irclog2html for #asterisk on 20061022

`00:03.13`	`goodbot`	`hopefully someone gets to handling those errors sooner or later :(. Thanks anyway folks.`
`00:03.22`	`goodbot`	`I'll log a bug now I know where to do it.`
`00:03.25`	`*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)`
`00:10.11`	`*** join/#asterisk xnon (n=xnon@200.8.30.161)`
`00:10.29`	`xnon`	`amigos`
`00:10.31`	`xnon`	`friends`
`00:10.41`	`xnon`	`i have a warning about MOH`
`00:10.56`	`xnon`	`Oct 22 02:07:16 WARNING[4883]: res_musiconhold.c:838 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.`
`00:11.05`	`xnon`	`what kind of warning is it?`
`00:11.18`	`xnon`	`i can to understand`
`00:11.24`	`macTijn`	`you have no zap device`
`00:11.30`	`xnon`	`emmm no`
`00:11.31`	`macTijn`	`so there's no real timing`
`00:12.01`	`xnon`	`but this warning it is important?`
`00:12.09`	`macTijn`	`so all sound stuff is happening based on what asterisk thinks is a timely fashion`
`00:12.13`	`macTijn`	`well`
`00:12.23`	`macTijn`	`does your moh sounds choppy ?`
`00:12.39`	`xnon`	`i dont know what is choppy! :p`
`00:12.41`	`xnon`	`jejejee`
`00:12.44`	`xnon`	`sorry`
`00:12.47`	`tzafrir_laptop`	`ztdummy should do`
`00:12.55`	`macTijn`	`tzafrir_laptop: true`
`00:13.01`	`macTijn`	`xnon: jittery`
`00:13.03`	`tzafrir_laptop`	`It should provide a pseudo channe`
`00:13.22`	`xnon`	`i have son songs in a directory /var/lib/asterisk/mohmp3`
`00:14.49`	`macTijn`	`xnon: how does the moh sound ?`
`00:15.03`	`macTijn`	`does it sound it's playing "weird" ?`
`00:15.09`	`macTijn`	`or does it sound OK ?`
`00:15.18`	`xnon`	`sound ok i think so`
`00:15.24`	`macTijn`	`ok`
`00:15.30`	`xnon`	`but my asterisk console say this error`
`00:15.32`	`xnon`	`:(`
`00:15.39`	`macTijn`	`then my opinion is you can ignore the warning`
`00:15.41`	`macTijn`	`it's not an error`
`00:15.44`	`macTijn`	`it's a warning`
`00:15.48`	`xnon`	`and others errors that i cant understand`
`00:15.59`	`xnon`	`ok`
`00:16.04`	`macTijn`	`try googling`
`00:16.07`	`xnon`	`macTijn, look at this`
`00:16.10`	`xnon`	`Oct 22 02:07:17 WARNING[4883]: chan_iax2.c:9572 load_module: Unable to open IAX timing interface: No such file or directory`
`00:16.18`	`*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)`
`00:16.21`	`xnon`	`this warning it is important?`
`00:16.30`	`macTijn`	`same shit`
`00:17.06`	`xnon`	`:S`
`00:17.08`	`xnon`	`what?`
`00:17.10`	`xnon`	`jejejej`
`00:17.29`	`Dovid`	`warnings in asterisk are info - not soo much that ur gona crash`
`00:18.03`	`xnon`	`ok but why say this warning my console?`
`00:18.17`	`xnon`	`chan_iax2.c cant find it`
`00:18.27`	`xnon`	`where are this files?`
`00:20.03`	`hads`	`xnon: That warning is caused by the same thing as the other one.`
`00:20.11`	`*** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net)`
`00:21.03`	`xnon`	`ummmm ok`
`00:21.12`	`xnon`	`so isnt important really?`
`00:23.08`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`00:23.47`	`Defraz`	`I know this is asterisk but while I was here does anyone know the name of that song that there is just a bunch of laughing in. It is a newer song.`
`00:23.57`	`Defraz`	`I hear it on the radio but I never hear the dang name of it.`
`00:28.48`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
`00:31.33`	`*** join/#asterisk saftsack (n=oliver@p54A7FB13.dip.t-dialin.net)`
`00:36.16`	`saftsack`	`hi`
`00:43.43`	`*** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`00:44.02`	`teknoprep`	`i am at a loss... looking for some info on running high volume asterisk box in vmware`
`00:44.24`	`teknoprep`	`maby some pointers on keeping stutter and timing issues to a minimum`
`00:45.36`	`*** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it)`
`00:45.41`	`*** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`00:47.31`	`saftsack`	`teknoprep, do you have * running on an openwrt?`
`00:47.39`	`teknoprep`	`no`
`00:47.42`	`teknoprep`	`inside of vmware`
`00:47.48`	`saftsack`	`ok`
`00:47.59`	`saftsack`	`why that? do you have more than one asterisk, or waht?`
`00:48.05`	`teknoprep`	`nope`
`00:48.06`	`saftsack`	`i mean more than one in one computer?`
`00:48.48`	`teknoprep`	`no... i run all my servers except of course the file servers inside of vmware`
`00:48.55`	`tzafrir_laptop`	`Asterisk is generally not the ideal application to run inside vmware`
`00:48.57`	`teknoprep`	`everything from fireall's to active directory...`
`00:49.06`	`teknoprep`	`well i do not run anything except VoIP`
`00:49.21`	`tzafrir_laptop`	`Asterisk, and real-time application in general, need timely reaction from the CPU`
`00:49.35`	`saftsack`	`tzafrir_laptop, do you know anything about * on openwrt?`
`00:49.47`	`teknoprep`	`but i am starting to see there are a few problems... like the stuttering... which is pretty much the only application i am worried about`
`00:50.11`	`tzafrir_laptop`	`teknoprep, get a dedicated box. It will be just as good`
`00:50.29`	`tzafrir_laptop`	`Putting Asterisk on the same box assomething else is not good`
`00:51.37`	`tzafrir_laptop`	`hmmm... firewall inside vmware? hmmm ... so you trust both the vmware host setup and the unbreakability of the vmware virtualization ...`
`00:52.10`	`tzafrir_laptop`	`saftsack, sorry. NO`
`00:52.12`	`saftsack`	`ok`
`00:52.15`	`tzafrir_laptop`	`no, that is`
`00:52.43`	`saftsack`	`is it possible that asterisk strictly divorces the media from the other signals in sip?`
`00:53.22`	`*** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`00:55.31`	`teknoprep`	`tzafrir_laptop, the NSA wrote up a report on it.. that it can be trusted.. so yes i do also`
`00:57.15`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`00:58.10`	`tzafrir_laptop`	`teknoprep, but what does it give you?`
`00:58.57`	`teknoprep`	`tzafrir_laptop, what does what give me? vmware? well if you don't really know.. you should check itout... ease of backups, High Availability, Load Balancing, the list goes on`
`00:59.34`	`teknoprep`	`tzafrir_laptop, reverting to snapshots is easy too.. which is a very nice feature for planned upgrades that go bad`
`00:59.39`	`tzafrir_laptop`	`teknoprep, for special-purpose machines such as a firewall`
`00:59.57`	`teknoprep`	`tzafrir_laptop, but low CPU usage applications`
`01:01.39`	`teknoprep`	`tzafrir_laptop, lets agree to disagree... only thing i am agreeing with you on.. is that asterisk because of its audio and the human ear can detect the imprefections of audio.. i probably will need to put it on another box... BUT, vmware for most applications run them perfectly...`
`01:03.02`	`teknoprep`	`tzafrir_laptop, the only imperfections i can hear and/or anyone else with the setup i have now is transcoding IVR's / VoiceMail or any other pre-recorded applications... transcoding live audio sounds the same... its the only thing i am having a problem with and i don't know if it is becuase i store my OS on an iscsi target and i am adding a delay factor there or if it is the Virtualization proccess`
`01:04.23`	`teknoprep`	`hmmmm you know what.. at home i never hear the delay.. maby it is the storing of the OS on an iSCSI target... at home my IVR's or VM's never lag and i also run them inside of vmware.. but i run them on the localhost's array`
`01:04.50`	`tzafrir_laptop`	`teknoprep, the OS practically sits in the memory for the time of the operation. Sound files may be on a disk. So to isolate the problem, create a small ramdrivve and put the sound files in it (using loopback mounting)`
`01:05.11`	`tzafrir_laptop`	`I believe that this should eliminate most disk accesses`
`01:05.45`	`tzafrir_laptop`	`(a bit complex, but can help for testing)`
`01:06.00`	`teknoprep`	`naw i can get that done easy`
`01:06.46`	`teknoprep`	`why not just mount all VM and IVR recordings to a local RAM Disk ?`
`01:06.46`	`teknoprep`	`the directorys`
`01:06.46`	`teknoprep`	`that wouldn't be good on a failure tho`
`01:09.09`	`saftsack`	`does someone of you both have experience with patton products?`
`01:15.42`	`*** join/#asterisk drehlecom (n=user5@p54815020.dip.t-dialin.net)`
`01:25.01`	`*** join/#asterisk ESCulapio__ (n=ESCulapi@206stb68.codetel.net.do)`
`01:26.50`	`Defraz`	`I know this is asterisk but while I was here does anyone know the name of that song that there is just a bunch of laughing in. It is a newer song.`
`01:26.56`	`Defraz`	`anyone heard of a song like this?`
`01:27.00`	`Defraz`	`Driving me nuts haha`
`01:27.50`	`*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)`
`01:28.35`	`napkin`	`Defraz: don't worry, sounds like you're already nuts :)`
`01:28.46`	`Defraz`	`haha thanks`
`01:29.04`	`Defraz`	`Just heard it and I want to find it but I can't remember the names.`
`01:29.08`	`Defraz`	`It is so funny to me.`
`01:30.29`	`*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)`
`01:30.36`	`*** part/#asterisk drehlecom (n=user5@p54815020.dip.t-dialin.net)`
`01:38.37`	`SuPrSluG`	`saftsack:i did * on wrt`
`01:38.38`	`saftsack`	`SuPrSluG, hi`
`01:38.38`	`saftsack`	`did it work? :)`
`01:38.38`	`saftsack`	`can you tell me something about it?`
`01:39.50`	`SuPrSluG`	`yes. only issue is with voicemail storage. if you want to do it right get an asus geluxe w/ usb ports for storage. I tried nfs but it was a bit choppy ymmv`
`01:40.10`	`saftsack`	`sounds quite well :)`
`01:40.17`	`saftsack`	`what was with the performance?`
`01:40.26`	`saftsack`	`how many clients did you manage on your openwrt?`
`01:40.36`	`SuPrSluG`	`well i wouldn't go over 4-5 users`
`01:40.51`	`SuPrSluG`	`couple`
`01:40.59`	`saftsack`	`asus deluxe or what? :>`
`01:41.10`	`saftsack`	`but isnt asterisk to big for openwrt? ^^`
`01:41.12`	`SuPrSluG`	`they only have 200mz processors`
`01:41.14`	`saftsack`	`i mean for 6mb`
`01:41.20`	`SuPrSluG`	`no`
`01:41.49`	`SuPrSluG`	`you dump alot of non used modules`
`01:42.00`	`saftsack`	`sounds great :) but i dont know why i can just manage 5 users :> i mean asterisk doesnt do any voice processing does it do it?`
`01:42.44`	`SuPrSluG`	`if your using ulaw no. gsm or a compressed codec yes`
`01:43.22`	`saftsack`	`in the case that i use alaw i think its the same in processing as ulaw is it possible to handle about 15 people?`
`01:43.44`	`SuPrSluG`	`or i should say as long as * doesn't have to transcode`
`01:44.01`	`SuPrSluG`	`maybe`
`01:44.21`	`saftsack`	`sounds good so i will try this :)`
`01:44.23`	`SuPrSluG`	`i didn't stress test it`
`01:44.25`	`saftsack`	`do you know the fritz boxes?`
`01:44.46`	`SuPrSluG`	`no what ate they?`
`01:45.07`	`saftsack`	`this are boxes from a german manufactor which are newly supported`
`01:45.18`	`saftsack`	`i will try asterisk on it the next days i think`
`01:45.46`	`saftsack`	`this would be great if i have all my pbx on embedded devices which are working properly :)`
`01:46.04`	`SuPrSluG`	`you can always flash it back. but they do warn you that you can toast it`
`01:46.24`	`saftsack`	`hrhr`
`01:46.49`	`SuPrSluG`	`look at the asus deluxe router. it has 2 usb ports`
`01:47.01`	`*** join/#asterisk anthonyc (n=ac@adsl-75-31-162-88.dsl.frs2ca.sbcglobal.net)`
`01:47.23`	`napkin`	`can you guys assure me i'm not on crack, so i don't waste $300? i'm switching our office to voip, and we need 4 lines - that means we need 4 fxs ports right? :)`
`01:47.44`	`SuPrSluG`	`fxo`
`01:47.48`	`saftsack`	`ASUS WL-500g Deluxe`
`01:47.51`	`saftsack`	`do you mean this?`
`01:48.18`	`napkin`	`SuPrSluG: you're joking right?`
`01:48.33`	`SuPrSluG`	`that's the one they rave about. i did it on a linksys. nfs not great for voicemail`
`01:48.50`	`napkin`	`we're not getting any land-lines... everything will go over the internet`
`01:48.53`	`SuPrSluG`	`u have 4 incoming lines?`
`01:49.11`	`napkin`	`nah just 4 orgs who need different lines out`
`01:49.24`	`saftsack`	`sumasuma, rave? you mean this is the good one with usb ports?`
`01:49.28`	`*** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu)`
`01:49.36`	`b11d`	`hello`
`01:49.49`	`b11d`	`should I be wasting my time with an extensions.conf, or should I go with this "ael" stuff?`
`01:50.40`	`SuPrSluG`	`fxo modules connects to lines fxs connects to phones`
`01:51.06`	`napkin`	`k i just had to ask someone, even after reading about it, to make sure i wasn't surprised. :)`
`01:51.24`	`saftsack`	`b11d, what is ael?=`
`01:51.43`	`SuPrSluG`	`~ael`
`01:51.45`	`jbot`	`i guess ael is Asterisk Extension Language - a dialplan language with 'c like' syntax?`
`01:52.07`	`saftsack`	`are there any advantages why i should use ael?`
`01:52.15`	`SuPrSluG`	`so if your a programmer you might like it`
`01:52.49`	`b11d`	`so no one is really using it then?`
`01:53.23`	`napkin`	`b11d: well there are ~100 people in this channel, and 3 people answered.`
`01:54.12`	`b11d`	`wow, i totally didnt notice that.`
`01:54.21`	`b11d`	`thanks for "awakening" me`
`01:54.25`	`saftsack`	`here are exactly 244 people`
`01:54.28`	`SuPrSluG`	`i would imagine they are. took a hell of a lot of effort to write it, some somebody must use it`
`01:54.29`	`saftsack`	`including bots and so on`
`01:54.59`	`*** join/#asterisk TheMafia (n=TheMafia@74.135.181.228)`
`01:55.00`	`SuPrSluG`	`they're all out having a smoke`
`01:55.07`	`anthonyc`	`im eating a hamburger`
`01:55.19`	`TheMafia`	`is there some service that I can register my sip phone with?`
`01:55.35`	`SuPrSluG`	`free or pay?`
`01:55.51`	`saftsack`	`umts telephony support video calls, or?`
`01:55.57`	`TheMafia`	`free`
`01:56.06`	`SuPrSluG`	`free world dialup`
`01:56.24`	`saftsack`	`what is if i have a sip phone with a camera? are there gateways for calling a umts user with a video call?`
`01:56.44`	`TheMafia`	`yep, that was the one I was trying to remember, thanks`
`01:56.56`	`SuPrSluG`	`never tried video`
`01:57.17`	`SuPrSluG`	`anytime`
`01:57.34`	`saftsack`	`i go to bed now`
`01:57.40`	`saftsack`	`it is 3:59 here ^^`
`01:57.48`	`napkin`	`b11d: sorry, so many people join a channel, ask a question, and only wait a few minutes before jumping to the assumption that they won't be answered and leave...`
`01:58.49`	`b11d`	`yeah I know.. I hate that`
`01:58.57`	`b11d`	`its their loss though`
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`02:46.09`	`doolph`	`help`
`02:46.15`	`doolph`	`I am having this message after upgrade`
`02:46.16`	`doolph`	`Oct 21 21:44:56 VERBOSE[25104] logger.c: [cdr_pgsql.so]Oct 21 21:44:56 WARNING[25104] loader.c: libpq.so.4: cannot open shared object file: No such file or directory`
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`02:48.13`	`b11d`	`is anyone here familar with using MusicOnHold?`
`02:48.38`	`b11d`	`I can set the hold music I hear when someone else puts me on hold, but I'd like to define what music they hear when I put them on hold.. and I cant seem to do this with SetMusicOnHold`
`02:54.59`	`doolph`	`hi`
`02:55.25`	`b11d`	`hi`
`02:56.16`	`*** join/#asterisk Parvaresh (i=bartali@213.207.218.66)`
`02:57.08`	`Parvaresh`	`any recommendation on an embedded pc for asterisk?`
`02:57.18`	`Parvaresh`	`want to able to install full-length pci also on it`
`02:57.27`	`Parvaresh`	`as small and customize lookng as possible`
`02:58.02`	`napkin`	`hmmm soekris? should look it up.`
`02:58.03`	`SuPrSluG`	`doolph:do you have the postres module? if not go to /etc/asterisk/modules.conf and insert noload => cdr_pgsql.so`
`02:58.35`	`napkin`	`i guess soekris only does half-length`
`02:58.43`	`Parvaresh`	`yep`
`02:58.49`	`Parvaresh`	`and also not sure about its performance`
`02:59.02`	`napkin`	`well, full length pci and embedded might be a pipe dream?`
`02:59.07`	`Parvaresh`	`i am actually looking to make asterisk to a good looking hardware`
`02:59.25`	`napkin`	`does mini-itx have full length pci?`
`03:00.18`	`SuPrSluG`	`Parvaresh:http://www.x100p.com/products_5.htm`
`03:00.20`	`napkin`	`it's probably your best bet`
`03:00.46`	`Parvaresh`	`i hope you got my idea`
`03:01.00`	`Parvaresh`	`looking for something gives the customer more feeling of a PBX rather than a PC :)`
`03:02.29`	`napkin`	`there are probably some affordable mini-itx cases out there, too`
`03:02.37`	`napkin`	`that look less like big pcs`
`03:05.41`	`Parvaresh`	`expensive...`
`03:07.43`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`03:08.10`	`Defraz`	`Anyone happen to know the feature in Sendmail.mc file to deny mail from domains that don't exist?`
`03:09.45`	`Defraz`	`why not go with an external spa3000`
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`05:58.38`	`stephane_`	`jour`
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`07:23.18`	`vietasterisk`	`hi there`
`07:24.25`	`vietasterisk`	`hello , anyone worked with asterisk ACD hereeeeee`
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`07:46.44`	`bsdfreak`	`hi`
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`09:42.45`	`jgoo`	`hey guys, OT, but no network chan on here`
`09:43.28`	`jgoo`	`kinda on topic, as is about bandwidth, and asterisk. I have 4 adsl connections. I google for ADSL connection pooling, but found nothing.`
`09:44.13`	`jgoo`	`anyone have an idea how to pool these connections? or is it just some traffic shaping app to choose which connection to go through? (as I will have 4 ip's...)`
`09:46.17`	`jgoo`	`yeah... desperate measures`
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`09:48.39`	`qdk`	`jgoo: naah, it just always supprises me that novice ppl thinks that everyone on a IRC-channel are mind readers.`
`09:49.08`	`Qorky`	`can anyone help me with registering my g729 codecs please?`
`09:50.11`	`jgoo`	`qdk: mind readers how?`
`09:51.15`	`qdk`	`jgoo: dunno, havent meet anyone yet, but now that you are looking for one you could get back to me with an update.`
`09:51.19`	`Qorky`	`i can get the register program to run.. but just does nothing for a while then drops back to prompt.`
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`09:53.20`	`jgoo`	`qdk: I mean, why do you think that I think that everyone are mind readers :p`
`09:53.44`	`Qorky`	`can anyone help please?`
`09:56.39`	`qdk`	`jgoo: how many in here do you think knows about your hardware, OS and software?`
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`11:17.13`	`*** join/#asterisk Givur (n=mail@p54BCF0B6.dip.t-dialin.net)`
`11:20.39`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`11:20.41`	`hi365`	`vm emails not getteing sent. how do i trobleshoot?`
`11:21.19`	`ramtha`	`look in you mail.log`
`11:21.26`	`eliXier`	`sendmail is installed?`
`11:21.38`	`eliXier`	`mailcmd = sendmail -t ?`
`11:21.39`	`ramtha`	`do you have "attache mail" option aktivated?`
`11:26.45`	`hi365`	`ramtha: /var/log/maillog is blank`
`11:27.53`	`hi365`	`ramtha: 201 => 8899,xxxxxx,xxxxxxxxx@gmail.com,,attach=yes\|saycid=no\|envelope=no\|delete=yes`
`11:28.14`	`hi365`	`eliXier: yes`
`11:29.08`	`hi365`	`[root@asterisk1 ~]# service sendmail status`
`11:29.08`	`hi365`	`sendmail (pid 3098 3090) is running...`
`11:29.13`	`hi365`	`eliXier:`
`11:29.20`	`hi365`	`[root@asterisk1 ~]# service sendmail status`
`11:29.20`	`hi365`	`sendmail (pid 3098 3090) is running...`
`11:30.57`	`eliXier`	`try to test on ssh:`
`11:31.07`	`eliXier`	`sendmail -t your@mailaddy.com`
`11:31.09`	`eliXier`	`test`
`11:31.11`	`eliXier`	`.`
`11:31.21`	`TheMafia`	`Is the rtp the acutal audio in a sip to sip call?`
`11:31.26`	`eliXier`	`show if you receive the mail`
`11:31.27`	`TheMafia`	`5060 is the control right?`
`11:32.23`	`*** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net)`
`11:32.25`	`*** join/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it)`
`11:33.16`	`hi365`	`eliXier: sendmail -t nachmic@gmail.com test`
`11:33.23`	`hi365`	`like that?`
`11:33.28`	`*** part/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it)`
`11:33.46`	`hi365`	`it sort of hangs. how long should it take?`
`11:34.11`	`eliXier`	`nope`
`11:34.13`	`eliXier`	`start:`
`11:34.22`	`eliXier`	`sendmail -t xxxx@gmail.com`
`11:34.23`	`eliXier`	`test`
`11:34.24`	`eliXier`	`.`
`11:34.27`	`eliXier`	`end`
`11:34.52`	`qdk`	`TheMafia: yes, RTP carries the media and SIP is the signaling.`
`11:35.29`	`*** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com)`
`11:35.43`	`Mattwj2005`	`hey guys :)`
`11:36.04`	`TheMafia`	`qdk what is the standard port/range for rtp?`
`11:36.24`	`hi365`	`eliXier: no email recived!`
`11:36.26`	`Mattwj2005`	`I believe the range for sip is 10000-20000`
`11:36.51`	`eliXier`	`now, it's a problem of sendmail, not asterisk`
`11:37.12`	`qdk`	`TheMafia: whatever you pick.`
`11:37.17`	`TheMafia`	`thanks`
`11:37.31`	`hi365`	`eliXier: good point. any advice anyway?`
`11:38.06`	`eliXier`	`hi365: reinstall/reconfigure sendmail again... the standard-configuration of sendmail is ok`
`11:38.19`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`11:38.36`	`hi365`	`eliXier: thanks ill try`
`11:38.36`	`eliXier`	`anotherway.. gmail has a spamfilter?`
`11:38.46`	`eliXier`	`np`
`11:39.03`	`hi365`	`it has worked in the past so i dont think thats the problem`
`11:39.17`	`eliXier`	`ok`
`11:40.47`	`coppice`	`anyone around who has used the codec simply called ADPCM in asterisk?`
`11:41.03`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`11:45.28`	`hi365`	`eliXier: /var/log/sendmail here. although its not asterisk, would please have a look? http://pastebin.ca/215099`
`11:46.05`	`TheMafia`	`I have a vpn connection between two networks with sip configured on both sides, phones ring as should however there is no audio, should I have to do anything netowrk related since I am using a vpn?`
`11:47.40`	`*** join/#asterisk flot (n=flot@87.251.134.36)`
`11:49.40`	`flot`	`hi all. I use asterisk 1.4 (SVN) In last version NOT WORKING transfer. Say: [Oct 22 15:42:16] WARNING[4718]: res_features.c:771 builtin_atxfer: Did not read data.`
`11:49.40`	`flot`	`<PROTECTED>`
`11:50.29`	`ramtha`	`TheMafia: what vpn you use? ipsec?`
`11:51.09`	`ramtha`	`TheMafia: sounds that signaling goes over the vpn tunnel but the rtp goes directly to the endpoint ips?`
`11:51.28`	`TheMafia`	`yes ipsec`
`11:51.45`	`ramtha`	`use tcpdump to see where the paces go`
`11:51.50`	`ramtha`	`or etherreal`
`11:52.10`	`TheMafia`	`192.168.15.11.6004 > 192.168.0.30.8000 is that a typical rtp port or is there no range at all?`
`11:52.36`	`ramtha`	`you can set rtp ports in rtp.conf of asterisk`
`11:52.39`	`ramtha`	`8000 is typical`
`11:52.53`	`ramtha`	`and you see the error?`
`11:53.23`	`ramtha`	`can you ping from 192.168.15.11 to 192.168.0.30?`
`11:54.19`	`TheMafia`	`yes I can`
`11:54.48`	`ramtha`	`no look where 192.168.0.30 send its rtp stream`
`11:56.57`	`TheMafia`	`192.168.0.30 is a softphone an I have no mic setup so I assumed there was no stream to send, I cannot hear any audio from 192.168.15.11, I am duiling a fax machine`
`11:57.51`	`*** join/#asterisk mrg82 (n=na@dsl82-163-126-23.as15444.net)`
`12:00.49`	`TheMafia`	`I can plug in my grandstream phone but it seems to do the same thing and I can see traffic in both directions then`
`12:02.17`	`*** join/#asterisk brif8 (n=brif8@67.78.24.178)`
`12:03.57`	`brif8`	`Hi All, reading I see that a "SIP Proxy Server can request to stay in the communication path" how is this done within the * env. Can phone A be set for call1 to be in the loop and call2 out of the loop or what ??`
`12:05.23`	`*** join/#asterisk AbsTradELic (n=vldmr@201.79.188.89)`
`12:06.26`	`AbsTradELic`	`good times for all !`
`12:06.39`	`AbsTradELic`	`X:)`
`12:06.55`	`AbsTradELic`	`SuPrSluG: hi !`
`12:08.10`	`TheMafia`	`when I dump the grandstream sip phone this is what I see over and over`
`12:08.12`	`TheMafia`	`07:07:35.001548 IP 192.168.15.11.6004 > 192.168.0.185.5004: UDP, length: 172`
`12:08.32`	`TheMafia`	`07:07:34.972617 IP 192.168.0.185.5004 > 192.168.15.11.6004: UDP, length: 172`
`12:08.44`	`TheMafia`	`Is that rtp to and from my grandstream?`
`12:12.47`	`festr__`	`rtp`
`12:12.56`	`festr__`	`try to use tethereal`
`12:14.16`	`*** join/#asterisk DataCompBoy (n=datacomp@217.151.230.182)`
`12:14.19`	`TheMafia`	`I can't from there, so if the rtp is getting to and from why would there be no audio?`
`12:14.40`	`DataCompBoy`	`Hi everybody! Is digium.com down? Or that only for me?`
`12:15.26`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`12:17.05`	`TheMafia`	`I am confident that the codecs will mate up`
`12:18.22`	`*** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)`
`12:18.26`	`*** part/#asterisk brif8 (n=brif8@67.78.24.178)`
`12:20.59`	`*** join/#asterisk wl0 (n=ge@88-196-80-95-dsl.krw.estpak.ee)`
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`12:25.20`	`hi365`	`DataCompBoy: same here`
`12:33.14`	`AbsTradELic`	`i have all packages installed and loaded now!`
`12:33.23`	`AbsTradELic`	`is allright !`
`12:33.36`	`AbsTradELic`	`i'm a beginner`
`12:33.50`	`AbsTradELic`	`and now I'll adjust the /etc/asterisk files`
`12:35.26`	`AbsTradELic`	`I didn't have nothing more that my computer and softphone x-lite installed... no voip hardwares`
`12:36.03`	`DataCompBoy`	`edit /etc/sip.conf and /etc/extensions.ael or /etc/extensions.conf`
`12:36.18`	`DataCompBoy`	`i mean, {edit (/etc/sip.conf and (/etc/extensions.ael or /etc/extensions.conf))}`
`12:36.37`	`AbsTradELic`	`hum... ok`
`12:36.50`	`AbsTradELic`	`wait a few`
`12:38.22`	`*** join/#asterisk axscode (n=axscode@124.217.42.206)`
`12:39.32`	`AbsTradELic`	`DataCompBoy: ok... I view the files on /etc/asterisk`
`12:40.35`	`DataCompBoy`	`edit them :)`
`12:41.01`	`DataCompBoy`	`carefully read comments and asterisk pages @ http://www.voip-info.org/wiki-Asterisk`
`12:45.13`	`*** join/#asterisk Shedoks (n=nn@82.117.206.193)`
`12:45.18`	`Shedoks`	`hi`
`12:45.47`	`Shedoks`	`i need a little help for asterisk`
`12:46.35`	`DataCompBoy`	`well... try to ask`
`12:49.57`	`Shedoks`	`:)`
`12:50.41`	`Shedoks`	`ok i want to make asterix work just over the internet right now, i don't need voip phones or connecting with the fixed telephony`
`12:50.42`	`DataCompBoy`	`... welll?`
`12:50.49`	`Shedoks`	`what i s confusing me is`
`12:51.08`	`Shedoks`	`do i have to make sip.conf for every user i made ?`
`12:51.24`	`Shedoks`	`or i can put every user into [username] brackets`
`12:51.56`	`DataCompBoy`	`you put all your users (if you have not much users) in sip.conf`
`12:52.07`	`DataCompBoy`	`every user info after [username]`
`12:52.50`	`DataCompBoy`	`define at least type, username, secret or md5secret, host and nat`
`12:53.42`	`DataCompBoy`	`(hm... nat can be global, if all your users possible behind nat)`
`12:54.09`	`Shedoks`	`well for testing purposes it will be`
`12:54.21`	`Shedoks`	`but how do i make works with nat`
`12:54.34`	`DataCompBoy`	`for testing just put several blocks to sip.conf and bee happy`
`12:54.39`	`Shedoks`	`becouse asterisk is behind nat and i would like to use it globaly later`
`12:54.54`	`DataCompBoy`	`asterisk behind nat? how -- did you port forward UDP to it?`
`12:55.19`	`Shedoks`	`well i plan to do that`
`12:55.24`	`Shedoks`	`which one 5060?`
`12:55.27`	`Shedoks`	`or 8000`
`12:55.29`	`DataCompBoy`	`All :)))`
`12:55.55`	`DataCompBoy`	`the best is forward all UDP to asterisk, but will be success only 5060 and RTP ports defined in rtp.conf`
`12:56.39`	`Shedoks`	`ok thanks :)`
`12:56.44`	`Shedoks`	`just one more thing`
`12:56.51`	`Shedoks`	`how to add user trought CLI`
`12:57.01`	`DataCompBoy`	`no way`
`12:57.16`	`DataCompBoy`	`edit sip.conf and do in cli 'sip reload'`
`12:57.41`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.21)`
`12:57.57`	`Shedoks`	`thanks man :)`
`12:58.34`	`robin_sz`	`meep?`
`12:59.14`	`DataCompBoy`	`robin_sz: meep?!`
`12:59.39`	`robin_sz`	`so, this asterisk thing .. and the CLI. Did one of you steal my "Dial" command?`
`13:00.34`	`robin_sz`	`I dont think I did ./configure --dont-bother-with-a-dail-command-for-CLI`
`13:01.22`	`robin_sz`	`does it omit the CLI dial command from compile of it cant findany suitable local audio hardware?`
`13:01.53`	`DataCompBoy`	`robin_sz: nope, it compile dial command always, since it can dial on any channel`
`13:01.56`	`DataCompBoy`	`even on LOCAL/`
`13:02.20`	`robin_sz`	`DataCompBoy, ok, so what do I have to do to enable it then?`
`13:02.54`	`TheMafia`	`If I have determined that rtp data is getting to and from both sides, and I still only get rings etc on the phone and no audio, what else shoudl I look at?`
`13:03.20`	`DataCompBoy`	`robin_sz: look at build log, and see what problem was when it tried to compile`
`13:03.49`	`robin_sz`	`I dont thin I had any problems with the compile at all ...`
`13:05.00`	`DataCompBoy`	`TheMafia: try to add nat=yes to phone configuration`
`13:05.00`	`DataCompBoy`	`TheMafia: i have saw that some times`
`13:05.00`	`DataCompBoy`	`robin_sz: look at build tree, is dial.o there? :)`
`13:05.02`	`DataCompBoy`	`robin_sz: what os?`
`13:05.09`	`robin_sz`	`Linux, obviously ;)`
`13:05.15`	`Rhizome`	`Anyone know why DTMF wouldnt work calling from asterisk to asterisk to PSTN? even with IAX it doesnt work.`
`13:05.19`	`DataCompBoy`	`robin_sz: distro?`
`13:05.28`	`DataCompBoy`	`robin_sz: what asterisk sources?`
`13:05.31`	`robin_sz`	`debian`
`13:05.38`	`robin_sz`	`latest bristuffed`
`13:05.48`	`DataCompBoy`	`robin_sz: how you build? dpkg-buildpackage ?`
`13:06.03`	`robin_sz`	`but its the same with other variants eg 1.2.9.1`
`13:06.08`	`robin_sz`	`or 1.2.7`
`13:06.26`	`robin_sz`	`obvioulsy the Dial() in extensions.conf works fine`
`13:06.32`	`robin_sz`	`its just from the CLI ..`
`13:06.37`	`DataCompBoy`	`robin_sz: Stop.`
`13:06.45`	`DataCompBoy`	`robin_sz: there NO cli command dial() !`
`13:07.12`	`robin_sz`	`you certain?`
`13:07.24`	`DataCompBoy`	`there Dial() command from cli here only for chan_oss :)`
`13:07.34`	`DataCompBoy`	`when you use soundcard of asterisk server.`
`13:07.42`	`robin_sz`	`so ...`
`13:08.02`	`robin_sz`	`when I said:`
`13:08.04`	`robin_sz`	`does it omit the CLI dial command from compile of it cant findany suitable local audio hardware?`
`13:08.07`	`robin_sz`	`the answer was:`
`13:08.26`	`robin_sz`	`"yeah thats right, if theres no local audio hardware, you cant dial from the CLI"`
`13:08.31`	`robin_sz`	`???`
`13:08.32`	`DataCompBoy`	`robin_sz: there CLI dial() command is non-standard, and only suitable for chan_oss.`
`13:09.17`	`DataCompBoy`	`if you want to initiate call, read http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out or http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate`
`13:09.32`	`robin_sz`	`OK`
`13:10.09`	`robin_sz`	`hmmm`
`13:10.16`	`robin_sz`	`neither of those seem useful`
`13:10.49`	`DataCompBoy`	`you are not right -- last one very useful and practical.`
`13:11.25`	`DataCompBoy`	`never tried first, since always use AMI :)`
`13:11.42`	`robin_sz`	`uses the manager API,`
`13:12.01`	`*** join/#asterisk ukh (i=ukh@visionary.svansen.se)`
`13:12.04`	`robin_sz`	`so, from the CLI I type what ?`
`13:13.06`	`wl0`	`hi, advice me any cheap but stable and quality sip phones for work with asterisk`
`13:13.09`	`wl0`	`please`
`13:13.23`	`DataCompBoy`	`robin_sz: what you want to do? how do you imagine your doints?`
`13:13.54`	`robin_sz`	`normally, in testing, I just type dial xxxx@<context> and watch the progress of the call, so I can see what happens from the CLI, watch timeouts. routings etc`
`13:14.04`	`robin_sz`	`I dont need any audio`
`13:14.24`	`robin_sz`	`I think the other boxes I have used must have had oss support by accident`
`13:15.12`	`robin_sz`	`the manager API is nto really an option, as the things you can connect to it all take a day or more to configure from what ive seen so far.`
`13:15.26`	`robin_sz`	`I just need a simple way of initiating a call and monitoring the progress`
`13:16.20`	`DataCompBoy`	`robin_sz: so, you want to dial xxxx@<context>, context/exten/prio right?`
`13:16.28`	`robin_sz`	`uh uhuh`
`13:16.57`	`DataCompBoy`	`robin_sz: you can just create test file as Asterisk+auto-dial+out. thats easy`
`13:18.08`	`TheMafia`	`is port 5060 that is used for control udp or tcp?`
`13:18.35`	`DataCompBoy`	`TheMafia: 5060 udp is control port, and 10000-20000 (by default) are RTP ports.`
`13:18.46`	`TheMafia`	`thanks`
`13:20.17`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
`13:20.37`	`robin_sz`	`hmmm ...`
`13:21.16`	`robin_sz`	`I guess I'll need some sort of script then to create call files for the various extensions and contexts`
`13:21.50`	`robin_sz`	`ah, stuff it, I'll just go and buy a sound card tomorow and stick it in`
`13:22.22`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
`13:26.23`	`Druken`	`robin_sz: what do you need an audio card for?`
`13:27.06`	`robin_sz`	`Druken, to test extensions.conf`
`13:27.17`	`DataCompBoy`	`robin_sz: why not just run oss for empty ? :) you not need hardware to run it.`
`13:27.38`	`robin_sz`	`explain that sentence please`
`13:27.45`	`robin_sz`	`why not just run oss for empty`
`13:27.48`	`robin_sz`	`??`
`13:27.52`	`DataCompBoy`	`robin_sz: also, it's pretty easy to write Dial() app that will do connect of origitated and local extension`
`13:28.14`	`DataCompBoy`	`robin_sz: you don't need hardware to run OSS mixer.`
`13:28.17`	`DataCompBoy`	`and chan_oss too`
`13:28.23`	`Druken`	`robin_sz: why not just use a phone?`
`13:28.48`	`DataCompBoy`	`robin_sz: also, Druken right, why not use simple softphone`
`13:29.00`	`robin_sz`	`Druken: I thought phones where restricted in context?`
`13:29.36`	`Druken`	`uhmm... asterisk is a telephone system... no??`
`13:29.45`	`Druken`	`how do you think everyone else will access the system ?`
`13:29.53`	`robin_sz`	`so can I call xxxx@incoming and it will originate a call and place it in the incoming context, as it it had come from the PSTN?`
`13:31.00`	`robin_sz`	`Druken, this is called "testing" ... before it goes live, I like to test ... normally I do stuff like "dial xxxx@incoming" and watch how the call prgresses through the timeouts etc`
`13:31.09`	`Druken`	`make an extension, something like exten => 987,1,goto(xxx,local,1)`
`13:31.16`	`robin_sz`	`make sure the logic is correct`
`13:31.38`	`DataCompBoy`	`robin_sz: do, you already used Dial cli command?`
`13:31.49`	`robin_sz`	`DataCompBoy, on other systems yes, all the time`
`13:31.50`	`DataCompBoy`	`robin_sz: why then not load chan_oss and use it now ?`
`13:32.03`	`robin_sz`	`because I dont understand that statement?`
`13:32.10`	`Druken`	`testing is usually a good idea... me, i edit my system WHILE it's live.. hehehe thankfully i'm very good at logic`
`13:33.17`	`DataCompBoy`	`robin_sz: load chan_oss, and you will be happy`
`13:33.48`	`robin_sz`	`modules/chan_oss: cannot open shared object file: No such file or directory`
`13:33.54`	`robin_sz`	`next?`
`13:34.14`	`DataCompBoy`	`hm. so, you have not build it?`
`13:34.28`	`robin_sz`	`probably not ...`
`13:34.34`	`robin_sz`	`no sound support on this machine ..`
`13:34.35`	`DataCompBoy`	`then, build it! :)`
`13:34.45`	`DataCompBoy`	`install liboss, that enough`
`13:34.50`	`robin_sz`	`ahhh`
`13:34.57`	`robin_sz`	`now you begin to make sense :)`
`13:37.47`	`robin_sz`	`libossp-sa12 - Abstraction library for the Unix socket API`
`13:37.51`	`robin_sz`	`that one?`
`13:38.15`	`DataCompBoy`	`nope`
`13:38.41`	`robin_sz`	`libossp-uuid12 - OSSP uuid ISO-C - shared library ??`
`13:39.03`	`DataCompBoy`	`install audiooss`
`13:39.13`	`DataCompBoy`	`(if you talk about debian)`
`13:39.38`	`robin_sz`	`i do ...`
`13:39.43`	`robin_sz`	`they hide things`
`13:39.45`	`robin_sz`	`:)`
`13:40.46`	`robin_sz`	`obviously, they dotn let that package show up for a search of liboss .. that wwould be too easy :)`
`13:40.46`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`13:44.36`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
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`13:48.50`	`TheMafia`	`i can see traffic traversing the nat on the two rtp ports defined on both ends, however there is still only rings and no audio. What shoudl I look at? if a codec cannot be negotiated the ring would or would not take place?`
`13:49.25`	`DarKnesS_WolF`	`TheMafia: in the sip.conf canreinvite=no`
`13:49.56`	`DarKnesS_WolF`	`TheMafia: for each extention`
`13:50.10`	`TheMafia`	`pcmu is the same as g.711 right?`
`13:55.19`	`DarKnesS_WolF`	`i'm not sure about that i know that ulaw =G.711U`
`13:55.47`	`Qwell`	`TheMafia: yes, I'm pretty sure it is`
`13:58.53`	`DarKnesS_WolF`	`Qwell: so pcmu is g.711 and it is ulaw and alaw?`
`13:59.06`	`Qwell`	`no`
`13:59.10`	`TheMafia`	`pcma is a law g.711 I think`
`13:59.22`	`Qwell`	`g711u is ulaw is pcmu, g711a is alaw is pcma`
`13:59.36`	`DarKnesS_WolF`	`ah goti t`
`13:59.39`	`DarKnesS_WolF`	`thx Qwell :-)`
`14:01.12`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net)`
`14:06.39`	`DarKnesS_WolF`	`what is the best asterisk billing system ?`
`14:09.18`	`*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)`
`14:09.30`	`Qwell`	`~best`
`14:09.31`	`jbot`	`best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes!`
`14:09.45`	`*** join/#asterisk Seggy (i=rbutler@tsss.org)`
`14:09.50`	`*** join/#asterisk Inez (i=faceoff@devel4.net)`
`14:09.53`	`Inez`	`Hi`
`14:09.59`	`hegemoOn`	`plop`
`14:10.37`	`Inez`	`Is it good idea to use app_sql_postgres to receive data from pg database in dialplan or maybe better way is to use AGI and perl/python script?`
`14:14.00`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-169-13.buff.east.verizon.net)`
`14:14.04`	`Qwell`	`Inez: use func_odbc`
`14:14.20`	`SuPrSluG`	`hell`
`14:14.23`	`SuPrSluG`	`hello`
`14:14.47`	`SuPrSluG`	`although i am in zaptel hell right now`
`14:15.50`	`DarKnesS_WolF`	`SuPrSluG: TDM400P ?`
`14:15.57`	`SuPrSluG`	`i keep getting a message when trying to call zt_handle_event: Ring/Off-hook in strange state 6 on channel 4`
`14:16.00`	`SuPrSluG`	`yep`
`14:16.34`	`Inez`	`Qwell but its only possible to select one row and one field?`
`14:16.46`	`Qwell`	`Inez: one row, as many fields as you tell it`
`14:18.09`	`DarKnesS_WolF`	`SuPrSluG: try from command like ztcfg -vvvvvvv`
`14:18.37`	`Inez`	`Qwell how, I cant find it on http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc Is it standard in asterisk 1.2?`
`14:18.45`	`Qwell`	`no`
`14:19.00`	`SuPrSluG`	`i look in dmesg and proc see this http://pastebin.ca/215241`
`14:19.06`	`Qwell`	`there is a backport of it though on svncommunity.digium.com`
`14:20.35`	`DarKnesS_WolF`	`SuPrSluG: what is the output from ztcfg -vvvv ?`
`14:21.12`	`SuPrSluG`	`i have a single fxo and a tdm04b i removed the wcfxo module but it want to keep the 1st port`
`14:22.03`	`Inez`	`Qwell I should configure sql resource in res_config_odbc.conf or res_odbc.conf, right?`
`14:22.14`	`AbsTradELic`	`SuPrSluG: Hi!`
`14:22.19`	`SuPrSluG`	`hi`
`14:22.26`	`AbsTradELic`	`X:)`
`14:23.07`	`Druken`	`func_odbc is FUN :)`
`14:23.15`	`SuPrSluG`	`DarKnesS_WolF:http://pastebin.ca/215250`
`14:23.30`	`Druken`	`i gave up on it, and use realtime....`
`14:23.54`	`*** join/#asterisk Vahram (n=VX@client-231-114.xter.net)`
`14:24.19`	`SuPrSluG`	`as you can see from proc it still thinks the single fxo port is 1 and using only the 1st 3 on the tdv04b board`
`14:25.42`	`DarKnesS_WolF`	`SuPrSluG: u have 4 FXO ports ?`
`14:25.53`	`*** join/#asterisk ESCulapio__ (n=ESCulapi@188stb68.codetel.net.do)`
`14:26.51`	`SuPrSluG`	`yes 1 single and a 1 4 port. it won't let go of the single and zap channel 1`
`14:28.13`	`SuPrSluG`	`so a total of 5 fxo. no line is plugged into the single.`
`14:28.40`	`DarKnesS_WolF`	`5 FXO !?`
`14:28.54`	`SuPrSluG`	`yes`
`14:28.58`	`DarKnesS_WolF`	`how come ?`
`14:29.15`	`DarKnesS_WolF`	`i know that the card has 4 places for modules`
`14:29.52`	`Inez`	`i have problem with svncommunity.digium.com, is it work ok for you? for making co`
`14:30.34`	`SuPrSluG`	`2 cards a single and a 4 port`
`14:30.57`	`DarKnesS_WolF`	`ah so u have 2 cards`
`14:31.01`	`DarKnesS_WolF`	`hm`
`14:31.01`	`SuPrSluG`	`yes`
`14:32.07`	`DarKnesS_WolF`	`SuPrSluG: ok check ur /etc/zaptel.conf`
`14:32.08`	`SuPrSluG`	`and the single is grabbing zap channel 1 and i don't want it to`
`14:32.28`	`SuPrSluG`	`want me to post on pastebin?`
`14:34.15`	`AbsTradELic`	`ok`
`14:34.47`	`AbsTradELic`	`SuPrSluG: I have dificulties to understand english!`
`14:34.49`	`ManxPower`	`SuPrSluG: what specific cards do you have?`
`14:34.53`	`AbsTradELic`	`X;)`
`14:35.06`	`AbsTradELic`	`ok`
`14:35.08`	`ManxPower`	`and what modules are on those cards?`
`14:35.14`	`SuPrSluG`	`generic clone and tdm04b`
`14:35.33`	`SuPrSluG`	`all fxo`
`14:35.49`	`ManxPower`	`SuPrSluG: so a Generic X100P and a TDM400P with 4xFXO (RED) modules?`
`14:36.14`	`AbsTradELic`	`can I put only one ip addres on sip.conf like this: 'localnet=192.168.1.1/255.255.255.0' ?`
`14:36.40`	`AbsTradELic`	`its correct ?`
`14:36.41`	`Qwell`	`AbsTradELic: That would be technically invalid`
`14:37.04`	`Qwell`	`192.168.1.0/255.255.255.0 or for just one single IP, 192.168.1.1/255.255.255.255`
`14:37.36`	`SuPrSluG`	`the generic is in red (has no line attached) and TDM04B has 4 lines attached. the generic won't give up zap channel 1`
`14:37.54`	`ManxPower`	`SuPrSluG: define "won't give up"`
`14:38.13`	`AbsTradELic`	`Qwell: ok... the asterisk work like a server ?`
`14:38.21`	`Qwell`	`huh?`
`14:38.27`	`*** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz)`
`14:38.37`	`asdx`	`lol`
`14:39.06`	`DarKnesS_WolF`	`Qwell: i think AbsTradELic means it can run as an init.d`
`14:39.12`	`ManxPower`	`SuPrSluG: also what is the order of the drivers loading? wcfxo wctdm or wctdm wcfxo ?`
`14:39.13`	`DarKnesS_WolF`	`AbsTradELic: yes u can run it like mysql`
`14:39.16`	`SuPrSluG`	`so if i pull the wcfxo module and ztcfg -v it says 4 channels configured. i look in proc and http://pastebin.ca/215241`
`14:40.08`	`AbsTradELic`	`Qwell: in one local network, I just need to put asterisk only one machine and be usefull to others computers clients ?`
`14:40.13`	`SuPrSluG`	`do i need to physically pull the single fxo to correct this?`
`14:40.24`	`ManxPower`	`SuPrSluG: you should not need to.`
`14:40.50`	`ManxPower`	`RED in the output means "no line connected"`
`14:40.59`	`AbsTradELic`	`ok`
`14:41.36`	`ManxPower`	`SuPrSluG: I still don't understand that the PROBLEM is. Can you not dial out, can you not receive calls, etc?`
`14:41.37`	`SuPrSluG`	`ztcfg will configure 5 channels. when i zap show channel 1 it is alway in alarm`
`14:41.55`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`14:41.56`	`ManxPower`	`SuPrSluG: correct. if a phone line is not plugged into the X100P it will show an alarm.`
`14:42.08`	`SuPrSluG`	`can't receive`
`14:42.18`	`ManxPower`	`you should not be alarmed by an alarm if you are expecting an alarm.`
`14:42.45`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
`14:42.48`	`ManxPower`	`SuPrSluG: so someone calls in on a line connected to channel 2 (port 1 on the TDM400P) and nothing shows up on the console?`
`14:43.18`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`14:43.39`	`SuPrSluG`	`yes. but when i configure it to use only the tdm04b it shows zap channel 1 in red alarm and there is a line plugged in`
`14:44.00`	`ManxPower`	`SuPrSluG: then the line is not working. Can you plug a normal analog phone into that line and get a dialltone?`
`14:44.43`	`Inez`	`Qwell can you tell me how to select two vwo fields from database (in one row).`
`14:45.07`	`SuPrSluG`	`here's what shows Executing BackGround("Zap/4-1", "night-greet-eea") in new stack`
`14:45.08`	`Qwell`	`select foo, bar from table where baz='blah'`
`14:45.09`	`SuPrSluG`	`<PROTECTED>`
`14:45.10`	`SuPrSluG`	`<PROTECTED>`
`14:45.37`	`ManxPower`	`SuPrSluG: that is channel FOUR, you were talking about chanel ONE`
`14:46.10`	`ManxPower`	`(09:43:21) ManxPower: SuPrSluG: then the line is not working. Can you plug a normal analog phone into the line plugged into port ONE and get a dialltone?`
`14:46.18`	`SuPrSluG`	`you can't hear the message and when you hangup it continues`
`14:46.28`	`ManxPower`	`the ring/offhook message is normal and not a problem.`
`14:46.36`	`ManxPower`	`SuPrSluG: what country are you in>?`
`14:47.03`	`Inez`	`Qwell but how to using func_odbc`
`14:47.13`	`ManxPower`	`I'm sorry, but I cannot help you. You are trying to diagnose the issues, but not following my advice or questions.`
`14:49.37`	`SuPrSluG`	`sorry that's one of the other numbers. when i call channel one it just rings. unless i configure the single card and 5 channels the zap channel 2 gets the same message`
`14:49.47`	`SuPrSluG`	`us`
`14:50.02`	`SuPrSluG`	`sorry i had to ssh back in`
`14:51.22`	`SuPrSluG`	`Manxpower i'm not phyically there. Although i can call to get access to the building`
`14:51.33`	`*** join/#asterisk kilobit2001 (n=locid@206-248-134-27.dsl.teksavvy.com)`
`14:52.15`	`ManxPower`	`you have a wiring problem. You need to be there to diagnose it. bring an analog phone so you can test the lines outside of Asterisk,`
`14:53.23`	`SuPrSluG`	`we have a butt set and anolog phones are attached to a splitter for emergencies`
`14:55.11`	`SuPrSluG`	`why would it try to set rbs . in dmesg i get this -> zt_rbs: Tried to set RBS hook state 0 on channel WCFXO/0/0 while span WCFXO/0 lacks rbsbits or hooksig function`
`14:55.25`	`ManxPower`	`nod Anytime you see a RED Alarm you can assume that means "I don't see the voltage I would expect on a phone line"`
`14:55.40`	`SuPrSluG`	`robbed bit is for T's right?`
`14:56.00`	`ManxPower`	`SuPrSluG: I would assume the driver tries to set the RBS bits on all Digium cards, but only T-1/E-1 cards support that function.`
`14:56.09`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`14:56.15`	`kilobit2001`	`hi`
`14:56.23`	`ManxPower`	`put your /etc/zaptel.conf on pastebin and I'll glance at it.`
`14:56.27`	`kilobit2001`	`does asterisk support dial by name?`
`14:56.40`	`SuPrSluG`	`so that'll affect all the other numbers too?`
`14:56.45`	`ManxPower`	`kilobit2001: voicemail does.`
`14:56.50`	`ManxPower`	`as does the Directory() app`
`14:57.05`	`ManxPower`	`SuPrSluG: I don't believe the RBS message is an issue.`
`14:58.16`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`14:58.45`	`kilobit2001`	`is the following valid: _xxxxxxxxxx,1,goto(${exten},1,1)`
`14:59.35`	`*** join/#asterisk saftsack (n=oliver@p54A7F8DD.dip.t-dialin.net)`
`15:01.34`	`ManxPower`	`only if your context is ${EXTEN}`
`15:01.55`	`ManxPower`	`The format is Goto([context],[extenation],priority)`
`15:02.09`	`ManxPower`	`context and extenstion are optional, the priority is not.`
`15:03.11`	`*** join/#asterisk af_ (n=af@ip-173-17.sn1.eutelia.it)`
`15:04.41`	`ManxPower`	`kilobit2001: of course _xxxxxxxxxx,1,goto(${exten},1) would cause a loop and prolly make Asterisk crash`
`15:05.19`	`kilobit2001`	`is there a limit on the amount of contexts allowed?`
`15:05.34`	`ManxPower`	`kilobit2001: I don't think so,`
`15:05.55`	`ManxPower`	`there really isn't any need to have zillions of contexts`
`15:05.58`	`kilobit2001`	`in this setup, there'll be a jump to a different context, based on the number dialed.`
`15:06.17`	`ManxPower`	`kilobit2001: why?`
`15:06.37`	`kilobit2001`	`manx-- how would you implement a multiple menu ivr system?`
`15:07.10`	`kilobit2001`	`lets say 5 users, each with 4 nested menus.`
`15:08.52`	`*** join/#asterisk Rhizome (n=rhizome@rhizome.boldlygoingnowhere.org) [NETSPLIT VICTIM]`
`15:08.52`	`*** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM]`
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`15:08.57`	`kilobit2001`	`anyway to do this within one context?`
`15:09.13`	`ManxPower`	`5 users?`
`15:09.24`	`ManxPower`	`I understand the 4 nested menus.`
`15:09.57`	`TheMafia`	`i can see traffic traversing the nat on the two rtp ports defined on both ends, however there is still only rings and no audio. What shoudl I look at? if a codec cannot be negotiated the ring would or would not take place? I am not using asterisk, but I thought that this would be a common problem`
`15:10.36`	`ManxPower`	`TheMafia: it's not a common problem`
`15:10.44`	`kilobit2001`	`5 users each with their own menus.. incoming calls are jumped to different context based on dialed numbers...`
`15:10.47`	`ManxPower`	`it happens with Asterisk, but for a variety of reasons`
`15:11.07`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`15:12.49`	`DarKnesS_WolF`	`TheMafia: i told u`
`15:12.54`	`DarKnesS_WolF`	`canreinvite= no`
`15:13.04`	`DarKnesS_WolF`	`TheMafia: server and the 2 nodes in same lan ?`
`15:13.12`	`ManxPower`	`kilobit2001: so, you mean something like this pastebin. [incoming] is where calls from the PSTN come into. http://pastebin.ca/215297`
`15:13.25`	`ManxPower`	`DarKnesS_WolF: he said he's not using Asterisk`
`15:13.35`	`DarKnesS_WolF`	`ManxPower: what he is using !?`
`15:13.41`	`ManxPower`	`DarKnesS_WolF: no idea`
`15:13.44`	`DarKnesS_WolF`	`ManxPower: sorry i didn't notice and why he is here then !`
`15:13.56`	`DarKnesS_WolF`	`oh i have to go to break the fast`
`15:14.01`	`DarKnesS_WolF`	`it almost sunset`
`15:14.04`	`ManxPower`	`DarKnesS_WolF: We have gotten a repuation of being able to fix all things VoIP.`
`15:14.16`	`ManxPower`	`I've heard rumors that we use dark magic to accomplish this.`
`15:14.22`	`DarKnesS_WolF`	`ManxPower: haha good for us :D`
`15:14.24`	`DarKnesS_WolF`	`lol`
`15:14.30`	`DarKnesS_WolF`	`ManxPower: leave the goat alone :P`
`15:14.39`	`DarKnesS_WolF`	`ok gtg will be in 1 hour or so i'm starving`
`15:15.12`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`15:15.12`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`15:15.21`	`wmandra`	`hey all, is anyone else having trouble getting their cisco phones to display the correct time today???`
`15:15.28`	`ManxPower`	`kilobit2001: yes, your current design will work, but it is a bad design and confusing. Manually put in a Goto for each number.`
`15:16.28`	`ManxPower`	`wmandra: It would not surprise me. Apparently some daylight savings functions think that October will NEVER have 5 sundays.`
`15:16.43`	`wmandra`	`that what it's looking like`
`15:16.55`	`kilobit2001`	`manx-- that example has one context per user.`
`15:16.57`	`ManxPower`	`wmandra: any chance on updating the cisco firmware?`
`15:17.14`	`ManxPower`	`kilobit2001: stop using the word "user". User means "person".`
`15:17.17`	`wmandra`	`maybe next weekend, but not today`
`15:17.35`	`wmandra`	`i really should though, i'm still using 6.3`
`15:18.40`	`ManxPower`	`kilobit2001: what it looks like is that you are trying to set up IVRs for each company or department.`
`15:18.51`	`ManxPower`	`wmandra: by next weekend it should no longer be a problem`
`15:18.56`	`ManxPower`	`well, at least until 2011`
`15:19.49`	`TheMafia`	`DarKnesS_WolF, I am not using asterisk so I can't set canreinvite=no, they are not in the same lan`
`15:20.19`	`kilobit2001`	`manx-- yes. and to do that, the only way I have found is to create multiple contexts for called number.`
`15:20.19`	`wmandra`	`lol`
`15:20.29`	`ManxPower`	`kilobit2001: correct.`
`15:27.58`	`saftsack`	`SuPrSluG, are you here?`
`15:29.40`	`*** join/#asterisk lorinc (n=ang@caracas-2158.adsl.interware.hu)`
`15:40.50`	`robin_sz`	`is it possible to mute a warning in the CLI ??`
`15:41.08`	`robin_sz`	`I get this:`
`15:41.10`	`robin_sz`	`Oct 22 16:40:23 WARNING[4871]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 4 as D-channel anyway!`
`15:41.21`	`robin_sz`	`every second. pian in the ass.`
`15:42.03`	`robin_sz`	`it will (hopefully) go away when my telco connects the ISDN`
`15:42.40`	`SuPrSluG`	`yes`
`15:43.10`	`ManxPower`	`robin_sz: unload chan_zap.so`
`15:44.06`	`SuPrSluG`	`ManxPower: got it to work by pulling wcfxo out of /etc/mod stuff. it finally let go of that 1st card and everything is working`
`15:44.35`	`SuPrSluG`	`saftsack:yes`
`15:44.49`	`robin_sz`	`ManxPower, not a bad plan ...`
`15:45.34`	`robin_sz`	`ManxPower, of course, thats killed the X100P as well, but hey .. its shut the damn thing up :)`
`15:45.57`	`*** part/#asterisk markdrago (n=mdrago@ool-182d1b14.dyn.optonline.net)`
`15:46.35`	`ManxPower`	`robin_sz: you can remove the pro channels from /etc/asterisk/zapata.conf`
`15:47.31`	`ManxPower`	`pro == pri`
`15:50.31`	`ManxPower`	`SuPrSluG: in that case, I'll be you are sharing IRQs`
`15:50.35`	`ManxPower`	`cat /proc/interrupts`
`15:52.16`	`SuPrSluG`	`yeah but the 2 cards aren't. 10: 808758 XT-PIC wctdm, VIA8233`
`15:52.18`	`SuPrSluG`	`<PROTECTED>`
`15:53.08`	`SuPrSluG`	`ManxPower,what the best way to make sure they're on their own irq`
`15:53.43`	`SuPrSluG`	`eth0 and wcfxo are on 10`
`15:54.52`	`saftsack`	`SuPrSluG, how many bogomips do you have with your box? or do you know how many bogomips are needed?`
`15:55.02`	`*** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no)`
`15:55.22`	`ManxPower`	`SuPrSluG: that would do it.`
`15:55.37`	`SuPrSluG`	`what's a bogomip?`
`15:55.49`	`SuPrSluG`	`i'm looking it up now`
`15:55.53`	`saftsack`	`its an indicator for the cpu speed`
`15:55.57`	`saftsack`	`cat /proc/cpuinfo`
`15:57.02`	`SuPrSluG`	`well in that case i have bogomips : 2940.92`
`15:57.13`	`saftsack`	`on your router? Oo`
`15:58.12`	`SuPrSluG`	`saftsack:are you up and running now`
`15:58.48`	`*** join/#asterisk xnon (n=xnon@200.8.30.161)`
`15:58.51`	`saftsack`	`yes im up and running but with the original firmware yet`
`15:58.55`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.161)`
`15:58.55`	`saftsack`	`i will install openwrt now`
`15:59.10`	`saftsack`	`but do i need something more than the image of the buildchain?`
`16:01.41`	`SuPrSluG`	`the one they give you works. they even have a asterisk pkg pre-made. you can get rid of things you don't want to mak more room.`
`16:02.01`	`SuPrSluG`	`what router do you have?`
`16:13.10`	`saftsack`	`speedport w501v`
`16:13.14`	`saftsack`	`it is an ar7 device`
`16:17.31`	`*** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net)`
`16:20.39`	`asdx`	`can you install asterisk in openwrt and put it in a router?`
`16:21.40`	`*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)`
`16:22.10`	`saftsack`	`yes`
`16:22.15`	`saftsack`	`this is possible`
`16:23.48`	`*** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0)`
`16:25.34`	`SuPrSluG`	`they have all supported hardware on their site`
`16:25.53`	`SuPrSluG`	`gotta walk the dog talk to ya soon`
`16:28.26`	`asdx`	`and how will you connect the phone to the router?`
`16:36.41`	`creativx`	`tcp ip`
`16:37.20`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.47)`
`16:38.36`	`*** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net)`
`16:41.02`	`robin_sz`	`heh, so I thought a 2 channel ATA would be useless ... I just found a use for the second channel! the big external bell inthe workshop`
`16:44.04`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg)`
`16:51.59`	`*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)`
`16:57.30`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`16:57.54`	`hi365`	`is rtp tcp or udt?`
`16:58.09`	`*** join/#asterisk Malawar_ (n=Malawar@adsl-75-21-165-54.dsl.sgnwmi.sbcglobal.net)`
`16:58.12`	`Malawar_`	`Oct 22 08:57:23 WARNING[25718]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/line2-081a5d00(4) to IAX2/voxee-1(256)`
`16:58.12`	`Malawar_`	`Oct 22 08:57:23 WARNING[25718]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make SIP/line2-081a5d00 compatible with IAX2/voxee-1`
`16:58.13`	`Malawar_`	`:(`
`16:58.33`	`Malawar_`	`oh`
`16:58.36`	`*** join/#asterisk lukketto (n=lukketto@host129-133-dynamic.57-82-r.retail.telecomitalia.it)`
`16:58.39`	`Malawar_`	`might have set up codecs wrong..`
`17:00.07`	`*** part/#asterisk lukketto (n=lukketto@host129-133-dynamic.57-82-r.retail.telecomitalia.it)`
`17:01.34`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`17:04.01`	`ManxPower`	`Malawar_: "show codecs"`
`17:04.02`	`Malawar_`	`yeah`
`17:04.05`	`Malawar_`	`it was trying to use g729`
`17:04.09`	`Malawar_`	`but i don't have it installed/etc`
`17:06.13`	`Malawar_`	`hmm`
`17:06.20`	`Malawar_`	`i have an IAX connection to voxee`
`17:06.24`	`Malawar_`	`i set disallow=all`
`17:06.27`	`Malawar_`	`then allow=ulaw`
`17:06.32`	`Malawar_`	`but it's still trying to use g729`
`17:07.09`	`Malawar_`	`voxee supports ulaaw, btw.`
`17:07.38`	`Malawar_`	`oops`
`17:07.38`	`Malawar_`	`nm`
`17:07.49`	`Malawar_`	`missed thde disallow line :(`
`17:15.28`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`17:28.04`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg)`
`17:31.55`	`*** join/#asterisk ramtha (n=tk@p5088C21E.dip0.t-ipconnect.de)`
`17:32.55`	`Malawar_`	`voxee is skipping :(`
`17:38.23`	`ManxPower`	`you need the device to not be allowed g729`
`17:38.36`	`Malawar_`	`i got it`
`17:38.54`	`Malawar_`	`i was pasting the block of disallow/allows into all my devices but i missed the disallow line going out to voxee`
`17:39.23`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg)`
`17:41.51`	`*** join/#asterisk napkin (n=izaak@bas10-montrealak-1096753512.dsl.bell.ca)`
`17:42.51`	`napkin`	`does anyone use linksys pap2(t) devices as fxs devices for asterisk? for a while i was set on buying a digium card, then i found these things. I only need 4 fxs ports so 2 would suffice.`
`17:43.46`	`Strom_C`	`napkin: I use both`
`17:44.08`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`17:44.18`	`*** join/#asterisk findlay (n=justin@67.137.24.115)`
`17:44.37`	`findlay`	`are there any softphones that are scriptble?`
`17:44.43`	`findlay`	`s/scriptable/`
`17:44.44`	`napkin`	`Strom_C: what is your recommendation? are they pretty flexible?`
`17:45.19`	`findlay`	`I want to be able to call a series of numbers without manually looking them up in a directory`
`17:46.19`	`*** join/#asterisk DataCompBoy (i=data@217.151.230.182)`
`17:46.23`	`DataCompBoy`	`Pfffr... Hi all!`
`17:46.31`	`DataCompBoy`	`Now, i'm need a some help with asterisk...`
`17:46.48`	`Strom_C`	`napkin: the tdm400 has slightly better audio quality and behaves more like a traditional telephone line. The only times I would recommend using the PAP2 over the TDM400 are either when you need an analog phone in a location where there is ethernet cabling but running new wiring for an analog pair is impractical, or in situations where the TDM400 is just too expensive.`
`17:48.29`	`napkin`	`thanks Strom_C. we are 4 orgs sharing a small office. we can probably handle the price difference. i am trying to get the highest quality service because a couple people here have been scared by voip and i want it to be good :)`
`17:49.17`	`napkin`	`but when i do workshops for local non profits i'll definitely recommend an asterisk+pap2 setup as it's so much cheaper...`
`17:49.17`	`DataCompBoy`	`asterisk won't send RTP packets. can't understand why...`
`17:49.43`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`17:49.49`	`DataCompBoy`	`and looks like when i'm connect to inet with external IP -- nothing receive. when i'm connect thru NAT -- everything ok`
`17:49.56`	`DataCompBoy`	`in asterisk i'm have on peer set nat=yes`
`17:49.57`	`hi365`	`Strom_C: hi`
`17:50.07`	`DataCompBoy`	`rtp debug doesn't show anything, absolutely!`
`17:50.21`	`hi365`	`Strom_c: my pap2 wont stop beeping. any ideas?`
`17:50.23`	`*** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it)`
`17:50.29`	`*** join/#asterisk ESCulapio_ (n=ESCulapi@213stb68.codetel.net.do)`
`17:50.34`	`hi365`	`i.e. what setting is it?`
`17:50.36`	`Strom_C`	`hi365: no. ive never encountered that problem.`
`17:54.01`	`Qwell`	`anybody know anything about buying hardware from the UK, and having it shipped to the US? :D`
`17:54.17`	`Strom_C`	`hi qwell :)`
`17:54.21`	`Qwell`	`Are there customs charges involved?`
`17:54.23`	`Qwell`	`Strom_C: hey`
`17:55.33`	`Strom_C`	`qwell: I don't know about shipping from the UK, but the one time I had something small shipped from hong kong, either the import duties were covered in the price of the postage, or they didn't exist at all`
`17:55.44`	`Qwell`	`ok`
`17:55.51`	`Qwell`	`? == euro, or gbp?`
`17:56.09`	`Strom_C`	`hong kong? hong kong dollar`
`17:56.21`	`Qwell`	`no, ?.. as in ?619.93`
`17:56.42`	`Strom_C`	`its just showing up as a question mark`
`17:56.47`	`Strom_C`	`stupid client`
`17:57.16`	`Qwell`	`.co.uk == euro or gbp? heh`
`17:57.28`	`Strom_C`	`gbp`
`17:57.31`	`Qwell`	`lame`
`17:57.42`	`Qwell`	`That means it costs more`
`17:57.58`	`Strom_C`	`so buy it from france or germany or something :)`
`17:58.10`	`Qwell`	`that's the thing...`
`17:58.24`	`Qwell`	`Acer went and made different configurations for different countries`
`17:58.37`	`Qwell`	`So, they only sell the configuration I want in the UK`
`17:58.50`	`Strom_C`	`what is it a configuration of?`
`17:58.53`	`Qwell`	`laptop`
`17:58.58`	`Strom_C`	`ah`
`17:59.01`	`Qwell`	`5103wlmi`
`17:59.16`	`Qwell`	`in the US, we only have 5102...`
`18:01.05`	`Qwell`	`I'm gonna have to call them on Monday, and tell them to stop being nubs :p`
`18:01.55`	`Strom_C`	`im trying to figure out what the difference is`
`18:02.40`	`Malawar_`	`does the caller id number format matter?`
`18:02.50`	`Malawar_`	`i.e. can I do "Name" <555-5555>`
`18:03.04`	`Malawar_`	`or does it have to be 12223334444`
`18:03.06`	`*** join/#asterisk Darthclue (n=Darthclu@adsl-71-149-167-182.dsl.snantx.sbcglobal.net)`
`18:03.17`	`Strom_C`	`Malawar_: are you in north america?`
`18:03.22`	`Malawar_`	`yeah.`
`18:03.37`	`Strom_C`	`you ideally want to put the number in format NXXNXXXXXX`
`18:06.05`	`Strom_C`	`i.e. just as NANPA states you should :)`
`18:07.25`	`Qwell`	`Strom_C: with what?`
`18:07.57`	`Strom_C`	`qwell: the difference between the UK-spec model and the US-spec model`
`18:08.04`	`Qwell`	`several things`
`18:08.12`	`*** join/#asterisk holmie (i=holm@blackedge.org)`
`18:08.27`	`holmie`	`so, the MWI NOTIFICATION stuff, Cisco or Digiums fault?`
`18:08.28`	`holmie`	`:-)`
`18:08.30`	`Qwell`	`first, we don't have the 5103wlmi here, which has the TL52 instead of the TL50 (both are Turion X2, but the 52 has double the cache)`
`18:08.42`	`holmie`	`(Cisco IP phones saying "399 Bad MWI NOTIFY")`
`18:08.42`	`DataCompBoy`	`Can anybody describe how to debug RTP path?`
`18:09.00`	`Qwell`	`second, their 5102wlmi model has a camera on the 512 and 1gb models, plus, I think theirs can do 4GB instead of 2GB`
`18:09.08`	`Strom_C`	`ah, i see`
`18:09.28`	`Qwell`	`our 5102wlmi with a camera has 2gb, and for some stupid reason, uses a 4200rpm drive instead of a 5400rpm drive like the lower models`
`18:09.44`	`*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`18:09.48`	`*** join/#asterisk advantage (i=steinar@8.d.a.1.1.0.0.2.ip6.arpa)`
`18:09.54`	`Qwell`	`it's just weird that it varies so much`
`18:10.57`	`Qwell`	`holmie: good luck with that...`
`18:11.02`	`holmie`	`;-D`
`18:11.14`	`Qwell`	`holmie: MWI should work just fine on Cisco phones...it's just a SIP notify or something`
`18:11.29`	`holmie`	`I think it's a cisco bug`
`18:11.40`	`holmie`	`I found a page with lots of people complaining about it`
`18:11.41`	`Qwell`	`however, if it isn't working, then it's probably a bug (either in Asterisk, or more likely, Cisco not conforming to spec...which isn't unusual)`
`18:11.53`	`hi365`	`my pap2 is beeping like a pager. how do i shut it up?`
`18:12.02`	`Juggie`	`throw it against the wall.`
`18:12.03`	`Qwell`	`hi365: unplug it? heh`
`18:12.15`	`Qwell`	`hi365: constantly, or every couple minutes?`
`18:12.46`	`Qwell`	`hi365: it's probably telling you that you have VM`
`18:12.54`	`holmie`	`Qwell: I think the c7970G don't handle the correct MWI notification from asterisk properly`
`18:13.05`	`Qwell`	`holmie: quite likely`
`18:13.32`	`holmie`	`Funky.`
`18:13.32`	`hi365`	`Qwell: every 30 sec. or so`
`18:13.45`	`Qwell`	`hi365: It's telling you that you have voicemail. There are options to disable it`
`18:15.23`	`Strom_C`	`man, you have to tell me what that option is so I can use it to irritate my most annoying clients`
`18:16.20`	`Qwell`	`Strom_C: sip.conf, mailbox=, I think`
`18:16.45`	`Qwell`	`I think the problem, is that both the ata and asterisk try to do mwi stuff`
`18:16.55`	`Qwell`	`something funky like that, and it ends up b0rking itself`
`18:16.58`	`hi365`	`Ring On No New VM = no`
`18:17.01`	`Strom_C`	`no no, the beeping option`
`18:17.05`	`Darthclue`	`How old is the 1.2 branch?`
`18:17.14`	`Strom_C`	`i know how to set mailbox= in sip.conf, qwell :)`
`18:17.26`	`Qwell`	`Darthclue: the last commit was probably a few days ago. That's a fairly loaded question`
`18:17.59`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`18:18.19`	`hi365`	`Qwell; Ring On No New VM = no`
`18:18.33`	`Darthclue`	`I like asking loaded questions. Makes things more interesting. Can't get trunk to compile cleanly so i'm going with 1.2.`
`18:18.35`	`TripleFFFF`	`can i record in an agi ?`
`18:18.42`	`Qwell`	`Darthclue: define cleanly?`
`18:18.42`	`hi365`	`VMWI Ring Splash Len: 0`
`18:18.51`	`Qwell`	`TripleFFFF: should be able to`
`18:18.57`	`Qwell`	`TripleFFFF: I can't see why not`
`18:19.01`	`TripleFFFF`	`how lol`
`18:19.10`	`Qwell`	`dunno, I've never done AGI, heh`
`18:19.12`	`TripleFFFF`	`i wanna use manager to call an agi that records`
`18:19.14`	`Strom_C`	`Darthclue: well if it's any indication, I still regularly use 1.2 for all my professional installs`
`18:19.17`	`Qwell`	`I only code - I don't use the stuff`
`18:19.33`	`Qwell`	`(I actually do use it...)`
`18:20.13`	`TripleFFFF`	`wahts net sec lol`
`18:20.31`	`hi365`	`and i dont even have new voicemail!`
`18:20.52`	`*** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it)`
`18:20.57`	`Qwell`	`TripleFFFF: it's a branch that let's you control firewall hardware from...umm...that company`
`18:21.07`	`*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)`
`18:21.07`	`TripleFFFF`	`lol`
`18:21.10`	`Qwell`	`I'm completely spacing on the name`
`18:21.21`	`Darthclue`	`make clean errors out, make install generates a couple of invalid modules (on zaptel). The last time I used Asterisk, it was always a cvs-head install.`
`18:21.46`	`Qwell`	`TripleFFFF: Ranch Networks`
`18:21.51`	`TripleFFFF`	`k heeh`
`18:22.01`	`Qwell`	`but, it uses an open protocol or something, so other vendors may support it too...dunno`
`18:22.45`	`Qwell`	`basically, it tells the firewall that it needs exactly X bandwidth, and it allocates it for you`
`18:22.45`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`18:22.48`	`drcode`	`hi all`
`18:22.51`	`Qwell`	`pretty cool stuff`
`18:23.10`	`TripleFFFF`	`hmm so i cant find monitor in the agi libs`
`18:23.16`	`TripleFFFF`	`i guess, its exec(monitor`
`18:23.23`	`Qwell`	`TripleFFFF: should be able to call it like any other application`
`18:23.55`	`TripleFFFF`	`yeah tought theres was an agi command for it`
`18:25.54`	`drcode`	`is there sip client in php / java that I can use from web site?`
`18:26.05`	`TripleFFFF`	`not java lol it sucks resources`
`18:26.10`	`drcode`	`k`
`18:26.13`	`TripleFFFF`	`drcode let me know im looking for one`
`18:26.36`	`drcode`	`I want to let users run client from web , I know there is some in activex`
`18:27.08`	`hi365`	`Qwell: any final thoughts on pap2 chirps?`
`18:27.33`	`*** join/#asterisk p1tst0p (n=p1tst0p@82-38-105-141.cable.ubr03.donc.blueyonder.co.uk)`
`18:28.52`	`DataCompBoy`	`Peeeeeeeopppleeee!! What I'm need to check to find out why no RTP sent from asterisk?!`
`18:29.28`	`hi365`	`port?`
`18:29.33`	`hi365`	`?ports`
`18:29.48`	`DataCompBoy`	`ports open on both ends.`
`18:30.00`	`DataCompBoy`	`i'm even not receive "Sent rtp packet"`
`18:30.05`	`DataCompBoy`	`nothing!`
`18:30.06`	`DataCompBoy`	`:/`
`18:30.13`	`hi365`	`which ports?`
`18:30.18`	`DataCompBoy`	`10000-20000`
`18:30.37`	`DataCompBoy`	`open on firewall on server, i have disabled firewall`
`18:30.54`	`DataCompBoy`	`Yesterday everything worked :(`
`18:31.36`	`hi365`	`:(`
`18:31.48`	`*** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)`
`18:31.48`	`hi365`	`rtp.conf fine?`
`18:31.54`	`holmie`	`Qwell: can I keep asterisk from sending the MWI notification?`
`18:31.59`	`DataCompBoy`	`untouched, and yes - fine`
`18:36.10`	`*** join/#asterisk lrizzo (n=luigi@88-149-142-59.f5.ngi.it)`
`18:36.35`	`clyrrad`	`I have a question about DTMF - I have phones set to INBAND+INFO, and it works well with Asterisk as well as most IVR systems. Howerver some IVR systems will not accept the DTMF's - Just wondering if anyone has had this issue - or knows of a work around?`
`18:37.44`	`DataCompBoy`	`the best, i'm get is one line: "Sent RTP packet to 217.151.230.182:9000 (type 0, seq 8001, ts 160, len 160)" and nothing more...`
`18:37.51`	`*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)`
`18:37.55`	`Qwell`	`holmie: unset the mailbox= option in sip.conf`
`18:39.25`	`DataCompBoy`	`oohmmm`
`18:39.38`	`DataCompBoy`	`looks like phone receive always on one port, send from other ?!`
`18:39.58`	`DataCompBoy`	`is that normal?!`
`18:42.09`	`Qwell`	`DataCompBoy: yes`
`18:42.43`	`DataCompBoy`	`Qwell: strange, but looks like Asterisk doesn't receive such packets.`
`18:42.55`	`Qwell`	`check your firewall`
`18:43.21`	`clyrrad`	`makesure your firewall has open the ports you defined in rtp.conf`
`18:43.32`	`DataCompBoy`	`firewall have open them`
`18:43.42`	`clyrrad`	`Qwell: do you know the answer to my DTMF query?`
`18:43.50`	`DataCompBoy`	`8:43:15.314965 IP 217.151.230.182.9001 > orange.11469: UDP, length 84 --- packets arrives`
`18:44.33`	`clyrrad`	`you see that on your firewall?`
`18:44.42`	`DataCompBoy`	`yes`
`18:44.48`	`clyrrad`	`does anyting hit your CLI?`
`18:44.59`	`DataCompBoy`	`Sent RTP packet to 217.151.230.182:9000 (type 0, seq 26403, ts 160, len 160)`
`18:45.11`	`DataCompBoy`	`and i'm got that packet on firewall -- and it passed`
`18:46.40`	`DataCompBoy`	`and on server i'm receive reply RTP packets (server have sent one packet from 17892 to mine 9000, from my machine sent back from 9001 to 17893).`
`18:46.58`	`DataCompBoy`	`but no more RTP packets from asterisk, and no sound heared...`
`18:48.57`	`DataCompBoy`	`looks like after no rtp phone tries to send reply from 9000 back -- also without luck :/`
`18:49.52`	`*** join/#asterisk Wall (n=mnose@host129.200-117-63.telecom.net.ar)`
`18:49.59`	`Wall`	`Hi`
`18:50.12`	`Wall`	`alguna que sepa español ?`
`18:50.18`	`Wall`	`alguno que sepa español ?`
`18:52.14`	`holmie`	`Qwell: figured, thanks. :-)`
`18:52.23`	`holmie`	`Qwell: I ask too fast!`
`18:54.26`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`18:54.45`	`DataCompBoy`	`What more I can check?...`
`19:03.27`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`19:03.40`	`Druken`	`anyone used asterfax?`
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`19:18.21`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.161)`
`19:18.47`	`TripleFFFF`	`anyway to see ip trying to connect as anonymous via dialplan like noop("IP") ?`
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`19:32.37`	`TripleFFFF`	`any reason my cisco 7960 when i talk it cuts the other one from talking .. i mean i cant hear him.. like VAD?`
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`19:40.27`	`TripleFFFF`	`??`
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`19:41.47`	`TripleFFFF`	`anyone ?`
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`19:51.56`	`enots`	`.t`
`19:52.00`	`enots`	`oops`
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`20:09.26`	`*** join/#asterisk DEader (i=DEader@ool-44c64639.dyn.optonline.net)`
`20:09.28`	`DEader`	`hello`
`20:09.37`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)`
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`20:09.44`	`DEader`	`does any one know how to tweek fromuser in asterisk`
`20:10.21`	`DEader`	`just like setting callerid in extension.conf`
`20:10.31`	`DEader`	`i want to be able to set fromuser there also`
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`20:50.16`	`eido`	`hey folks, i'm sort of exploring using wifi voip phones as convention / hotel-type radios. i know the concept of 'PTT' is still not really available to the mases (except it seems via Vocera's 'badges') - is there a recommendation of a type of VOIP wifi phone that is inexpensive but capable? I see the ZyXEL P-2000W's around on ebay for decent prices?`
`20:52.20`	`[TK]D-Fender`	`WiFi VoIP phones currently SUCK. ALL of them.`
`20:52.37`	`eido`	`heh.`
`20:55.56`	`robin_sz`	`The Zyxel WiFi phones are great ...`
`20:56.16`	`robin_sz`	`you can use them to hold most sorts of doors open`
`20:56.51`	`robin_sz`	`they do have one teeny weeny little drawback in terms of using them as a phone`
`20:56.56`	`eido`	`alrighty then :)`
`20:57.18`	`robin_sz`	`their complete inability to connect to other WiFi devices for short periods ...`
`20:57.30`	`[hC]`	`Ive tried the zyxel and linksys wip300 extensively`
`20:57.38`	`robin_sz`	`typically for the first hour or so after you turn them on ...`
`20:57.41`	`[hC]`	`im now having to eat a sale i did to a client of wip300's cause they suck`
`20:57.43`	`[hC]`	`and they dont want em`
`20:57.53`	`robin_sz`	`after that, the battery is flat and the problem goes away :)`
`20:59.27`	`robin_sz`	`best advice is to use DECT phones and ATAs`
`20:59.28`	`eido`	`ew.`
`20:59.40`	`robin_sz`	`theres a dutch SIP to DECT gateway ...`
`20:59.41`	`eido`	`er, the ew was on the various ocmments, not on the last one.`
`21:00.43`	`robin_sz`	`I think it takes up to 6 phones per base station`
`21:00.46`	`eido`	`well, here's what i'm tyring to do, see if i'm loopy. i do convention services - registration, badging, IT stuff, etc. i see so many events that use FRS radios as their communication net. they suck. "I can do better!" - i'm trying to figure out -how- :)`
`21:01.00`	`robin_sz`	`hmmm`
`21:01.02`	`eido`	`the best iv'e seen are the Vocera badges - that's pretty damned slick, but i bet it costs an arm and a leg.`
`21:01.18`	`[TK]D-Fender`	`WIP300 is nice for browsing networks, but the phone has vitually 0 functionality, and the interface is slow`
`21:01.24`	`robin_sz`	`wifi is unsuitable ...`
`21:01.40`	`robin_sz`	`due to jitter amongst other stuff`
`21:01.56`	`robin_sz`	`the tdm soon clogs up with a few active channels`
`21:01.56`	`eido`	`well, audio quality on the wifi network will be a -vast- improvement over FRS. really :)`
`21:03.29`	`robin_sz`	`does that matter?`
`21:04.23`	`eido`	`does which matter?`
`21:04.29`	`robin_sz`	`audio quality`
`21:04.56`	`robin_sz`	`is that high on the list of your clients priorities?`
`21:05.04`	`robin_sz`	`or is range and reliability more important?`
`21:05.39`	`robin_sz`	`woudl wired phones be too hard to do?`
`21:06.04`	`eido`	`range and coverage are the kicker.`
`21:06.11`	`eido`	`we need to be able to bridge between disparate locations.`
`21:07.11`	`robin_sz`	`out in the fields?`
`21:07.11`	`eido`	`so folks in the ballroom, vs folks down in registration vs folks on the 40th floor of the hotel.`
`21:07.12`	`robin_sz`	`in some woods?`
`21:07.12`	`eido`	`no, primarily in the same building.`
`21:07.12`	`robin_sz`	`in africa?`
`21:07.12`	`eido`	`we can carpet the place with our own wifi hardware.`
`21:07.20`	`robin_sz`	`or europe?`
`21:07.29`	`eido`	`ideally, a PTT solution for all the users - similar to FRS would be great. private communications (direct dial) is a nice, but general 'group chat' is really what we need.`
`21:07.31`	`eido`	`no, US.`
`21:07.43`	`robin_sz`	`in europe all the decent conference venues have networking spread all over them`
`21:08.09`	`eido`	`yah, most of my events are not in large commercial venues.`
`21:08.23`	`eido`	`they tend to be in at best, conference hotels (1500-room hotels)`
`21:08.42`	`eido`	`at worst, small cheezy hotels that have poor wireless. at very worst, its small cheezy hotels with commercial wireless services :)`
`21:08.42`	`robin_sz`	`and they aint networked?`
`21:08.51`	`eido`	`we'd put in our own wireless.`
`21:09.06`	`eido`	`i don't mind buying a dozen APs to set our own network and control it.`
`21:09.14`	`robin_sz`	`wireless trunks to router/gateways .. wired phoens from there`
`21:09.14`	`eido`	`it' the mobile units that are the problem.`
`21:09.17`	`robin_sz`	`or DECT`
`21:09.34`	`*** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se)`
`21:09.54`	`eido`	`i don't know DECT phones well... how do they handle hopping? can i move one from one conference room to another, or upstairs, and it'll stay on-network?`
`21:10.19`	`robin_sz`	`no idea`
`21:10.32`	`eido`	`yah, that's the problem. and we're still stuck with no PTT tech.`
`21:10.43`	`robin_sz`	`sounds like you need a UHF base station and a bunch of Motorola HTs`
`21:10.50`	`eido`	`thought of that too.`
`21:10.58`	`robin_sz`	`probably the way to go`
`21:11.22`	`robin_sz`	`VOIP is almost certainly NOT the way to go`
`21:11.25`	`shy_guy`	`eido:whats this hopping feature like to the end user?`
`21:11.34`	`eido`	`shy_guy: in my ideal world, seemless.`
`21:11.41`	`eido`	`see`
`21:11.47`	`eido`	`frs or gmrs or the like are fine for small venues.`
`21:11.55`	`eido`	`ala, everyone within, say, 1500 feet of each other.`
`21:11.57`	`shy_guy`	`eido:if u have an account on the conference server,cant u join from anywhere? whats missing`
`21:12.03`	`eido`	`but go beyond that, and the whole mechanism breaks.`
`21:12.47`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`21:13.01`	`shy_guy`	`i cant really get u. are you from the marketing? :D`
`21:13.06`	`eido`	`shy - that would mean a wireless SIP phone that's dialled in / 'off hook' permanently. PTT problems are already an issue - doing it that way would bring on battery problems.`
`21:13.24`	`eido`	`shy_guy: no. by the way, something is wrong with your 'yo' keys.`
`21:14.06`	`shy_guy`	`we do such wifi stuff`
`21:14.06`	`robin_sz`	`eido, as I thought. Motorola have it well covered.`
`21:14.11`	`robin_sz`	`check out iDEN`
`21:14.12`	`shy_guy`	`pretty seamlessly`
`21:14.14`	`robin_sz`	`http://idenphones.motorola.com/idenHome/common/what_is_iden.jsp`
`21:14.28`	`*** join/#asterisk saftsack (n=bla@pD9E04CF7.dip.t-dialin.net)`
`21:14.33`	`robin_sz`	`and some pretty cute phones too`
`21:14.34`	`robin_sz`	`http://idenphones.motorola.com/idenProducts/phonesHome.do?phones=1`
`21:15.12`	`saftsack`	`hi`
`21:15.24`	`eido`	`shy_guy: your product is not what i'm trying to do.`
`21:15.35`	`eido`	`iden is what nextel uss, yes?`
`21:16.38`	`shy_guy`	`eido:whats this ptt issue? tell me`
`21:16.39`	`eido`	`yah, iden is neat, but is a lot higher level / more costly than what i'm looking for. i may be hosed here :(`
`21:16.45`	`eido`	`shy_guy: ever use a walkie talkie?`
`21:16.54`	`robin_sz`	`yeah, lots.`
`21:16.55`	`eido`	`like an frs or gmrs radio?`
`21:17.10`	`shy_guy`	`keep talkin eido`
`21:17.24`	`eido`	`er, that sentence ended with a ? - that implies "its your turn to talk"`
`21:17.46`	`robin_sz`	`i use Motorola 300's a LOT ... coverage is typically 1000s of metres`
`21:17.56`	`robin_sz`	`well, used`
`21:17.57`	`eido`	`robin_sz: they require a commercial license, yes?`
`21:18.00`	`robin_sz`	`not in that game anymore`
`21:18.03`	`robin_sz`	`yeah`
`21:18.26`	`eido`	`i have a bunch of motorola VHF radios tat i was going to adapt to GMRS frequencies - they can do it, but they're not "fcc approved" equipment. It's a little edgy. but the radios are cheap.`
`21:18.47`	`shy_guy`	`i worked on ultra wide band radio`
`21:18.54`	`robin_sz`	`2watts on VHF goes a LONG way`
`21:19.10`	`eido`	`yeah, no kidding.`
`21:19.10`	`shy_guy`	`and now after four years of working when i left, its ericsson whose implementing it`
`21:19.27`	`shy_guy`	`god damn fcc and the club`
`21:19.35`	`shy_guy`	`politcs players`
`21:19.47`	`robin_sz`	`iDEN looks to be exactly what you need`
`21:19.58`	`*** join/#asterisk xnon_ (n=xnon@200.8.30.161)`
`21:20.09`	`robin_sz`	`im pretty certain voip is not what you want`
`21:20.16`	`eido`	`yeah, but iden is a large scale network - it's a digital transport. i'd need radio coverage in the venue, plus the phones - plus the fcc licenseing to do it, plus a server to contorl / route the phones.`
`21:20.24`	`eido`	`nifty if i had a zillion dollars :)`
`21:20.44`	`robin_sz`	`wired phoens could work ...`
`21:20.49`	`robin_sz`	`but you dotn want that either`
`21:20.58`	`eido`	`yah, i know i could do voip + wired.`
`21:21.02`	`shy_guy`	`eido:lol, who the hell wud wish for iden`
`21:21.17`	`*** join/#asterisk DanTMG (n=danielga@124-168-3-90.dyn.iinet.net.au)`
`21:21.18`	`robin_sz`	`radios would work ..`
`21:21.19`	`eido`	`that could link all the stations / departments / whatever together, without stringing pots lines (or using the hotel's PBX, which is traditoinally -garbage-`
`21:21.21`	`*** join/#asterisk Gunde (n=spamyous@82.153.170.213)`
`21:21.32`	`eido`	`i'm unfortunately heading back to radios with some form of repeater.`
`21:21.39`	`robin_sz`	`rounds simplest`
`21:21.44`	`robin_sz`	`sounds`
`21:21.44`	`eido`	`robin_sz: did you say you used gp300's?`
`21:21.51`	`shy_guy`	`whats wrong with wifi and wifi phones doing all this`
`21:21.58`	`eido`	`shy_guy: listen to the convo please.`
`21:22.04`	`robin_sz`	`yeah, on vhf and uhf`
`21:22.05`	`eido`	`mostly because they can't.`
`21:22.14`	`eido`	`gp300's look inexpensive. hm.`
`21:22.16`	`shy_guy`	`by 2008, 80% mobile phones would have wifi support anyway`
`21:22.20`	`eido`	`i should look at an FCC commercial license.`
`21:22.21`	`Greek-Boy`	`if the wifi phones were reliable atleast you would have a covnerged solution but u gotta use the radios for now`
`21:22.22`	`robin_sz`	`shy_guy, there are no wifi phones`
`21:22.44`	`robin_sz`	`gp300s are solid and reliable and very configurable ... and multi use`
`21:23.01`	`eido`	`heh. i could... rig up one of the Gp300's to an audio gateway, subscribe it into an asterisk hosted conference.`
`21:23.03`	`robin_sz`	`you can use them to talk on, or to beat punters to death with`
`21:23.10`	`eido`	`folks could listen / chat in from VOIP clients on PC's.`
`21:23.11`	`eido`	`HAH`
`21:23.13`	`eido`	`that's getting interesting.`
`21:23.35`	`*** part/#asterisk Gunde (n=spamyous@82.153.170.213)`
`21:23.36`	`eido`	`can the GP300's do freq offset stuff? ala ham radio repeater style?`
`21:23.46`	`robin_sz`	`oh yeah`
`21:23.47`	`robin_sz`	`easy`
`21:23.52`	`robin_sz`	`fully synthed`
`21:23.54`	`shy_guy`	`Greek-Boy:i dont want to market our products, but wifi works. we have a patent pending technology on that.`
`21:24.08`	`saftsack`	`SuPrSluG hi are you still here?`
`21:24.17`	`eido`	`i could... hmmmmmmmmmmmmmmm. interesting. FCC would love this. what if i had some form of simple audio gateway. hmm. tht won't work. i can't find out when it should transmit. or could it. Hm.`
`21:24.30`	`robin_sz`	`shy_guy, you'll be rich then, because as yet, all WiFI phoens suck`
`21:24.49`	`eido`	`i was just thinking of attaching a radio via a simple PTT trigger, coneting it via VOIP gateway to an asterisk hosted conference. VOIP clients on PC's would be able to hear the radio conversation.`
`21:24.56`	`shy_guy`	`robin_sz:why exactly do they suck? can u point to the core issues`
`21:24.57`	`eido`	`but how could they answer back? Hmm.`
`21:24.59`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`21:24.59`	`*** mode/#asterisk [+o mog] by ChanServ`
`21:25.12`	`robin_sz`	`shy_guy, point 1, inabality to connect to gateways`
`21:25.21`	`robin_sz`	`shy_guy, poitn 2 battery live measured in hours`
`21:25.25`	`eido`	`hah. freq offset would be on permanent trnasmit? hm.`
`21:25.36`	`robin_sz`	`shy_guy, point 3 see poitn 1`
`21:26.01`	`shy_guy`	`robin_sz:is that all?`
`21:26.07`	`robin_sz`	`ALL?`
`21:26.18`	`eido`	`uhh. i'd say that's "enough". it means i can't use them for this function.`
`21:26.23`	`robin_sz`	`I would have thought an inablity to connect to a gateway was failry serious no?`
`21:26.43`	`robin_sz`	`a MINIMUM pocket life of 8 hours is a MUST HAVE`
`21:26.46`	`eido`	`also, the voip phones don't AP-hop well, as i'm understanding.`
`21:26.51`	`eido`	`minimum, yes.`
`21:27.00`	`robin_sz`	`absolute minimum`
`21:27.03`	`eido`	`and for radio usage, they would have to be on permanet 'listen' mode.`
`21:27.07`	`robin_sz`	`the zyxel is ~2 hours`
`21:27.10`	`shy_guy`	`i have heard and answered more criticism than that robin_sz`
`21:27.14`	`eido`	`iden folks work because the y'wake up' when a connection / msg comes in.`
`21:27.24`	`eido`	`most voip phones are not that smart.`
`21:27.26`	`eido`	`er.`
`21:27.34`	`eido`	`i'd say all of them - since they' renot designed to work that way.`
`21:27.46`	`robin_sz`	`shy_guy, well, show me one that works then ...`
`21:28.23`	`shy_guy`	`robin_sz:you are livin in wifi stone age like prolly everyone else. but trust me there is a product that exceeds your expectations`
`21:28.30`	`eido`	`shy_guy: name it.`
`21:28.32`	`robin_sz`	`bollocks`
`21:28.43`	`shy_guy`	`it works on same wifi networks`
`21:28.46`	`eido`	`shy_guy:`
`21:28.49`	`eido`	`name what you're talking about.`
`21:28.55`	`eido`	`and stop fishing around like a marketing idiot.`
`21:28.56`	`shy_guy`	`runs on windows mobile and symbian platforms`
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`21:29.12`	`shy_guy`	`hehe`
`21:29.33`	`eido`	`anyway, shy_guy guys idiocies aside, i think i'm hosed on this.`
`21:29.34`	`shy_guy`	`eido: u...`
`21:29.38`	`eido`	`the -only-thing i saw was vocera.`
`21:29.50`	`eido`	`i'll drop them a aline and see if they can do something for something under the US military budget.`
`21:30.14`	`shy_guy`	`eido:you like to discuss things in which i have no interest. microwave,radios... um not even hearing.`
`21:30.29`	`robin_sz`	`shy_guy, so you going to name this mystery product then or what?`
`21:30.33`	`eido`	`shy_guy: and you keep spouting drivel with no information.`
`21:30.42`	`eido`	`so unless you ave actual facts, information, and details, just stfu.`
`21:30.56`	`saftsack`	`is it possible to build asterisk < 2mb?`
`21:31.10`	`Qwell`	`saftsack: sure`
`21:31.22`	`Qwell`	`saftsack: You'll need to remove quite a few modules, but...yeah`
`21:31.32`	`saftsack`	`and < 0.5mb in a sqaushfs system?`
`21:31.34`	`*** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu)`
`21:31.41`	`Qwell`	`500k? probably not`
`21:31.59`	`Qwell`	`at least, not with any usability`
`21:32.06`	`saftsack`	`hmm but this 500k are compressed (squashfs)`
`21:32.39`	`eido`	`hmmmmmmmmmmmm.`
`21:33.02`	`saftsack`	`Qwell do you know the compression factor of squashfs?`
`21:33.13`	`Qwell`	`no clue`
`21:33.58`	`saftsack`	`ok`
`21:40.03`	`shy_guy`	`robin_sz:guess you liked it, later mate!`
`21:40.36`	`eido`	`robin_sz: did shy_guy point you at his swedish whatever site?`
`21:40.49`	`robin_sz`	`interesting idea ...`
`21:40.57`	`eido`	`yes, and i told him it was not what i needed, but he wasn't listening.`
`21:41.10`	`eido`	`it's an auto-switch from voip to cellular net and back`
`21:41.12`	`robin_sz`	`shrug .. .its stil an interesting idea`
`21:41.13`	`eido`	`bfd?`
`21:41.51`	`robin_sz`	`it assumes you have some windows enabled, gsm and wifi capable PDA`
`21:42.08`	`robin_sz`	`do those things last 8 hours on in your pocket?`
`21:42.17`	`robin_sz`	`in WiFi mode>`
`21:42.18`	`robin_sz`	`?`
`21:42.20`	`eido`	`not with the radio active.`
`21:42.22`	`eido`	`wifi is heavy.`
`21:42.30`	`robin_sz`	`oh`
`21:42.38`	`robin_sz`	`thats my minimum requirement`
`21:51.14`	`Greek-Boy`	`anyone here from Philippines?`
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`22:01.38`	`DEader`	`is it posible to set asterisk user setting example fromuser and context from the extension.conf file`
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