irclog2html for #asterisk on 20061118

`00:02.54`	`ucfMethod`	`can anyone provide a bit of explaination on this error msg "res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!"`
`00:03.01`	`Juggie`	`if your not behind a FW, all you have to do is make sure rtp.conf's rtp range and the range you allow to the server is thre same.`
`00:03.59`	`TommyTheKid`	`ucfMethod: I think that has something to do with not having a "timing device" (IMO one of the major shortcomings of *) ... do you have ztdummy loaded and configured (?)`
`00:04.33`	`TommyTheKid`	`Juggie: then you are suggesting "b" as well.. so far B is in the lead :)`
`00:05.39`	`ucfMethod`	`TommyTheKid: I have it loaded, otherwise conf rooms (MeetMe) wouldn't work`
`00:05.59`	`ucfMethod`	`TommyTheKid: configured may be the word? what am I looking for.`
`00:06.40`	`TommyTheKid`	`I have no idea, my two servers both have digium cards... my lappy (OSX admittedly) was giving that error, I never really pushed it since I dont think zaptel would compile for OSX :)`
`00:06.45`	`ucfMethod`	`TommyTheKid: I have my musiconhold.conf correct.... I want to randomly play those sample mp3 that came with *`
`00:07.21`	`ucfMethod`	`i don't think its crucial to have, but its one of those things that people will bitch bitch bitch if it doesnt`
`00:07.28`	`ucfMethod`	`employees I mean`
`00:07.28`	`TommyTheKid`	`http://www.asteriskguru.com/tutorials/request_schedule_past.html`
`00:08.33`	`TommyTheKid`	`ucfMethod: are you using NTP to keep your clock in sync?`
`00:15.17`	`ucfMethod`	`no`
`00:15.37`	`ucfMethod`	`* should be using kernel for timing right?`
`00:15.52`	`ucfMethod`	`the date and time are accurate, just synced to time.nist.gov`
`00:15.56`	`ucfMethod`	`hwclock too`
`00:16.30`	`ucfMethod`	`whatever... illl work on it Monday.. time to drink.... thanks again ... ill be back Monday morning.`
`00:18.05`	`xheliox`	`I should be able to transfer a call from SIP (Polycom 430) to an IAX2 extension without any issues, right? :)`
`00:20.11`	`CunningPike`	`xheliox: Correct`
`00:23.05`	`xheliox`	`Hrm. Using SVN 1.4... r47782... and that seems not to work very well. In one case the call was xfer'd and then dropped within 10 seconds. In other cases, the IAX device rings, but when the call is answered, there's no audio on either end.`
`00:23.59`	`*** join/#asterisk saftsack (n=saftsack@pD9E04A2B.dip.t-dialin.net)`
`00:23.59`	`*** join/#asterisk nextime (n=nextime@unaffiliated/nextime)`
`00:24.03`	`xheliox`	`[Nov 17 19:12:40] WARNING[21692]: io.c:234 ast_io_remove: Asked to remove NULL? -- Bunch of these came up a time or two as well. Thoughts?`
`00:24.28`	`nextime`	`how many stable actually is 1.4 beta3?`
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`00:24.53`	`TommyTheKid`	`xheliox: you will get herpies if you use it (I think?)`
`00:24.59`	`*** part/#asterisk ctooley (n=ctooley@jc1-111.moment.net)`
`00:25.01`	`xheliox`	`I didn't ask that. :P`
`00:25.03`	`TommyTheKid`	`er.. nextime i mean :)`
`00:25.27`	`xheliox`	`I'm fully aware of the risks involved with running a beta. That's why I'm doing it at home and not work. ;)`
`00:26.15`	`TommyTheKid`	`damn.. even with my phone directly on the Internet (no FW, no NAT) I still can't get qualify to go thru.. I wonder if I need to allow outbound sip connections?`
`00:26.24`	`TommyTheKid`	`(from the corp fw)`
`00:26.33`	`nextime`	`TommyTheKid : i have some stability issues with 1.4 on a test machine, expecially when i use queues, so, i'm asking if it shuld be considerable unstable only for me`
`00:27.18`	`xheliox`	`nextime: Just by definition, it's prone to have stability issues.`
`00:29.04`	`nextime`	`xheliox : i known, but my question is something different, i was asking "i'm the only one with a serius stability issue?"`
`00:29.24`	`xheliox`	`nextime: I've had my 1.4 install crash at least once a day.`
`00:29.42`	`nextime`	`xheliox : ok, this is the answer, thanks`
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`00:32.10`	`*** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net)`
`00:34.59`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
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`00:36.10`	`TommyTheKid`	`curious, if I comment out the "qualify=1000" line for my "user" entry, the call goes thru fine... its like the "sip ping" is getting lost.. how does "qualify" work?`
`00:41.47`	`*** join/#asterisk gerphimum (n=trekkie@cpe-70-114-42-210.satx.res.rr.com)`
`00:42.31`	`*** join/#asterisk bjohnson_ (n=bjohnson@i209-195-81-138.cia.com)`
`00:45.27`	`*** join/#asterisk andresmujica (n=andresmu@201.244.247.15)`
`00:45.35`	`andresmujica`	`hello there`
`00:45.53`	`andresmujica`	`anyone knows if i can use call progress detection over a pri zap channel?`
`00:47.33`	`nextime`	`is there a way to get CDR(billsec) from a DeadAGI in 1.2.13?`
`01:06.59`	`*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)`
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`01:11.08`	`gerphimum`	`.`
`01:11.23`	`FuriousGeorge`	`anyone know anything about videoconferencing with asterisk?`
`01:19.11`	`FuriousGeorge`	`is sip video the standard?`
`01:23.37`	`andresmujica`	`ss`
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`01:29.47`	`*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)`
`01:40.33`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`01:51.47`	`*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net)`
`01:51.52`	`Idle`	`how can I close off a channel?`
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`01:53.09`	`Qwell`	`Idle: /part #channel`
`01:53.14`	`Idle`	`:P`
`01:53.31`	`Idle`	`seriously, I have both of my external lines that bridged themselves`
`01:53.49`	`Idle`	`3 bridged to 4... no idea why`
`01:55.49`	`Idle`	`god, all of these commands are 100% useless`
`01:57.51`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`02:01.11`	`Idle`	`http://uuoc.com/1650`
`02:01.18`	`Idle`	`Qwell: which line originated that call?`
`02:04.02`	`Idle`	`ugh`
`02:04.11`	`Idle`	`how can I enable it so it logs _EVERYTHING_ to a file`
`02:06.40`	`hads`	`soft hangup`
`02:06.58`	`Idle`	`yea, I got it eventually`
`02:08.36`	`*** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn)`
`02:08.44`	`Idle`	`I just need to figure out why it happened`
`02:08.59`	`Idle`	`I think its probably to do with voicemail... cause nothing else could do that`
`02:16.55`	`*** join/#asterisk ma-requin (n=youssef@dyn10.tcom.ou.edu)`
`02:17.50`	`ma-requin`	`anybody familiar with siproxd`
`02:20.18`	`ma-requin`	`???`
`02:27.45`	`*** join/#asterisk asdx (n=diego@200.61.236.33)`
`02:33.40`	`*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)`
`02:37.10`	`*** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com)`
`02:42.27`	`sivana`	`could a channel not show in Asterisk, however, signalling is still passing through?`
`02:51.23`	`*** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk)`
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`03:02.43`	`*** mode/#asterisk [+o mog] by ChanServ`
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`03:11.47`	`*** part/#asterisk voicetech (n=lotusscr@marnock.com)`
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`03:21.12`	`hoobastooba`	`can you guys help me with a lame question... If i want to apply a patch file to manager.c what would the syntax be... I thought it would be "patch manager.c <patchname" not working for me... can anyone help me correct this?`
`03:28.16`	`TommyTheKid`	`hoobastooba: its something like patch -p0 < patchfile`
`03:28.44`	`TommyTheKid`	`if you look at the patch file, it probably specifies manager.c in it.. and it may have path/to/manager.c`
`03:29.01`	`*** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net)`
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`03:32.06`	`hoobastooba`	`so you mean patch -p0 manager.c < patchfile?`
`03:32.17`	`hoobastooba`	`cuz i have to specify the file to patch right?`
`03:32.33`	`hoobastooba`	`i am trying to figure this out from the man pages and i am completely confused.`
`03:32.48`	`hoobastooba`	`i get the output that only garbage was found....`
`03:32.56`	`hoobastooba`	`i know that is because I am typing it wrong.`
`03:39.46`	`hoobastooba`	`TommyTheKid: i understand what you are saying... that worked. patch -p0 <patchfile`
`03:40.05`	`hoobastooba`	`the patch file finds what it is supposed to patch... thanks,;) first time`
`03:40.11`	`TommyTheKid`	`:)`
`03:40.26`	`TommyTheKid`	`sometimes a patchfile touches MANY files`
`03:41.06`	`TommyTheKid`	`I pussy'd out.. I was gonna upgrade to 1.4.0-b3, but here I sit compiling 1.2.13`
`03:41.24`	`TommyTheKid`	`all that talk about herpies earlier`
`03:42.36`	`hoobastooba`	`TommyTheKid: for prod environment?`
`03:44.04`	`TommyTheKid`	`yea`
`03:44.23`	`hoobastooba`	`smart move`
`03:44.42`	`TommyTheKid`	`i mean, we all have alternatives (cell, land lines) but we are "ITCTO" we are supposed to be on the bleeding edge.. I use OpenSoalris for a desktop :)`
`03:47.13`	`Juggie`	`there is enough bleeding in 1.2.13`
`03:47.41`	`TommyTheKid`	`more or less than 1.2.10 ? :)`
`03:47.50`	`Juggie`	`not sure`
`03:47.55`	`Juggie`	`i'm still running 1.2.9.1`
`03:48.13`	`mog`	`1.4 Juggie`
`03:49.14`	`Juggie`	`heh, i run in production.`
`03:49.16`	`*** join/#asterisk fnordus (n=dnall@24.85.128.203)`
`03:49.23`	`Juggie`	`i have like 5 * boxes`
`03:49.35`	`Juggie`	`the only one i upgraded past 1.2.9.1 is the one which is accessible publically`
`03:49.40`	`mog`	`yeah buts its just the candian goverment`
`03:49.46`	`mog`	`its not like its anything important`
`03:49.47`	`Juggie`	`heh.`
`03:49.54`	`Juggie`	`i got 10 new servers today :)`
`03:50.58`	`aptura`	`you or your department`
`03:51.40`	`aptura`	`I want to see a open souce gsm gateway built ;)`
`03:52.06`	`aptura`	`my wifes bosses kit racked up a 500 dollars cell phone bill and most of it was for roaming.`
`03:52.22`	`aptura`	`kid not kit :)`
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`03:56.45`	`Juggie`	`aptura, my dept.`
`03:56.57`	`Juggie`	`which indirectally means me`
`03:57.02`	`aptura`	`:)`
`03:57.03`	`Juggie`	`since i maintain all the linux servers`
`03:57.19`	`Juggie`	`thought i'm supposed to train some ppl in on installing at least next week.`
`03:57.48`	`aptura`	`Do you relize how many teeth I had to pull just to get anything faster then a pentium 200 while working at microsoft in 1999? It made my job difficult testing software on such slow hardware.`
`03:58.00`	`aptura`	`train people on what?`
`03:58.14`	`Juggie`	`well, i'm not really infrastructure`
`03:58.31`	`Juggie`	`so, i'm supposed to just walk one of our normal network guys through installing linux`
`03:58.36`	`Juggie`	`we are normally a microsoft shop.`
`03:58.53`	`aptura`	`well dont loose your job over it.`
`03:58.58`	`Juggie`	`why would i.`
`03:59.01`	`Juggie`	`i work for the goverment`
`03:59.05`	`aptura`	`good`
`03:59.06`	`aptura`	`:)`
`03:59.06`	`Juggie`	`i have to murder someone to get fired`
`03:59.24`	`aptura`	`state or federal?`
`03:59.25`	`Juggie`	`nah, they can probally figure out the centos installer to build pc's`
`03:59.35`	`Juggie`	`the main thing is what packages to install and what not to install`
`03:59.40`	`Juggie`	`thats the part they probally woudnt be sure on`
`03:59.44`	`Juggie`	`canadian federal.`
`03:59.57`	`aptura`	`ohh yea your up here what province?`
`04:00.08`	`Juggie`	`Ottawa, Ontario`
`04:00.13`	`aptura`	`okay`
`04:00.14`	`aptura`	`:)`
`04:00.30`	`Juggie`	`u?`
`04:00.34`	`aptura`	`Are we BC residents making you rich?`
`04:00.45`	`Juggie`	`no.`
`04:00.49`	`aptura`	`We have been over taxed for a long time.`
`04:00.57`	`Juggie`	`hah, i do work in Revenue :)`
`04:01.03`	`Juggie`	`but i dont know anything about that.`
`04:01.10`	`Juggie`	`my real job outside of being jack of all trades.`
`04:01.18`	`Juggie`	`is Toll Free Networks`
`04:01.25`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`04:01.27`	`aptura`	`I see`
`04:01.28`	`Juggie`	`we maintain all the 1-800 (toll free numbers) for revenue canada.`
`04:01.36`	`aptura`	`so you work at CRA?`
`04:01.52`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`04:02.00`	`Juggie`	`we get a realtime CDR/CMP feed from bell and its my job to take the cdr's and process them into meaningful information`
`04:02.05`	`Juggie`	`yes.`
`04:02.08`	`aptura`	`I see`
`04:02.33`	`aptura`	`Are most of the CRA buildings going voip? I was aware that the one in downtown Vancouver is going this route.`
`04:03.03`	`Juggie`	`so, i have sql jobs running to generate 15/hourly/daily stats, and those all get dumped as processed stats, and some other processed stats`
`04:03.14`	`Juggie`	`i also maintain the web interface to build reports from this pre-processed data.`
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`04:03.55`	`Juggie`	`aptura, we have been voip for years and years`
`04:03.57`	`Juggie`	`probally 3-4`
`04:04.24`	`Juggie`	`but, my last job (and my new job falls under the same division so i still work closely w/ my old section) was called Emerging Telephony Section`
`04:04.30`	`Juggie`	`so it is our job to do that stuff.`
`04:04.47`	`Juggie`	`we tested cisco/nortel/mitel/alcatel i think`
`04:04.52`	`Juggie`	`and maybe some others voip systems`
`04:04.52`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`04:04.57`	`Juggie`	`and we ended up going w/ mitel.`
`04:05.24`	`Juggie`	`aptura, do you know what that vancouver building was installing?`
`04:05.46`	`aptura`	`I know`
`04:05.58`	`aptura`	`I saw the old mitel pbx servers before.`
`04:06.06`	`aptura`	`Well`
`04:06.09`	`Juggie`	`probally mitel ip then?`
`04:06.14`	`Juggie`	`you probally saw SX2000`
`04:06.21`	`aptura`	`yes`
`04:06.25`	`Juggie`	`w/ 4025 Smartsets`
`04:06.26`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
`04:06.31`	`aptura`	`but still had some legacy wiring in the back?`
`04:06.43`	`aptura`	`blue/grey mitel boxes`
`04:06.44`	`Juggie`	`oh yah, those SX2000's are digital`
`04:06.52`	`Juggie`	`but they are over single pair like analog`
`04:07.01`	`aptura`	`I see`
`04:07.13`	`aptura`	`so did thay scrap the old mitel digital boxes?`
`04:07.23`	`Juggie`	`in my building?`
`04:07.29`	`Juggie`	`we still maintain it as a test environment`
`04:07.33`	`aptura`	`no in cra like in bc for example`
`04:07.38`	`Juggie`	`oh, i have no idea.`
`04:07.59`	`Juggie`	`we are responsible for all the call centers across canada for CRA`
`04:08.10`	`Juggie`	`so we have to have every pbx that we have installed in our lab.`
`04:08.12`	`aptura`	`ahh thats interesting`
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`04:08.19`	`Juggie`	`so we have like 4-5 running pbx's that dont even get used`
`04:08.20`	`aptura`	`very cool`
`04:08.26`	`Juggie`	`they are just there as test enviromenents`
`04:08.30`	`aptura`	`use them as room heaters`
`04:08.31`	`aptura`	`:)`
`04:08.40`	`Juggie`	`hah no kidding.`
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`04:09.04`	`Juggie`	`you have to test stuff when you get alot of calls per day.`
`04:09.08`	`aptura`	`Well thats interesting.`
`04:09.19`	`aptura`	`how did you hear about this position?`
`04:09.28`	`Juggie`	`i started w/ CRA as a student`
`04:09.34`	`aptura`	`ahhh`
`04:09.39`	`Juggie`	`i did 3 workterms there, and they hired me before i finished my last one`
`04:09.44`	`Juggie`	`so i finished school and moved to ottawa`
`04:09.48`	`aptura`	`nifty`
`04:09.57`	`Juggie`	`i'm originally a newfie :)`
`04:10.28`	`aptura`	`btw what does cra do with its older servers?`
`04:11.06`	`Juggie`	`well, the official thing to do is to send them to crown assets`
`04:11.28`	`Juggie`	`which means crown assets will put them on a pallet, leave them there for 5 years and finally go to sell them when they are totally worthless.`
`04:11.41`	`Juggie`	`so, i think sometimes we donate them to computers for schools or something`
`04:11.49`	`Juggie`	`i am not 100% sure i dont work in infrastructure`
`04:11.58`	`Juggie`	`it probally varies alot between different lab management`
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`04:12.35`	`aptura`	`:)`
`04:12.40`	`Crumbles`	`my attended transfers still doesn't work after an upgrade to 1.2.11 :(`
`04:12.43`	`aptura`	`minus the hard drives of course`
`04:12.58`	`linagee`	`you never have to worry about a heater in a computer lab. just turn down the A/C`
`04:13.54`	`aptura`	`juggie, can you listen to a channel into a call on one of those 1800 numbers?`
`04:14.06`	`Juggie`	`aptura, i have no idea, i am sure they are disposed of carefully ;)`
`04:14.20`	`Juggie`	`there is a standard for overwritting HDD's its a US DOD spec.`
`04:14.33`	`Juggie`	`its a 7pass format`
`04:14.54`	`Juggie`	`aptura, i dont know, i dont work in the call center stuff, other then generating stats`
`04:15.07`	`Juggie`	`but yes, when the call is on the line w/ an agent, then yes, they can barge into a call`
`04:15.13`	`Juggie`	`but when its just the ivr, i dont think so.`
`04:15.33`	`aptura`	`okay`
`04:15.35`	`Juggie`	`all our systems play the typical this call may be recorded or monitored message.`
`04:15.42`	`aptura`	`yea`
`04:15.44`	`Juggie`	`just like any other agent system`
`04:16.07`	`Crumbles`	`is there some trick to transfers that i could be totally missing here?`
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`04:18.59`	`sivana`	`does anyone have a clue which library this is? /bin/ld: cannot find -lGL`
`04:19.12`	`sivana`	`I'm missing one.. -lGL`
`04:20.24`	`Crumbles`	`sweet, thanks guys.`
`04:20.33`	`Crumbles`	`i'll go try that.`
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`04:21.37`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`04:22.49`	`bluregard`	`hey guys`
`04:25.36`	`*** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)`
`04:27.35`	`bluregard`	`quite in here tonight`
`04:27.54`	`bluregard`	`quiet even`
`04:30.43`	`*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap)`
`04:33.24`	`TommyTheKid`	`I got rid of my weird double-ring by upgrading zaptel and ast, go figure :)`
`04:33.58`	`Crumbles`	`can someone help me get DISA and atxfer working? :( neither of them do ANYTHING`
`04:34.55`	`ManxPower`	`Crumbles: Does DTMF Transfers work without DISA?`
`04:35.02`	`Crumbles`	`no`
`04:35.05`	`Crumbles`	`nothing works`
`04:35.11`	`Crumbles`	`i mean nothing in features.conf`
`04:35.29`	`ManxPower`	`Crumbles: then you do not have a transfer issue and do not have a DISA issue. You have a DTMF issue.`
`04:35.41`	`Crumbles`	`okay?`
`04:36.04`	`ManxPower`	`IF you dial into an Asterisk IVR, does it hear the DTMF?`
`04:36.13`	`Crumbles`	`ivr?`
`04:36.20`	`ManxPower`	`Crumbles: voice menu`
`04:36.24`	`Crumbles`	`yes.`
`04:36.34`	`ManxPower`	`how is the call connecting to Asterisk? Zap card, IAX2 provider, SIP provider?`
`04:36.50`	`Crumbles`	`it's an internal sip to sip call`
`04:37.16`	`Crumbles`	`the DISA call is a call coming from the outside world through my sip provider.`
`04:37.18`	`ManxPower`	`Well, in SIP DTMF is not sound, it is messages`
`04:37.32`	`ManxPower`	`Crumbles: does it work if you call DISA from the SIP phones?`
`04:37.49`	`Crumbles`	`i haven't tried that i've mostly been trying to get atxfer working`
`04:37.50`	`Crumbles`	`hold on`
`04:37.54`	`TommyTheKid`	`the next fork of asterisk needs to be called octothorpe, i dcree`
`04:38.59`	`ManxPower`	`Crumbles: get it working internal first.`
`04:39.07`	`Crumbles`	`yes DISA works if i dial it from an internal extension, but not if i dial in from outside.`
`04:39.27`	`ManxPower`	`Crumbles: does atxfer work if you dial from inside into DISA and then dial out from DISA?`
`04:40.22`	`Crumbles`	`if i press anything at the DISA dialtone i get a turkey tone`
`04:40.40`	`ManxPower`	`Crumbles: define "at the DISA"`
`04:40.40`	`*** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net)`
`04:40.48`	`TommyTheKid`	`turkey? thanksgiving turkey?`
`04:40.50`	`Crumbles`	`i dial 611 for disa`
`04:40.53`	`Crumbles`	`i hear a dialtone`
`04:41.01`	`ManxPower`	`Crumbles: you need to get it working from an internal phone before trying to get ti working via a provider`
`04:41.08`	`Crumbles`	`if i press any number on my keypad i get short beep tones`
`04:41.21`	`ManxPower`	`Crumbles: does the asterisk CLI show anything?`
`04:41.27`	`Crumbles`	`yes`
`04:41.33`	`TommyTheKid`	`.. in verbose 3+ :)`
`04:41.36`	`ManxPower`	`put it on pastebin.ca`
`04:42.25`	`Crumbles`	`it's just "executing answer" "executing DISA" and "spawn extension exited non-zero"`
`04:42.28`	`Crumbles`	`that's all it shows.`
`04:43.05`	`bluregard`	`does anyone else have problems calling other FWD users when using IAX?`
`04:43.54`	`Crumbles`	`http://pastebin.ca/251450`
`04:45.35`	`Crumbles`	`figured out what was wrong internally`
`04:45.46`	`TommyTheKid`	`I assume you specified a context ?`
`04:45.51`	`Crumbles`	`i did now`
`04:45.55`	`TommyTheKid`	`:)`
`04:46.06`	`Crumbles`	`that's what fixed that but it still won't work coming in from my sip provider`
`04:46.17`	`TommyTheKid`	`I don't use default either`
`04:46.44`	`ManxPower`	`Crumbles: but it DOES work coming from your SIP phone?`
`04:46.50`	`Crumbles`	`yes`
`04:46.52`	`Crumbles`	`works great.`
`04:46.58`	`Juggie`	`hmmmm`
`04:47.03`	`ManxPower`	`Crumbles: what dtmfmode= do you have set for that provider?`
`04:47.08`	`Crumbles`	`inband`
`04:47.09`	`Juggie`	`offtopic: http://www.virtualnes.com`
`04:47.14`	`ManxPower`	`Crumbles: and what codec?`
`04:47.18`	`Crumbles`	`sec`
`04:47.36`	`Crumbles`	`none specified`
`04:47.39`	`ManxPower`	`inband will ONLY work with ulaw or alaw codecs`
`04:47.47`	`Crumbles`	`i only allow those codecs`
`04:47.52`	`Crumbles`	`from [general]`
`04:47.56`	`TommyTheKid`	`http://pastebin.ca/251453 - I lied.. I sorta don't use default :)`
`04:48.18`	`ManxPower`	`Crumbles: what does the SIP provider recommend?`
`04:48.24`	`ManxPower`	`and who is the sip provider?`
`04:48.34`	`Crumbles`	`sunrocket`
`04:48.44`	`Crumbles`	`the sip provider recommends i stop using asterisk`
`04:49.24`	`ManxPower`	`set the debug to 3, do a test call, pastebin the cli output`
`04:49.34`	`Crumbles`	`okay...`
`04:49.36`	`ManxPower`	`you should see the incoming DTMF.`
`04:49.58`	`TommyTheKid`	`vnes is cool`
`04:51.14`	`TommyTheKid`	`it even works on a REMOTE Sun Ray on Solaris 11 x64`
`04:51.38`	`*** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)`
`04:51.39`	`Crumbles`	`http://pastebin.ca/251454`
`04:51.54`	`Crumbles`	`i don't get a dialtone or anything`
`04:52.06`	`*** join/#asterisk vasterisk (n=yliu@c-67-170-232-240.hsd1.ca.comcast.net)`
`04:52.12`	`ManxPower`	`Crumbles: start asterisk as "asterisk -rvvvddd"`
`04:52.23`	`ManxPower`	`then in the CLI do "set verbose 3" and "set debug 3"`
`04:52.28`	`Crumbles`	`oh sorry`
`04:52.42`	`*** join/#asterisk qwluhbear (n=kvirc@dsl-203-113-238-76.SA.netspace.net.au)`
`04:52.50`	`ManxPower`	`Oh, make sure you have a /etc/asterisk/indications.conf the default one is fine`
`04:52.53`	`hoobastooba`	`ok, i have looked at gnudialer... it blows, I have looked at vicidialer, I cannot figure it out. I need to have a dialer that can do voicemail detection and a couple of styles of dialing ie predictive, power, preview.... what is out there? I understand that they wont be free... but I need recommendations. needs to be able to do up to 300 agents.`
`04:53.21`	`Crumbles`	`i have it`
`04:53.27`	`ManxPower`	`hoobastooba: Don't expect much help here. Most of us do not help with autodialer problems.`
`04:53.58`	`hoobastooba`	`just wondering if anyone else is using anything they like and can make a recommendation.`
`04:54.05`	`qwluhbear`	`Hi.. Can I get help with Asterisk here?`
`04:54.07`	`ManxPower`	`Crumbles: BTW, you can easily write a short dialplan entry to sort of emulate DISA`
`04:54.22`	`ManxPower`	`qwluhbear: sometimes`
`04:54.28`	`hoobastooba`	`qwluhbear: what do you need help with?`
`04:54.38`	`Crumbles`	`http://pastebin.ca/251456`
`04:55.02`	`qwluhbear`	`I think i'm missing something really really obvious.. I've got a Voicetronix OpenPCI card in my system, I just can't get it to interface with Asterisk`
`04:55.19`	`ManxPower`	`Crumbles: pastebin your /etc/asterisk/logger.conf`
`04:56.02`	`ManxPower`	`qwluhbear: not many people use those cards with Asterisk, but the drivers are updated fairly often. Did you do a google search?`
`04:56.04`	`ManxPower`	`~docs`
`04:56.07`	`jbot`	`rumour has it, 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, or http://www.astmasters.net/howtos.html`
`04:56.18`	`ManxPower`	`..er...`
`04:56.20`	`ManxPower`	`~mailinglist`
`04:56.21`	`jbot`	`Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm`
`04:56.23`	`ManxPower`	`that's what I wantede`
`04:56.26`	`Crumbles`	`http://pastebin.ca/251457`
`04:56.50`	`qwluhbear`	`yep.. the drivers are installed, everything seems to work fine - when I ring the phone, it comes up with ringing in /var/log/messages, I'm just lost as to how it actually blends in with Asterisk.. It's as though I can only use a zap card?`
`04:57.03`	`ManxPower`	`Crumbles: change it to console => notice,warning,error,debug,verbose`
`04:58.05`	`Crumbles`	`same output`
`04:58.08`	`Crumbles`	`with the new logger.conf`
`04:58.09`	`ManxPower`	`qwluhbear: I don't know how it interfaces with Asterisik. see if there are any sample configs in /path/to/src/asterisk/configs or /path/to/src/asterisk/docs`
`04:58.20`	`ManxPower`	`Crumbles: you need to do a "reload" or a "logger reload"`
`04:58.26`	`Crumbles`	`i did reload`
`04:58.42`	`ManxPower`	`Crumbles: you should be getting MASSIVE amounts of output.`
`04:59.03`	`Crumbles`	`ok`
`04:59.05`	`*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)`
`04:59.08`	`Crumbles`	`after logger reload it gave me tons`
`04:59.09`	`Crumbles`	`hold on`
`05:00.17`	`qwluhbear`	`ManxPower: don't have a file/folder like that. I'm using Trixbox.. how do you know that a connected zap card is working? The output of ztcfg -vvv?`
`05:00.42`	`ManxPower`	`qwluhbear: trixbox has it's own config system, totally unlike the standard Asterisk one. I cannot help you firther.`
`05:00.48`	`ManxPower`	`or further either`
`05:01.00`	`Juggie`	`ManxPower, i have to say i've been looking @ your nickname for years now`
`05:01.11`	`brookshire`	`juggie!!!!!!!!!!!!!!!!!!!!!!!`
`05:01.14`	`Juggie`	`and every time i do, it reminds me of when homer changed his name to Max Power`
`05:01.23`	`ManxPower`	`Juggie: it is a reference to that.`
`05:01.33`	`Juggie`	`:)`
`05:01.34`	`Juggie`	`figures.`
`05:01.36`	`ManxPower`	`My former nick was WereCat`
`05:01.39`	`Juggie`	`that was a good episode.`
`05:01.41`	`Juggie`	`brookshire!`
`05:01.43`	`ManxPower`	`so I kept the cat reference`
`05:01.49`	`brookshire`	`manx: you had a former nick?`
`05:01.59`	`Juggie`	`ManxPower, the man formerly known as.`
`05:02.02`	`ManxPower`	`brookshire: I've had MANY former nicks.`
`05:02.09`	`Juggie`	`i've allways been Juggie`
`05:02.13`	`Juggie`	`since i was 14.`
`05:02.17`	`brookshire`	`manx: so you switch nicks faster than bfs`
`05:02.19`	`ManxPower`	`ManxPower, WereCat, d'WereCat, ImaGeek, Recluse, CyberMonk`
`05:02.35`	`brookshire`	`:)`
`05:02.37`	`brookshire`	`<3`
`05:02.43`	`Crumbles`	`http://pastebin.ca/251459`
`05:03.23`	`brookshire`	`omg.. this is like the first time i've been on irc in a whole week`
`05:04.03`	`ManxPower`	`brookshire: I don't switch BFs very often, silly.`
`05:04.11`	`ManxPower`	`Crumbles: No indication DTMF is received.`
`05:04.24`	`Crumbles`	`i dialed like 500 things`
`05:04.25`	`ManxPower`	`can you pastebin [secret] for me?`
`05:04.30`	`Crumbles`	`and screamed in to the phone like a lunatic`
`05:04.38`	`ManxPower`	`Crumbles: we need to try rfc2833 as well`
`05:05.12`	`Crumbles`	`i don't even hear a dialtone when i dial in`
`05:05.21`	`ManxPower`	`Crumbles: that is not unusual`
`05:06.00`	`ManxPower`	`pastebin sip.conf [general] and the incoming sip.conf [section] change as little as you can, but remember to change the password.`
`05:06.22`	`ManxPower`	`Crumbles: how long did it take to get NAT working?`
`05:06.27`	`Crumbles`	`like 3 seconds`
`05:06.29`	`Crumbles`	`:)`
`05:06.45`	`ManxPower`	`Learn fast, you will.`
`05:06.54`	`Crumbles`	`hold on a sec for me`
`05:07.40`	`Crumbles`	`it doesn't see the dtmf even if i change it to rfc2833`
`05:07.57`	`Crumbles`	`sip.conf coming up`
`05:10.13`	`ManxPower`	`Crumbles: If I still don't see anything wrong I can some up with a easy dialplan replacement for DISA.`
`05:10.32`	`Crumbles`	`http://pastebin.ca/251463`
`05:10.44`	`Crumbles`	`:(`
`05:11.05`	`Crumbles`	`how would you be able to get a dialplan to accept incoming dtmf if disa can't even see it?`
`05:11.18`	`Juggie`	`Disa is one of those apps that really has no point.`
`05:11.28`	`Juggie`	`you can do it in dialplan with almost no effort.`
`05:11.34`	`ManxPower`	`Crumbles: We still need to fix the DTMF problem, of course.`
`05:12.00`	`*** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)`
`05:12.18`	`Crumbles`	`well i want it so when i call in with my cell it gives me a way to dial out under the sipphone context`
`05:12.26`	`Crumbles`	`so that i can dial any extension i want`
`05:12.35`	`ManxPower`	`Juggie: The whole Max Power episode is great social engineering.`
`05:12.40`	`Crumbles`	`and i also would like to be able to transfer calls hits his head against a wall`
`05:12.59`	`ManxPower`	`Crumbles: nod most of that is NOT something only in DISA`
`05:13.31`	`ManxPower`	`Crumbles: do this, dial into something that plays an intro message Background(/path/to/message/file) no extension`
`05:13.46`	`Crumbles`	`i've got something like that`
`05:13.50`	`Crumbles`	`but with a playback application`
`05:13.52`	`Crumbles`	`and it works great`
`05:13.52`	`ManxPower`	`Maybe throw a WaitExten() after it and see if you can call in and at least dial an extensions`
`05:14.02`	`Crumbles`	`wait what?`
`05:14.12`	`Crumbles`	`what's waitexten?`
`05:14.18`	`ManxPower`	`Crumbles: So DTMF IS working via the SIP provider`
`05:14.25`	`Crumbles`	`no no i'm confused`
`05:14.39`	`Crumbles`	`i CAN make it so when i dial in i hear a sound played`
`05:14.56`	`Crumbles`	`in fact i put my cable company on a list so they just get tt-monkeys`
`05:15.07`	`ManxPower`	`Crumbles: it manually waits for you to dial something rather than just falling off the end of the dialplan, in 1.2 the "wait for extension after background" is deprecated`
`05:15.27`	`Crumbles`	`so background automatically waits for you to dial?`
`05:15.42`	`ManxPower`	`Crumbles: in 1.0 and 1.2 it does`
`05:15.49`	`ManxPower`	`well in 1.2 if you have some option set`
`05:16.04`	`Juggie`	`ManxPower, in 1.2 autofallthrough is enabled by default.`
`05:16.05`	`Crumbles`	`so i should do`
`05:16.15`	`Crumbles`	`exten => blahabladjlkasjf,Background(tt-monkeys)`
`05:16.25`	`Crumbles`	`exten => alskdfjasldkjf,WaitExten()`
`05:16.25`	`Crumbles`	`?`
`05:16.37`	`Juggie`	`correct.`
`05:16.47`	`Juggie`	`except w/ a real extension, and priorities`
`05:16.50`	`ManxPower`	`well the blahabladjlkasjf would have to be something you could dial froma phone`
`05:16.58`	`ManxPower`	`and actually`
`05:17.10`	`ManxPower`	`exten -> 611,1,Background(tt-monkeys)`
`05:17.20`	`ManxPower`	`exten => 611,2,WaitExten`
`05:17.27`	`ManxPower`	`that is a => of course`
`05:17.50`	`Corydon76-home`	`and a length argument to WaitExten`
`05:17.54`	`Crumbles`	`do i have to hit # or something?`
`05:18.09`	`Crumbles`	`the monkeys played and while they were playing i dialed 2000`
`05:18.12`	`*** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu)`
`05:18.14`	`Crumbles`	`but 2000 did not ring`
`05:18.18`	`ManxPower`	`no, it will either timeout or if it has an erxact match it will just jump to that extension`
`05:18.50`	`Juggie`	`Crumbles, add a exten => _XXXX,1,Noop(${EXTEN}`
`05:18.59`	`Juggie`	`then when you dial anything 4 digit, it should print to your console`
`05:18.59`	`Crumbles`	`why?`
`05:19.00`	`ManxPower`	`Crumbles: I'm afraid we now need to get into SIP debug`
`05:19.07`	`ManxPower`	`set debug 0`
`05:19.16`	`Crumbles`	`right but this is an incoming call from my provider`
`05:19.21`	`Crumbles`	`so _XXXX won't catch it`
`05:19.39`	`Juggie`	`i thought you said you wanted to get input`
`05:19.40`	`*** join/#asterisk laborat (n=ariel@cypher.punk.net)`
`05:19.41`	`Juggie`	`and do a transfer`
`05:20.01`	`Crumbles`	`i am dialing in from an outside line`
`05:20.11`	`Juggie`	`and * is answering right?`
`05:20.12`	`ManxPower`	`then sip debug peer sunrocket-out`
`05:20.14`	`Crumbles`	`yes`
`05:20.16`	`Juggie`	`you hear tt-monkeys`
`05:20.19`	`ManxPower`	`then do a call, patebin the CLI`
`05:20.20`	`Crumbles`	`right`
`05:20.24`	`ManxPower`	`d`
`05:20.54`	`Juggie`	`ok, then WaitExten runs. putting _XXXX,1,blah() in the same context will allow it to trap whatever the user types once the waitexten runs.`
`05:20.55`	`ManxPower`	`Juggie: I'm pretty sure his ITSP is not sending the DTMF`
`05:21.33`	`Crumbles`	`right but it doesn't see the dtmf or something`
`05:21.33`	`Crumbles`	`hold on a sec`
`05:21.34`	`Crumbles`	`i have an idea`
`05:25.05`	`*** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com)`
`05:25.11`	`Crumbles`	`yeah my provider isn't sending the dtmf tones or something`
`05:25.15`	`*** join/#asterisk asdx (n=diego@200.61.236.33)`
`05:25.20`	`Crumbles`	`i just made voicemailmain answer`
`05:25.24`	`Crumbles`	`it can't hear my password`
`05:25.42`	`ManxPower`	`nod As I suspected.`
`05:25.56`	`ManxPower`	`hence my looking for the sip debug`
`05:26.01`	`Crumbles`	`okay now it heard it this time`
`05:26.02`	`Crumbles`	`hold on`
`05:26.59`	`Crumbles`	`now it doesn't again`
`05:27.00`	`Crumbles`	`>-(`
`05:28.17`	`Crumbles`	`i don't get it why would voicemailmain hear the incoming dtmf but not DISA or waitexten`
`05:28.52`	`WildPikachu`	`"codec_speex.c:278 speextolin_framein: Out of buffer space" <= anyone seen this before?`
`05:29.09`	`ManxPower`	`Crumbles: are you using inband or rfc2833?`
`05:29.21`	`Crumbles`	`inband lets voicemailmain see it`
`05:29.28`	`Crumbles`	`but rfc2833 it can't see it`
`05:29.51`	`ManxPower`	`n ot surpizing. I'll bet it will if you disallow=all and allow=g726`
`05:29.53`	`Crumbles`	`but waitexten never sees it`
`05:30.00`	`Crumbles`	`g726?`
`05:30.06`	`Crumbles`	`is that high quality?`
`05:30.19`	`ManxPower`	`Crumbles: try it an see/hear`
`05:30.32`	`Crumbles`	`should i disallow ulaw and alaw?`
`05:30.44`	`ManxPower`	`no, the disallow=all will take care of that`
`05:30.54`	`Crumbles`	`yeah but i mean`
`05:30.58`	`Crumbles`	`i've got allow lines for them`
`05:31.01`	`Crumbles`	`should i comment them?`
`05:31.07`	`ManxPower`	`correct, comment them out.`
`05:31.42`	`Crumbles`	`and set dtmf=rfc2833`
`05:31.43`	`Crumbles`	`?`
`05:31.48`	`ManxPower`	`no!`
`05:31.50`	`Crumbles`	`k`
`05:31.52`	`*** join/#asterisk klaus7 (n=josh@cpe-69-203-212-182.nyc.res.rr.com)`
`05:31.56`	`ManxPower`	`stmfmode=rfc2833`
`05:31.59`	`ManxPower`	`..er..`
`05:32.01`	`Crumbles`	`right`
`05:32.04`	`Crumbles`	`that's what i meant`
`05:32.05`	`ManxPower`	`dtmfmode=rfc2833`
`05:32.13`	`ManxPower`	`there is no option dtmf=`
`05:32.18`	`Crumbles`	`i know`
`05:32.21`	`Crumbles`	`i'm just a lazy typer`
`05:32.30`	`ManxPower`	`some people thing there is.`
`05:32.37`	`ManxPower`	`think there is.`
`05:33.20`	`Crumbles`	`now the call isn't going through`
`05:33.26`	`Corydon76-home`	`The problem with being inexact is that there are people in here who lurk and pay attention to things that aren't quite correct`
`05:33.47`	`klaus7`	`hello, is it possible to have unique moh for each station/extension?`
`05:34.05`	`ManxPower`	`Crumbles: try allow=gsm instead of allow=g726`
`05:34.24`	`ManxPower`	`klaus7: in theory yes`
`05:34.51`	`Crumbles`	`call won't go through anymore`
`05:35.00`	`ManxPower`	`Crumbles: even with GSM?`
`05:35.02`	`Crumbles`	`sunrocket like only supports ulaw/alaw or something lame`
`05:35.03`	`Crumbles`	`yes`
`05:35.20`	`ManxPower`	`Crumbles: Have you considered using a less lame provider?`
`05:35.34`	`Crumbles`	`i pay less than 8 bucks a month`
`05:36.04`	`Juggie`	`which provider?`
`05:36.08`	`Crumbles`	`sunrocket`
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`05:36.18`	`ManxPower`	`Yeah, but if there DTMF does not work well with Asterisk.....`
`05:36.44`	`klaus7`	`What would a sip message from a UA look like that is starts moh?`
`05:37.29`	`ManxPower`	`klaus7: no idea. I'd use "sip debug" to see.`
`05:37.55`	`Crumbles`	`but why would voicemailmain have no problem hearing dtmf if sunrocket were sending it wrong?`
`05:37.57`	`olinux`	`whats a good vendor for receiving incoming calls on a dedicated number (USA), all i have is internet connection?`
`05:38.00`	`klaus7`	`ah good advice, is there a way to filter sip debug?`
`05:38.20`	`ManxPower`	`olinux: All providers suck. Teliax seems to (usually) suck less than most.`
`05:38.23`	`olinux`	`i use my cell as primary number but have no signal in my apartment`
`05:38.42`	`Crumbles`	`ManxPower if all providers suck then why bother with voip at all?`
`05:39.02`	`ManxPower`	`Crumbles: VoIP does not have to go across the internet`
`05:39.47`	`Crumbles`	`yeah but POTS = insane long distance charges of lame`
`05:40.03`	`olinux`	`just cancelled my landline when i activated cable internet (i only had landline for DSL)`
`05:40.28`	`ManxPower`	`Uh, you can get like 3cents/min`
`05:40.42`	`Crumbles`	`or you can get ... 0/min with voip`
`05:40.55`	`ManxPower`	`You still pay for toll calling for VoIP.`
`05:41.02`	`hads`	`You get what you pay for.`
`05:41.05`	`ManxPower`	`most people you want to talk to will not have VoIP`
`05:41.13`	`Crumbles`	`so?`
`05:41.29`	`Crumbles`	`less than 8 a month for unlimited local and long distance within the united states`
`05:41.52`	`hads`	`That's not 0c :)`
`05:42.03`	`Crumbles`	`the 8 a month is for local calls`
`05:42.19`	`Crumbles`	`and yes it is... i can talk all i want and the cost will not go up`
`05:42.21`	`ManxPower`	`it is $8.00/number of mins usage in the month`
`05:42.34`	`Crumbles`	`wrong`
`05:42.38`	`Crumbles`	`it's $8 for the month`
`05:42.41`	`Crumbles`	`no matter how much i call`
`05:42.43`	`Crumbles`	`no matter how often`
`05:42.46`	`Crumbles`	`no matter what time`
`05:42.58`	`hads`	`Hah, we know, but ManxPower is still right.`
`05:43.03`	`olinux`	`you dont call anyone, so they win`
`05:43.05`	`ManxPower`	`Crumbles: all those plans have caps, some companies just don't disclose the cap. Granted, their caps -- most people won;t hit them.`
`05:43.26`	`Crumbles`	`plus $3 credit each month toward international calls`
`05:43.42`	`ManxPower`	`and $8/month is a good deal.`
`05:43.44`	`Crumbles`	`he's asked them specificly`
`05:43.46`	`Crumbles`	`there is no cap`
`05:44.05`	`olinux`	`it's like the $4/month hosts that promise 1000 gigs of transfer, you go ahead and use it and next month you'll have no service`
`05:44.06`	`Juggie`	`ManxPower, did you have him try inband dtmf?`
`05:44.08`	`ManxPower`	`Crumbles: nail the line up 24/7 and see how long that "no cap" lasts.`
`05:44.30`	`ManxPower`	`Juggie: he was using inband when I started helping him.`
`05:44.45`	`ManxPower`	`rfc2833 was just something I wanted to see if it would work`
`05:45.24`	`ManxPower`	`For HOME use, much of VoIP is about cost savings on toll calling`
`05:45.33`	`Juggie`	`i'm reading some forums some people are reporting it working with rfc2833`
`05:45.38`	`ManxPower`	`for corporate that is not so much a driving force for many companies`
`05:46.36`	`Juggie`	`sunrocket officially doesnt even support external sip devices.`
`05:47.09`	`ManxPower`	`Juggie: He's falking the User-Agent: in sip.conf`
`05:48.11`	`Juggie`	`yah i figured that`
`05:48.41`	`ManxPower`	`I won't give my business to companies that try to lock their customers into hardware.`
`05:48.48`	`Crumbles`	`can't i make 2 different extensions ring at the same time whenever i get an incoming call? :(`
`05:49.10`	`ManxPower`	`Crumbles: real The Gook`
`05:49.11`	`ManxPower`	`~book`
`05:49.14`	`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`
`05:49.16`	`Crumbles`	`there's a book?`
`05:49.30`	`Crumbles`	`i tried blah&blah and it didn't do anything`
`05:51.14`	`Juggie`	`Dial(SIP/extension1&SIP/extension2)`
`05:51.54`	`ManxPower`	`Juggie: Crumbles got Asterisk working behind NAT in a very short time.`
`05:52.20`	`Crumbles`	`yeah i found it already`
`05:52.26`	`Crumbles`	`i typoed something stupid`
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`05:53.27`	`Juggie`	`ManxPower,?`
`05:53.32`	`Juggie`	`whats that have to do with anything`
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`06:26.38`	`FuriousGeorge`	`hey all`
`06:27.23`	`FuriousGeorge`	`i was asking about this earlier today and no one knew what i was talking about: when i transfer an extension on the PSTN to a meetme from my remote sip phone, asterisk loses the PSTN parties audi`
`06:27.27`	`FuriousGeorge`	`audio`
`06:27.38`	`FuriousGeorge`	`s/parties/party's`
`06:28.19`	`Supaplex`	`fun`
`06:28.23`	`FuriousGeorge`	`so ill assume that netsplit pushed my ? off the page`
`06:28.32`	`Supaplex`	`no`
`06:28.53`	`Supaplex`	`pstn has extensions?`
`06:28.58`	`FuriousGeorge`	`when i forward a pstn party to a meetme their audio gets lost, but only when i do it from my remote sip phone`
`06:29.07`	`FuriousGeorge`	`Supaplex: yes`
`06:29.42`	`Supaplex`	`to clarify, you try to connect a call from pstn into a meetme using your phone?`
`06:29.44`	`FuriousGeorge`	`and only when the extension is on the pstn`
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`06:30.09`	`FuriousGeorge`	`Supaplex: yeah, i transfer it there`
`06:30.16`	`FuriousGeorge`	`if the party is local then its fine`
`06:30.24`	`FuriousGeorge`	`or if the phone is local it is fine`
`06:30.32`	`Supaplex`	`what do you mean by extension on the pstn?`
`06:30.43`	`FuriousGeorge`	`Supaplex: i just mean a number`
`06:30.49`	`Supaplex`	`ok`
`06:30.50`	`FuriousGeorge`	`a did`
`06:31.00`	`Supaplex`	`ahh`
`06:31.03`	`Supaplex`	`click`
`06:31.21`	`FuriousGeorge`	`well let me try rebooting the server`
`06:31.38`	`Supaplex`	`strange. I'm not sure what to look into on that one.`
`06:31.54`	`FuriousGeorge`	`Supaplex: and last time i tried it it worked`
`06:32.47`	`FuriousGeorge`	`so something is very odd. it almost seems like a nat issue but nat traversal works with every other call`
`06:33.30`	`Supaplex`	`well, time for that 7-8hr nap we call sleep/bed. nite.`
`06:33.32`	`Supaplex`	`best of luck.`
`06:33.36`	`FuriousGeorge`	`thx`
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`07:23.35`	`bluregard`	`hi all`
`07:23.52`	`bluregard`	`anyone alive?`
`07:24.19`	`FuriousGeorge`	`depends on your question`
`07:24.28`	`bluregard`	`don't have a question`
`07:24.39`	`bluregard`	`just bored out of my skull`
`07:25.08`	`[TK]D-Fender`	`"Boring is between your ears"`
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`07:25.50`	`bluregard`	`right`
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`07:36.19`	`nextime`	`is function agent available on 1.2.13?`
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`08:07.05`	`wdsatr`	`hello`
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`08:07.12`	`wdsatr`	`need help`
`08:07.29`	`wdsatr`	`anyone here`
`08:08.29`	`SheriF_SpacE`	`wdsatr: what is ur problem ?`
`08:12.00`	`wdsatr`	`I am a windows person and new to fedore and having trouble installing asterisk`
`08:12.39`	`wdsatr`	`after installing is typed export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot`
`08:12.47`	`SheriF_SpacE`	`wdsatr: what is the problem ? how do u need to isntal it ? compiling or YUM`
`08:12.49`	`wdsatr`	`and I get unknow host`
`08:12.55`	`SheriF_SpacE`	`wdsatr: why ur using the CVS ?`
`08:13.26`	`SheriF_SpacE`	`wdsatr: to be easy i think u can YUM install ASterisk zaptel and ur done " but i don't know which verstion fedora include ...`
`08:13.32`	`wdsatr`	`i dont know i am just following what it said on http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation`
`08:13.57`	`SheriF_SpacE`	`wdsatr: leave that link for now ... try install it 1st with ur pacakge manger`
`08:15.09`	`wdsatr`	`ok I have fedore core 6. could you let me know how to install. all the steps if you can please`
`08:20.04`	`SheriF_SpacE`	`wdsatr: i'm not a fedora user u can ask how to setup ur package manager in #fedora`
`08:20.25`	`wdsatr`	`thanks`
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`08:55.28`	`hads`	`That page is out of date if it mentions cvs.`
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`09:18.18`	`M_at`	`Try downloading the tar.gz files rather than taking it from CVS`
`09:18.18`	`wdsatr`	`I just want to install asterisk and play with it`
`09:18.19`	`M_at`	`There's also a vmware package with it pre-installed that you can use on windows if you want to do it that way`
`09:18.19`	`wdsatr`	`I have downloaded it but do now know what to do with it`
`09:18.19`	`hads`	`~book`
`09:18.27`	`jbot`	`i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`09:18.27`	`M_at`	`I'd suggest getting used to Linux before getting into * - Asterisk is quite complex and spending a few weeks working with Linux and just getting used to the command line and file editting will serve you in very good stead`
`09:18.27`	`wdsatr`	`cool`
`09:18.27`	`M_at`	`Or download trixbox and play with that for a month`
`09:18.27`	`M_at`	`That's Linux + * + FreePBX + More pre configured`
`09:18.27`	`JT`	`yeah but you don't learn much taking that route`
`09:19.27`	`M_at`	`I dunno, I learnt quite a bit digging through the config files it created`
`09:19.28`	`JT`	`which is great if you know how to dig around config files`
`09:19.28`	`M_at`	`But whatever you do you need to understand how to compile software and edit text files, where libraries live and where to find packages and source code that may not be installed as standard`
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`09:28.37`	`patrickvox`	`anyone can give me a hand on voice translation-rule on cisco ios as5400 ?`
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`11:15.18`	`wdsatr`	`question what does in mean when it says "The configure script must be excuted before running make". how do is do it`
`11:18.57`	`Ftexcom`	`wdsatr, ./configure`
`11:29.56`	`WildPikachu`	`doesn't appear speex wants to work with me`
`11:33.54`	`xheliox`	`I told you not to disrespect the speex gods.`
`11:34.00`	`xheliox`	`Now look what's happened.`
`11:37.01`	`WildPikachu`	`disaster!`
`11:37.06`	`WildPikachu`	`"codec_speex.c:278 speextolin_framein: Out of buffer space"`
`11:37.29`	`WildPikachu`	`the person i'm trying to call says i sound like donald duck :(`
`11:37.52`	`xheliox`	`Perhaps there's nothing wrong with the codec and the truth just hurts?`
`11:38.19`	`WildPikachu`	`yea`
`12:04.09`	`*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es)`
`12:04.10`	`kink0`	`hi`
`12:04.47`	`kink0`	`how to return a BUSY,CONGESTION or NOANSWER to the caller party ? I need to return some cause codes to the peer`
`12:06.24`	`kink0`	`I set an extension like s,1,goto(s-${DIALSTATUS}) but what to place on s-BUSY or s-CONGESTION extensions in a manner the call is hanged returning the cause to the sender peer ?`
`12:06.28`	`*** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it)`
`12:21.15`	`kink0`	`I set an extension like s,1,goto(s-${DIALSTATUS}) but what to place on s-BUSY or s-CONGESTION extensions in a manner the call is hanged returning the cause to the sender peer ?`
`12:37.16`	`*** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch)`
`12:41.00`	`*** join/#asterisk Zork_ (n=Zork_@j214090.upc-j.chello.nl)`
`12:43.50`	`Zork_`	`Hi. I'm having some trouble getting chan_misdn working. When an outgoing call comes in through my ISDN, my SIP phone rings only once. If I pick it up, everything is fine. If I don't pick it up fast enough, it stops ringing like the other party hanged up (but didn't). As for dialing out though ISDN, it seems like only L1 gets activated, but never L2. Never really dials out, and gives up in 4-6 seconds. I've been fiddeling with this for quite a while`
`12:50.20`	`kink0`	`Zork_ paste your Dial() extension line, may be just timeout`
`12:50.41`	`Zork_`	`exten => _8X.,2,Dial(mISDN/1/${EXTEN:1},40,Tr)`
`12:52.24`	`Zork_`	`But I do get output on the console (like P[ 1] MGMT: SSTATUS: L1_ACTIVATED), so I think it must be something else.`
`12:55.43`	`Zork_`	`B.t.w. This pops up in my message log at restart: kernel: MDL_ERROR\|REQ (tei_l2)`
`12:58.25`	`Zork_`	`Seems to be the cause of all evil (I think), as it is layer 2 that doesn't seem to be working correctly.`
`12:58.38`	`Zork_`	`Can this be caused by a configuration error on my part?`
`12:59.32`	`*** join/#asterisk t_shravan (n=shravan@59.93.116.186)`
`12:59.47`	`t_shravan`	`HI ALL I AM FACING THE PROBLEM OF VOICE QUALITY IN ASTERISK`
`12:59.53`	`t_shravan`	`USING SIP`
`12:59.57`	`t_shravan`	`CAN ANY ONE HELP ME OUT`
`13:01.10`	`Zork_`	`B.t.w.: The MDL_ERROR only pops up in pmtp mode, if in ptp mode, that error is not present. But instead "kernel: l2mgr: addr:40000102 prim 23082 G" is shown (though the problem remains).`
`13:01.47`	`SheriF_SpacE`	`t_shravan: how do u make the call , what coceds , and the connection ?`
`13:02.11`	`kink0`	`Zork_, has you any absoutetimeout sentence in your extension.conf ?`
`13:02.56`	`Zork_`	`kink0: Nope.`
`13:03.32`	`kink0`	`Zork_ what happens if you dial your SIP phone from the console instead from incoming from the ISDN ? is ok ? still ringing ?`
`13:04.20`	`Zork_`	`How do I do that from console?`
`13:05.44`	`kink0`	`place your SIP phone on one local extension and then type dial <exten> in the CLI`
`13:06.03`	`t_shravan`	`I AM USING G729`
`13:06.12`	`t_shravan`	`FOR THE OUTSIDE AND AGENTS ARE USING ULAW`
`13:06.22`	`kink0`	`in that way you can issolate the problem, if still one ring, then may be some timeout in your SIP phone configuration`
`13:07.16`	`Zork_`	`I get a "Unable to re-open DSP device /dev/dsp: Permission denied" if I do that...`
`13:08.01`	`Zork_`	`t_shravan: No need for caps...`
`13:08.33`	`t_shravan`	`sorry`
`13:08.38`	`Zork_`	`NP`
`13:09.05`	`kink0`	`Zork_ stop all other apps that are ussing your sound device. If still, I would try to re-execute alsaconf`
`13:09.23`	`kink0`	`( I suposse you are ussing ALSA and you have one sound card )`
`13:09.32`	`Zork_`	`kink0: However, I can call SIP -> SIP. And it keeps ringing...`
`13:09.47`	`Zork_`	`Is that good enough?`
`13:09.49`	`kink0`	`ahh ok, then is the same test.`
`13:09.54`	`kink0`	`yes, enough`
`13:10.20`	`kink0`	`then the problem is just when an ISDN call arrives , have you try to debug and seek for some mistake ?`
`13:10.59`	`Zork_`	`yes... but I didn't quite get it... :-) It just seems to hang up.`
`13:11.11`	`Zork_`	`Perhaps it is easier to concentrate on dialing out through ISDN?`
`13:12.20`	`Zork_`	`When dialing out, a few seconds after seeing "P[ 1] MGMT: SSTATUS: L1_ACTIVATED", I get the following:`
`13:12.21`	`Zork_`	`P[ 1] handle_frm: frm->addr:42000103 frm->prim:3f182`
`13:12.22`	`Zork_`	`P[ 1] --> lib: RELEASE_CR Ind with l3id:30002`
`13:12.22`	`Zork_`	`P[ 1] --> lib: CLEANING UP l3id: 30002`
`13:12.59`	`Zork_`	`(and a bit more, but I figured this might be the important lines)`
`13:13.44`	`Zork_`	`Is there a "paste-site" used here? Where I can paste the full output?`
`13:14.27`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)`
`13:15.48`	`Zork_`	`I posted the full output when dialing out here: http://pastebin.com/827307`
`13:19.54`	`Zork_`	`And that "kernel: l2mgr: addr:40000102 prim 23082 G" popped up in my message log again.`
`13:20.58`	`*** join/#asterisk spr1te (n=spr1te@213.227.193.75)`
`13:24.12`	`Zork_`	`any idea kink0 ?`
`13:25.35`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`13:32.41`	`*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)`
`13:32.48`	`FuriousGeorge`	`hey all`
`13:33.03`	`FuriousGeorge`	`anyone know what people use for a video conferencing solution`
`13:35.13`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
`13:40.13`	`t_shravan`	`where can i get the non commercial g729`
`13:40.39`	`ManxPower`	`t_shravan: there is no such thing as non-commercial G729`
`13:41.30`	`Zork_`	`Can someone tell me what goes wrong here: http://pastebin.com/827307`
`13:41.54`	`Zork_`	`I'm trying to dial out through an ISDN line. But after L1_ACTIVATED it seems to go wrong...`
`13:42.49`	`*** join/#asterisk apardo (n=apardo@87.217.146.11)`
`13:53.20`	`*** join/#asterisk zotz (n=zotz@208.196.247.175)`
`13:54.24`	`coppice`	`is the commercial G.729 the one that keeps interrupting your conversation to tell you about shampoo or soap?`
`13:57.54`	`xheliox`	`say what?`
`14:04.17`	`Zork_`	`Is there an easy way to test misdn without using asterisk?`
`14:06.12`	`ManxPower`	`Zork_: What card do you have?`
`14:06.28`	`Zork_`	`Dynalink SP64+`
`14:06.39`	`ManxPower`	`Ah.`
`14:06.46`	`Zork_`	`which is a w6692pci card.`
`14:07.03`	`ManxPower`	`Zork_: you must not be in the USA/Canada.`
`14:07.15`	`Zork_`	`True, I'm from The Netherlands.`
`14:07.29`	`ManxPower`	`Zork_: Also try #asterisk-drinkers There are a fair number of EuroPeople there.`
`14:07.37`	`Zork_`	`drinkers ;-) lol`
`14:07.50`	`ManxPower`	`Zork_: I liked the netherlands when I was there visiting.`
`14:08.35`	`Zork_`	`I like living there :-)`
`14:09.15`	`ManxPower`	`Amsterdam, Delft, and the ciy the Phillips factory was.`
`14:09.23`	`Zork_`	`Eindhoven`
`14:09.47`	`Zork_`	`I presume?`
`14:09.51`	`coppice`	`"ciy the Phillips factory" would be ShenZhen or ShangHai`
`14:10.05`	`Zork_`	`;-)`
`14:11.55`	`ManxPower`	`Eindhoven`
`14:13.36`	`ManxPower`	`I'm disappointed about the growing anti-immigration movement in the Nederlands`
`14:14.29`	`Zork_`	`Yes, they have been pulling up some barriers lately. Whether or not that's a good thing, I don't know.`
`14:15.06`	`ManxPower`	`Before Katrina I wanted to move to the Netherlands`
`14:15.52`	`Zork_`	`Where are you from?`
`14:16.09`	`ManxPower`	`I was from New Orleans / Mississippi Coast`
`14:16.47`	`Zork_`	`Ah, I see. So Katrina hit you personally I guess...`
`14:17.15`	`ManxPower`	`.6m of floowing`
`14:17.19`	`ManxPower`	`flooding that is`
`14:17.26`	`Zork_`	`That sucks.`
`14:18.11`	`ManxPower`	`nod`
`14:19.46`	`ManxPower`	`Zork_: ISDN BRI is very uncommon in the USA, so you won't find much help from people there.`
`14:20.06`	`Zork_`	`:-(`
`14:20.20`	`Zork_`	`Oh well, I'll keep struggling ;-)`
`14:20.35`	`ManxPower`	`Zork_: more and more people in Europe are using ISDN BRI with Asterisk. Don't give up.`
`14:20.35`	`Zork_`	`There must be a way to get it to work eventually :-)`
`14:21.17`	`Zork_`	`Though I am considering to trade my ISDN line in for a SIP line...`
`14:21.33`	`ManxPower`	`Zork_: ISDN will always be more reliable than VoIP`
`14:21.39`	`Zork_`	`Yes I know.`
`14:21.55`	`Zork_`	`But I don't use it for data.`
`14:22.44`	`Zork_`	`And if I trade it in for a VOIP line, then I get to call for free to all landlines in The Netherlands, for the same price I'm paying now.`
`14:23.07`	`Zork_`	`But I'm not sure yet.`
`14:23.34`	`Zork_`	`If I do, then I still want to use my ISDN phones. So I guess I will have to see if I can get the ISDN card working in NT mode right?`
`14:23.51`	`ManxPower`	`nod`
`14:24.24`	`Zork_`	`B.t.w.: Would I have to change a lot in the wiring if I would like to do that?`
`14:24.40`	`Zork_`	`Or just put a terminator on the end which used to lead to the ISDN out?`
`14:25.08`	`ManxPower`	`ISDN BRI normally uses 1 pair (2 wires), you should not have to change the wiring.`
`14:25.59`	`Zork_`	`And the terminator? Would that be required?`
`14:26.05`	`kaldemar`	`erm.. the U interface uses one pair, but the S/T uses two pairs.`
`14:28.58`	`Zork_`	`Well, if I plug out the line, the phone's display goes dead. And if I plug in a terminator. It's still dead :-) So I guess it's not that easy :-)`
`14:29.09`	`Zork_`	`Can every ISDN card be set in NT mode?`
`14:29.48`	`*** join/#asterisk peka (n=pk@ACB35783.ipt.aol.com)`
`14:31.13`	`peka`	`hi everybody`
`14:31.15`	`kaldemar`	`don't know about every, but you can find a small list here:`
`14:31.24`	`kaldemar`	`http://isdn.jolly.de/cards.html`
`14:32.18`	`kaldemar`	`i've tried junghanns 4 port model and some hfc model (don't know the manufacturer), and they both worked with bristuff.`
`14:32.56`	`peka`	`does anyone know the default value and the range of possible values for the silencethreshold setting in voicemail.conf?`
`14:35.34`	`SheriF_SpacE`	`i have a question .. now i have 2 lines comming out from the PSTN over PRI or BRI "not sure don't know " to a Cisco VG " as i think " routed to 2 analog phones " as i understand this are extensions lines " so the user has to dial 9 before the call .. now i want to plug the 2 lines to an asterisk box . which`
`14:35.45`	`SheriF_SpacE`	`<PROTECTED>`
`14:37.10`	`tzafrir_laptop`	`SheriF_SpacE, just asking the same question again and again is not a good idea`
`14:37.37`	`tzafrir_laptop`	`For instance: is it PRI or BRI? This is something you could easily check`
`14:37.42`	`kaldemar`	`first of all you have to find out whether your line is BRI or PRI. 2 lines suggests that it is BRI.`
`14:38.16`	`SheriF_SpacE`	`tzafrir_laptop: i ask it only when i see new active member`
`14:38.20`	`kaldemar`	`then the case is that you need hardware that supports your line, then you can start thinking about modules.`
`14:38.21`	`SheriF_SpacE`	`may be someone will help me.`
`14:38.44`	`SheriF_SpacE`	`kaldemar: i don't have control over the lines / cisco VG`
`14:39.08`	`tzafrir_laptop`	`SheriF_SpacE, start by helping yourself. I believe I started answering this yesterday. I figure others have as well. Did your question evolve?`
`14:39.23`	`SheriF_SpacE`	`kaldemar: the BRI is not mine. i only have 2 extention from this BRI want to hock them up in asterisk box what hardware i should use ?`
`14:39.48`	`tzafrir_laptop`	`Frankly, I can't easily parse your question with all of those quote marks in non-obvious places`
`14:40.06`	`SheriF_SpacE`	`tzafrir_laptop: nop ur answer didn't help :( cuz i can't use a digital card cuz i only have 2 extentions from the BRI round`
`14:40.10`	`tzafrir_laptop`	`Right. So how can you connect to that extra box?`
`14:40.13`	`ManxPower`	`SheriF_SpacE: I would just use SIP between the Cisco and Asterisk`
`14:40.42`	`SheriF_SpacE`	`ManxPower: can't i don't have access to teh Cisco thing .. it's like i'm renting to extention lines and thats it.`
`14:40.51`	`kaldemar`	`SheriF_SpacE: i'm not quite sure about the scenario. find out exactly what the line is and how you plan on connecting to it, then someone could be able to help you.`
`14:41.25`	`tzafrir_laptop`	`A simple single-port BRI card is much cheaper than even a single FXO module for the TDM card. And you'll need two. Not to mention the better quality.`
`14:41.39`	`SheriF_SpacE`	`kaldemar: it's not about the line the idea is what i'm using is not a nromal PSTN line it's an exntesions in the VG deviec so i should use FXS or FXO modules ?`
`14:42.13`	`tzafrir_laptop`	`An FXO module acts like a phone. If you can hook up a normal analog phone there, you need an FXO port.`
`14:42.27`	`kaldemar`	`if they're analog extensions meant for analog telephones, you should use fxo.`
`14:42.53`	`kaldemar`	`we're having a stereo transmission here. :)`
`14:43.09`	`tzafrir_laptop`	`just some non-perfect echo`
`14:43.10`	`tzafrir_laptop`	`echo`
`14:43.23`	`peka`	`seems like nobody's at their keyboards`
`14:43.29`	`SheriF_SpacE`	`tzafrir_laptop: kaldemar: this is my problem i know that FXO modules for PSTN lines. and i use this senario in the office . but this 2 new extension lines i thought i should plug it to FXS..`
`14:43.33`	`kaldemar`	`non-perfect echo. how rude.`
`14:43.34`	`SheriF_SpacE`	`lol`
`14:44.06`	`SheriF_SpacE`	`kaldemar: yes i can hock normal analog phone to the line`
`14:44.08`	`tzafrir_laptop`	`kaldemar, I didn't tell the direction...`
`14:44.09`	`SheriF_SpacE`	`and to call i dial 9`
`14:44.16`	`SheriF_SpacE`	`then i get a tone and i dial the number i want`
`14:44.22`	`kaldemar`	`tzafrir_laptop: hehe`
`14:44.30`	`SheriF_SpacE`	`so to plug same line into asteriks u think i need FXO module not FXS ?`
`14:44.46`	`kaldemar`	`SheriF_SpacE: you've had your answer twice already.`
`14:46.41`	`peka`	`i've seen a default value of 128 being mentioned for silencethreshold. other sources claim it to be 1000. does anyone know for sure?`
`14:47.04`	`*** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`14:47.06`	`*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)`
`14:47.57`	`SheriF_SpacE`	`kaldemar: tzafrir_laptop okay thx :-) and sorry but ti's very confusing for me`
`14:56.16`	`*** join/#asterisk peka (n=pk@ACB35783.ipt.aol.com)`
`14:56.29`	`peka`	`or could you point me to an url where i might find an authoritative answer?`
`14:57.07`	`SheriF_SpacE`	`peka: did u cehck voip-info in voicemail.conf page?`
`14:58.12`	`tzafrir_laptop`	`peka, frankly I'm not sure which is the config file you're talking about and why you suddenly need to change it.`
`14:58.45`	`peka`	`i looked for it at voip-info.org maybe i missed something`
`14:59.33`	`peka`	`it's in voicemail.conf`
`15:01.53`	`peka`	`i'm just interested in this because setting it to a value of 1000 might either be fine or if the range is 0-255 somehow be mapped to fit or might be reset to a hardcoded default`
`15:02.12`	`peka`	`just want to be sure ...`
`15:07.29`	`xnon`	`hello`
`15:07.47`	`tzafrir_laptop`	`What does this parameter mean? What are the units?`
`15:07.57`	`xnon`	`i have some question`
`15:08.21`	`xnon`	`if the Port Range 10000 - 20000 is a RTP PORT`
`15:08.31`	`xnon`	`why in my phone say 8000?`
`15:09.01`	`peka`	`silencethreshold - When using the maxsilence setting, it is sometimes necessary to adjust the silence detection threshold to eliminate false triggering on background noise. - Silencethreshold allows the adminstrator to do just that. The default silencethreshold value is 128. Higher numbers raise the threshold so that more background noise is needed to cause the silence detector to reset. When employing this setting, some experimentation will be necessary`
`15:09.03`	`tzafrir_laptop`	`Asterisk uses this port range for ports it opens. The other party may use other values.`
`15:09.42`	`xnon`	`can i change the port 8000 for the otherone in the asterisk conf files?`
`15:10.04`	`xnon`	`what kind port is it? UDP or TCP`
`15:10.17`	`tzafrir_laptop`	`It shouldn't matter. But if it does, you need to configure the phone.`
`15:10.32`	`tzafrir_laptop`	`RTP is generally UDP`
`15:10.38`	`xnon`	`ok`
`15:10.42`	`peka`	`that's from voip-info. but it doesnt give any information about the units of the setting`
`15:11.05`	`peka`	`i thought you knew ... :)`
`15:11.24`	`xnon`	`so i can chancge the port 8000 in my phone and the same work?`
`15:11.38`	`xnon`	`8001 maybe!`
`15:14.28`	`*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)`
`15:15.02`	`*** join/#asterisk ixx (n=ixx@cpe-70-122-50-222.austin.res.rr.com)`
`15:16.43`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
`15:17.03`	`tzafrir_laptop`	`xnon, does it really matter? You should not assume in your filewall that all RTP traffic to remote hosts is to ports 10000:20000`
`15:17.36`	`xnon`	`yes but i have 2 Asterisk server in my local net!`
`15:18.04`	`tzafrir_laptop`	`Because that port number is sent as part of the handshaking, and is not set to stone in the SIP specifications.`
`15:18.16`	`xnon`	`i have one FreeBSD Machine with Asterisk direct connect in a cablemodem with a 1 IP Static Address`
`15:18.47`	`xnon`	`and other Asterisk Server Trixbox behind this FreeBSD`
`15:19.13`	`xnon`	`the FreeBSD Server its a Asterisk PBX and Gateway Firewall`
`15:19.32`	`xnon`	`but i can use 2 servers in the same ports`
`15:20.39`	`xnon`	`how i can make 2 asterisk server in my local area network`
`15:21.30`	`xnon`	`and register near the extranet with a IP Phone`
`15:21.30`	`xnon`	`??????????`
`15:23.53`	`*** join/#asterisk ixx (i=foobar@cpe-70-122-50-222.austin.res.rr.com)`
`15:24.31`	`peka`	`is it possible to record voice messages in mp3 format? i have seen information on using mp3 for music on hold but not for recording`
`15:27.10`	`*** join/#asterisk RamsesII (n=ekjhqkwj@core1-gw.net.cubemedia.it)`
`15:27.13`	`RamsesII`	`hi`
`15:37.25`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`15:39.19`	`*** join/#asterisk saftsack (n=saftsack@pD9E04B29.dip.t-dialin.net)`
`15:39.19`	`Xen^`	`mog : arround ?`
`15:48.39`	`*** join/#asterisk Dovid (n=Dovid@85.159.160.207)`
`15:57.23`	`*** join/#asterisk Cinen (n=Cinen@208.70.20.33)`
`16:05.50`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`16:06.13`	`kink0`	`how is possible I got this:"14","0","NO ANSWER" .... "s-CHANUNAVAI" ?`
`16:06.45`	`kink0`	`in the case PRI has not any available channels, how I get a time for dial and ringing the call ?`
`16:10.15`	`*** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net)`
`16:12.28`	`kink0`	`ahhh ok !!! I Think I understand...... CHANUNAVAL is that the Asterisk channels is not up, I think I had interpreted as there no any channel available in the PRI ( like Congestion )`
`16:14.56`	`awannabe`	`is there a way if a SIP phone is on a call that when you ring that phone it will go stright to voicemail?`
`16:15.35`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`16:16.03`	`kink0`	`awannabe, use $DIALSTATUS, if BUSY then go to voicemail`
`16:16.30`	`awannabe`	`kink0: BUSY is the same as on a call?`
`16:17.18`	`[TK]D-Fender`	`"Busy" is a bad concept with multi-line phones in mind....`
`16:17.35`	`awannabe`	`yeah, i agree, thats why im confused i guess, heh`
`16:17.44`	`awannabe`	`so a macro would be the best for this situtation then?`
`16:17.54`	`[TK]D-Fender`	`awannabe : what do you want to do exactly, and pastebin what you've done.`
`16:18.25`	`awannabe`	`ive done nothing yet, heh, i got a few phones that if they are on a call and they get a 2nd call they want it to go stright to voicemail`
`16:19.31`	`*** join/#asterisk Dovid (n=Dovid@85.159.160.207)`
`16:19.33`	`[TK]D-Fender`	`awannabe : I take it their phones can accept addition calls then?`
`16:19.40`	`Zork_`	`Besides my PCI isdn card, I also have a serial CAPI capable ISDN modem (Dynalink IS128 I believe). Would it be possible to hook that one up to asterisk (I presume via CAPI)?`
`16:19.52`	`awannabe`	`[TK]D-Fender: yeah, snom 360s`
`16:20.35`	`[TK]D-Fender`	`awannabe : thats a sad request.... You'll want to do a ChanIsAvail on the target phone with the "s" option and gotif out of the way then.`
`16:20.56`	`awannabe`	`i agree, these people are stupid, heh`
`16:21.45`	`awannabe`	`ive also got to figure a way for the recepontist to transfer calls stright to voicemail without using extensions, they _only_ want to use short buttons on the phone`
`16:21.54`	`[TK]D-Fender`	`First picking a bleh phone, then CRIPPLING it to boot....`
`16:22.15`	`awannabe`	`i like the snoms!`
`16:22.48`	`[TK]D-Fender`	`"Without using extensions" ? Clarify please.`
`16:22.48`	`awannabe`	`ive got to have DSS keys, polycoms dont suport that in a way that most old skool PBX customers want`
`16:23.06`	`[TK]D-Fender`	`DSS?`
`16:23.13`	`awannabe`	`one touch transfers, you know using DSS keys / shortcut keys on the phones themself`
`16:23.49`	`[TK]D-Fender`	`awannabe : Sure you can. Add as a speedial and it'll show up on the line-keys. Then you get expansion modules for the 601.`
`16:24.03`	`*** join/#asterisk ManxPower (n=manxpowe@200.sub-75-202-103.myvzw.com)`
`16:24.10`	`awannabe`	`those are stupid pricey, and take up WAY to much desk room`
`16:24.22`	`[TK]D-Fender`	`technically it'd be a 3 key transfer : [transfer] [blind] [target person's key]`
`16:24.38`	`awannabe`	`the snoms can 54 extensions for 300 bucks, and the polycom can do 52 extensiosn for around 900 bucks, heh`
`16:25.01`	`awannabe`	`what do you mean [blind] ?`
`16:25.24`	`awannabe`	`well imean, i understand that, but wondeirng how you would do that on the phone, heh`
`16:26.14`	`[TK]D-Fender`	`awannabe : Not that bad.... 800$ :)`
`16:26.36`	`[TK]D-Fender`	`awannabe : assuiming that the receptioninst wouldn't want to do attended transfers all the time....`
`16:26.40`	`awannabe`	`but the desk space it takes up is stupid, polycom needs to rethink that setup, its just to big, heh`
`16:26.59`	`awannabe`	`yeah, see that the prob, she wants to do attended, and then sometimes just send them stright to their voicemail`
`16:27.03`	`[TK]D-Fender`	`awannabe : Well they leave you lots of nice space to see the name :)`
`16:27.23`	`awannabe`	`with normal PBXs you can just hit "voicemail" then the extension or whatever, heh`
`16:27.51`	`awannabe`	`i was going to use polycom with tihs customer originally, but they said they were cheap looking and didnt like them, this customer is a odd bunch, ill say that much, lol`
`16:28.46`	`[TK]D-Fender`	`awannabe : yeah ... Polycom's are still my favourite, but I have it on good authority that Aastra is about to come out with something REALLY nice pretty soon :)`
`16:29.08`	`ManxPower`	`Polycoms are in my opinion the best phone considering price, sound quality, poliocies, features`
`16:29.13`	`*** join/#asterisk andresmujica (n=andresmu@200.75.38.138)`
`16:29.38`	`*** join/#asterisk tRSS (n=tRSS@124.29.204.178)`
`16:29.51`	`tRSS`	`has anyone successfully used iCall with asterisk?`
`16:29.53`	`awannabe`	`polycoms are great, i just hate the receponisit solution is so big and pricey, heh`
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`16:30.02`	`[TK]D-Fender`	`awannabe : I was at a clients last night and had some Polycoms that kinda crapped out and I've got working next to me right now. I brought their IP 601 back and now have every desktop model currently available at home :)`
`16:30.43`	`awannabe`	`nice!`
`16:31.09`	`awannabe`	`can you set that up on the polycom to where with 2 buttons you can transfer the call to voiucemail?`
`16:31.33`	`awannabe`	`all my voicemailboxes are 9xxx, gotta be a way, heh`
`16:31.59`	`[TK]D-Fender`	`ManxPower : I just something TOTALLY whacked you've got to try!`
`16:33.06`	`[TK]D-Fender`	`ManxPower : Using 2 Polys call from #1 to #2 twice, and hit "join" on both sides, then make any noise your want :)`
`16:33.20`	`[TK]D-Fender`	`ManxPower : And watch the bouncing begin!`
`16:33.34`	`Zork_`	`hehe...`
`16:33.53`	`[TK]D-Fender`	`awannabe : You can, but you'd have to set those speed dials to the # witht he "9" in front, and you'd lose the ability to use that key to call the actual person.`
`16:35.32`	`awannabe`	`damn`
`16:36.00`	`awannabe`	`i wonder if i can assign a speed dial to have "9" then press the 2nd speeddial with the extensions, and have it complete the call somehow`
`16:36.21`	`Zork_`	`Hey Fender. I have a problem with dialing out through an ISDN line. I've got the debug output on http://pastebin.com/827307 . If you have the time, perhaps you can take a look to see if you know what's wrong?`
`16:36.36`	`Zork_`	`Been fighting this all day to no avail :-(`
`16:37.13`	`[TK]D-Fender`	`awannabe : nope.`
`16:37.31`	`Zork_`	`(incoming calls also don't work that great, but that's a different story)`
`16:37.41`	`[TK]D-Fender`	`Zork_ : Sorry, never worked with BRI`
`16:37.52`	`Zork_`	`k, NP. Thanks.`
`16:40.31`	`Zork_`	`Just wondering... What exactly is BRI, and how did you figure out that I'm using that?`
`16:40.56`	`awannabe`	`BRI is two channel ISDN, PRI is multi channel`
`16:41.49`	`Zork_`	`Ah... okay... Thanks.`
`16:47.11`	`tRSS`	`is there a service similar to ipkall that would allow me to make outbound calls when configured with asterisk? ;)`
`16:47.27`	`tRSS`	`for free ? :oP`
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`17:03.36`	`tzanger`	`offtopic question`
`17:03.38`	`tzanger`	`anyone know how to find the network security settings on an xp system as guest?`
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`17:11.38`	`[TK]D-Fender`	`tzanger : Why login as administrator of course!`
`17:18.10`	`TommyTheKid`	`So, I asked a couple days ago about a "double-ringing" sound when I dial out via PRI lines (not SIP). I have seen a few posts around the net on various mailing lists that ask about it, but no one ever seemed to find a resolution? I just updated to the latest (GA) Asterisk and Zaptel to make sure that wasn't the cause. I have verified that we have 4 individual PRI's, not NFAS.. I have tried changing the LBO number (no effect). Any suggestions..`
`17:19.20`	`TommyTheKid`	`of course we would have a netsplit right as I am typing that :)`
`17:19.44`	`[TK]D-Fender`	`TommyTheKid : Are you using the "r" optin in your Dial command?`
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`17:19.56`	`TommyTheKid`	`no, (that was in the pastebin comment) :)`
`17:20.12`	`[TK]D-Fender`	`TommyTheKid : PB your zapata.conf`
`17:20.20`	`TommyTheKid`	`its up there`
`17:21.00`	`TommyTheKid`	`"priindication = outofband" was a suggestion last time (sorry I don't remember who) .. but that didn't seem to work either`
`17:21.03`	`[TK]D-Fender`	`TommyTheKid : I don't see it...`
`17:21.34`	`TommyTheKid`	`starts about line 21`
`17:21.34`	`TommyTheKid`	`I "excluded" the comments`
`17:21.34`	`[TK]D-Fender`	`TommyTheKid : I don't see your pastebin..... link it please.`
`17:21.38`	`TommyTheKid`	`<PROTECTED>`
`17:21.56`	`TommyTheKid`	`I proably overflowed a buffer that was a pretty long message ;)`
`17:22.17`	`[TK]D-Fender`	`TommyTheKid : Why the large LBO?`
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`17:23.07`	`TommyTheKid`	`I am not sure what to put there... We have a cross wired T1 that goes from our server room thru the building wiring (patch panels) to the main server room (two buildings away, and across the courtyard) to the main PBX`
`17:23.26`	`TommyTheKid`	`it was a "SWAG" .. I admit :)`
`17:23.46`	`TommyTheKid`	`it was "0" before, trying "3" .. neither seems to have an effect, positive or negative`
`17:24.47`	`[TK]D-Fender`	`TommyTheKid : You only use LBO when your wiring is very long and needs a boost. Get rid of the "priindication" line.`
`17:24.57`	`TommyTheKid`	`All the "network" connectivity is via 1000SX, or I would say the 300ft one`
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`17:25.31`	`[TK]D-Fender`	`TommyTheKid : And I'd suggest "pridialplan=local" and prilocaldialplan=local" as well.`
`17:26.12`	`TommyTheKid`	`what exactly does overlapdial do?`
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`17:26.58`	`TommyTheKid`	`pri(local)dialplan said it was only very rarely used, but I can add it and try it :)`
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`17:28.09`	`TommyTheKid`	`still double ringing`
`17:28.21`	`[TK]D-Fender`	`TommyTheKid : Did you kill * and reload zaptel then restart *?`
`17:28.27`	`TommyTheKid`	`yea "restart now"`
`17:28.32`	`TommyTheKid`	`er.. now`
`17:29.54`	`TommyTheKid`	`heh, guh, you can't rmmod as "asterisk" :)`
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`17:32.17`	`TommyTheKid`	`ok, afer a full stop, remove modules, reload modules, start asterisk, still have double ring :)`
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`17:35.17`	`TommyTheKid`	`http://pastebin.ca/251791`
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`17:35.45`	`TommyTheKid`	`unaffiliated? :`
`17:36.09`	`Qwell`	`yeah...something like that`
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`17:53.32`	`zippytech`	`any one know anything about dundi`
`17:54.16`	`xheliox`	`[TK]D-Fender: Around?`
`17:55.24`	`[TK]D-Fender`	`xheliox : yup`
`17:56.29`	`xheliox`	`When using the forward function on the Polycom's, it by default sends the call to Local/<number>@current-inboundcontext.`
`17:56.37`	`xheliox`	`How can I force it to use a different context?`
`17:57.13`	`[TK]D-Fender`	`xheliox : can't. Forwarding isn't a Polycom thing, its a SIP thing. Contextx don't exist, only users. and the polycom user has a context. thats where call's go. Period.`
`17:57.41`	`[TK]D-Fender`	`xheliox : And since the Polycom is doing the forwarding its based on where IT is allowed to go.`
`17:59.07`	`xheliox`	`Yeah, but it's not going to the context that the Polycom user is in, it's going to the context which the call was last sent to...`
`17:59.31`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`17:59.41`	`xheliox`	`So if it comes in on [default] and I use goto(inbound,s,1)... it tries to dial out over [inbound].`
`18:00.53`	`[TK]D-Fender`	`xheliox : It goes out wherever that person would normally. Which means employees can abuse work to call LD relatives, etc in many implementations.`
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`18:02.57`	`xheliox`	`Okay. Thanks.`
`18:07.10`	`TommyTheKid`	`xheliox: forward (at least on the cisco) uses the context that the user (of the phone) is in, not the inbound context`
`18:07.32`	`xheliox`	`TommyTheKid: Yeah, that's what I thought, but I'm positive that's not what's happening.`
`18:07.39`	`[TK]D-Fender`	`TommyTheKid : Thats exactly what was clarified`
`18:07.45`	`TommyTheKid`	`maybe you need "insecure=invite" :)`
`18:08.20`	`TommyTheKid`	`can the phone in question make that call without forwarding in play?`
`18:09.03`	`TommyTheKid`	`fuck I really hate the 186XX extensions.. i get a flood of inbound calls at the top of every hour`
`18:13.22`	`xheliox`	`TommyTheKid: Yes, without any issue. And I've not normally noticed this, because my contexts are usually less hairy. :)`
`18:13.25`	`TommyTheKid`	`xheliox: another good test (IMO) is to make your inbound calls be on an "inbound" context, that way if something ends up in "default" you know something has gone awry`
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`18:16.48`	`TommyTheKid`	`also just tested with my polycom 500 .. it gets a sip 302 "moved temporarily" back`
`18:17.19`	`TommyTheKid`	`I think I had to add something to allow for the stupid EBS servers to do that "promiscuously"`
`18:18.08`	`TommyTheKid`	`"promiscredir = yes"`
`18:18.20`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`18:18.28`	`TommyTheKid`	`but I don't know if thats needed for this :)`
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`18:22.44`	`file`	`[TK]D-Fender: eep!`
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`18:25.07`	`[TK]D-Fender`	`file : I don't want to work today!`
`18:25.51`	`*** part/#asterisk Kamu (n=kamu@ns1.liquidexploit.com)`
`18:26.25`	`file`	`[TK]D-Fender: disappear off the map!`
`18:27.35`	`[TK]D-Fender`	`gedit http://maps.google.com`
`18:28.35`	`file`	`exactly.`
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`18:31.36`	`asdx`	`where can i get the book?`
`18:32.06`	`TommyTheKid`	`your local library?`
`18:32.17`	`TommyTheKid`	`amazon.com?`
`18:33.41`	`asdx`	`the asterisk free book i mean`
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`18:37.05`	`svenna_`	`asdx: google: http://www.ute.edu.ec/walc2006/track5/AsteriskTFOT.pdf`
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`18:40.03`	`[TK]D-Fender`	`Coming to a duplexing laser printer near you!`
`18:41.44`	`asdx`	`svenna_: thanks`
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`18:47.58`	`[TK]D-Fender`	`TommyTheKid : Printed nice for me and I don't mind the dimensioning marks personally.`
`18:48.05`	`[TK]D-Fender`	`TommyTheKid : makes me feel "on taget" :)`
`18:52.22`	`TommyTheKid`	`heh, and I was just thinking for $20, it might actually be less expensive to just buy it on Amazon.. of course you can read the PDF for instant gratification :)`
`18:55.06`	`TommyTheKid`	`starfish was a sweet "logo" tho :)`
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`18:55.54`	`mog`	`<PROTECTED>`
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`18:57.12`	`[TK]D-Fender`	`TommyTheKid : 300 pg @ 1.2 c/ page + 500pg ream @ 4$ (500) means the book costs < 7$ to print yourself (CDN)`
`18:57.37`	`TommyTheKid`	`then you really ought to have it bound :)`
`18:57.40`	`mog`	`and you dont support the authors`
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`18:57.45`	`mog`	`who are good guys`
`18:57.52`	`TommyTheKid`	`mog: http://www.oreilly.com/catalog/apacheckbk/colophon.html <-- moose already taken`
`18:57.53`	`mog`	`and gave you book for free anyways`
`18:58.11`	`mog`	`hmm freaking apache`
`18:58.15`	`[TK]D-Fender`	`mog : What are you talking about.... I drove file around town while he was down here last year! :)`
`18:58.27`	`[TK]D-Fender`	`mog : Blitzrage too!`
`18:58.34`	`mog`	`file didnt write the book`
`18:58.39`	`mog`	`blitzrage did`
`18:58.52`	`[TK]D-Fender`	`mog : And I think my support in HERE is somewhat counts, no?`
`18:59.30`	`mog`	`id just prefer people pay for book`
`18:59.35`	`mog`	`but this is my personal preference`
`18:59.39`	`mog`	`people do what they want`
`18:59.50`	`mog`	`i own 4 coppies so i guess i make up for some other people`
`19:00.05`	`file`	`I only own 1, I'm lame >_<`
`19:00.52`	`mog`	`well 3 of them were gifts....`
`19:01.48`	`mog`	`people keep thinking, hey you love asterisk, ill get you the asterisk book`
`19:02.10`	`file`	`mog: internet status?!?`
`19:02.24`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`19:02.31`	`mog`	`20% packet loss`
`19:02.35`	`mog`	`reasonable speed`
`19:02.38`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`19:02.41`	`mog`	`very annoyed`
`19:03.39`	`*** join/#asterisk sloth (n=josh@pool-162-83-156-97.ny5030.east.verizon.net)`
`19:05.34`	`mog`	`and because i added a tuner to my myth box`
`19:05.37`	`mog`	`i need more ram`
`19:05.41`	`mog`	`which is depressing`
`19:05.45`	`mog`	`but meh had to do it`
`19:06.46`	`file`	`:(`
`19:08.03`	`*** join/#asterisk jim_sunnyvale (n=chatzill@c-67-161-114-204.hsd1.wa.comcast.net)`
`19:08.15`	`jim_sunnyvale`	`good morning, i've been around asterisk for a couple of years, but first time on IRC`
`19:09.49`	`jim_sunnyvale`	`can i ask a question of the group regarding Polycom and NAT traversal?`
`19:10.35`	`Xen^`	`mog : you arround ?`
`19:11.00`	`mog`	`yes`
`19:11.01`	`mog`	`but busy`
`19:11.08`	`Xen^`	`humm`
`19:11.17`	`Xen^`	`have you fixed that thing which i showed you last time ?`
`19:12.54`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`19:14.39`	`*** part/#asterisk rene1 (n=rene1@189.140.216.252)`
`19:15.47`	`jim_sunnyvale`	`did my question make it out to the group?`
`19:16.19`	`jim_sunnyvale`	`i'm new to irc`
`19:16.19`	`file`	`yes, and it's better to just ask your question`
`19:16.19`	`file`	`that way if someone has an answer/wants to answer, they will answer`
`19:16.24`	`jim_sunnyvale`	`ok, thx`
`19:17.01`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`19:23.24`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`19:25.18`	`*** part/#asterisk asdx (n=nhoNhi@200.61.236.33)`
`19:25.51`	`file`	`[TK]D-Fender: how about that weather eh?`
`19:27.00`	`jim_sunnyvale`	`i am trying to install a customer with Linksys RV082 and 5 Polycom IP501 (firmware is 1.6.6)`
`19:27.14`	`jim_sunnyvale`	`the asterisk system is on a public IP at a colo`
`19:27.36`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`19:27.58`	`jim_sunnyvale`	`the phones are marked NAT=YES and QUALIFY=YES but the issue is the phones become UNREACHABLE all the time`
`19:28.17`	`jim_sunnyvale`	`i've even gone so far as to trun the firewall on the RV082 off.`
`19:28.45`	`jim_sunnyvale`	`problem occurs on CISCO 7940 too, but LINKSYS PAP2 stays registered just fine`
`19:28.51`	`*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`19:29.16`	`jim_sunnyvale`	`these same phones on the same IP connection work just fine behind a Linksys WRT54G`
`19:29.34`	`jim_sunnyvale`	`any ideas would be appreciated, sorry for the long question`
`19:29.38`	`EmleyMoor`	`Can someone direct me to a softphone that works well on a network behind a NAT, when the asterisk box to which it connects is at a distance but on a public IP?`
`19:30.07`	`EmleyMoor`	`Er, a SIP softphone, that is`
`19:35.07`	`jim_sunnyvale`	`sjphone is a good sip phone`
`19:35.08`	`jim_sunnyvale`	`idefisk is a good iax phone which is better at nat traveral`
`19:39.57`	`*** join/#asterisk [jwb] (i=jwb@jwb.sh)`
`19:40.58`	`EmleyMoor`	`OK - How about a good truly-free SIP phone that would work in my circumstances?`
`19:41.22`	`EmleyMoor`	`(I know iax is better for nat traversal - trying to get some comparison in quality)`
`19:41.54`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`19:42.51`	`*** join/#asterisk sloth (n=josh@64.3.170.41.ptr.us.xo.net)`
`19:44.40`	`*** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`19:48.21`	`*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`19:48.41`	`EmleyMoor`	`Sorry about that - hosed my audio badly!`
`19:49.11`	`EmleyMoor`	`Any suggestions for a good truly-free SIP phone that can cope with being the wrong side of NAT?`
`19:49.17`	`*** join/#asterisk kristalino (i=kristali@gateway/tor/x-16ed2f0970e306a9)`
`19:49.19`	`kaldemar`	`EmleyMoor: have you tried firefly?`
`19:49.44`	`kaldemar`	`http://www.freshtel.net/firefly/download/`
`19:50.32`	`*** part/#asterisk kristalino (i=kristali@gateway/tor/x-16ed2f0970e306a9)`
`19:51.02`	`EmleyMoor`	`Um... no...`
`19:51.07`	`kaldemar`	`x-lite is also widely used.`
`19:51.11`	`EmleyMoor`	`I should also point out that I want it for Linux`
`19:51.22`	`EmleyMoor`	`Widely used but only narrowly free :-(`
`19:51.52`	`kaldemar`	`http://www.voip-info.org/wiki-VOIP+Phones`
`19:52.12`	`kaldemar`	`there's quite a few soft phones also.`
`19:53.50`	`EmleyMoor`	`Yes - a lot of tinkerig on both the client and service sides`
`19:57.05`	`*** join/#asterisk Tili (n=tili@202.133.67.188)`
`19:57.23`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`19:58.40`	`*** join/#asterisk [Wiebel] (i=wiebel@wiebel.nl)`
`19:58.45`	`[Wiebel]`	`Hi there`
`19:59.31`	`[Wiebel]`	`in asterisk 1.0 you could use the "H" flag with the dial cmd. When I then called my asterisk box and press * while ringing it did a +101`
`19:59.49`	`[Wiebel]`	`now in 1.2 this doesnt work anymore unless you answer the call`
`19:59.50`	`robin_sz`	`EmleyMoor, regarding soft phones .... you have two choices: 1) commercial non-free ones that work nicely, but you have to pay a few $ for .. 2) truly free ones that seem never to work, but, you COULD make them work .. apparently;`
`19:59.58`	`[Wiebel]`	`is there a way to get this working ?`
`20:03.52`	`*** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com)`
`20:07.17`	`EmleyMoor`	`Is there a reasonably good listing of Linux SIP softphones?`
`20:07.18`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`20:07.34`	`SheriF_SpacE`	`EmleyMoor: what ?`
`20:07.35`	`*** join/#asterisk ToyMan (n=stuq@dpc6714368169.direcpc.com)`
`20:08.08`	`EmleyMoor`	`I'm trying to investigate what softphones are available for Linux`
`20:08.36`	`*** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net)`
`20:08.48`	`SheriF_SpacE`	`EmleyMoor: ahh there is a softphone page in voip-info check it out`
`20:09.20`	`EmleyMoor`	`Yes, but that's a list of softphones generally - some of which are only for Windows...`
`20:09.47`	`SheriF_SpacE`	`then there is no linux version of this phones ;-)`
`20:10.01`	`*** join/#asterisk t_shravan (n=shravan@59.93.112.243)`
`20:10.30`	`EmleyMoor`	`... yes - but I want to see at a glance which ones do exist for Linux - not in the smallprintg`
`20:12.45`	`EmleyMoor`	`It's like this... Somename Softphone - described but no OS mentioned...`
`20:12.57`	`EmleyMoor`	`Anothername Softphone - described - for Windows...`
`20:13.03`	`EmleyMoor`	`And so on`
`20:13.20`	`EmleyMoor`	`It needs a spade to dig out which are for Linux`
`20:17.42`	`EmleyMoor`	`Hmmm... found a useful list of VoIP stuff available for Linux`
`20:18.56`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
`20:19.08`	`SheriF_SpacE`	`EmleyMoor: great`
`20:21.51`	`EmleyMoor`	`How do I set up kphone to use a specific audio device?`
`20:23.44`	`EmleyMoor`	`(can it be done)`
`20:25.49`	`mpruett`	`Guys - When I try to send a sip notify messsage to one of my Sipura devices, I get "401 Unauthorixed" message back. DOes anyone know why this would happen? THe device is working fine otherwise - registration, term, orig, etc..`
`20:28.35`	`*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)`
`20:35.24`	`EmleyMoor`	`Are all iax softphones "mostly the same quality"?`
`20:37.07`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`20:39.20`	`SheriF_SpacE`	`EmleyMoor: i think so`
`20:39.37`	`SheriF_SpacE`	`EmleyMoor: i wish to find a decent open source SIP/IAX phone .. i think Ekiga very promising`
`20:39.57`	`EmleyMoor`	`Ekiga refuses to run for me - complains about a missing audio plugin`
`20:42.53`	`EmleyMoor`	`Or rather, no usable audio plugin`
`20:44.03`	`SheriF_SpacE`	`EmleyMoor: what is ur distro ?`
`20:44.10`	`SheriF_SpacE`	`EmleyMoor: and how did u install ekiga ?`
`20:44.26`	`EmleyMoor`	`Debian - went into aptitude and selected it`
`20:46.46`	`SheriF_SpacE`	`EmleyMoor: u can do aptitude install ekiga and should get all dependancies`
`20:47.03`	`EmleyMoor`	`That is the equivalent of what I did`
`20:47.33`	`SheriF_SpacE`	`yes`
`20:47.47`	`SheriF_SpacE`	`EmleyMoor: but what u mean with missing audio plugin /what is the error message ?`
`20:48.19`	`EmleyMoor`	`No usable audio plugin detected`
`20:48.54`	`SheriF_SpacE`	`strange`
`20:51.21`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-20-221.red.bezeqint.net)`
`20:51.46`	`*** join/#asterisk Ken_rom (n=sdfgas@63.175.158.33)`
`20:51.59`	`Ken_rom`	`hi guys:)`
`20:52.07`	`*** join/#asterisk dasenjo (n=dasenjo@201.228.128.10)`
`20:52.36`	`Ken_rom`	i`m looking for some help
`20:52.46`	`Ken_rom`	`coding a php script`
`20:53.06`	`[Wiebel]`	`what is used for logging asterisk calls nowadays?`
`20:53.12`	`[Wiebel]`	`frontend that is`
`20:55.21`	`Ken_rom`	`can anyone help me plz`
`20:56.11`	`Zork_`	`Ken_rom: Is this #php? ;-)`
`20:56.18`	`Ken_rom`	`no`
`20:56.29`	`Ken_rom`	`but i need help linking the script to the dailer`
`20:56.42`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-20-221.red.bezeqint.net)`
`20:57.08`	`Zork_`	`Not that I can help you with that, but it might help specifying what you want.`
`20:57.51`	`wunderkin`	`Ken_rom, you probably want to use the manager`
`20:58.12`	`Ken_rom`	`huh? what manager?`
`20:59.14`	`wunderkin`	`asterisk manager, there should be some info on the wiki`
`21:01.12`	`SheriF_SpacE`	`Ken_rom: there is asterisk-php thing`
`21:02.17`	`*** join/#asterisk Ken_rom (n=sdfgas@63.175.158.33)`
`21:02.22`	`Ken_rom`	`sorry about that i ot dc`
`21:02.38`	`Ken_rom`	`what was that u guys were telling me about some type of amanger`
`21:03.33`	`wunderkin`	`~wiki`
`21:04.16`	`wunderkin`	`ok well i forgot what the command is now`
`21:04.56`	`*** join/#asterisk sloth (n=josh@cpe-74-64-52-106.nyc.res.rr.com)`
`21:05.15`	`[Wiebel]`	`anyone using skinny.so for cisco 79xx phones by any chance?`
`21:05.19`	`SheriF_SpacE`	`Ken_rom: http://www.voip-info.org/wiki/view/Asterisk+PHP`
`21:05.40`	`Ken_rom`	`ook..ty`
`21:06.02`	`Ken_rom`	`<PROTECTED>`
`21:07.15`	`sloth`	`anyone know how i might create the ability for each extension to have their own music on hold?`
`21:09.20`	`sloth`	`Perhaps would i use SetMusicOnHold before dialing the extension?`
`21:09.58`	`EmleyMoor`	`Ah, sorted it`
`21:10.13`	`EmleyMoor`	`I now get a security failure :-(`
`21:10.40`	`EmleyMoor`	`Security check failed when I try to make a call`
`21:11.57`	`*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)`
`21:15.17`	`EmleyMoor`	`Now, why would that be?`
`21:16.49`	`*** join/#asterisk tRSS (n=tRSS@124.29.204.178)`
`21:18.25`	`tRSS`	`i need some help with firewall and natting. i have successfully port forwarded my udp 5060 to my asterisk box. but the user can't hear any sound. my asterisk is behind a nat and so is the user. rtp debug showed me the following several similar lines: Sent RTP packet to 193.XX2.2XX.2:49154 (type 3, seq 51417, ts 160, len 33)`
`21:18.46`	`tRSS`	`i mean the user is behind another nat.`
`21:19.00`	`wunderkin`	`5060 is not rtp`
`21:19.05`	`tRSS`	`i know`
`21:19.19`	`tRSS`	`i meant, i have port forwarded 10000-10003 on my firewall as well`
`21:19.26`	`tRSS`	`for rtp`
`21:19.38`	`tRSS`	`testing with limited ports initially`
`21:23.09`	`mishehu`	`sip + rtp is a pain when NAT is involved, but then again, NAT itself is a glorious hack.`
`21:23.36`	`*** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net)`
`21:23.46`	`mishehu`	`I see that the linux kernel netfilter modules includes ip_nat_sip as of kernel 2.6.18`
`21:24.34`	`tRSS`	`mishehu: you are right and for that reason, i have my iax working perfectly. i just want to be able to run sip as well over the nat, because a lot of my users will demand sip initially`
`21:25.09`	`EmleyMoor`	`I want to get sip working over nat too - just in case`
`21:25.27`	`[TK]D-Fender`	`tRSS : Don't limit your ports while you're just trying to get it working period. Please pastebin the [general] section of your sip.conf`
`21:25.58`	`tRSS`	`[TK]D-Fender: good to see you around. pasting now.. give me a sec.`
`21:26.02`	`*** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net)`
`21:27.25`	`tRSS`	`[TK]D-Fender: here you go: http://pastebin.ca/251938`
`21:29.29`	`[TK]D-Fender`	`tRSS : externip should not have anything but an IP in there. if youare only trying to get basic phones working onthe outside, then I'd suggest you pull out realm (BIG suspicion on that, esp as its a PRIVATE one), and why is bindport at 6050 and not SIP standard 5060?`
`21:30.01`	`EmleyMoor`	`What does "Security check failed" mean in ekiga? How do I make it work right with asterisk?`
`21:30.08`	`[TK]D-Fender`	`tRSS : And I'll trust that your preference level on G.729 is backed up licence wise on both ends if involved.`
`21:30.25`	`tRSS`	`bindport is 6050 because my isp is blocking 5060, and this machine is purely experimental machine to see how nattting and firewalls fit in my scenario`
`21:30.45`	`tRSS`	`g.729 is licensed on both ends`
`21:31.35`	`tRSS`	`i was also under the impression that i can use the hostname in externip, since, my external ip keeps changing, hence, i am using dyndns (homelinux.org)`
`21:34.02`	`[TK]D-Fender`	`tRSS : No, thats what externhost is for, and because you have BOTH, I'm not sure that that is not interfereing. For the purpose of testing I suggest you take out the host and manually pulg in the IP.`
`21:34.30`	`tRSS`	`[TK]D-Fender: sure, let me try that.`
`21:35.36`	`*** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com)`
`21:36.11`	`*** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`21:36.43`	`tRSS`	`tried it and got the following again (with rtp debug on): Sent RTP packet to 193.XX2.2XX.2:49156 (type 3, seq 28856, ts 160, len 33)`
`21:39.32`	`[TK]D-Fender`	`tRSS : What NAT router?`
`21:40.15`	`tRSS`	`I using a linksys wrt54gs + my adsl modem ( i know, its not the best setup).`
`21:42.07`	`tRSS`	`i think i would have use some public stun server to avoid all the problems but I still believe the rtp issue would still haunt me.`
`21:43.02`	`mpruett`	`Guys - When I try to send a sip notify messsage to one of my Sipura devices, I get "401 Unauthorized" message back. Does anyone know why this would happen? The device is working fine otherwise - registration, term, orig, etc..`
`21:45.32`	`*** join/#asterisk pygrammer (n=pygramme@ip68-100-96-205.dc.dc.cox.net)`
`21:45.36`	`pygrammer`	`Hey all`
`21:46.35`	`pygrammer`	`I'm curious about receiving text messages on my Asterisk box from cell phones`
`21:46.37`	`[TK]D-Fender`	`tRSS : An I geuss killing off "fromdomain couldn't hurt either.`
`21:47.15`	`pygrammer`	`Is this possible via IP, or do I have to go through the PSTN?`
`21:47.23`	`tRSS`	`[TK]D-Fender: true. that is there for registering when asterisk is acting as a sip client.`
`21:47.46`	`*** join/#asterisk Dovid (n=Dovid@85.159.160.207)`
`21:47.49`	`[TK]D-Fender`	`tRSS : You shouldn't put that in [general], only withing your peer & users setups`
`21:48.04`	`*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)`
`21:48.26`	`tRSS`	`sure. i think i will go with the tutorials at fridu.org if nothing works out. they seem to fit my scenario.`
`21:50.06`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`21:52.17`	`pygrammer`	`so, how can I set up receiving text messages from cell phones on my Asterisk box... I'm in the U.S., so most of the documentation out there doesn't apply`
`21:54.26`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
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`22:06.00`	`Dovid`	`test\`
`22:08.07`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
`22:09.32`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
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`22:46.03`	`xheliox`	`sho' is quiet`
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`23:11.09`	`olinux`	`trying to compile the 1.4 beta on centos 4.4 (my first asterisk install)`
`23:11.14`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`23:11.27`	`olinux`	`install instructs say make && make install`
`23:11.40`	`olinux`	`but asterisk complains that i need to fun ./configure`
`23:11.44`	`FuriousGeorge`	`hey all`
`23:12.27`	`olinux`	`hmm ./configure passed this time`
`23:12.31`	`olinux`	`hi FG`
`23:12.44`	`FuriousGeorge`	`im looking at voip info under "asterisk consultants" in the NY / NJ area. webpages dont work, phone numbers are disconnected, and when i do get through the pbx is kinda crappy sounding :) can anyone recommend anyone in the tri-state area`
`23:13.14`	`FuriousGeorge`	`ive installed a few systems freelance over the last few years and i need a "support partner" so i can go to europe for a while`
`23:14.25`	`*** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com)`
`23:21.56`	`Dovid`	`FuriousGeorge: we are loacted in central NJ`
`23:22.22`	`FuriousGeorge`	`Dovid: mind if i pm you to ask you some ?`
`23:22.30`	`Dovid`	`np`
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`23:24.30`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`23:25.30`	`*** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com)`
`23:30.31`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`23:36.41`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`23:43.24`	`FuriousGeorge`	`~lasttseen shmaltz`
`23:43.38`	`FuriousGeorge`	`~lastspoke shmaltz`
`23:43.51`	`FuriousGeorge`	`~lastspoke schmaltz`
`23:44.06`	`FuriousGeorge`	`~seen FuriousGeorge`
`23:44.21`	`jbot`	`furiousgeorge is currently on #asterisk (10h 11m 39s). Has said a total of 10 messages. Is idling for 14s, last said: '~seen FuriousGeorge'.`
`23:44.23`	`FuriousGeorge`	`wadup Sup`
`23:44.34`	`Supaplex`	`sup :)`
`23:44.51`	`FuriousGeorge`	`~seen shmaltz`
`23:44.55`	`jbot`	`shmaltz <n=mybox@mail.dmaven.com> was last seen on IRC in channel #asterisk, 1d 20h 30m 12s ago, saying: 'you are right'.`
`23:48.09`	`Supaplex`	`can my dialplan fetch modify dates of files? what about the current line number?`
`23:48.17`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`23:50.05`	`mog`	`sure`
`23:50.53`	`Qwell`	`mog: y0`
`23:51.00`	`mog`	`oy`
`23:51.25`	`Supaplex`	`spiffy. I'll chew on the wiki a bit.`
`23:54.11`	`bluregard`	`hi all`
`23:55.27`	`bluregard`	`I was looking at the admin guide for the polycom IP 501, can those phones pull down a directory from voicemail.conf or do you need a separate file like the xml file it talks about?`

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