`	`foobar778`	`from the lan`
`23:49.11`	`perd`	`gotta be my asterisk configuration i guess?`
`23:49.16`	`perd`	`that's the only thing similar`
`23:49.24`	`perd`	`even have them on separate switches`
`23:49.24`	`perd`	`heh`
`23:49.26`	`mercestes`	`I'm guessing.`
`23:49.39`	`perd`	`well i just put the samples in place, gonna test with this and see if i get messed up audio`
`23:49.39`	`foobar778`	`no luck heramtic?`
`23:49.53`	`k-man`	`is there a channel for the discussion of networks and networking?`
`23:50.36`	`foobar778`	`hermatitec sorry for mispelling ur nick`
`23:51.26`	`perd`	`ok so....`
`23:51.36`	`perd`	`it appears that my audio works well if i dont have my config in place`
`23:51.48`	`perd`	`which is bad because that means i f'd up my server config somehow`
`23:52.04`	`perd`	`i guess i have to go through everything by hand now arg`
`23:52.51`	`foobar778`	`another issue I have is that softphone will dial and the anolaog phone get not a sip header huh is this dtmfsetting?`
`23:53.22`	`mercestes`	`good luck`
`23:53.29`	`thekidrio`	`anyone here use sipplan.com?`
`23:53.34`	`Waverly360`	`Ok..another question. Is it possible to dial an extension on another pbx from the current pbx's CLI?`
`23:53.36`	`thekidrio`	`i am having problems getting iax2 to work with them`
`23:56.31`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`23:56.36`	`elriah`	`Anyone running Cisco 7941's? I can't get them to recognize a SIP firmware ugprade from tftp, they never even look, they just keep grabbing the sep<mac>.cfg files...`
`23:56.42`	`riddlebox`	`is anyone running fedora core 6, I need to find out what package contains zapscan.conf`
`23:58.22`	`*** join/#asterisk JT_ (n=jon@unaffiliated/jt)`

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